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Diffstat (limited to 'plugins/MacVST/VariMu/source/VariMuProc.cpp')
| -rwxr-xr-x | plugins/MacVST/VariMu/source/VariMuProc.cpp | 518 |
1 files changed, 518 insertions, 0 deletions
diff --git a/plugins/MacVST/VariMu/source/VariMuProc.cpp b/plugins/MacVST/VariMu/source/VariMuProc.cpp new file mode 100755 index 0000000..b966cd8 --- /dev/null +++ b/plugins/MacVST/VariMu/source/VariMuProc.cpp @@ -0,0 +1,518 @@ +/* ======================================== + * VariMu - VariMu.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __VariMu_H +#include "VariMu.h" +#endif + +void VariMu::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 = 2.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + + double threshold = 1.001 - (1.0-pow(1.0-A,3)); + double muMakeupGain = sqrt(1.0 / threshold); + muMakeupGain = (muMakeupGain + sqrt(muMakeupGain))/2.0; + muMakeupGain = sqrt(muMakeupGain); + double outGain = sqrt(muMakeupGain); + //gain settings around threshold + double release = pow((1.15-B),5)*32768.0; + release /= overallscale; + double fastest = sqrt(release); + //speed settings around release + double coefficient; + double output = outGain * C; + double wet = D; + long double squaredSampleL; + long double squaredSampleR; + + // µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~ + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + 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 aVariMu. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + if (fabs(inputSampleL) > fabs(previousL)) squaredSampleL = previousL * previousL; + else squaredSampleL = inputSampleL * inputSampleL; + previousL = inputSampleL; + inputSampleL *= muMakeupGain; + + if (fabs(inputSampleR) > fabs(previousR)) squaredSampleR = previousR * previousR; + else squaredSampleR = inputSampleR * inputSampleR; + previousR = inputSampleR; + inputSampleR *= muMakeupGain; + + //adjust coefficients for L + if (flip) + { + if (fabs(squaredSampleL) > threshold) + { + muVaryL = threshold / fabs(squaredSampleL); + muAttackL = sqrt(fabs(muSpeedAL)); + muCoefficientAL = muCoefficientAL * (muAttackL-1.0); + if (muVaryL < threshold) + { + muCoefficientAL = muCoefficientAL + threshold; + } + else + { + muCoefficientAL = muCoefficientAL + muVaryL; + } + muCoefficientAL = muCoefficientAL / muAttackL; + } + else + { + muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0); + muCoefficientAL = muCoefficientAL + 1.0; + muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL); + } + muNewSpeedL = muSpeedAL * (muSpeedAL-1); + muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest; + muSpeedAL = muNewSpeedL / muSpeedAL; + } + else + { + if (fabs(squaredSampleL) > threshold) + { + muVaryL = threshold / fabs(squaredSampleL); + muAttackL = sqrt(fabs(muSpeedBL)); + muCoefficientBL = muCoefficientBL * (muAttackL-1); + if (muVaryL < threshold) + { + muCoefficientBL = muCoefficientBL + threshold; + } + else + { + muCoefficientBL = muCoefficientBL + muVaryL; + } + muCoefficientBL = muCoefficientBL / muAttackL; + } + else + { + muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0); + muCoefficientBL = muCoefficientBL + 1.0; + muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL); + } + muNewSpeedL = muSpeedBL * (muSpeedBL-1); + muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest; + muSpeedBL = muNewSpeedL / muSpeedBL; + } + //got coefficients, adjusted speeds for L + + //adjust coefficients for R + if (flip) + { + if (fabs(squaredSampleR) > threshold) + { + muVaryR = threshold / fabs(squaredSampleR); + muAttackR = sqrt(fabs(muSpeedAR)); + muCoefficientAR = muCoefficientAR * (muAttackR-1.0); + if (muVaryR < threshold) + { + muCoefficientAR = muCoefficientAR + threshold; + } + else + { + muCoefficientAR = muCoefficientAR + muVaryR; + } + muCoefficientAR = muCoefficientAR / muAttackR; + } + else + { + muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0); + muCoefficientAR = muCoefficientAR + 1.0; + muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR); + } + muNewSpeedR = muSpeedAR * (muSpeedAR-1); + muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest; + muSpeedAR = muNewSpeedR / muSpeedAR; + } + else + { + if (fabs(squaredSampleR) > threshold) + { + muVaryR = threshold / fabs(squaredSampleR); + muAttackR = sqrt(fabs(muSpeedBR)); + muCoefficientBR = muCoefficientBR * (muAttackR-1); + if (muVaryR < threshold) + { + muCoefficientBR = muCoefficientBR + threshold; + } + else + { + muCoefficientBR = muCoefficientBR + muVaryR; + } + muCoefficientBR = muCoefficientBR / muAttackR; + } + else + { + muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0); + muCoefficientBR = muCoefficientBR + 1.0; + muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR); + } + muNewSpeedR = muSpeedBR * (muSpeedBR-1); + muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest; + muSpeedBR = muNewSpeedR / muSpeedBR; + } + //got coefficients, adjusted speeds for R + + if (flip) + { + coefficient = (muCoefficientAL + pow(muCoefficientAL,2))/2.0; + inputSampleL *= coefficient; + coefficient = (muCoefficientAR + pow(muCoefficientAR,2))/2.0; + inputSampleR *= coefficient; + } + else + { + coefficient = (muCoefficientBL + pow(muCoefficientBL,2))/2.0; + inputSampleL *= coefficient; + coefficient = (muCoefficientBR + pow(muCoefficientBR,2))/2.0; + inputSampleR *= coefficient; + } + //applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~ + //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 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 VariMu::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 = 2.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + + double threshold = 1.001 - (1.0-pow(1.0-A,3)); + double muMakeupGain = sqrt(1.0 / threshold); + muMakeupGain = (muMakeupGain + sqrt(muMakeupGain))/2.0; + muMakeupGain = sqrt(muMakeupGain); + double outGain = sqrt(muMakeupGain); + //gain settings around threshold + double release = pow((1.15-B),5)*32768.0; + release /= overallscale; + double fastest = sqrt(release); + //speed settings around release + double coefficient; + double output = outGain * C; + double wet = D; + long double squaredSampleL; + long double squaredSampleR; + + // µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~ + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + 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 aVariMu. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + if (fabs(inputSampleL) > fabs(previousL)) squaredSampleL = previousL * previousL; + else squaredSampleL = inputSampleL * inputSampleL; + previousL = inputSampleL; + inputSampleL *= muMakeupGain; + + if (fabs(inputSampleR) > fabs(previousR)) squaredSampleR = previousR * previousR; + else squaredSampleR = inputSampleR * inputSampleR; + previousR = inputSampleR; + inputSampleR *= muMakeupGain; + + //adjust coefficients for L + if (flip) + { + if (fabs(squaredSampleL) > threshold) + { + muVaryL = threshold / fabs(squaredSampleL); + muAttackL = sqrt(fabs(muSpeedAL)); + muCoefficientAL = muCoefficientAL * (muAttackL-1.0); + if (muVaryL < threshold) + { + muCoefficientAL = muCoefficientAL + threshold; + } + else + { + muCoefficientAL = muCoefficientAL + muVaryL; + } + muCoefficientAL = muCoefficientAL / muAttackL; + } + else + { + muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0); + muCoefficientAL = muCoefficientAL + 1.0; + muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL); + } + muNewSpeedL = muSpeedAL * (muSpeedAL-1); + muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest; + muSpeedAL = muNewSpeedL / muSpeedAL; + } + else + { + if (fabs(squaredSampleL) > threshold) + { + muVaryL = threshold / fabs(squaredSampleL); + muAttackL = sqrt(fabs(muSpeedBL)); + muCoefficientBL = muCoefficientBL * (muAttackL-1); + if (muVaryL < threshold) + { + muCoefficientBL = muCoefficientBL + threshold; + } + else + { + muCoefficientBL = muCoefficientBL + muVaryL; + } + muCoefficientBL = muCoefficientBL / muAttackL; + } + else + { + muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0); + muCoefficientBL = muCoefficientBL + 1.0; + muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL); + } + muNewSpeedL = muSpeedBL * (muSpeedBL-1); + muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest; + muSpeedBL = muNewSpeedL / muSpeedBL; + } + //got coefficients, adjusted speeds for L + + //adjust coefficients for R + if (flip) + { + if (fabs(squaredSampleR) > threshold) + { + muVaryR = threshold / fabs(squaredSampleR); + muAttackR = sqrt(fabs(muSpeedAR)); + muCoefficientAR = muCoefficientAR * (muAttackR-1.0); + if (muVaryR < threshold) + { + muCoefficientAR = muCoefficientAR + threshold; + } + else + { + muCoefficientAR = muCoefficientAR + muVaryR; + } + muCoefficientAR = muCoefficientAR / muAttackR; + } + else + { + muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0); + muCoefficientAR = muCoefficientAR + 1.0; + muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR); + } + muNewSpeedR = muSpeedAR * (muSpeedAR-1); + muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest; + muSpeedAR = muNewSpeedR / muSpeedAR; + } + else + { + if (fabs(squaredSampleR) > threshold) + { + muVaryR = threshold / fabs(squaredSampleR); + muAttackR = sqrt(fabs(muSpeedBR)); + muCoefficientBR = muCoefficientBR * (muAttackR-1); + if (muVaryR < threshold) + { + muCoefficientBR = muCoefficientBR + threshold; + } + else + { + muCoefficientBR = muCoefficientBR + muVaryR; + } + muCoefficientBR = muCoefficientBR / muAttackR; + } + else + { + muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0); + muCoefficientBR = muCoefficientBR + 1.0; + muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR); + } + muNewSpeedR = muSpeedBR * (muSpeedBR-1); + muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest; + muSpeedBR = muNewSpeedR / muSpeedBR; + } + //got coefficients, adjusted speeds for R + + if (flip) + { + coefficient = (muCoefficientAL + pow(muCoefficientAL,2))/2.0; + inputSampleL *= coefficient; + coefficient = (muCoefficientAR + pow(muCoefficientAR,2))/2.0; + inputSampleR *= coefficient; + } + else + { + coefficient = (muCoefficientBL + pow(muCoefficientBL,2))/2.0; + inputSampleL *= coefficient; + coefficient = (muCoefficientBR + pow(muCoefficientBR,2))/2.0; + inputSampleR *= coefficient; + } + //applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~ + //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. +} |