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author | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
commit | 633be2e22c6648c901f08f3b4cd4e8e14ea86443 (patch) | |
tree | 1e272c3d2b5bd29636b9f9f521af62734e4df012 /plugins/WinVST/Pressure4/Pressure4Proc.cpp | |
parent | 057757aa8eb0a463caf0cdfdb5894ac5f723ff3f (diff) | |
download | airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.gz airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.bz2 airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.zip |
Updates (in case my plane crashes)
Diffstat (limited to 'plugins/WinVST/Pressure4/Pressure4Proc.cpp')
-rwxr-xr-x | plugins/WinVST/Pressure4/Pressure4Proc.cpp | 449 |
1 files changed, 449 insertions, 0 deletions
diff --git a/plugins/WinVST/Pressure4/Pressure4Proc.cpp b/plugins/WinVST/Pressure4/Pressure4Proc.cpp new file mode 100755 index 0000000..02b8a78 --- /dev/null +++ b/plugins/WinVST/Pressure4/Pressure4Proc.cpp @@ -0,0 +1,449 @@ +/* ======================================== + * Pressure4 - Pressure4.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Pressure4_H +#include "Pressure4.h" +#endif + +void Pressure4::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* inputL = inputs[0]; + float* inputR = inputs[1]; + float* outputL = outputs[0]; + float* outputR = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + double threshold = 1.0 - (A * 0.95); + double muMakeupGain = 1.0 / threshold; + //gain settings around threshold + double release = pow((1.28-B),5)*32768.0; + release /= overallscale; + double fastest = sqrt(release); + //speed settings around release + long double bridgerectifier; + double coefficient; + double inputSense; + double mewiness = (C*2.0)-1.0; + double unmewiness; + double outputGain = D; + bool positivemu; + if (mewiness >= 0) + { + positivemu = true; + unmewiness = 1.0-mewiness; + } + else + { + positivemu = false; + mewiness = -mewiness; + unmewiness = 1.0-mewiness; + } + // µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~ + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + + while (--sampleFrames >= 0) + { + inputSampleL = *inputL; + inputSampleR = *inputR; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + + inputSampleL = inputSampleL * muMakeupGain; + inputSampleR = inputSampleR * muMakeupGain; + + inputSense = fabs(inputSampleL); + if (fabs(inputSampleR) > inputSense) + inputSense = fabs(inputSampleR); + //we will take the greater of either channel and just use that, then apply the result + //to both stereo channels. + + if (flip) + { + if (inputSense > threshold) + { + muVary = threshold / inputSense; + muAttack = sqrt(fabs(muSpeedA)); + muCoefficientA = muCoefficientA * (muAttack-1.0); + if (muVary < threshold) + { + muCoefficientA = muCoefficientA + threshold; + } + else + { + muCoefficientA = muCoefficientA + muVary; + } + muCoefficientA = muCoefficientA / muAttack; + } + else + { + muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0); + muCoefficientA = muCoefficientA + 1.0; + muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA); + } + muNewSpeed = muSpeedA * (muSpeedA-1); + muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; + muSpeedA = muNewSpeed / muSpeedA; + } + else + { + if (inputSense > threshold) + { + muVary = threshold / inputSense; + muAttack = sqrt(fabs(muSpeedB)); + muCoefficientB = muCoefficientB * (muAttack-1); + if (muVary < threshold) + { + muCoefficientB = muCoefficientB + threshold; + } + else + { + muCoefficientB = muCoefficientB + muVary; + } + muCoefficientB = muCoefficientB / muAttack; + } + else + { + muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0); + muCoefficientB = muCoefficientB + 1.0; + muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB); + } + muNewSpeed = muSpeedB * (muSpeedB-1); + muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; + muSpeedB = muNewSpeed / muSpeedB; + } + //got coefficients, adjusted speeds + + if (flip) + { + if (positivemu) coefficient = pow(muCoefficientA,2); + else coefficient = sqrt(muCoefficientA); + coefficient = (coefficient*mewiness)+(muCoefficientA*unmewiness); + inputSampleL *= coefficient; + inputSampleR *= coefficient; + } + else + { + if (positivemu) coefficient = pow(muCoefficientB,2); + else coefficient = sqrt(muCoefficientB); + coefficient = (coefficient*mewiness)+(muCoefficientB*unmewiness); + inputSampleL *= coefficient; + 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 + + if (outputGain != 1.0) { + inputSampleL *= outputGain; + inputSampleR *= outputGain; + } + + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0){inputSampleL = bridgerectifier;} + else {inputSampleL = -bridgerectifier;} + //second stage of overdrive to prevent overs and allow bloody loud extremeness + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0){inputSampleR = bridgerectifier;} + else {inputSampleR = -bridgerectifier;} + //second stage of overdrive to prevent overs and allow bloody loud extremeness + + //noise shaping to 32-bit floating point + if (flip) { + fpTemp = inputSampleL; + fpNShapeAL = (fpNShapeAL*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeAL; + + fpTemp = inputSampleR; + fpNShapeAR = (fpNShapeAR*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeAR; + } + else { + fpTemp = inputSampleL; + fpNShapeBL = (fpNShapeBL*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeBL; + + fpTemp = inputSampleR; + fpNShapeBR = (fpNShapeBR*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeBR; + } + flip = !flip; + //end noise shaping on 32 bit output + + *outputL = inputSampleL; + *outputR = inputSampleR; + + *inputL++; + *inputR++; + *outputL++; + *outputR++; + } +} + +void Pressure4::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* inputL = inputs[0]; + double* inputR = inputs[1]; + double* outputL = outputs[0]; + double* outputR = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + double threshold = 1.0 - (A * 0.95); + double muMakeupGain = 1.0 / threshold; + //gain settings around threshold + double release = pow((1.28-B),5)*32768.0; + release /= overallscale; + double fastest = sqrt(release); + //speed settings around release + long double bridgerectifier; + double coefficient; + double inputSense; + double mewiness = (C*2.0)-1.0; + double unmewiness; + double outputGain = D; + bool positivemu; + if (mewiness >= 0) + { + positivemu = true; + unmewiness = 1.0-mewiness; + } + else + { + positivemu = false; + mewiness = -mewiness; + unmewiness = 1.0-mewiness; + } + // µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~ + double fpTemp; //this is different from singlereplacing + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + + + while (--sampleFrames >= 0) + { + inputSampleL = *inputL; + inputSampleR = *inputR; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + + inputSampleL = inputSampleL * muMakeupGain; + inputSampleR = inputSampleR * muMakeupGain; + + inputSense = fabs(inputSampleL); + if (fabs(inputSampleR) > inputSense) + inputSense = fabs(inputSampleR); + //we will take the greater of either channel and just use that, then apply the result + //to both stereo channels. + + if (flip) + { + if (inputSense > threshold) + { + muVary = threshold / inputSense; + muAttack = sqrt(fabs(muSpeedA)); + muCoefficientA = muCoefficientA * (muAttack-1.0); + if (muVary < threshold) + { + muCoefficientA = muCoefficientA + threshold; + } + else + { + muCoefficientA = muCoefficientA + muVary; + } + muCoefficientA = muCoefficientA / muAttack; + } + else + { + muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0); + muCoefficientA = muCoefficientA + 1.0; + muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA); + } + muNewSpeed = muSpeedA * (muSpeedA-1); + muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; + muSpeedA = muNewSpeed / muSpeedA; + } + else + { + if (inputSense > threshold) + { + muVary = threshold / inputSense; + muAttack = sqrt(fabs(muSpeedB)); + muCoefficientB = muCoefficientB * (muAttack-1); + if (muVary < threshold) + { + muCoefficientB = muCoefficientB + threshold; + } + else + { + muCoefficientB = muCoefficientB + muVary; + } + muCoefficientB = muCoefficientB / muAttack; + } + else + { + muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0); + muCoefficientB = muCoefficientB + 1.0; + muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB); + } + muNewSpeed = muSpeedB * (muSpeedB-1); + muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; + muSpeedB = muNewSpeed / muSpeedB; + } + //got coefficients, adjusted speeds + + if (flip) + { + if (positivemu) coefficient = pow(muCoefficientA,2); + else coefficient = sqrt(muCoefficientA); + coefficient = (coefficient*mewiness)+(muCoefficientA*unmewiness); + inputSampleL *= coefficient; + inputSampleR *= coefficient; + } + else + { + if (positivemu) coefficient = pow(muCoefficientB,2); + else coefficient = sqrt(muCoefficientB); + coefficient = (coefficient*mewiness)+(muCoefficientB*unmewiness); + inputSampleL *= coefficient; + 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 + + if (outputGain != 1.0) { + inputSampleL *= outputGain; + inputSampleR *= outputGain; + } + + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0){inputSampleL = bridgerectifier;} + else {inputSampleL = -bridgerectifier;} + //second stage of overdrive to prevent overs and allow bloody loud extremeness + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0){inputSampleR = bridgerectifier;} + else {inputSampleR = -bridgerectifier;} + //second stage of overdrive to prevent overs and allow bloody loud extremeness + + //noise shaping to 32-bit floating point + if (flip) { + fpTemp = inputSampleL; + fpNShapeAL = (fpNShapeAL*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeAL; + + fpTemp = inputSampleR; + fpNShapeAR = (fpNShapeAR*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeAR; + } + else { + fpTemp = inputSampleL; + fpNShapeBL = (fpNShapeBL*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeBL; + + fpTemp = inputSampleR; + fpNShapeBR = (fpNShapeBR*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeBR; + } + flip = !flip; + //end noise shaping on 32 bit output + + *outputL = inputSampleL; + *outputR = inputSampleR; + + *inputL++; + *inputR++; + *outputL++; + *outputR++; + } +}
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