diff options
Diffstat (limited to 'plugins/MacVST/OneCornerClip/source/OneCornerClipProc.cpp')
| -rwxr-xr-x | plugins/MacVST/OneCornerClip/source/OneCornerClipProc.cpp | 383 |
1 files changed, 383 insertions, 0 deletions
diff --git a/plugins/MacVST/OneCornerClip/source/OneCornerClipProc.cpp b/plugins/MacVST/OneCornerClip/source/OneCornerClipProc.cpp new file mode 100755 index 0000000..0398b60 --- /dev/null +++ b/plugins/MacVST/OneCornerClip/source/OneCornerClipProc.cpp @@ -0,0 +1,383 @@ +/* ======================================== + * OneCornerClip - OneCornerClip.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __OneCornerClip_H +#include "OneCornerClip.h" +#endif + +void OneCornerClip::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputGain = pow(10.0,(((A*36.0)-12.0)/20.0)); + double posThreshold = B; + double posTargetL = posThreshold; + double posTargetR = posThreshold; + double negThreshold = -C; + double negTargetL = negThreshold; + double negTargetR = negThreshold; + double voicing = D; + if (voicing == 0.618) voicing = 0.618033988749894848204586; + //special case: we will do a perfect golden ratio as the default 0.618 + //just 'cos magic universality sauce (seriously, it seems a sweetspot) + if (overallscale > 0.0) voicing /= overallscale; + //translate to desired sample rate, 44.1K is the base + if (voicing < 0.0) voicing = 0.0; + if (voicing > 1.0) voicing = 1.0; + //some insanity checking + double inverseHardness = 1.0 - voicing; + bool clipEngage = false; + + double wet = E; + double dry = 1.0 - wet; + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (inputGain != 1.0) + { + inputSampleL *= inputGain; + inputSampleR *= inputGain; + clipEngage = true; + //if we are altering gain we will always process + } + else + { + clipEngage = false; + //if we are not touching gain, we will bypass unless + //a clip is actively being softened. + } + + + if (inputSampleL > posTargetL) + { + inputSampleL = (lastSampleL * voicing) + (posThreshold * inverseHardness); + posTargetL = inputSampleL; + clipEngage = true; + } + else + { + posTargetL = posThreshold; + } + + if (inputSampleR > posTargetR) + { + inputSampleR = (lastSampleR * voicing) + (posThreshold * inverseHardness); + posTargetR = inputSampleR; + clipEngage = true; + } + else + { + posTargetR = posThreshold; + } + + if (inputSampleL < negTargetL) + { + inputSampleL = (lastSampleL * voicing) + (negThreshold * inverseHardness); + negTargetL = inputSampleL; + clipEngage = true; + } + else { + negTargetL = negThreshold; + } + + if (inputSampleR < negTargetR) + { + inputSampleR = (lastSampleR * voicing) + (negThreshold * inverseHardness); + negTargetR = inputSampleR; + clipEngage = true; + } + else { + negTargetR = negThreshold; + } + + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + if (clipEngage == false) + { + inputSampleL = *in1; + inputSampleR = *in2; + } + //fall back to raw passthrough if at all possible + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void OneCornerClip::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputGain = pow(10.0,(((A*36.0)-12.0)/20.0)); + double posThreshold = B; + double posTargetL = posThreshold; + double posTargetR = posThreshold; + double negThreshold = -C; + double negTargetL = negThreshold; + double negTargetR = negThreshold; + double voicing = D; + if (voicing == 0.618) voicing = 0.618033988749894848204586; + //special case: we will do a perfect golden ratio as the default 0.618 + //just 'cos magic universality sauce (seriously, it seems a sweetspot) + if (overallscale > 0.0) voicing /= overallscale; + //translate to desired sample rate, 44.1K is the base + if (voicing < 0.0) voicing = 0.0; + if (voicing > 1.0) voicing = 1.0; + //some insanity checking + double inverseHardness = 1.0 - voicing; + bool clipEngage = false; + + double wet = E; + double dry = 1.0 - wet; + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (inputGain != 1.0) + { + inputSampleL *= inputGain; + inputSampleR *= inputGain; + clipEngage = true; + //if we are altering gain we will always process + } + else + { + clipEngage = false; + //if we are not touching gain, we will bypass unless + //a clip is actively being softened. + } + + + if (inputSampleL > posTargetL) + { + inputSampleL = (lastSampleL * voicing) + (posThreshold * inverseHardness); + posTargetL = inputSampleL; + clipEngage = true; + } + else + { + posTargetL = posThreshold; + } + + if (inputSampleR > posTargetR) + { + inputSampleR = (lastSampleR * voicing) + (posThreshold * inverseHardness); + posTargetR = inputSampleR; + clipEngage = true; + } + else + { + posTargetR = posThreshold; + } + + if (inputSampleL < negTargetL) + { + inputSampleL = (lastSampleL * voicing) + (negThreshold * inverseHardness); + negTargetL = inputSampleL; + clipEngage = true; + } + else { + negTargetL = negThreshold; + } + + if (inputSampleR < negTargetR) + { + inputSampleR = (lastSampleR * voicing) + (negThreshold * inverseHardness); + negTargetR = inputSampleR; + clipEngage = true; + } + else { + negTargetR = negThreshold; + } + + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //noise shaping to 64-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 64 bit output + + if (clipEngage == false) + { + inputSampleL = *in1; + inputSampleR = *in2; + } + //fall back to raw passthrough if at all possible + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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