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Diffstat (limited to 'plugins/MacVST/Point/source/PointProc.cpp')
| -rwxr-xr-x | plugins/MacVST/Point/source/PointProc.cpp | 312 |
1 files changed, 312 insertions, 0 deletions
diff --git a/plugins/MacVST/Point/source/PointProc.cpp b/plugins/MacVST/Point/source/PointProc.cpp new file mode 100755 index 0000000..a00263d --- /dev/null +++ b/plugins/MacVST/Point/source/PointProc.cpp @@ -0,0 +1,312 @@ +/* ======================================== + * Point - Point.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Point_H +#include "Point.h" +#endif + +void Point::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(); + + double gaintrim = pow(10.0,((A*24.0)-12.0)/20); + double nibDiv = 1 / pow(C+0.2,7); + nibDiv /= overallscale; + double nobDiv; + if (((B*2.0)-1.0) > 0) nobDiv = nibDiv / (1.001-((B*2.0)-1.0)); + else nobDiv = nibDiv * (1.001-pow(((B*2.0)-1.0)*0.75,2)); + double nibnobFactor = 0.0; //start with the fallthrough value, why not + double absolute; + + 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 = *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. + } + + inputSampleL *= gaintrim; + absolute = fabs(inputSampleL); + if (fpFlip) + { + nibAL = nibAL + (absolute / nibDiv); + nibAL = nibAL / (1 + (1/nibDiv)); + nobAL = nobAL + (absolute / nobDiv); + nobAL = nobAL / (1 + (1/nobDiv)); + if (nobAL > 0) + { + nibnobFactor = nibAL / nobAL; + } + } + else + { + nibBL = nibBL + (absolute / nibDiv); + nibBL = nibBL / (1 + (1/nibDiv)); + nobBL = nobBL + (absolute / nobDiv); + nobBL = nobBL / (1 + (1/nobDiv)); + if (nobBL > 0) + { + nibnobFactor = nibBL / nobBL; + } + } + inputSampleL *= nibnobFactor; + + + inputSampleR *= gaintrim; + absolute = fabs(inputSampleR); + if (fpFlip) + { + nibAR = nibAR + (absolute / nibDiv); + nibAR = nibAR / (1 + (1/nibDiv)); + nobAR = nobAR + (absolute / nobDiv); + nobAR = nobAR / (1 + (1/nobDiv)); + if (nobAR > 0) + { + nibnobFactor = nibAR / nobAR; + } + } + else + { + nibBR = nibBR + (absolute / nibDiv); + nibBR = nibBR / (1 + (1/nibDiv)); + nobBR = nobBR + (absolute / nobDiv); + nobBR = nobBR / (1 + (1/nobDiv)); + if (nobBR > 0) + { + nibnobFactor = nibBR / nobBR; + } + } + inputSampleR *= nibnobFactor; + + //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 + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Point::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 gaintrim = pow(10.0,((A*24.0)-12.0)/20); + double nibDiv = 1 / pow(C+0.2,7); + nibDiv /= overallscale; + double nobDiv; + if (((B*2.0)-1.0) > 0) nobDiv = nibDiv / (1.001-((B*2.0)-1.0)); + else nobDiv = nibDiv * (1.001-pow(((B*2.0)-1.0)*0.75,2)); + double nibnobFactor = 0.0; //start with the fallthrough value, why not + double absolute; + + 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 = *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. + } + + inputSampleL *= gaintrim; + absolute = fabs(inputSampleL); + if (fpFlip) + { + nibAL = nibAL + (absolute / nibDiv); + nibAL = nibAL / (1 + (1/nibDiv)); + nobAL = nobAL + (absolute / nobDiv); + nobAL = nobAL / (1 + (1/nobDiv)); + if (nobAL > 0) + { + nibnobFactor = nibAL / nobAL; + } + } + else + { + nibBL = nibBL + (absolute / nibDiv); + nibBL = nibBL / (1 + (1/nibDiv)); + nobBL = nobBL + (absolute / nobDiv); + nobBL = nobBL / (1 + (1/nobDiv)); + if (nobBL > 0) + { + nibnobFactor = nibBL / nobBL; + } + } + inputSampleL *= nibnobFactor; + + + inputSampleR *= gaintrim; + absolute = fabs(inputSampleR); + if (fpFlip) + { + nibAR = nibAR + (absolute / nibDiv); + nibAR = nibAR / (1 + (1/nibDiv)); + nobAR = nobAR + (absolute / nobDiv); + nobAR = nobAR / (1 + (1/nobDiv)); + if (nobAR > 0) + { + nibnobFactor = nibAR / nobAR; + } + } + else + { + nibBR = nibBR + (absolute / nibDiv); + nibBR = nibBR / (1 + (1/nibDiv)); + nobBR = nobBR + (absolute / nobDiv); + nobBR = nobBR / (1 + (1/nobDiv)); + if (nobBR > 0) + { + nibnobFactor = nibBR / nobBR; + } + } + inputSampleR *= nibnobFactor; + + //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 + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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