diff options
Diffstat (limited to 'plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp')
-rwxr-xr-x | plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp | 271 |
1 files changed, 271 insertions, 0 deletions
diff --git a/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp b/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp new file mode 100755 index 0000000..b594b05 --- /dev/null +++ b/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp @@ -0,0 +1,271 @@ +/* ======================================== + * ToneSlant - ToneSlant.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __ToneSlant_H +#include "ToneSlant.h" +#endif + +void ToneSlant::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputSampleL; + double inputSampleR; + double correctionSampleL; + double correctionSampleR; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double overallscale = (A*99.0)+1.0; + double applySlant = (B*2.0)-1.0; + + + f[0] = 1.0 / overallscale; + //count to f(gain) which will be 0. f(0) is x1 + for (int count = 1; count < 102; count++) { + if (count <= overallscale) { + f[count] = (1.0 - (count / overallscale)) / overallscale; + //recalc the filter and don't change the buffer it'll apply to + } else { + bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops + bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops + } + } + + while (--sampleFrames >= 0) + { + for (int count = overallscale; count >= 0; count--) { + bL[count+1] = bL[count]; + bR[count+1] = bR[count]; + } + + 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. + } + + bL[0] = accumulatorSampleL = drySampleL = inputSampleL; + bR[0] = accumulatorSampleR = drySampleR = inputSampleR; + + accumulatorSampleL *= f[0]; + accumulatorSampleR *= f[0]; + + for (int count = 1; count < overallscale; count++) { + accumulatorSampleL += (bL[count] * f[count]); + accumulatorSampleR += (bR[count] * f[count]); + } + + correctionSampleL = inputSampleL - (accumulatorSampleL*2.0); + correctionSampleR = inputSampleR - (accumulatorSampleR*2.0); + //we're gonna apply the total effect of all these calculations as a single subtract + + inputSampleL += (correctionSampleL * applySlant); + inputSampleR += (correctionSampleR * applySlant); + //our one math operation on the input data coming in + + //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 ToneSlant::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double fpTemp; //this is different from singlereplacing + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputSampleL; + double inputSampleR; + double correctionSampleL; + double correctionSampleR; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double overallscale = (A*99.0)+1.0; + double applySlant = (B*2.0)-1.0; + + f[0] = 1.0 / overallscale; + //count to f(gain) which will be 0. f(0) is x1 + for (int count = 1; count < 102; count++) { + if (count <= overallscale) { + f[count] = (1.0 - (count / overallscale)) / overallscale; + //recalc the filter and don't change the buffer it'll apply to + } else { + bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops + bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops + } + } + + while (--sampleFrames >= 0) + { + for (int count = overallscale; count >= 0; count--) { + bL[count+1] = bL[count]; + bR[count+1] = bR[count]; + } + + 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. + } + + bL[0] = accumulatorSampleL = drySampleL = inputSampleL; + bR[0] = accumulatorSampleR = drySampleR = inputSampleR; + + accumulatorSampleL *= f[0]; + accumulatorSampleR *= f[0]; + + for (int count = 1; count < overallscale; count++) { + accumulatorSampleL += (bL[count] * f[count]); + accumulatorSampleR += (bR[count] * f[count]); + } + + correctionSampleL = inputSampleL - (accumulatorSampleL*2.0); + correctionSampleR = inputSampleR - (accumulatorSampleR*2.0); + //we're gonna apply the total effect of all these calculations as a single subtract + + inputSampleL += (correctionSampleL * applySlant); + inputSampleR += (correctionSampleR * applySlant); + //our one math operation on the input data coming in + + //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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