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Diffstat (limited to 'plugins/MacVST/uLawDecode/source/uLawDecodeProc.cpp')
| -rwxr-xr-x | plugins/MacVST/uLawDecode/source/uLawDecodeProc.cpp | 228 |
1 files changed, 228 insertions, 0 deletions
diff --git a/plugins/MacVST/uLawDecode/source/uLawDecodeProc.cpp b/plugins/MacVST/uLawDecode/source/uLawDecodeProc.cpp new file mode 100755 index 0000000..a78b3c2 --- /dev/null +++ b/plugins/MacVST/uLawDecode/source/uLawDecodeProc.cpp @@ -0,0 +1,228 @@ +/* ======================================== + * uLawDecode - uLawDecode.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __uLawDecode_H +#include "uLawDecode.h" +#endif + +void uLawDecode::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 gain = A; + double wet = B; + double dry = 1.0 - wet; + + 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 again. We want a 'air' hiss + double drySampleL = inputSampleL; + double drySampleR = inputSampleR; + + if (gain != 1.0) { + inputSampleL *= gain; + inputSampleR *= gain; + } + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + + if (inputSampleL > 0) inputSampleL = (pow(256,fabs(inputSampleL))-1.0) / 255; + if (inputSampleL < 0) inputSampleL = -(pow(256,fabs(inputSampleL))-1.0) / 255; + + if (inputSampleR > 0) inputSampleR = (pow(256,fabs(inputSampleR))-1.0) / 255; + if (inputSampleR < 0) inputSampleR = -(pow(256,fabs(inputSampleR))-1.0) / 255; + + if (wet != 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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 uLawDecode::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 gain = A; + double wet = B; + double dry = 1.0 - wet; + + 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 again. We want a 'air' hiss + double drySampleL = inputSampleL; + double drySampleR = inputSampleR; + + if (gain != 1.0) { + inputSampleL *= gain; + inputSampleR *= gain; + } + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + + if (inputSampleL > 0) inputSampleL = (pow(256,fabs(inputSampleL))-1.0) / 255; + if (inputSampleL < 0) inputSampleL = -(pow(256,fabs(inputSampleL))-1.0) / 255; + + if (inputSampleR > 0) inputSampleR = (pow(256,fabs(inputSampleR))-1.0) / 255; + if (inputSampleR < 0) inputSampleR = -(pow(256,fabs(inputSampleR))-1.0) / 255; + + if (wet != 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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. +} |