mirror of
https://github.com/airwindows/airwindows.git
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228 lines
7.4 KiB
C++
Executable file
228 lines
7.4 KiB
C++
Executable file
/* ========================================
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* uLawEncode - uLawEncode.h
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* Copyright (c) 2016 airwindows, All rights reserved
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* ======================================== */
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#ifndef __uLawEncode_H
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#include "uLawEncode.h"
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#endif
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void uLawEncode::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
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{
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float* in1 = inputs[0];
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float* in2 = inputs[1];
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float* out1 = outputs[0];
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float* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double gain = A;
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double wet = B;
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double dry = 1.0 - wet;
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while (--sampleFrames >= 0)
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{
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long double inputSampleL = *in1;
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long double inputSampleR = *in2;
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static int noisesourceL = 0;
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static int noisesourceR = 850010;
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int residue;
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double applyresidue;
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noisesourceL = noisesourceL % 1700021; noisesourceL++;
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residue = noisesourceL * noisesourceL;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleL += applyresidue;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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inputSampleL -= applyresidue;
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}
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noisesourceR = noisesourceR % 1700021; noisesourceR++;
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residue = noisesourceR * noisesourceR;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleR += applyresidue;
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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inputSampleR -= applyresidue;
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}
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//for live air, we always apply the dither noise. Then, if our result is
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//effectively digital black, we'll subtract it auLawEncode. We want a 'air' hiss
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double drySampleL = inputSampleL;
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double drySampleR = inputSampleR;
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if (gain != 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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}
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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if (inputSampleL > 0) inputSampleL = log(1.0+(255*fabs(inputSampleL))) / log(256);
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if (inputSampleL < 0) inputSampleL = -log(1.0+(255*fabs(inputSampleL))) / log(256);
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if (inputSampleR > 0) inputSampleR = log(1.0+(255*fabs(inputSampleR))) / log(256);
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if (inputSampleR < 0) inputSampleR = -log(1.0+(255*fabs(inputSampleR))) / log(256);
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if (wet !=1.0) {
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inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
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inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
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}
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//noise shaping to 32-bit floating point
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float fpTemp = inputSampleL;
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fpNShapeL += (inputSampleL-fpTemp);
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inputSampleL += fpNShapeL;
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//if this confuses you look at the wordlength for fpTemp :)
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fpTemp = inputSampleR;
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fpNShapeR += (inputSampleR-fpTemp);
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inputSampleR += fpNShapeR;
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//for deeper space and warmth, we try a non-oscillating noise shaping
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//that is kind of ruthless: it will forever retain the rounding errors
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//except we'll dial it back a hair at the end of every buffer processed
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//end noise shaping on 32 bit output
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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*in1++;
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*in2++;
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*out1++;
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*out2++;
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}
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fpNShapeL *= 0.999999;
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fpNShapeR *= 0.999999;
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//we will just delicately dial back the FP noise shaping, not even every sample
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//this is a good place to put subtle 'no runaway' calculations, though bear in mind
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//that it will be called more often when you use shorter sample buffers in the DAW.
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//So, very low latency operation will call these calculations more often.
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}
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void uLawEncode::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
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{
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double gain = A;
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double wet = B;
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double dry = 1.0 - wet;
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while (--sampleFrames >= 0)
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{
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long double inputSampleL = *in1;
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long double inputSampleR = *in2;
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static int noisesourceL = 0;
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static int noisesourceR = 850010;
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int residue;
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double applyresidue;
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noisesourceL = noisesourceL % 1700021; noisesourceL++;
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residue = noisesourceL * noisesourceL;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleL += applyresidue;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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inputSampleL -= applyresidue;
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}
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noisesourceR = noisesourceR % 1700021; noisesourceR++;
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residue = noisesourceR * noisesourceR;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleR += applyresidue;
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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inputSampleR -= applyresidue;
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}
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//for live air, we always apply the dither noise. Then, if our result is
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//effectively digital black, we'll subtract it auLawEncode. We want a 'air' hiss
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double drySampleL = inputSampleL;
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double drySampleR = inputSampleR;
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if (gain != 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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}
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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if (inputSampleL > 0) inputSampleL = log(1.0+(255*fabs(inputSampleL))) / log(256);
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if (inputSampleL < 0) inputSampleL = -log(1.0+(255*fabs(inputSampleL))) / log(256);
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if (inputSampleR > 0) inputSampleR = log(1.0+(255*fabs(inputSampleR))) / log(256);
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if (inputSampleR < 0) inputSampleR = -log(1.0+(255*fabs(inputSampleR))) / log(256);
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if (wet !=1.0) {
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inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
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inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
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}
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//noise shaping to 64-bit floating point
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double fpTemp = inputSampleL;
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fpNShapeL += (inputSampleL-fpTemp);
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inputSampleL += fpNShapeL;
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//if this confuses you look at the wordlength for fpTemp :)
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fpTemp = inputSampleR;
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fpNShapeR += (inputSampleR-fpTemp);
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inputSampleR += fpNShapeR;
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//for deeper space and warmth, we try a non-oscillating noise shaping
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//that is kind of ruthless: it will forever retain the rounding errors
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//except we'll dial it back a hair at the end of every buffer processed
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//end noise shaping on 64 bit output
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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*in1++;
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*in2++;
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*out1++;
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*out2++;
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}
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fpNShapeL *= 0.999999;
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fpNShapeR *= 0.999999;
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//we will just delicately dial back the FP noise shaping, not even every sample
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//this is a good place to put subtle 'no runaway' calculations, though bear in mind
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//that it will be called more often when you use shorter sample buffers in the DAW.
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//So, very low latency operation will call these calculations more often.
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}
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