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https://github.com/airwindows/airwindows.git
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270 lines
No EOL
9.6 KiB
C++
Executable file
270 lines
No EOL
9.6 KiB
C++
Executable file
/* ========================================
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* Highpass - Highpass.h
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* Copyright (c) 2016 airwindows, All rights reserved
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* ======================================== */
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#ifndef __Highpass_H
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#include "Highpass.h"
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#endif
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void Highpass::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 iirAmount = pow(A,3)/overallscale;
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double tight = (B*2.0)-1.0;
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double wet = C;
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double dry = 1.0 - wet;
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double offset;
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double inputSampleL;
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double inputSampleR;
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double outputSampleL;
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double outputSampleR;
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iirAmount += (iirAmount * tight * tight);
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if (tight > 0) tight /= 1.5;
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else tight /= 3.0;
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//we are setting it up so that to either extreme we can get an audible sound,
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//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
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if (iirAmount <= 0.0) iirAmount = 0.0;
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if (iirAmount > 1.0) iirAmount = 1.0;
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//handle the change in cutoff frequency
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while (--sampleFrames >= 0)
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{
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inputSampleL = *in1;
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inputSampleR = *in2;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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static int noisesource = 0;
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//this declares a variable before anything else is compiled. It won't keep assigning
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//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
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//but it lets me add this denormalization fix in a single place rather than updating
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//it in three different locations. The variable isn't thread-safe but this is only
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//a random seed and we can share it with whatever.
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noisesource = noisesource % 1700021; noisesource++;
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int residue = noisesource * noisesource;
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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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double 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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}
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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static int noisesource = 0;
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noisesource = noisesource % 1700021; noisesource++;
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int residue = noisesource * noisesource;
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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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double 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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//this denormalization routine produces a white noise at -300 dB which the noise
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//shaping will interact with to produce a bipolar output, but the noise is actually
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//all positive. That should stop any variables from going denormal, and the routine
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//only kicks in if digital black is input. As a final touch, if you save to 24-bit
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//the silence will return to being digital black again.
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}
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outputSampleL = inputSampleL;
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outputSampleR = inputSampleR;
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if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
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else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
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if (offset < 0) offset = 0;
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if (offset > 1) offset = 1;
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if (fpFlip)
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{
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iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
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outputSampleL = outputSampleL - iirSampleAL;
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}
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else
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{
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iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
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outputSampleL = outputSampleL - iirSampleBL;
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}
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if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
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else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
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if (offset < 0) offset = 0;
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if (offset > 1) offset = 1;
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if (fpFlip)
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{
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iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
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outputSampleR = outputSampleR - iirSampleAR;
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}
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else
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{
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iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
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outputSampleR = outputSampleR - iirSampleBR;
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}
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fpFlip = !fpFlip;
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if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
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if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
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//stereo 32 bit dither, made small and tidy.
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int expon; frexpf((float)inputSampleL, &expon);
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long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
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inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
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frexpf((float)inputSampleR, &expon);
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dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
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inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
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//end 32 bit dither
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*out1 = outputSampleL;
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*out2 = outputSampleR;
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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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}
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void Highpass::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 iirAmount = pow(A,3)/overallscale;
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double tight = (B*2.0)-1.0;
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double wet = C;
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double dry = 1.0 - wet;
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double offset;
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double inputSampleL;
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double inputSampleR;
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double outputSampleL;
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double outputSampleR;
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iirAmount += (iirAmount * tight * tight);
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if (tight > 0) tight /= 1.5;
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else tight /= 3.0;
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//we are setting it up so that to either extreme we can get an audible sound,
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//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
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if (iirAmount <= 0.0) iirAmount = 0.0;
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if (iirAmount > 1.0) iirAmount = 1.0;
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//handle the change in cutoff frequency
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while (--sampleFrames >= 0)
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{
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inputSampleL = *in1;
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inputSampleR = *in2;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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static int noisesource = 0;
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//this declares a variable before anything else is compiled. It won't keep assigning
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//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
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//but it lets me add this denormalization fix in a single place rather than updating
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//it in three different locations. The variable isn't thread-safe but this is only
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//a random seed and we can share it with whatever.
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noisesource = noisesource % 1700021; noisesource++;
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int residue = noisesource * noisesource;
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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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double 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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}
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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static int noisesource = 0;
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noisesource = noisesource % 1700021; noisesource++;
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int residue = noisesource * noisesource;
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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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double 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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//this denormalization routine produces a white noise at -300 dB which the noise
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//shaping will interact with to produce a bipolar output, but the noise is actually
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//all positive. That should stop any variables from going denormal, and the routine
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//only kicks in if digital black is input. As a final touch, if you save to 24-bit
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//the silence will return to being digital black again.
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}
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outputSampleL = inputSampleL;
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outputSampleR = inputSampleR;
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if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
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else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
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if (offset < 0) offset = 0;
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if (offset > 1) offset = 1;
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if (fpFlip)
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{
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iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
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outputSampleL = outputSampleL - iirSampleAL;
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}
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else
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{
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iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
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outputSampleL = outputSampleL - iirSampleBL;
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}
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if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
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else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
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if (offset < 0) offset = 0;
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if (offset > 1) offset = 1;
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if (fpFlip)
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{
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iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
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outputSampleR = outputSampleR - iirSampleAR;
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}
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else
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{
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iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
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outputSampleR = outputSampleR - iirSampleBR;
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}
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fpFlip = !fpFlip;
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if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
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if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
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//stereo 64 bit dither, made small and tidy.
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int expon; frexp((double)inputSampleL, &expon);
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long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
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dither /= 536870912.0; //needs this to scale to 64 bit zone
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inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
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frexp((double)inputSampleR, &expon);
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dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
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dither /= 536870912.0; //needs this to scale to 64 bit zone
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inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
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//end 64 bit dither
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*out1 = outputSampleL;
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*out2 = outputSampleR;
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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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} |