/* ========================================
* Highpass - Highpass.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Highpass_H
#include "Highpass.h"
#endif
void Highpass::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 iirAmount = pow(A,3)/overallscale;
double tight = (B*2.0)-1.0;
double wet = C;
double dry = 1.0 - wet;
double offset;
double inputSampleL;
double inputSampleR;
double outputSampleL;
double outputSampleR;
iirAmount += (iirAmount * tight * tight);
if (tight > 0) tight /= 1.5;
else tight /= 3.0;
//we are setting it up so that to either extreme we can get an audible sound,
//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
if (iirAmount <= 0.0) iirAmount = 0.0;
if (iirAmount > 1.0) iirAmount = 1.0;
//handle the change in cutoff frequency
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.
}
outputSampleL = inputSampleL;
outputSampleR = inputSampleR;
if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip)
{
iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
outputSampleL = outputSampleL - iirSampleAL;
}
else
{
iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
outputSampleL = outputSampleL - iirSampleBL;
}
if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip)
{
iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
outputSampleR = outputSampleR - iirSampleAR;
}
else
{
iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
outputSampleR = outputSampleR - iirSampleBR;
}
fpFlip = !fpFlip;
if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
//stereo 32 bit dither, made small and tidy.
int expon; frexpf((float)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexpf((float)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 32 bit dither
*out1 = outputSampleL;
*out2 = outputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void Highpass::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 iirAmount = pow(A,3)/overallscale;
double tight = (B*2.0)-1.0;
double wet = C;
double dry = 1.0 - wet;
double offset;
double inputSampleL;
double inputSampleR;
double outputSampleL;
double outputSampleR;
iirAmount += (iirAmount * tight * tight);
if (tight > 0) tight /= 1.5;
else tight /= 3.0;
//we are setting it up so that to either extreme we can get an audible sound,
//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
if (iirAmount <= 0.0) iirAmount = 0.0;
if (iirAmount > 1.0) iirAmount = 1.0;
//handle the change in cutoff frequency
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.
}
outputSampleL = inputSampleL;
outputSampleR = inputSampleR;
if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip)
{
iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
outputSampleL = outputSampleL - iirSampleAL;
}
else
{
iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
outputSampleL = outputSampleL - iirSampleBL;
}
if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip)
{
iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
outputSampleR = outputSampleR - iirSampleAR;
}
else
{
iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
outputSampleR = outputSampleR - iirSampleBR;
}
fpFlip = !fpFlip;
if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
//stereo 64 bit dither, made small and tidy.
int expon; frexp((double)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexp((double)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 64 bit dither
*out1 = outputSampleL;
*out2 = outputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
