mirror of
https://github.com/airwindows/airwindows.git
synced 2026-05-15 22:01:19 -06:00
188 lines
6.9 KiB
C++
Executable file
188 lines
6.9 KiB
C++
Executable file
/* ========================================
|
|
* Shape - Shape.h
|
|
* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
|
|
* ======================================== */
|
|
|
|
#ifndef __Gain_H
|
|
#include "Shape.h"
|
|
#endif
|
|
|
|
void Shape::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
|
|
{
|
|
float* in1 = inputs[0];
|
|
float* in2 = inputs[1];
|
|
float* out1 = outputs[0];
|
|
float* out2 = outputs[1];
|
|
|
|
double shape = -((A*2.0)-1.0);
|
|
double gainstage = fabs(shape)+0.01; //no divide by zero
|
|
double offset = (B*2.0)-1.0;
|
|
double postOffset = 0.0;
|
|
if (shape > 0) {
|
|
gainstage += 0.99;
|
|
postOffset = sin(offset);
|
|
}
|
|
if (shape < 0) postOffset = asin(offset);
|
|
double cutoff = 25000.0 / getSampleRate();
|
|
if (cutoff > 0.49) cutoff = 0.49; //don't crash if run at 44.1k
|
|
fixA[fix_freq] = cutoff;
|
|
fixA[fix_reso] = 0.70710678; //butterworth Q
|
|
double K = tan(M_PI * fixA[fix_freq]); //lowpass
|
|
double norm = 1.0 / (1.0 + K / fixA[fix_reso] + K * K);
|
|
fixA[fix_a0] = K * K * norm;
|
|
fixA[fix_a1] = 2.0 * fixA[fix_a0];
|
|
fixA[fix_a2] = fixA[fix_a0];
|
|
fixA[fix_b1] = 2.0 * (K * K - 1.0) * norm;
|
|
fixA[fix_b2] = (1.0 - K / fixA[fix_reso] + K * K) * norm;
|
|
fixA[fix_sL1] = 0.0;
|
|
fixA[fix_sL2] = 0.0;
|
|
fixA[fix_sR1] = 0.0;
|
|
fixA[fix_sR2] = 0.0;
|
|
//define filters here: on VST you can't define them in reset 'cos getSampleRate isn't returning good information yet
|
|
|
|
while (--sampleFrames >= 0)
|
|
{
|
|
double inputSampleL = *in1;
|
|
double inputSampleR = *in2;
|
|
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
|
|
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
|
|
double drySampleL = inputSampleL;
|
|
double drySampleR = inputSampleR;
|
|
|
|
double outSample = (inputSampleL * fixA[fix_a0]) + fixA[fix_sL1];
|
|
fixA[fix_sL1] = (inputSampleL * fixA[fix_a1]) - (outSample * fixA[fix_b1]) + fixA[fix_sL2];
|
|
fixA[fix_sL2] = (inputSampleL * fixA[fix_a2]) - (outSample * fixA[fix_b2]);
|
|
inputSampleL = outSample; //fixed biquad filtering ultrasonics
|
|
|
|
inputSampleL *= gainstage;
|
|
inputSampleL += offset;
|
|
if (inputSampleL > 1.0) inputSampleL = 1.0;
|
|
if (inputSampleL < -1.0) inputSampleL = -1.0;
|
|
if (shape > 0) inputSampleL = sin(inputSampleL);
|
|
if (shape < 0) inputSampleL = asin(inputSampleL);
|
|
inputSampleL -= postOffset;
|
|
inputSampleL = ((inputSampleL/gainstage)*fabs(shape))+(drySampleL*(1.0-fabs(shape)));
|
|
|
|
|
|
outSample = (inputSampleR * fixA[fix_a0]) + fixA[fix_sR1];
|
|
fixA[fix_sR1] = (inputSampleR * fixA[fix_a1]) - (outSample * fixA[fix_b1]) + fixA[fix_sR2];
|
|
fixA[fix_sR2] = (inputSampleR * fixA[fix_a2]) - (outSample * fixA[fix_b2]);
|
|
inputSampleR = outSample; //fixed biquad filtering ultrasonics
|
|
|
|
inputSampleR *= gainstage;
|
|
inputSampleR += offset;
|
|
if (inputSampleR > 1.0) inputSampleR = 1.0;
|
|
if (inputSampleR < -1.0) inputSampleR = -1.0;
|
|
if (shape > 0) inputSampleR = sin(inputSampleR);
|
|
if (shape < 0) inputSampleR = asin(inputSampleR);
|
|
inputSampleR -= postOffset;
|
|
inputSampleR = ((inputSampleR/gainstage)*fabs(shape))+(drySampleR*(1.0-fabs(shape)));
|
|
|
|
//begin 32 bit stereo floating point dither
|
|
int expon; frexpf((float)inputSampleL, &expon);
|
|
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
|
|
inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
|
|
frexpf((float)inputSampleR, &expon);
|
|
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
|
|
inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
|
|
//end 32 bit stereo floating point dither
|
|
|
|
*out1 = inputSampleL;
|
|
*out2 = inputSampleR;
|
|
|
|
in1++;
|
|
in2++;
|
|
out1++;
|
|
out2++;
|
|
}
|
|
}
|
|
|
|
void Shape::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
|
|
{
|
|
double* in1 = inputs[0];
|
|
double* in2 = inputs[1];
|
|
double* out1 = outputs[0];
|
|
double* out2 = outputs[1];
|
|
|
|
double shape = -((A*2.0)-1.0);
|
|
double gainstage = fabs(shape)+0.01; //no divide by zero
|
|
double offset = (B*2.0)-1.0;
|
|
double postOffset = 0.0;
|
|
if (shape > 0) {
|
|
gainstage += 0.99;
|
|
postOffset = sin(offset);
|
|
}
|
|
if (shape < 0) postOffset = asin(offset);
|
|
double cutoff = 25000.0 / getSampleRate();
|
|
if (cutoff > 0.49) cutoff = 0.49; //don't crash if run at 44.1k
|
|
fixA[fix_freq] = cutoff;
|
|
fixA[fix_reso] = 0.70710678; //butterworth Q
|
|
double K = tan(M_PI * fixA[fix_freq]); //lowpass
|
|
double norm = 1.0 / (1.0 + K / fixA[fix_reso] + K * K);
|
|
fixA[fix_a0] = K * K * norm;
|
|
fixA[fix_a1] = 2.0 * fixA[fix_a0];
|
|
fixA[fix_a2] = fixA[fix_a0];
|
|
fixA[fix_b1] = 2.0 * (K * K - 1.0) * norm;
|
|
fixA[fix_b2] = (1.0 - K / fixA[fix_reso] + K * K) * norm;
|
|
fixA[fix_sL1] = 0.0;
|
|
fixA[fix_sL2] = 0.0;
|
|
fixA[fix_sR1] = 0.0;
|
|
fixA[fix_sR2] = 0.0;
|
|
//define filters here: on VST you can't define them in reset 'cos getSampleRate isn't returning good information yet
|
|
|
|
while (--sampleFrames >= 0)
|
|
{
|
|
double inputSampleL = *in1;
|
|
double inputSampleR = *in2;
|
|
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
|
|
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
|
|
double drySampleL = inputSampleL;
|
|
double drySampleR = inputSampleR;
|
|
|
|
double outSample = (inputSampleL * fixA[fix_a0]) + fixA[fix_sL1];
|
|
fixA[fix_sL1] = (inputSampleL * fixA[fix_a1]) - (outSample * fixA[fix_b1]) + fixA[fix_sL2];
|
|
fixA[fix_sL2] = (inputSampleL * fixA[fix_a2]) - (outSample * fixA[fix_b2]);
|
|
inputSampleL = outSample; //fixed biquad filtering ultrasonics
|
|
|
|
inputSampleL *= gainstage;
|
|
inputSampleL += offset;
|
|
if (inputSampleL > 1.0) inputSampleL = 1.0;
|
|
if (inputSampleL < -1.0) inputSampleL = -1.0;
|
|
if (shape > 0) inputSampleL = sin(inputSampleL);
|
|
if (shape < 0) inputSampleL = asin(inputSampleL);
|
|
inputSampleL -= postOffset;
|
|
inputSampleL = ((inputSampleL/gainstage)*fabs(shape))+(drySampleL*(1.0-fabs(shape)));
|
|
|
|
|
|
outSample = (inputSampleR * fixA[fix_a0]) + fixA[fix_sR1];
|
|
fixA[fix_sR1] = (inputSampleR * fixA[fix_a1]) - (outSample * fixA[fix_b1]) + fixA[fix_sR2];
|
|
fixA[fix_sR2] = (inputSampleR * fixA[fix_a2]) - (outSample * fixA[fix_b2]);
|
|
inputSampleR = outSample; //fixed biquad filtering ultrasonics
|
|
|
|
inputSampleR *= gainstage;
|
|
inputSampleR += offset;
|
|
if (inputSampleR > 1.0) inputSampleR = 1.0;
|
|
if (inputSampleR < -1.0) inputSampleR = -1.0;
|
|
if (shape > 0) inputSampleR = sin(inputSampleR);
|
|
if (shape < 0) inputSampleR = asin(inputSampleR);
|
|
inputSampleR -= postOffset;
|
|
inputSampleR = ((inputSampleR/gainstage)*fabs(shape))+(drySampleR*(1.0-fabs(shape)));
|
|
|
|
//begin 64 bit stereo floating point dither
|
|
//int expon; frexp((double)inputSampleL, &expon);
|
|
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
|
|
//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
|
|
//frexp((double)inputSampleR, &expon);
|
|
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
|
|
//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
|
|
//end 64 bit stereo floating point dither
|
|
|
|
*out1 = inputSampleL;
|
|
*out2 = inputSampleR;
|
|
|
|
in1++;
|
|
in2++;
|
|
out1++;
|
|
out2++;
|
|
}
|
|
}
|