/* ========================================
* Baxandall - Baxandall.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Baxandall_H
#include "Baxandall.h"
#endif
void Baxandall::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0);
double trebleFreq = (4410.0*trebleGain)/getSampleRate();
if (trebleFreq > 0.45) trebleFreq = 0.45;
trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq;
double bassGain = pow(10.0,((B*30.0)-15.0)/20.0);
double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0);
bassFreq = (4410.0*bassFreq)/getSampleRate();
if (bassFreq > 0.45) bassFreq = 0.45;
bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq;
trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4;
bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2;
double output = pow(10.0,((C*30.0)-15.0)/20.0);
double K = tan(M_PI * trebleAL[0]);
double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K);
trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm;
trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2];
trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2];
trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm;
trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm;
K = tan(M_PI * bassAL[0]);
norm = 1.0 / (1.0 + K / bassAL[1] + K * K);
bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm;
bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2];
bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2];
bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm;
bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[1] + K * K) * norm;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
if (output != 1.0) {
inputSampleL *= output;
inputSampleR *= output;
}//gain trim in front of plugin, in case Console stage clips
inputSampleL = sin(inputSampleL);
inputSampleR = sin(inputSampleR);
//encode Console5: good cleanness
long double trebleSampleL;
long double bassSampleL;
long double trebleSampleR;
long double bassSampleR;
if (flip)
{
trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7];
trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8];
trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]);
trebleSampleL = inputSampleL - trebleSampleL;
bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7];
bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8];
bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]);
trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7];
trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8];
trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7];
bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8];
bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]);
}
else
{
trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7];
trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8];
trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]);
trebleSampleL = inputSampleL - trebleSampleL;
bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7];
bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8];
bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]);
trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7];
trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8];
trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7];
bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8];
bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]);
}
flip = !flip;
trebleSampleL *= trebleGain;
bassSampleL *= bassGain;
inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
trebleSampleR *= trebleGain;
bassSampleR *= bassGain;
inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleL = asin(inputSampleL);
//amplitude aspect
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleR = asin(inputSampleR);
//amplitude aspect
//begin 32 bit stereo floating point dither
int expon; frexpf((float)inputSampleL, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
frexpf((float)inputSampleR, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleR += ((double(fpd)-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 Baxandall::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0);
double trebleFreq = (4410.0*trebleGain)/getSampleRate();
if (trebleFreq > 0.45) trebleFreq = 0.45;
trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq;
double bassGain = pow(10.0,((B*30.0)-15.0)/20.0);
double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0);
bassFreq = (4410.0*bassFreq)/getSampleRate();
if (bassFreq > 0.45) bassFreq = 0.45;
bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq;
trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4;
bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2;
double output = pow(10.0,((C*30.0)-15.0)/20.0);
double K = tan(M_PI * trebleAL[0]);
double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K);
trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm;
trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2];
trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2];
trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm;
trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm;
K = tan(M_PI * bassAL[0]);
norm = 1.0 / (1.0 + K / bassAL[1] + K * K);
bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm;
bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2];
bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2];
bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm;
bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[1] + K * K) * norm;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
if (output != 1.0) {
inputSampleL *= output;
inputSampleR *= output;
}//gain trim in front of plugin, in case Console stage clips
inputSampleL = sin(inputSampleL);
inputSampleR = sin(inputSampleR);
//encode Console5: good cleanness
long double trebleSampleL;
long double bassSampleL;
long double trebleSampleR;
long double bassSampleR;
if (flip)
{
trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7];
trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8];
trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]);
trebleSampleL = inputSampleL - trebleSampleL;
bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7];
bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8];
bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]);
trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7];
trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8];
trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7];
bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8];
bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]);
}
else
{
trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7];
trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8];
trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]);
trebleSampleL = inputSampleL - trebleSampleL;
bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7];
bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8];
bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]);
trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7];
trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8];
trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7];
bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8];
bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]);
}
flip = !flip;
trebleSampleL *= trebleGain;
bassSampleL *= bassGain;
inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
trebleSampleR *= trebleGain;
bassSampleR *= bassGain;
inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleL = asin(inputSampleL);
//amplitude aspect
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleR = asin(inputSampleR);
//amplitude aspect
//begin 64 bit stereo floating point dither
int expon; frexp((double)inputSampleL, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
frexp((double)inputSampleR, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleR += ((double(fpd)-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++;
}
}
