/* ========================================
 *  Baxandall - Baxandall.h
 *  Copyright (c) 2016 airwindows, Airwindows uses the MIT license
 * ======================================== */

#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 = (8820.0*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)
    {
		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;
		
		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
		
		double trebleSampleL;
		double bassSampleL;
		double trebleSampleR;
		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);
		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 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 = (8820.0*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)
    {
		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;
		
		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
		
		double trebleSampleL;
		double bassSampleL;
		double trebleSampleR;
		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);
		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++;
    }
}
