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

#ifndef __Sweeten_H
#include "Sweeten.h"
#endif

void Sweeten::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();
	int cycleEnd = floor(overallscale);
	if (cycleEnd < 1) cycleEnd = 1;
	if (cycleEnd > 4) cycleEnd = 4;
	//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
	
	int sweetBits = 10-floor(A*10.0);
	double sweet = 1.0;
	switch (sweetBits)
	{
		case 11: sweet = 0.00048828125; break;
		case 10: sweet = 0.0009765625; break;
		case 9: sweet = 0.001953125; break;
		case 8: sweet = 0.00390625; break;
		case 7: sweet = 0.0078125; break;
		case 6: sweet = 0.015625; break;
		case 5: sweet = 0.03125; break;
		case 4: sweet = 0.0625; break;
		case 3: sweet = 0.125; break;
		case 2: sweet = 0.25; break;
		case 1: sweet = 0.5; break;
		case 0: sweet = 1.0; break;
		case -1: sweet = 2.0; break;
	} //now we have our input trim
		
    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 sweetSample = inputSampleL;
		double sv = sweetSample; sweetSample = (sweetSample + savg[0]) * 0.5; savg[0] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[1]) * 0.5; savg[1] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[2]) * 0.5; savg[2] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[3]) * 0.5; savg[3] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
		sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
		sv = sweetSample; sweetSample = (sweetSample + savg[4]) * 0.5; savg[4] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[5]) * 0.5; savg[5] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[6]) * 0.5; savg[6] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[7]) * 0.5; savg[7] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
		inputSampleL -= sweetSample; //apply the filtered second harmonic correction
		
		sweetSample = inputSampleR;
		sv = sweetSample; sweetSample = (sweetSample + savg[8]) * 0.5; savg[8] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[9]) * 0.5; savg[9] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[10]) * 0.5; savg[10] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[11]) * 0.5; savg[11] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
		sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
		sv = sweetSample; sweetSample = (sweetSample + savg[12]) * 0.5; savg[12] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[13]) * 0.5; savg[13] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[14]) * 0.5; savg[14] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[15]) * 0.5; savg[15] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
		inputSampleR -= sweetSample; //apply the filtered second harmonic correction
		
		//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 Sweeten::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();
	int cycleEnd = floor(overallscale);
	if (cycleEnd < 1) cycleEnd = 1;
	if (cycleEnd > 4) cycleEnd = 4;
	//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
	
	int sweetBits = 10-floor(A*10.0);
	double sweet = 1.0;
	switch (sweetBits)
	{
		case 11: sweet = 0.00048828125; break;
		case 10: sweet = 0.0009765625; break;
		case 9: sweet = 0.001953125; break;
		case 8: sweet = 0.00390625; break;
		case 7: sweet = 0.0078125; break;
		case 6: sweet = 0.015625; break;
		case 5: sweet = 0.03125; break;
		case 4: sweet = 0.0625; break;
		case 3: sweet = 0.125; break;
		case 2: sweet = 0.25; break;
		case 1: sweet = 0.5; break;
		case 0: sweet = 1.0; break;
		case -1: sweet = 2.0; break;
	} //now we have our input trim
	
    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 sweetSample = inputSampleL;
		double sv = sweetSample; sweetSample = (sweetSample + savg[0]) * 0.5; savg[0] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[1]) * 0.5; savg[1] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[2]) * 0.5; savg[2] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[3]) * 0.5; savg[3] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
		sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
		sv = sweetSample; sweetSample = (sweetSample + savg[4]) * 0.5; savg[4] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[5]) * 0.5; savg[5] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[6]) * 0.5; savg[6] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[7]) * 0.5; savg[7] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
		inputSampleL -= sweetSample; //apply the filtered second harmonic correction
		
		sweetSample = inputSampleR;
		sv = sweetSample; sweetSample = (sweetSample + savg[8]) * 0.5; savg[8] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[9]) * 0.5; savg[9] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[10]) * 0.5; savg[10] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[11]) * 0.5; savg[11] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
		sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
		sv = sweetSample; sweetSample = (sweetSample + savg[12]) * 0.5; savg[12] = sv;
		if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[13]) * 0.5; savg[13] = sv;
			if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[14]) * 0.5; savg[14] = sv;
				if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[15]) * 0.5; savg[15] = sv;}
			} //if undersampling code is present, this gives a simple averaging that stacks up
		} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
		inputSampleR -= sweetSample; //apply the filtered second harmonic correction
		
		//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++;
    }
}