/* ========================================
 *  ToneSlant - ToneSlant.h
 *  Copyright (c) 2016 airwindows, All rights reserved
 * ======================================== */

#ifndef __ToneSlant_H
#include "ToneSlant.h"
#endif

void ToneSlant::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
{
    float* in1  =  inputs[0];
    float* in2  =  inputs[1];
    float* out1 = outputs[0];
    float* out2 = outputs[1];


	double inputSampleL;
	double inputSampleR;
	double correctionSampleL;
	double correctionSampleR;
	double accumulatorSampleL;
	double accumulatorSampleR;
	double drySampleL;
	double drySampleR;
	double overallscale = (A*99.0)+1.0;
	double applySlant = (B*2.0)-1.0;
	
	
	f[0] = 1.0 / overallscale;
	//count to f(gain) which will be 0. f(0) is x1
	for (int count = 1; count < 102; count++) {
		if (count <= overallscale) {
			f[count] = (1.0 - (count / overallscale)) / overallscale;
			//recalc the filter and don't change the buffer it'll apply to
		} else {
			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
		}
	}
    
    while (--sampleFrames >= 0)
    {
		for (int count = overallscale; count >= 0; count--) {
			bL[count+1] = bL[count];
			bR[count+1] = bR[count];
		}
		
		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.
		}
		
		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
		
		accumulatorSampleL *= f[0];
		accumulatorSampleR *= f[0];

		for (int count = 1; count < overallscale; count++) {
			accumulatorSampleL += (bL[count] * f[count]);
			accumulatorSampleR += (bR[count] * f[count]);
		}
		
		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
		//we're gonna apply the total effect of all these calculations as a single subtract
		
		inputSampleL += (correctionSampleL * applySlant);
		inputSampleR += (correctionSampleR * applySlant);
		//our one math operation on the input data coming in

		//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 = inputSampleL;
		*out2 = inputSampleR;

		*in1++;
		*in2++;
		*out1++;
		*out2++;
    }
}

void ToneSlant::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
{
    double* in1  =  inputs[0];
    double* in2  =  inputs[1];
    double* out1 = outputs[0];
    double* out2 = outputs[1];


	double inputSampleL;
	double inputSampleR;
	double correctionSampleL;
	double correctionSampleR;
	double accumulatorSampleL;
	double accumulatorSampleR;
	double drySampleL;
	double drySampleR;
	double overallscale = (A*99.0)+1.0;
	double applySlant = (B*2.0)-1.0;
	
	f[0] = 1.0 / overallscale;
	//count to f(gain) which will be 0. f(0) is x1
	for (int count = 1; count < 102; count++) {
		if (count <= overallscale) {
			f[count] = (1.0 - (count / overallscale)) / overallscale;
			//recalc the filter and don't change the buffer it'll apply to
		} else {
			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
		}
	}
	
    while (--sampleFrames >= 0)
    {
		for (int count = overallscale; count >= 0; count--) {
			bL[count+1] = bL[count];
			bR[count+1] = bR[count];
		}
		
		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.
		}
		
		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
		
		accumulatorSampleL *= f[0];
		accumulatorSampleR *= f[0];
		
		for (int count = 1; count < overallscale; count++) {
			accumulatorSampleL += (bL[count] * f[count]);
			accumulatorSampleR += (bR[count] * f[count]);
		}
		
		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
		//we're gonna apply the total effect of all these calculations as a single subtract
		
		inputSampleL += (correctionSampleL * applySlant);
		inputSampleR += (correctionSampleR * applySlant);
		//our one math operation on the input data coming in
		
		//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 = inputSampleL;
		*out2 = inputSampleR;

		*in1++;
		*in2++;
		*out1++;
		*out2++;
    }
}