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

#ifndef __Lowpass_H
#include "Lowpass.h"
#endif

void Lowpass::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();
	double iirAmount = (pow(A,2)+A)/2.0;
	iirAmount /= overallscale;
	double tight = (B*2.0)-1.0;
	double wet = C;
	double dry = 1.0 - wet;
	double offset;
	double inputSampleL;
	double inputSampleR;
	double outputSampleL;
	double outputSampleR;
	
	iirAmount += (iirAmount * tight * tight);
	if (tight > 0) tight /= 1.5;
	else tight /= 3.0;
	//we are setting it up so that to either extreme we can get an audible sound,
	//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
	if (iirAmount <= 0.0) iirAmount = 0.0;
	if (iirAmount > 1.0) iirAmount = 1.0;
	//handle the change in cutoff frequency
    
    while (--sampleFrames >= 0)
    {
		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.
		}
		
		outputSampleL = inputSampleL;
		outputSampleR = inputSampleR;
		
		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
		else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
		if (offset < 0) offset = 0;
		if (offset > 1) offset = 1;
		if (fpFlip)
		{
			iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
			outputSampleL = iirSampleAL;
		}
		else
		{
			iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
			outputSampleL = iirSampleBL;
		}

		
		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
		else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
		if (offset < 0) offset = 0;
		if (offset > 1) offset = 1;
		if (fpFlip)
		{
			iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
			outputSampleR = iirSampleAR;
		}
		else
		{
			iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
			outputSampleR = iirSampleBR;
		}
		fpFlip = !fpFlip;

		
		
		if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
		if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);

		//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 = outputSampleL;
		*out2 = outputSampleR;

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

void Lowpass::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();
	double iirAmount = (pow(A,2)+A)/2.0;
	iirAmount /= overallscale;
	double tight = (B*2.0)-1.0;
	double wet = C;
	double dry = 1.0 - wet;
	double offset;
	double inputSampleL;
	double inputSampleR;
	double outputSampleL;
	double outputSampleR;
	
	iirAmount += (iirAmount * tight * tight);
	if (tight > 0) tight /= 1.5;
	else tight /= 3.0;
	//we are setting it up so that to either extreme we can get an audible sound,
	//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
	if (iirAmount <= 0.0) iirAmount = 0.0;
	if (iirAmount > 1.0) iirAmount = 1.0;
	//handle the change in cutoff frequency
	
    while (--sampleFrames >= 0)
    {
		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.
		}
		
		outputSampleL = inputSampleL;
		outputSampleR = inputSampleR;
		
		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
		else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
		if (offset < 0) offset = 0;
		if (offset > 1) offset = 1;
		if (fpFlip)
		{
			iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
			outputSampleL = iirSampleAL;
		}
		else
		{
			iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
			outputSampleL = iirSampleBL;
		}
		
		
		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
		else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
		if (offset < 0) offset = 0;
		if (offset > 1) offset = 1;
		if (fpFlip)
		{
			iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
			outputSampleR = iirSampleAR;
		}
		else
		{
			iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
			outputSampleR = iirSampleBR;
		}
		fpFlip = !fpFlip;
		
		
		
		if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
		if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
		
		//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 = outputSampleL;
		*out2 = outputSampleR;
		
		*in1++;
		*in2++;
		*out1++;
		*out2++;
    }
}