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

#ifndef __ChorusEnsemble_H
#include "ChorusEnsemble.h"
#endif

void ChorusEnsemble::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 speed = pow(A,3) * 0.001;
	speed *= overallscale;
	int loopLimit = (int)(totalsamples * 0.499);
	int count;
	double range = pow(B,3) * loopLimit * 0.12;
	double wet = C;
	double modulation = range*wet;
	double dry = 1.0 - wet;
	double tupi = 3.141592653589793238 * 2.0;
	double offset;
	double start[4];

	long double inputSampleL;
	long double inputSampleR;
	double drySampleL;
	double drySampleR;
	//now we'll precalculate some stuff that needn't be in every sample
	start[0] = range;
	start[1] = range * 2;
	start[2] = range * 3;
	start[3] = range * 4;
    
    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.
		}
		drySampleL = inputSampleL;
		drySampleR = inputSampleR;

		airFactorL = airPrevL - inputSampleL;
		if (fpFlip) {airEvenL += airFactorL; airOddL -= airFactorL; airFactorL = airEvenL;}
		else {airOddL += airFactorL; airEvenL -= airFactorL; airFactorL = airOddL;}
		airOddL = (airOddL - ((airOddL - airEvenL)/256.0)) / 1.0001;
		airEvenL = (airEvenL - ((airEvenL - airOddL)/256.0)) / 1.0001;
		airPrevL = inputSampleL;
		inputSampleL += (airFactorL*wet);
		//air, compensates for loss of highs in flanger's interpolation
		
		airFactorR = airPrevR - inputSampleR;
		if (fpFlip) {airEvenR += airFactorR; airOddR -= airFactorR; airFactorR = airEvenR;}
		else {airOddR += airFactorR; airEvenR -= airFactorR; airFactorR = airOddR;}
		airOddR = (airOddR - ((airOddR - airEvenR)/256.0)) / 1.0001;
		airEvenR = (airEvenR - ((airEvenR - airOddR)/256.0)) / 1.0001;
		airPrevR = inputSampleR;
		inputSampleR += (airFactorR*wet);
		//air, compensates for loss of highs in flanger's interpolation
		
		if (gcount < 1 || gcount > loopLimit) {gcount = loopLimit;}
		count = gcount;
		dL[count+loopLimit] = dL[count] = inputSampleL;
		dR[count+loopLimit] = dR[count] = inputSampleR;
		gcount--;
		//double buffer
		
		offset = start[0] + (modulation * sin(sweep));
		count = gcount + (int)floor(offset);
		inputSampleL = dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us

		inputSampleR = dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		offset = start[1] + (modulation * sin(sweep + 1.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us

		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		
		offset = start[2] + (modulation * sin(sweep + 2.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us

		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		
		offset = start[3] + (modulation * sin(sweep + 3.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us

		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleL *= 0.125; //to get a comparable level
		inputSampleR *= 0.125; //to get a comparable level
		
		sweep += speed;
		if (sweep > tupi){sweep -= tupi;}
		//still scrolling through the samples, remember
		
		if (wet !=1.0) {
			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
		}
		fpFlip = !fpFlip;
		

		//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 ChorusEnsemble::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 speed = pow(A,3) * 0.001;
	speed *= overallscale;
	int loopLimit = (int)(totalsamples * 0.499);
	int count;
	double range = pow(B,3) * loopLimit * 0.12;
	double wet = C;
	double modulation = range*wet;
	double dry = 1.0 - wet;
	double tupi = 3.141592653589793238 * 2.0;
	double offset;
	double start[4];
	
	long double inputSampleL;
	long double inputSampleR;
	double drySampleL;
	double drySampleR;
	//now we'll precalculate some stuff that needn't be in every sample
	start[0] = range;
	start[1] = range * 2;
	start[2] = range * 3;
	start[3] = range * 4;
    
    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.
		}
		drySampleL = inputSampleL;
		drySampleR = inputSampleR;

		airFactorL = airPrevL - inputSampleL;
		if (fpFlip) {airEvenL += airFactorL; airOddL -= airFactorL; airFactorL = airEvenL;}
		else {airOddL += airFactorL; airEvenL -= airFactorL; airFactorL = airOddL;}
		airOddL = (airOddL - ((airOddL - airEvenL)/256.0)) / 1.0001;
		airEvenL = (airEvenL - ((airEvenL - airOddL)/256.0)) / 1.0001;
		airPrevL = inputSampleL;
		inputSampleL += (airFactorL*wet);
		//air, compensates for loss of highs in flanger's interpolation
		
		airFactorR = airPrevR - inputSampleR;
		if (fpFlip) {airEvenR += airFactorR; airOddR -= airFactorR; airFactorR = airEvenR;}
		else {airOddR += airFactorR; airEvenR -= airFactorR; airFactorR = airOddR;}
		airOddR = (airOddR - ((airOddR - airEvenR)/256.0)) / 1.0001;
		airEvenR = (airEvenR - ((airEvenR - airOddR)/256.0)) / 1.0001;
		airPrevR = inputSampleR;
		inputSampleR += (airFactorR*wet);
		//air, compensates for loss of highs in flanger's interpolation
		
		if (gcount < 1 || gcount > loopLimit) {gcount = loopLimit;}
		count = gcount;
		dL[count+loopLimit] = dL[count] = inputSampleL;
		dR[count+loopLimit] = dR[count] = inputSampleR;
		gcount--;
		//double buffer
		
		offset = start[0] + (modulation * sin(sweep));
		count = gcount + (int)floor(offset);
		inputSampleL = dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleR = dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		offset = start[1] + (modulation * sin(sweep + 1.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		
		offset = start[2] + (modulation * sin(sweep + 2.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		
		offset = start[3] + (modulation * sin(sweep + 3.0));
		count = gcount + (int)floor(offset);
		inputSampleL += dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleL += dL[count+1]; //we can assume always using this in one way or another?
		inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleR += dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0
		inputSampleR += dR[count+1]; //we can assume always using this in one way or another?
		inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0
		inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us
		
		inputSampleL *= 0.125; //to get a comparable level
		inputSampleR *= 0.125; //to get a comparable level
		
		sweep += speed;
		if (sweep > tupi){sweep -= tupi;}
		//still scrolling through the samples, remember
		
		if (wet !=1.0) {
			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
		}
		fpFlip = !fpFlip;
		
		
		//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++;
    }
}