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

#ifndef __ConsoleMCChannel_H
#include "ConsoleMCChannel.h"
#endif

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

	VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
	double overallscale = 1.0;
	overallscale /= 44100.0;
	overallscale *= getSampleRate(); //will be over 1.0848 when over 48k
	int cycleEnd = floor(overallscale);
	if (cycleEnd < 1) cycleEnd = 1;
	if (cycleEnd > 3) cycleEnd = 3;
	
	double fatTreble = (A*6.0)-3.0;
	bassA = bassB;
	bassB = (D*6.0)-3.0;
	//these should stack to go up to -3.0 to 3.0
	if (fatTreble < 0.0) fatTreble /= 3.0;
	if (bassB < 0.0) bassB /= 3.0;
	//and then become -1.0 to 3.0;
	//there will be successive sin/cos stages w. dry/wet in these
	double freqTreble = 0.853;
	double freqMid = 0.026912;
	switch (cycleEnd)
	{
		case 1: //base sample rate, no change
			break;
		case 2: //96k tier
			freqTreble = 0.4265;
			freqMid = 0.013456;
			break;
		case 3: //192k tier
			freqTreble = 0.21325;
			freqMid = 0.006728;
			break;
	}
	
	//begin ResEQ2 Mid Boost
	double freqMPeak = pow(B+0.16,3);
	double amountMPeak = pow(C,2);
	int maxMPeak = (amountMPeak*63.0)+1;
	if ((freqMPeak != prevfreqMPeak)||(amountMPeak != prevamountMPeak)) {
		for (int x = 0; x < maxMPeak; x++) {
			if (((double)x*freqMPeak) < M_PI_4) f[x] = sin(((double)x*freqMPeak)*4.0)*freqMPeak*sin(((double)(maxMPeak-x)/(double)maxMPeak)*M_PI_2);
			else f[x] = cos((double)x*freqMPeak)*freqMPeak*sin(((double)(maxMPeak-x)/(double)maxMPeak)*M_PI_2);
		}
		prevfreqMPeak = freqMPeak; prevamountMPeak = amountMPeak;
	}//end ResEQ2 Mid Boost
	
	int bitshiftL = 0;
	int bitshiftR = 0;
	double panControl = (E*2.0)-1.0; //-1.0 to 1.0
	double panAttenuation = (1.0-fabs(panControl));
	int panBits = 20; //start centered
	if (panAttenuation > 0.0) panBits = floor(1.0 / panAttenuation);
	if (panControl > 0.25) bitshiftL += panBits;
	if (panControl < -0.25) bitshiftR += panBits;
	if (bitshiftL < 0) bitshiftL = 0; if (bitshiftL > 17) bitshiftL = 17;
	if (bitshiftR < 0) bitshiftR = 0; if (bitshiftR > 17) bitshiftR = 17;
	double gainL = 1.0;
	double gainR = 1.0;
	switch (bitshiftL)
	{
		case 17: gainL = 0.0; break;
		case 16: gainL = 0.0000152587890625; break;
		case 15: gainL = 0.000030517578125; break;
		case 14: gainL = 0.00006103515625; break;
		case 13: gainL = 0.0001220703125; break;
		case 12: gainL = 0.000244140625; break;
		case 11: gainL = 0.00048828125; break;
		case 10: gainL = 0.0009765625; break;
		case 9: gainL = 0.001953125; break;
		case 8: gainL = 0.00390625; break;
		case 7: gainL = 0.0078125; break;
		case 6: gainL = 0.015625; break;
		case 5: gainL = 0.03125; break;
		case 4: gainL = 0.0625; break;
		case 3: gainL = 0.125; break;
		case 2: gainL = 0.25; break;
		case 1: gainL = 0.5; break;
		case 0: break;
	}
	switch (bitshiftR)
	{
		case 17: gainR = 0.0; break;
		case 16: gainR = 0.0000152587890625; break;
		case 15: gainR = 0.000030517578125; break;
		case 14: gainR = 0.00006103515625; break;
		case 13: gainR = 0.0001220703125; break;
		case 12: gainR = 0.000244140625; break;
		case 11: gainR = 0.00048828125; break;
		case 10: gainR = 0.0009765625; break;
		case 9: gainR = 0.001953125; break;
		case 8: gainR = 0.00390625; break;
		case 7: gainR = 0.0078125; break;
		case 6: gainR = 0.015625; break;
		case 5: gainR = 0.03125; break;
		case 4: gainR = 0.0625; break;
		case 3: gainR = 0.125; break;
		case 2: gainR = 0.25; break;
		case 1: gainR = 0.5; break;
		case 0: break;
	}
	
	gainA = gainB;
	gainB = F*2.0; //smoothed master fader from Z2 filters
	//BitShiftGain pre gain trim goes here
	double subTrim = 0.0046999 / overallscale;
	
    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 temp = (double)sampleFrames/inFramesToProcess;
		double gain = (gainA*temp)+(gainB*(1.0-temp));
		double bass = (bassA*temp)+(bassB*(1.0-temp));
		inputSampleL *= gain;
		inputSampleR *= gain;
		//for MCI consoles, the fader is before the EQ, which overdrives easily.
		//so we put the main fader here.
		
		//begin Pear filter stages
		double bassL = inputSampleL;
		double bassR = inputSampleR;
		double slew = ((bassL - pearA[0]) + pearA[1])*freqTreble*0.5;
		pearA[0] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[0] + pearA[1]));
		pearA[1] = slew; slew = ((bassR - pearA[2]) + pearA[3])*freqTreble*0.5;
		pearA[2] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[2] + pearA[3]));
		pearA[3] = slew; slew = ((bassL - pearA[4]) + pearA[5])*freqTreble*0.5;
		pearA[4] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[4] + pearA[5]));
		pearA[5] = slew; slew = ((bassR - pearA[6]) + pearA[7])*freqTreble*0.5;
		pearA[6] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[6] + pearA[7]));
		pearA[7] = slew; slew = ((bassL - pearA[8]) + pearA[9])*freqTreble*0.5;
		pearA[8] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[8] + pearA[9]));
		pearA[9] = slew; slew = ((bassR - pearA[10]) + pearA[11])*freqTreble*0.5;
		pearA[10] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[10] + pearA[11]));
		pearA[11] = slew; slew = ((bassL - pearA[12]) + pearA[13])*freqTreble*0.5;
		pearA[12] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[12] + pearA[13]));
		pearA[13] = slew; slew = ((bassR - pearA[14]) + pearA[15])*freqTreble*0.5;
		pearA[14] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[14] + pearA[15]));
		pearA[15] = slew;
		//unrolled mid/treble crossover (called bass to use fewer variables)		
		double trebleL = inputSampleL - bassL; inputSampleL = bassL;
		double trebleR = inputSampleR - bassR; inputSampleR = bassR;
		//at this point 'bass' is actually still mid and bass
		slew = ((bassL - pearB[0]) + pearB[1])*freqMid*0.5;
		pearB[0] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[0] + pearB[1]));
		pearB[1] = slew; slew = ((bassR - pearB[2]) + pearB[3])*freqMid*0.5;
		pearB[2] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[2] + pearB[3]));
		pearB[3] = slew; slew = ((bassL - pearB[4]) + pearB[5])*freqMid*0.5;
		pearB[4] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[4] + pearB[5]));
		pearB[5] = slew; slew = ((bassR - pearB[6]) + pearB[7])*freqMid*0.5;
		pearB[6] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[6] + pearB[7]));
		pearB[7] = slew; slew = ((bassL - pearB[8]) + pearB[9])*freqMid*0.5;
		pearB[8] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[8] + pearB[9]));
		pearB[9] = slew; slew = ((bassR - pearB[10]) + pearB[11])*freqMid*0.5;
		pearB[10] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[10] + pearB[11]));
		pearB[11] = slew; slew = ((bassL - pearB[12]) + pearB[13])*freqMid*0.5;
		pearB[12] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[12] + pearB[13]));
		pearB[13] = slew; slew = ((bassR - pearB[14]) + pearB[15])*freqMid*0.5;
		pearB[14] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[14] + pearB[15]));
		pearB[15] = slew; slew = ((bassL - pearB[16]) + pearB[17])*freqMid*0.5;
		pearB[16] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[16] + pearB[17]));
		pearB[17] = slew; slew = ((bassR - pearB[18]) + pearB[19])*freqMid*0.5;
		pearB[18] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[18] + pearB[19]));
		pearB[19] = slew;
		double midL = inputSampleL - bassL;
		double midR = inputSampleR - bassR;
		//we now have three bands out of two pear filters
		
		double w = 0.0; //filter into bands, apply the sin/cos to each band
		double avg = 0.0; //for the treble band, we're applying mild filtering
		if (overallscale > 1.1) {
			avg = (trebleL+avgAL)*0.5; avgAL = trebleL; trebleL = avg;
			avg = (trebleR+avgAR)*0.5; avgAR = trebleR;	trebleR = avg;
			if (overallscale > 2.1) {
				avg = (trebleL+avgBL)*0.5; avgBL = trebleL; trebleL = avg;
				avg = (trebleR+avgBR)*0.5; avgBR = trebleR;	trebleR = avg;
			}			
		}
		if (fatTreble > 0.0) {
			w = fatTreble; if (w > 1.0) w = 1.0;
			trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
			trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
			if (fatTreble > 1.0) {
				w = fatTreble-1.0; if (w > 1.0) w = 1.0;
				trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
				trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
				if (fatTreble > 2.0) {
					w = fatTreble-2.0;
					trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
					trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
				} //sine stages for EQ or compression
			}
		}
		if (fatTreble < 0.0) {
			if (trebleL > 1.0) trebleL = 1.0; if (trebleL < -1.0) trebleL = -1.0;
			if (trebleR > 1.0) trebleR = 1.0; if (trebleR < -1.0) trebleR = -1.0;
			w = -fatTreble; if (w > 1.0) w = 1.0;
			if (trebleL > 0) trebleL = (trebleL*(1.0-w))+((1.0-cos(trebleL))*sin(w));
			else trebleL = (trebleL*(1.0-w))+((-1.0+cos(-trebleL))*sin(w));
			if (trebleR > 0) trebleR = (trebleR*(1.0-w))+((1.0-cos(trebleR))*sin(w));
			else trebleR = (trebleR*(1.0-w))+((-1.0+cos(-trebleR))*sin(w));
		} //cosine stages for EQ or expansion
		if (overallscale > 1.1) {
			avg = (trebleL+avgCL)*0.5; avgCL = trebleL; trebleL = avg;
			avg = (trebleR+avgCR)*0.5; avgCR = trebleR;	trebleR = avg;
		}
		
		//begin ResEQ2 Mid Boost
		mpc++; if (mpc < 1 || mpc > 2001) mpc = 1;
		mpkL[mpc] = midL;
		mpkR[mpc] = midR;
		double midMPeakL = 0.0;
		double midMPeakR = 0.0;
		for (int x = 0; x < maxMPeak; x++) {
			int y = x*cycleEnd;
			switch (cycleEnd)
			{
				case 1: 
					midMPeakL += (mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x]);
					midMPeakR += (mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x]); break;
				case 2: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5); break;
				case 3: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); break;
				case 4: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); //break
			}
		}
		midL = (midMPeakL*amountMPeak)+((1.5-amountMPeak>1.0)?midL:midL*(1.5-amountMPeak));
		midR = (midMPeakR*amountMPeak)+((1.5-amountMPeak>1.0)?midR:midR*(1.5-amountMPeak));
		//end ResEQ2 Mid Boost
		
		if (bassL > 1.0) bassL = 1.0; if (bassL < -1.0) bassL = -1.0;
		if (bassR > 1.0) bassR = 1.0; if (bassR < -1.0) bassR = -1.0;
		if (bass > 0.0) {
			w = bass; if (w > 1.0) w = 1.0;
			bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.6);
			bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.6);
			if (bass > 1.0) {
				w = bass-1.0; if (w > 1.0) w = 1.0;
				bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.4);
				bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.4);
				if (bass > 2.0) {
					w = bass-2.0;
					bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.2);
					bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.2);
				} //sine stages for EQ or compression
			}
		}
		if (bass < 0.0) {
			w = -bass; if (w > 1.0) w = 1.0;	
			if (bassL > 0) bassL = (bassL*(1.0-w))+((1.0-cos(bassL))*sin(w));
			else bassL = (bassL*(1.0-w))+((-1.0+cos(-bassL))*sin(w));
			if (bassR > 0) bassR = (bassR*(1.0-w))+((1.0-cos(bassR))*sin(w));
			else bassR = (bassR*(1.0-w))+((-1.0+cos(-bassR))*sin(w));
		} //cosine stages for EQ or expansion
		//the sin() is further restricting output when fully attenuated
		
		//begin SubTight section
		double subSampleL = bassL * subTrim;
		double subSampleR = bassR * subTrim;
		double scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
		subAL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
		subAR = subSampleR*scale;
		scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
		subBL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
		subBR = subSampleR*scale;
		if (subSampleL > 0.25) subSampleL = 0.25;
		if (subSampleL < -0.25) subSampleL = -0.25;
		if (subSampleR > 0.25) subSampleR = 0.25;
		if (subSampleR < -0.25) subSampleR = -0.25;
		bassL = bassL - (subSampleL*16.0);
		bassR = bassR - (subSampleR*16.0);
		//end SubTight section
		
		inputSampleL = (bassL + midL + trebleL)*gainL;
		inputSampleR = (bassR + midR + trebleR)*gainR;
		//applies BitShiftPan pan section
		
		//begin sin() style Channel processing
		if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
		if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
		if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
		if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
		inputSampleL = sin(inputSampleL);
		inputSampleR = sin(inputSampleR);
		
		//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 ConsoleMCChannel::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
{
    double* in1  =  inputs[0];
    double* in2  =  inputs[1];
    double* out1 = outputs[0];
    double* out2 = outputs[1];

	VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
	double overallscale = 1.0;
	overallscale /= 44100.0;
	overallscale *= getSampleRate(); //will be over 1.0848 when over 48k
	int cycleEnd = floor(overallscale);
	if (cycleEnd < 1) cycleEnd = 1;
	if (cycleEnd > 3) cycleEnd = 3;
	
	double fatTreble = (A*6.0)-3.0;
	bassA = bassB;
	bassB = (D*6.0)-3.0;
	//these should stack to go up to -3.0 to 3.0
	if (fatTreble < 0.0) fatTreble /= 3.0;
	if (bassB < 0.0) bassB /= 3.0;
	//and then become -1.0 to 3.0;
	//there will be successive sin/cos stages w. dry/wet in these
	double freqTreble = 0.853;
	double freqMid = 0.026912;
	switch (cycleEnd)
	{
		case 1: //base sample rate, no change
			break;
		case 2: //96k tier
			freqTreble = 0.4265;
			freqMid = 0.013456;
			break;
		case 3: //192k tier
			freqTreble = 0.21325;
			freqMid = 0.006728;
			break;
	}
	
	//begin ResEQ2 Mid Boost
	double freqMPeak = pow(B+0.16,3);
	double amountMPeak = pow(C,2);
	int maxMPeak = (amountMPeak*63.0)+1;
	if ((freqMPeak != prevfreqMPeak)||(amountMPeak != prevamountMPeak)) {
		for (int x = 0; x < maxMPeak; x++) {
			if (((double)x*freqMPeak) < M_PI_4) f[x] = sin(((double)x*freqMPeak)*4.0)*freqMPeak*sin(((double)(maxMPeak-x)/(double)maxMPeak)*M_PI_2);
			else f[x] = cos((double)x*freqMPeak)*freqMPeak*sin(((double)(maxMPeak-x)/(double)maxMPeak)*M_PI_2);
		}
		prevfreqMPeak = freqMPeak; prevamountMPeak = amountMPeak;
	}//end ResEQ2 Mid Boost
	
	int bitshiftL = 0;
	int bitshiftR = 0;
	double panControl = (E*2.0)-1.0; //-1.0 to 1.0
	double panAttenuation = (1.0-fabs(panControl));
	int panBits = 20; //start centered
	if (panAttenuation > 0.0) panBits = floor(1.0 / panAttenuation);
	if (panControl > 0.25) bitshiftL += panBits;
	if (panControl < -0.25) bitshiftR += panBits;
	if (bitshiftL < 0) bitshiftL = 0; if (bitshiftL > 17) bitshiftL = 17;
	if (bitshiftR < 0) bitshiftR = 0; if (bitshiftR > 17) bitshiftR = 17;
	double gainL = 1.0;
	double gainR = 1.0;
	switch (bitshiftL)
	{
		case 17: gainL = 0.0; break;
		case 16: gainL = 0.0000152587890625; break;
		case 15: gainL = 0.000030517578125; break;
		case 14: gainL = 0.00006103515625; break;
		case 13: gainL = 0.0001220703125; break;
		case 12: gainL = 0.000244140625; break;
		case 11: gainL = 0.00048828125; break;
		case 10: gainL = 0.0009765625; break;
		case 9: gainL = 0.001953125; break;
		case 8: gainL = 0.00390625; break;
		case 7: gainL = 0.0078125; break;
		case 6: gainL = 0.015625; break;
		case 5: gainL = 0.03125; break;
		case 4: gainL = 0.0625; break;
		case 3: gainL = 0.125; break;
		case 2: gainL = 0.25; break;
		case 1: gainL = 0.5; break;
		case 0: break;
	}
	switch (bitshiftR)
	{
		case 17: gainR = 0.0; break;
		case 16: gainR = 0.0000152587890625; break;
		case 15: gainR = 0.000030517578125; break;
		case 14: gainR = 0.00006103515625; break;
		case 13: gainR = 0.0001220703125; break;
		case 12: gainR = 0.000244140625; break;
		case 11: gainR = 0.00048828125; break;
		case 10: gainR = 0.0009765625; break;
		case 9: gainR = 0.001953125; break;
		case 8: gainR = 0.00390625; break;
		case 7: gainR = 0.0078125; break;
		case 6: gainR = 0.015625; break;
		case 5: gainR = 0.03125; break;
		case 4: gainR = 0.0625; break;
		case 3: gainR = 0.125; break;
		case 2: gainR = 0.25; break;
		case 1: gainR = 0.5; break;
		case 0: break;
	}
	
	gainA = gainB;
	gainB = F*2.0; //smoothed master fader from Z2 filters
	//BitShiftGain pre gain trim goes here
	double subTrim = 0.0046999 / overallscale;
	
    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 temp = (double)sampleFrames/inFramesToProcess;
		double gain = (gainA*temp)+(gainB*(1.0-temp));
		double bass = (bassA*temp)+(bassB*(1.0-temp));
		inputSampleL *= gain;
		inputSampleR *= gain;
		//for MCI consoles, the fader is before the EQ, which overdrives easily.
		//so we put the main fader here.
		
		//begin Pear filter stages
		double bassL = inputSampleL;
		double bassR = inputSampleR;
		double slew = ((bassL - pearA[0]) + pearA[1])*freqTreble*0.5;
		pearA[0] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[0] + pearA[1]));
		pearA[1] = slew; slew = ((bassR - pearA[2]) + pearA[3])*freqTreble*0.5;
		pearA[2] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[2] + pearA[3]));
		pearA[3] = slew; slew = ((bassL - pearA[4]) + pearA[5])*freqTreble*0.5;
		pearA[4] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[4] + pearA[5]));
		pearA[5] = slew; slew = ((bassR - pearA[6]) + pearA[7])*freqTreble*0.5;
		pearA[6] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[6] + pearA[7]));
		pearA[7] = slew; slew = ((bassL - pearA[8]) + pearA[9])*freqTreble*0.5;
		pearA[8] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[8] + pearA[9]));
		pearA[9] = slew; slew = ((bassR - pearA[10]) + pearA[11])*freqTreble*0.5;
		pearA[10] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[10] + pearA[11]));
		pearA[11] = slew; slew = ((bassL - pearA[12]) + pearA[13])*freqTreble*0.5;
		pearA[12] = bassL = (freqTreble * bassL) + ((1.0-freqTreble) * (pearA[12] + pearA[13]));
		pearA[13] = slew; slew = ((bassR - pearA[14]) + pearA[15])*freqTreble*0.5;
		pearA[14] = bassR = (freqTreble * bassR) + ((1.0-freqTreble) * (pearA[14] + pearA[15]));
		pearA[15] = slew;
		//unrolled mid/treble crossover (called bass to use fewer variables)		
		double trebleL = inputSampleL - bassL; inputSampleL = bassL;
		double trebleR = inputSampleR - bassR; inputSampleR = bassR;
		//at this point 'bass' is actually still mid and bass
		slew = ((bassL - pearB[0]) + pearB[1])*freqMid*0.5;
		pearB[0] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[0] + pearB[1]));
		pearB[1] = slew; slew = ((bassR - pearB[2]) + pearB[3])*freqMid*0.5;
		pearB[2] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[2] + pearB[3]));
		pearB[3] = slew; slew = ((bassL - pearB[4]) + pearB[5])*freqMid*0.5;
		pearB[4] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[4] + pearB[5]));
		pearB[5] = slew; slew = ((bassR - pearB[6]) + pearB[7])*freqMid*0.5;
		pearB[6] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[6] + pearB[7]));
		pearB[7] = slew; slew = ((bassL - pearB[8]) + pearB[9])*freqMid*0.5;
		pearB[8] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[8] + pearB[9]));
		pearB[9] = slew; slew = ((bassR - pearB[10]) + pearB[11])*freqMid*0.5;
		pearB[10] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[10] + pearB[11]));
		pearB[11] = slew; slew = ((bassL - pearB[12]) + pearB[13])*freqMid*0.5;
		pearB[12] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[12] + pearB[13]));
		pearB[13] = slew; slew = ((bassR - pearB[14]) + pearB[15])*freqMid*0.5;
		pearB[14] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[14] + pearB[15]));
		pearB[15] = slew; slew = ((bassL - pearB[16]) + pearB[17])*freqMid*0.5;
		pearB[16] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[16] + pearB[17]));
		pearB[17] = slew; slew = ((bassR - pearB[18]) + pearB[19])*freqMid*0.5;
		pearB[18] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[18] + pearB[19]));
		pearB[19] = slew;
		double midL = inputSampleL - bassL;
		double midR = inputSampleR - bassR;
		//we now have three bands out of two pear filters
		
		double w = 0.0; //filter into bands, apply the sin/cos to each band
		double avg = 0.0; //for the treble band, we're applying mild filtering
		if (overallscale > 1.1) {
			avg = (trebleL+avgAL)*0.5; avgAL = trebleL; trebleL = avg;
			avg = (trebleR+avgAR)*0.5; avgAR = trebleR;	trebleR = avg;
			if (overallscale > 2.1) {
				avg = (trebleL+avgBL)*0.5; avgBL = trebleL; trebleL = avg;
				avg = (trebleR+avgBR)*0.5; avgBR = trebleR;	trebleR = avg;
			}			
		}
		if (fatTreble > 0.0) {
			w = fatTreble; if (w > 1.0) w = 1.0;
			trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
			trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
			if (fatTreble > 1.0) {
				w = fatTreble-1.0; if (w > 1.0) w = 1.0;
				trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
				trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
				if (fatTreble > 2.0) {
					w = fatTreble-2.0;
					trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*w);
					trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*w);
				} //sine stages for EQ or compression
			}
		}
		if (fatTreble < 0.0) {
			if (trebleL > 1.0) trebleL = 1.0; if (trebleL < -1.0) trebleL = -1.0;
			if (trebleR > 1.0) trebleR = 1.0; if (trebleR < -1.0) trebleR = -1.0;
			w = -fatTreble; if (w > 1.0) w = 1.0;
			if (trebleL > 0) trebleL = (trebleL*(1.0-w))+((1.0-cos(trebleL))*sin(w));
			else trebleL = (trebleL*(1.0-w))+((-1.0+cos(-trebleL))*sin(w));
			if (trebleR > 0) trebleR = (trebleR*(1.0-w))+((1.0-cos(trebleR))*sin(w));
			else trebleR = (trebleR*(1.0-w))+((-1.0+cos(-trebleR))*sin(w));
		} //cosine stages for EQ or expansion
		if (overallscale > 1.1) {
			avg = (trebleL+avgCL)*0.5; avgCL = trebleL; trebleL = avg;
			avg = (trebleR+avgCR)*0.5; avgCR = trebleR;	trebleR = avg;
		}
		
		//begin ResEQ2 Mid Boost
		mpc++; if (mpc < 1 || mpc > 2001) mpc = 1;
		mpkL[mpc] = midL;
		mpkR[mpc] = midR;
		double midMPeakL = 0.0;
		double midMPeakR = 0.0;
		for (int x = 0; x < maxMPeak; x++) {
			int y = x*cycleEnd;
			switch (cycleEnd)
			{
				case 1: 
					midMPeakL += (mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x]);
					midMPeakR += (mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x]); break;
				case 2: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.5); break;
				case 3: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.333); break;
				case 4: 
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); y--;
					midMPeakL += ((mpkL[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25);
					midMPeakR += ((mpkR[(mpc-y)+((mpc-y < 1)?2001:0)] * f[x])*0.25); //break
			}
		}
		midL = (midMPeakL*amountMPeak)+((1.5-amountMPeak>1.0)?midL:midL*(1.5-amountMPeak));
		midR = (midMPeakR*amountMPeak)+((1.5-amountMPeak>1.0)?midR:midR*(1.5-amountMPeak));
		//end ResEQ2 Mid Boost
		
		if (bassL > 1.0) bassL = 1.0; if (bassL < -1.0) bassL = -1.0;
		if (bassR > 1.0) bassR = 1.0; if (bassR < -1.0) bassR = -1.0;
		if (bass > 0.0) {
			w = bass; if (w > 1.0) w = 1.0;
			bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.6);
			bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.6);
			if (bass > 1.0) {
				w = bass-1.0; if (w > 1.0) w = 1.0;
				bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.4);
				bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.4);
				if (bass > 2.0) {
					w = bass-2.0;
					bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*w*1.2);
					bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*w*1.2);
				} //sine stages for EQ or compression
			}
		}
		if (bass < 0.0) {
			w = -bass; if (w > 1.0) w = 1.0;	
			if (bassL > 0) bassL = (bassL*(1.0-w))+((1.0-cos(bassL))*sin(w));
			else bassL = (bassL*(1.0-w))+((-1.0+cos(-bassL))*sin(w));
			if (bassR > 0) bassR = (bassR*(1.0-w))+((1.0-cos(bassR))*sin(w));
			else bassR = (bassR*(1.0-w))+((-1.0+cos(-bassR))*sin(w));
		} //cosine stages for EQ or expansion
		//the sin() is further restricting output when fully attenuated
		
		//begin SubTight section
		double subSampleL = bassL * subTrim;
		double subSampleR = bassR * subTrim;
		double scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
		subAL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
		subAR = subSampleR*scale;
		scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
		subBL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
		subBR = subSampleR*scale;
		if (subSampleL > 0.25) subSampleL = 0.25;
		if (subSampleL < -0.25) subSampleL = -0.25;
		if (subSampleR > 0.25) subSampleR = 0.25;
		if (subSampleR < -0.25) subSampleR = -0.25;
		bassL = bassL - (subSampleL*16.0);
		bassR = bassR - (subSampleR*16.0);
		//end SubTight section
		
		inputSampleL = (bassL + midL + trebleL)*gainL;
		inputSampleR = (bassR + midR + trebleR)*gainR;
		//applies BitShiftPan pan section
		
		//begin sin() style Channel processing
		if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
		if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
		if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
		if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
		inputSampleL = sin(inputSampleL);
		inputSampleR = sin(inputSampleR);
		
		//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++;
    }
}
