airwindows/plugins/MacVST/AverMatrix/source/AverMatrixProc.cpp

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

#ifndef __AverMatrix_H
#include "AverMatrix.h"
#endif

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

	double overalltaps = (A * 9.0)+1.0;
	double taps = overalltaps;
	//this is our averaging, which is not integer but continuous
	
	double overallpoles = (B * 9.0)+1.0;
	//this is the poles of the filter, also not integer but continuous
	int yLimit = floor(overallpoles)+1;
	double yPartial = overallpoles - floor(overallpoles);
	//now we can do a for loop, and also apply the final pole continuously
	
	double wet = (C * 2.0)-1.0;
	double dry = (1.0-wet);
	if (dry > 1.0) dry = 1.0;
	
	int xLimit = 1;
	for(int x = 0; x < 11; x++) {
		if (taps > 1.0) {
			f[x] = 1.0;
			taps -= 1.0;
			xLimit++;
		} else {
			f[x] = taps;
			taps = 0.0;
		}
	} //there, now we have a neat little moving average with remainders
	if (xLimit > 9) xLimit = 9;
	
	if (overalltaps < 1.0) overalltaps = 1.0;
	for(int x = 0; x < xLimit; x++) {
		f[x] /= overalltaps;
	} //and now it's neatly scaled, too	
    
    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 drySampleL = inputSampleL;
		double drySampleR = inputSampleR;
		
		double previousPoleL = 0;		
		double previousPoleR = 0;		
		for (int y = 0; y < yLimit; y++) {
			for (int x = xLimit; x >= 0; x--) {
				bL[x+1][y] = bL[x][y];
				bR[x+1][y] = bR[x][y];
			}
			bL[0][y] = previousPoleL = inputSampleL;
			bR[0][y] = previousPoleR = inputSampleR;
			inputSampleL = 0.0;
			inputSampleR = 0.0;
			for (int x = 0; x < xLimit; x++) {
				inputSampleL += (bL[x][y] * f[x]);
				inputSampleR += (bR[x][y] * f[x]);
			}
		}
		inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial);
		inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial);
		//in this way we can blend in the final pole
		
		inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
		inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
		//wet can be negative, in which case dry is always full volume and they cancel
		
		//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 AverMatrix::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
{
    double* in1  =  inputs[0];
    double* in2  =  inputs[1];
    double* out1 = outputs[0];
    double* out2 = outputs[1];

	
	double overalltaps = (A * 9.0)+1.0;
	double taps = overalltaps;
	//this is our averaging, which is not integer but continuous
	
	double overallpoles = (B * 9.0)+1.0;
	//this is the poles of the filter, also not integer but continuous
	int yLimit = floor(overallpoles)+1;
	double yPartial = overallpoles - floor(overallpoles);
	//now we can do a for loop, and also apply the final pole continuously
	
	double wet = (C * 2.0)-1.0;
	double dry = (1.0-wet);
	if (dry > 1.0) dry = 1.0;
	
	int xLimit = 1;
	for(int x = 0; x < 11; x++) {
		if (taps > 1.0) {
			f[x] = 1.0;
			taps -= 1.0;
			xLimit++;
		} else {
			f[x] = taps;
			taps = 0.0;
		}
	} //there, now we have a neat little moving average with remainders
	if (xLimit > 9) xLimit = 9;
	
	if (overalltaps < 1.0) overalltaps = 1.0;
	for(int x = 0; x < xLimit; x++) {
		f[x] /= overalltaps;
	} //and now it's neatly scaled, too	
	
    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 drySampleL = inputSampleL;
		double drySampleR = inputSampleR;

		double previousPoleL = 0;		
		double previousPoleR = 0;		
		for (int y = 0; y < yLimit; y++) {
			for (int x = xLimit; x >= 0; x--) {
				bL[x+1][y] = bL[x][y];
				bR[x+1][y] = bR[x][y];
			}
			bL[0][y] = previousPoleL = inputSampleL;
			bR[0][y] = previousPoleR = inputSampleR;
			inputSampleL = 0.0;
			inputSampleR = 0.0;
			for (int x = 0; x < xLimit; x++) {
				inputSampleL += (bL[x][y] * f[x]);
				inputSampleR += (bR[x][y] * f[x]);
			}
		}
		inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial);
		inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial);
		//in this way we can blend in the final pole
		
		inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
		inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
		//wet can be negative, in which case dry is always full volume and they cancel
		
		//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++;
    }
}