1 files changed, 191 insertions, 0 deletions
diff --git a/plugins/WinVST/AverMatrix/AverMatrixProc.cpp b/plugins/WinVST/AverMatrix/AverMatrixProc.cpp
new file mode 100755
index 0000000..381ea9c
--- /dev/null
+++ b/plugins/WinVST/AverMatrix/AverMatrixProc.cpp
@@ -0,0 +1,191 @@
+/* ========================================
+ *  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)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
+		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		long double previousPoleL = 0;		
+		long 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 * dry);
+		inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		//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);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		frexpf((float)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-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)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
+		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+
+		long double previousPoleL = 0;		
+		long 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 * dry);
+		inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		//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);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		frexp((double)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-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++;
+    }
+}