Updates (in case my plane crashes)

1 files changed, 271 insertions, 0 deletions

diff --git a/plugins/WinVST/ToneSlant/ToneSlantProc.cpp b/plugins/WinVST/ToneSlant/ToneSlantProc.cpp
new file mode 100755
index 0000000..b594b05
--- /dev/null
+++ b/plugins/WinVST/ToneSlant/ToneSlantProc.cpp
@@ -0,0 +1,271 @@
+/* ========================================
+ *  ToneSlant - ToneSlant.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __ToneSlant_H
+#include "ToneSlant.h"
+#endif
+
+void ToneSlant::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double inputSampleL;
+	double inputSampleR;
+	double correctionSampleL;
+	double correctionSampleR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double drySampleL;
+	double drySampleR;
+	double overallscale = (A*99.0)+1.0;
+	double applySlant = (B*2.0)-1.0;
+	
+	
+	f[0] = 1.0 / overallscale;
+	//count to f(gain) which will be 0. f(0) is x1
+	for (int count = 1; count < 102; count++) {
+		if (count <= overallscale) {
+			f[count] = (1.0 - (count / overallscale)) / overallscale;
+			//recalc the filter and don't change the buffer it'll apply to
+		} else {
+			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+		}
+	}
+    
+    while (--sampleFrames >= 0)
+    {
+		for (int count = overallscale; count >= 0; count--) {
+			bL[count+1] = bL[count];
+			bR[count+1] = bR[count];
+		}
+		
+		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.
+		}
+		
+		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
+		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleR *= f[0];
+
+		for (int count = 1; count < overallscale; count++) {
+			accumulatorSampleL += (bL[count] * f[count]);
+			accumulatorSampleR += (bR[count] * f[count]);
+		}
+		
+		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
+		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
+		//we're gonna apply the total effect of all these calculations as a single subtract
+		
+		inputSampleL += (correctionSampleL * applySlant);
+		inputSampleR += (correctionSampleR * applySlant);
+		//our one math operation on the input data coming in
+
+		//noise shaping to 32-bit floating point
+		if (fpFlip) {
+			fpTemp = inputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLA;
+			fpTemp = inputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = inputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLB;
+			fpTemp = inputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 32 bit output
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void ToneSlant::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	double fpTemp; //this is different from singlereplacing
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double inputSampleL;
+	double inputSampleR;
+	double correctionSampleL;
+	double correctionSampleR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double drySampleL;
+	double drySampleR;
+	double overallscale = (A*99.0)+1.0;
+	double applySlant = (B*2.0)-1.0;
+	
+	f[0] = 1.0 / overallscale;
+	//count to f(gain) which will be 0. f(0) is x1
+	for (int count = 1; count < 102; count++) {
+		if (count <= overallscale) {
+			f[count] = (1.0 - (count / overallscale)) / overallscale;
+			//recalc the filter and don't change the buffer it'll apply to
+		} else {
+			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+		}
+	}
+	
+    while (--sampleFrames >= 0)
+    {
+		for (int count = overallscale; count >= 0; count--) {
+			bL[count+1] = bL[count];
+			bR[count+1] = bR[count];
+		}
+		
+		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.
+		}
+		
+		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
+		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleR *= f[0];
+		
+		for (int count = 1; count < overallscale; count++) {
+			accumulatorSampleL += (bL[count] * f[count]);
+			accumulatorSampleR += (bR[count] * f[count]);
+		}
+		
+		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
+		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
+		//we're gonna apply the total effect of all these calculations as a single subtract
+		
+		inputSampleL += (correctionSampleL * applySlant);
+		inputSampleR += (correctionSampleR * applySlant);
+		//our one math operation on the input data coming in
+		
+		//noise shaping to 64-bit floating point
+		if (fpFlip) {
+			fpTemp = inputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLA;
+			fpTemp = inputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = inputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLB;
+			fpTemp = inputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 64 bit output
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
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