Aura

3 files changed, 692 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/Aura/Aura.cpp b/plugins/LinuxVST/src/Aura/Aura.cpp
new file mode 100755
index 0000000..d011564
--- /dev/null
+++ b/plugins/LinuxVST/src/Aura/Aura.cpp
@@ -0,0 +1,140 @@
+/* ========================================
+ *  Aura - Aura.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Aura_H
+#include "Aura.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Aura(audioMaster);}
+
+Aura::Aura(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 1.0;
+	for(int count = 0; count < 21; count++) {
+		bL[count] = 0.0;
+		bR[count] = 0.0;
+		f[count] = 0.0;
+	}
+	lastSampleL = 0.0;
+	previousVelocityL = 0.0;
+	lastSampleR = 0.0;
+	previousVelocityR = 0.0;
+
+	fpNShapeLA = 0.0;
+	fpNShapeLB = 0.0;
+	fpNShapeRA = 0.0;
+	fpNShapeRB = 0.0;
+	fpFlip = true;
+	//this is reset: values being initialized only once. Startup values, whatever they are.
+	
+    _canDo.insert("plugAsChannelInsert"); // plug-in can be used as a channel insert effect.
+    _canDo.insert("plugAsSend"); // plug-in can be used as a send effect.
+    _canDo.insert("x2in2out"); 
+    setNumInputs(kNumInputs);
+    setNumOutputs(kNumOutputs);
+    setUniqueID(kUniqueId);
+    canProcessReplacing();     // supports output replacing
+    canDoubleReplacing();      // supports double precision processing
+	programsAreChunks(true);
+    vst_strncpy (_programName, "Default", kVstMaxProgNameLen); // default program name
+}
+
+Aura::~Aura() {}
+VstInt32 Aura::getVendorVersion () {return 1000;}
+void Aura::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Aura::getProgramName(char *name) {vst_strncpy (name, _programName, kVstMaxProgNameLen);}
+//airwindows likes to ignore this stuff. Make your own programs, and make a different plugin rather than
+//trying to do versioning and preventing people from using older versions. Maybe they like the old one!
+
+static float pinParameter(float data)
+{
+	if (data < 0.0f) return 0.0f;
+	if (data > 1.0f) return 1.0f;
+	return data;
+}
+
+VstInt32 Aura::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	/* Note: The way this is set up, it will break if you manage to save settings on an Intel
+	 machine and load them on a PPC Mac. However, it's fine if you stick to the machine you 
+	 started with. */
+	
+	*data = chunkData;
+	return kNumParameters * sizeof(float);
+}
+
+VstInt32 Aura::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	/* We're ignoring byteSize as we found it to be a filthy liar */
+	
+	/* calculate any other fields you need here - you could copy in 
+	 code from setParameter() here. */
+	return 0;
+}
+
+void Aura::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float Aura::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        default: break; // unknown parameter, shouldn't happen!
+    } return 0.0; //we only need to update the relevant name, this is simple to manage
+}
+
+void Aura::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Voicing", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Dry/Wet", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void Aura::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Aura::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Aura::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Aura::getEffectName(char* name) {
+    vst_strncpy(name, "Aura", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Aura::getPlugCategory() {return kPlugCategEffect;}
+
+bool Aura::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Aura", kVstMaxProductStrLen); return true;
+}
+
+bool Aura::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Aura/Aura.h b/plugins/LinuxVST/src/Aura/Aura.h
new file mode 100755
index 0000000..557169d
--- /dev/null
+++ b/plugins/LinuxVST/src/Aura/Aura.h
@@ -0,0 +1,76 @@
+/* ========================================
+ *  Aura - Aura.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Aura_H
+#define __Aura_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+  kNumParameters = 2
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'aura';    //Change this to what the AU identity is!
+
+class Aura : 
+    public AudioEffectX 
+{
+public:
+    Aura(audioMasterCallback audioMaster);
+    ~Aura();
+    virtual bool getEffectName(char* name);                       // The plug-in name
+    virtual VstPlugCategory getPlugCategory();                    // The general category for the plug-in
+    virtual bool getProductString(char* text);                    // This is a unique plug-in string provided by Steinberg
+    virtual bool getVendorString(char* text);                     // Vendor info
+    virtual VstInt32 getVendorVersion();                          // Version number
+    virtual void processReplacing (float** inputs, float** outputs, VstInt32 sampleFrames);
+    virtual void processDoubleReplacing (double** inputs, double** outputs, VstInt32 sampleFrames);
+    virtual void getProgramName(char *name);                      // read the name from the host
+    virtual void setProgramName(char *name);                      // changes the name of the preset displayed in the host
+	virtual VstInt32 getChunk (void** data, bool isPreset);
+	virtual VstInt32 setChunk (void* data, VstInt32 byteSize, bool isPreset);
+    virtual float getParameter(VstInt32 index);                   // get the parameter value at the specified index
+    virtual void setParameter(VstInt32 index, float value);       // set the parameter at index to value
+    virtual void getParameterLabel(VstInt32 index, char *text);  // label for the parameter (eg dB)
+    virtual void getParameterName(VstInt32 index, char *text);    // name of the parameter
+    virtual void getParameterDisplay(VstInt32 index, char *text); // text description of the current value    
+    virtual VstInt32 canDo(char *text);
+private:
+    char _programName[kVstMaxProgNameLen + 1];
+    std::set< std::string > _canDo;
+    
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool fpFlip;
+	//default stuff
+	long double lastSampleL;
+	double previousVelocityL;
+	long double lastSampleR;
+	double previousVelocityR;
+	
+	double bL[21];
+	double bR[21];
+	double f[21];		
+	
+
+    float A;
+    float B;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Aura/AuraProc.cpp b/plugins/LinuxVST/src/Aura/AuraProc.cpp
new file mode 100755
index 0000000..daf54d6
--- /dev/null
+++ b/plugins/LinuxVST/src/Aura/AuraProc.cpp
@@ -0,0 +1,476 @@
+/* ========================================
+ *  Aura - Aura.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Aura_H
+#include "Aura.h"
+#endif
+
+void Aura::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 correctionL;
+	double correctionR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double velocityL;
+	double velocityR;
+	double trim = A;
+	double wet = B;
+	double dry = 1.0 - wet;
+	double overallscale = trim * 10.0;
+	double gain = overallscale + (pow(wet,3) * 0.187859642462067);
+	trim *= (1.0 - (pow(wet,3) * 0.187859642462067));
+	long double inputSampleL;
+	long double inputSampleR;
+	double drySampleL;
+	double drySampleR;
+	
+	if (gain < 1.0) gain = 1.0;
+	if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[10] = 1.0; gain -= 1.0;} else {f[10] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[11] = 1.0; gain -= 1.0;} else {f[11] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[12] = 1.0; gain -= 1.0;} else {f[12] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[13] = 1.0; gain -= 1.0;} else {f[13] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[14] = 1.0; gain -= 1.0;} else {f[14] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[15] = 1.0; gain -= 1.0;} else {f[15] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[16] = 1.0; gain -= 1.0;} else {f[16] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[17] = 1.0; gain -= 1.0;} else {f[17] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[18] = 1.0; gain -= 1.0;} else {f[18] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[19] = 1.0; gain -= 1.0;} else {f[19] = gain; gain = 0.0;}
+	
+	//there, now we have a neat little moving average with remainders
+	
+	if (overallscale < 1.0) overallscale = 1.0;
+	f[0] /= overallscale;
+	f[1] /= overallscale;
+	f[2] /= overallscale;
+	f[3] /= overallscale;
+	f[4] /= overallscale;
+	f[5] /= overallscale;
+	f[6] /= overallscale;
+	f[7] /= overallscale;
+	f[8] /= overallscale;
+	f[9] /= overallscale;
+	f[10] /= overallscale;
+	f[11] /= overallscale;
+	f[12] /= overallscale;
+	f[13] /= overallscale;
+	f[14] /= overallscale;
+	f[15] /= overallscale;
+	f[16] /= overallscale;
+	f[17] /= overallscale;
+	f[18] /= overallscale;
+	f[19] /= overallscale;
+	//and now it's neatly scaled, too
+    
+    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;
+
+		velocityL = lastSampleL - inputSampleL;
+		correctionL = previousVelocityL - velocityL;
+		
+		bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15];
+		bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11];
+		bL[11] = bL[10]; bL[10] = bL[9];
+		bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
+		bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = accumulatorSampleL = correctionL;
+		
+		//we are accumulating rates of change of the rate of change
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleL += (bL[1] * f[1]);
+		accumulatorSampleL += (bL[2] * f[2]);
+		accumulatorSampleL += (bL[3] * f[3]);
+		accumulatorSampleL += (bL[4] * f[4]);
+		accumulatorSampleL += (bL[5] * f[5]);
+		accumulatorSampleL += (bL[6] * f[6]);
+		accumulatorSampleL += (bL[7] * f[7]);
+		accumulatorSampleL += (bL[8] * f[8]);
+		accumulatorSampleL += (bL[9] * f[9]);
+		accumulatorSampleL += (bL[10] * f[10]);
+		accumulatorSampleL += (bL[11] * f[11]);
+		accumulatorSampleL += (bL[12] * f[12]);
+		accumulatorSampleL += (bL[13] * f[13]);
+		accumulatorSampleL += (bL[14] * f[14]);
+		accumulatorSampleL += (bL[15] * f[15]);
+		accumulatorSampleL += (bL[16] * f[16]);
+		accumulatorSampleL += (bL[17] * f[17]);
+		accumulatorSampleL += (bL[18] * f[18]);
+		accumulatorSampleL += (bL[19] * f[19]);
+		
+		velocityL = previousVelocityL + accumulatorSampleL;		
+		inputSampleL = lastSampleL + velocityL;
+		lastSampleL = inputSampleL;
+		previousVelocityL = -velocityL * pow(trim,2);	
+		//left channel done
+		
+		velocityR = lastSampleR - inputSampleR;
+		correctionR = previousVelocityR - velocityR;
+		
+		bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15];
+		bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11];
+		bR[11] = bR[10]; bR[10] = bR[9];
+		bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
+		bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = accumulatorSampleR = correctionR;
+		
+		//we are accumulating rates of change of the rate of change
+				
+		accumulatorSampleR *= f[0];
+		accumulatorSampleR += (bR[1] * f[1]);
+		accumulatorSampleR += (bR[2] * f[2]);
+		accumulatorSampleR += (bR[3] * f[3]);
+		accumulatorSampleR += (bR[4] * f[4]);
+		accumulatorSampleR += (bR[5] * f[5]);
+		accumulatorSampleR += (bR[6] * f[6]);
+		accumulatorSampleR += (bR[7] * f[7]);
+		accumulatorSampleR += (bR[8] * f[8]);
+		accumulatorSampleR += (bR[9] * f[9]);
+		accumulatorSampleR += (bR[10] * f[10]);
+		accumulatorSampleR += (bR[11] * f[11]);
+		accumulatorSampleR += (bR[12] * f[12]);
+		accumulatorSampleR += (bR[13] * f[13]);
+		accumulatorSampleR += (bR[14] * f[14]);
+		accumulatorSampleR += (bR[15] * f[15]);
+		accumulatorSampleR += (bR[16] * f[16]);
+		accumulatorSampleR += (bR[17] * f[17]);
+		accumulatorSampleR += (bR[18] * f[18]);
+		accumulatorSampleR += (bR[19] * f[19]);
+		//we are doing our repetitive calculations on a separate value
+		
+		velocityR = previousVelocityR + accumulatorSampleR;		
+		inputSampleR = lastSampleR + velocityR;
+		lastSampleR = inputSampleR;
+		previousVelocityR = -velocityR * pow(trim,2);		
+		//right channel done
+				
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		}		
+		
+		//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 Aura::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;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	
+	double correctionL;
+	double correctionR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double velocityL;
+	double velocityR;
+	double trim = A;
+	double wet = B;
+	double dry = 1.0 - wet;
+	double overallscale = trim * 10.0;
+	double gain = overallscale + (pow(wet,3) * 0.187859642462067);
+	trim *= (1.0 - (pow(wet,3) * 0.187859642462067));
+	long double inputSampleL;
+	long double inputSampleR;
+	double drySampleL;
+	double drySampleR;
+	
+	if (gain < 1.0) gain = 1.0;
+	if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[10] = 1.0; gain -= 1.0;} else {f[10] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[11] = 1.0; gain -= 1.0;} else {f[11] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[12] = 1.0; gain -= 1.0;} else {f[12] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[13] = 1.0; gain -= 1.0;} else {f[13] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[14] = 1.0; gain -= 1.0;} else {f[14] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[15] = 1.0; gain -= 1.0;} else {f[15] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[16] = 1.0; gain -= 1.0;} else {f[16] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[17] = 1.0; gain -= 1.0;} else {f[17] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[18] = 1.0; gain -= 1.0;} else {f[18] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[19] = 1.0; gain -= 1.0;} else {f[19] = gain; gain = 0.0;}
+	
+	//there, now we have a neat little moving average with remainders
+	
+	if (overallscale < 1.0) overallscale = 1.0;
+	f[0] /= overallscale;
+	f[1] /= overallscale;
+	f[2] /= overallscale;
+	f[3] /= overallscale;
+	f[4] /= overallscale;
+	f[5] /= overallscale;
+	f[6] /= overallscale;
+	f[7] /= overallscale;
+	f[8] /= overallscale;
+	f[9] /= overallscale;
+	f[10] /= overallscale;
+	f[11] /= overallscale;
+	f[12] /= overallscale;
+	f[13] /= overallscale;
+	f[14] /= overallscale;
+	f[15] /= overallscale;
+	f[16] /= overallscale;
+	f[17] /= overallscale;
+	f[18] /= overallscale;
+	f[19] /= overallscale;
+	//and now it's neatly scaled, too
+    
+    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;
+		
+		velocityL = lastSampleL - inputSampleL;
+		correctionL = previousVelocityL - velocityL;
+		
+		bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15];
+		bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11];
+		bL[11] = bL[10]; bL[10] = bL[9];
+		bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
+		bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = accumulatorSampleL = correctionL;
+		
+		//we are accumulating rates of change of the rate of change
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleL += (bL[1] * f[1]);
+		accumulatorSampleL += (bL[2] * f[2]);
+		accumulatorSampleL += (bL[3] * f[3]);
+		accumulatorSampleL += (bL[4] * f[4]);
+		accumulatorSampleL += (bL[5] * f[5]);
+		accumulatorSampleL += (bL[6] * f[6]);
+		accumulatorSampleL += (bL[7] * f[7]);
+		accumulatorSampleL += (bL[8] * f[8]);
+		accumulatorSampleL += (bL[9] * f[9]);
+		accumulatorSampleL += (bL[10] * f[10]);
+		accumulatorSampleL += (bL[11] * f[11]);
+		accumulatorSampleL += (bL[12] * f[12]);
+		accumulatorSampleL += (bL[13] * f[13]);
+		accumulatorSampleL += (bL[14] * f[14]);
+		accumulatorSampleL += (bL[15] * f[15]);
+		accumulatorSampleL += (bL[16] * f[16]);
+		accumulatorSampleL += (bL[17] * f[17]);
+		accumulatorSampleL += (bL[18] * f[18]);
+		accumulatorSampleL += (bL[19] * f[19]);
+		
+		velocityL = previousVelocityL + accumulatorSampleL;		
+		inputSampleL = lastSampleL + velocityL;
+		lastSampleL = inputSampleL;
+		previousVelocityL = -velocityL * pow(trim,2);	
+		//left channel done
+		
+		velocityR = lastSampleR - inputSampleR;
+		correctionR = previousVelocityR - velocityR;
+		
+		bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15];
+		bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11];
+		bR[11] = bR[10]; bR[10] = bR[9];
+		bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
+		bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = accumulatorSampleR = correctionR;
+		
+		//we are accumulating rates of change of the rate of change
+		
+		accumulatorSampleR *= f[0];
+		accumulatorSampleR += (bR[1] * f[1]);
+		accumulatorSampleR += (bR[2] * f[2]);
+		accumulatorSampleR += (bR[3] * f[3]);
+		accumulatorSampleR += (bR[4] * f[4]);
+		accumulatorSampleR += (bR[5] * f[5]);
+		accumulatorSampleR += (bR[6] * f[6]);
+		accumulatorSampleR += (bR[7] * f[7]);
+		accumulatorSampleR += (bR[8] * f[8]);
+		accumulatorSampleR += (bR[9] * f[9]);
+		accumulatorSampleR += (bR[10] * f[10]);
+		accumulatorSampleR += (bR[11] * f[11]);
+		accumulatorSampleR += (bR[12] * f[12]);
+		accumulatorSampleR += (bR[13] * f[13]);
+		accumulatorSampleR += (bR[14] * f[14]);
+		accumulatorSampleR += (bR[15] * f[15]);
+		accumulatorSampleR += (bR[16] * f[16]);
+		accumulatorSampleR += (bR[17] * f[17]);
+		accumulatorSampleR += (bR[18] * f[18]);
+		accumulatorSampleR += (bR[19] * f[19]);
+		//we are doing our repetitive calculations on a separate value
+		
+		velocityR = previousVelocityR + accumulatorSampleR;		
+		inputSampleR = lastSampleR + velocityR;
+		lastSampleR = inputSampleR;
+		previousVelocityR = -velocityR * pow(trim,2);		
+		//right channel done
+		
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		}		
+		
+		//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++;
+    }
+}
\ No newline at end of file