UnBox, and build folder cleanups

Note that I am compelled to break VST builds on Windows because my build folders have had some files in 'em: you'll have to find those yourself. Here's hoping anyone trying to develop for Windows VST (or any VST) is able to find what they need, but I cannot help you set up a build environment, I can only give you my code for the audio part.

4 files changed, 711 insertions, 0 deletions

diff --git a/plugins/LinuxVST/CMakeLists.txt b/plugins/LinuxVST/CMakeLists.txt
index 73e7ca2..9e8fa4e 100755
--- a/plugins/LinuxVST/CMakeLists.txt
+++ b/plugins/LinuxVST/CMakeLists.txt
@@ -57,6 +57,7 @@ add_airwindows_plugin(StereoFX)
 add_airwindows_plugin(SubsOnly)
 add_airwindows_plugin(TapeDither)
 add_airwindows_plugin(TPDFDither)
+add_airwindows_plugin(UnBox)
 add_airwindows_plugin(VoiceOfTheStarship)
 add_airwindows_plugin(VariMu)
 add_airwindows_plugin(VinylDither)
diff --git a/plugins/LinuxVST/src/UnBox/UnBox.cpp b/plugins/LinuxVST/src/UnBox/UnBox.cpp
new file mode 100755
index 0000000..bb2c7d4
--- /dev/null
+++ b/plugins/LinuxVST/src/UnBox/UnBox.cpp
@@ -0,0 +1,143 @@
+/* ========================================
+ *  UnBox - UnBox.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __UnBox_H
+#include "UnBox.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new UnBox(audioMaster);}
+
+UnBox::UnBox(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.0;
+	C = 0.5;
+	
+	for(int count = 0; count < 5; count++) {aL[count] = 0.0; bL[count] = 0.0; aR[count] = 0.0; bR[count] = 0.0; e[count] = 0.0;}
+	for(int count = 0; count < 11; count++) {cL[count] = 0.0; cR[count] = 0.0; f[count] = 0.0;}
+	iirSampleAL = 0.0;
+	iirSampleBL = 0.0;
+	iirSampleAR = 0.0;
+	iirSampleBR = 0.0;
+	
+	fpNShapeL = 0.0;
+	fpNShapeR = 0.0;
+	//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
+}
+
+UnBox::~UnBox() {}
+VstInt32 UnBox::getVendorVersion () {return 1000;}
+void UnBox::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void UnBox::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 UnBox::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	chunkData[2] = C;
+	/* 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 UnBox::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	C = pinParameter(chunkData[2]);
+	/* 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 UnBox::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float UnBox::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        case kParamC: return C; 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 UnBox::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Input", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "UnBox", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Output", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void UnBox::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A*2.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string (C*2.0, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void UnBox::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 UnBox::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool UnBox::getEffectName(char* name) {
+    vst_strncpy(name, "UnBox", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory UnBox::getPlugCategory() {return kPlugCategEffect;}
+
+bool UnBox::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows UnBox", kVstMaxProductStrLen); return true;
+}
+
+bool UnBox::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/UnBox/UnBox.h b/plugins/LinuxVST/src/UnBox/UnBox.h
new file mode 100755
index 0000000..61db142
--- /dev/null
+++ b/plugins/LinuxVST/src/UnBox/UnBox.h
@@ -0,0 +1,79 @@
+/* ========================================
+ *  UnBox - UnBox.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __UnBox_H
+#define __UnBox_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+	kParamC = 2,
+  kNumParameters = 3
+};
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'unbx';    //Change this to what the AU identity is!
+
+class UnBox : 
+    public AudioEffectX 
+{
+public:
+    UnBox(audioMasterCallback audioMaster);
+    ~UnBox();
+    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 fpNShapeL;
+	long double fpNShapeR;
+	//default stuff
+	long double aL[5];
+	long double bL[5];
+	long double cL[11];
+	long double aR[5];
+	long double bR[5];
+	long double cR[11];
+	long double e[5];
+	long double f[11];
+	long double iirSampleAL;
+	long double iirSampleBL;
+	long double iirSampleAR;
+	long double iirSampleBR;
+	
+
+    float A;
+    float B;
+    float C;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/UnBox/UnBoxProc.cpp b/plugins/LinuxVST/src/UnBox/UnBoxProc.cpp
new file mode 100755
index 0000000..12133dc
--- /dev/null
+++ b/plugins/LinuxVST/src/UnBox/UnBoxProc.cpp
@@ -0,0 +1,488 @@
+/* ========================================
+ *  UnBox - UnBox.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __UnBox_H
+#include "UnBox.h"
+#endif
+
+void UnBox::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+
+	double input = A*2.0;
+	double unbox = B+1.0;
+	unbox *= unbox; //let's get some more gain into this
+	double iirAmount = (unbox*0.00052)/overallscale;
+	double output = C*2.0;
+	
+	double treble = unbox; //averaging taps 1-4
+	double gain = treble;
+	if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;}
+	//there, now we have a neat little moving average with remainders
+	if (treble < 1.0) treble = 1.0;
+	e[0] /= treble;
+	e[1] /= treble;
+	e[2] /= treble;
+	e[3] /= treble;
+	e[4] /= treble;
+	//and now it's neatly scaled, too
+	
+	treble = unbox*2.0; //averaging taps 1-8
+	gain = treble;
+	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;}
+	//there, now we have a neat little moving average with remainders
+	if (treble < 1.0) treble = 1.0;
+	f[0] /= treble;
+	f[1] /= treble;
+	f[2] /= treble;
+	f[3] /= treble;
+	f[4] /= treble;
+	f[5] /= treble;
+	f[6] /= treble;
+	f[7] /= treble;
+	f[8] /= treble;
+	f[9] /= treble;
+	//and now it's neatly scaled, too
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;}
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1];
+		aL[1] = aL[0]; aL[0] = inputSampleL;
+		inputSampleL *= e[0];
+		inputSampleL += (aL[1] * e[1]);
+		inputSampleL += (aL[2] * e[2]);
+		inputSampleL += (aL[3] * e[3]);
+		inputSampleL += (aL[4] * e[4]);
+		//this is now an average of inputSampleL
+
+		aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1];
+		aR[1] = aR[0]; aR[0] = inputSampleR;
+		inputSampleR *= e[0];
+		inputSampleR += (aR[1] * e[1]);
+		inputSampleR += (aR[2] * e[2]);
+		inputSampleR += (aR[3] * e[3]);
+		inputSampleR += (aR[4] * e[4]);
+		//this is now an average of inputSampleR
+		
+		bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = inputSampleL;
+		inputSampleL *= e[0];
+		inputSampleL += (bL[1] * e[1]);
+		inputSampleL += (bL[2] * e[2]);
+		inputSampleL += (bL[3] * e[3]);
+		inputSampleL += (bL[4] * e[4]);
+		//this is now an average of an average of inputSampleL. Two poles
+		
+		bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = inputSampleR;
+		inputSampleR *= e[0];
+		inputSampleR += (bR[1] * e[1]);
+		inputSampleR += (bR[2] * e[2]);
+		inputSampleR += (bR[3] * e[3]);
+		inputSampleR += (bR[4] * e[4]);
+		//this is now an average of an average of inputSampleR. Two poles
+		
+		inputSampleL *= unbox;
+		inputSampleR *= unbox;
+		//clip to 1.2533141373155 to reach maximum output
+		if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
+		if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
+		inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
+		
+		if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
+		if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
+		inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
+		
+		inputSampleL /= unbox;	
+		inputSampleR /= unbox;	
+		//now we have a distorted inputSample at the correct volume relative to drySample
+		
+		long double accumulatorSampleL = (drySampleL - inputSampleL);
+		cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5];
+		cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1];
+		cL[1] = cL[0]; cL[0] = accumulatorSampleL;
+		accumulatorSampleL *= f[0];
+		accumulatorSampleL += (cL[1] * f[1]);
+		accumulatorSampleL += (cL[2] * f[2]);
+		accumulatorSampleL += (cL[3] * f[3]);
+		accumulatorSampleL += (cL[4] * f[4]);
+		accumulatorSampleL += (cL[5] * f[5]);
+		accumulatorSampleL += (cL[6] * f[6]);
+		accumulatorSampleL += (cL[7] * f[7]);
+		accumulatorSampleL += (cL[8] * f[8]);
+		accumulatorSampleL += (cL[9] * f[9]);
+		//this is now an average of all the recent variances from dry
+		
+		long double accumulatorSampleR = (drySampleR - inputSampleR);
+		cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5];
+		cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1];
+		cR[1] = cR[0]; cR[0] = accumulatorSampleR;
+		accumulatorSampleR *= f[0];
+		accumulatorSampleR += (cR[1] * f[1]);
+		accumulatorSampleR += (cR[2] * f[2]);
+		accumulatorSampleR += (cR[3] * f[3]);
+		accumulatorSampleR += (cR[4] * f[4]);
+		accumulatorSampleR += (cR[5] * f[5]);
+		accumulatorSampleR += (cR[6] * f[6]);
+		accumulatorSampleR += (cR[7] * f[7]);
+		accumulatorSampleR += (cR[8] * f[8]);
+		accumulatorSampleR += (cR[9] * f[9]);
+		//this is now an average of all the recent variances from dry
+		
+		iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
+		accumulatorSampleL -= iirSampleAL;
+		//two poles of IIR
+
+		iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
+		accumulatorSampleR -= iirSampleAR;
+		//two poles of IIR
+
+		iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
+		accumulatorSampleL -= iirSampleBL;
+		//highpass section
+
+		iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
+		accumulatorSampleR -= iirSampleBR;
+		//highpass section
+		//this is now a highpassed average of all the recent variances from dry
+		
+		inputSampleL = drySampleL - accumulatorSampleL;
+		inputSampleR = drySampleR - accumulatorSampleR;
+		//we apply it as one operation, to get the result.
+		
+		if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;}
+		
+		//noise shaping to 32-bit floating point
+		float fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 32 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}
+
+void UnBox::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	
+	double input = A*2.0;
+	double unbox = B+1.0;
+	unbox *= unbox; //let's get some more gain into this
+	double iirAmount = (unbox*0.00052)/overallscale;
+	double output = C*2.0;
+	
+	double treble = unbox; //averaging taps 1-4
+	double gain = treble;
+	if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;}
+	//there, now we have a neat little moving average with remainders
+	if (treble < 1.0) treble = 1.0;
+	e[0] /= treble;
+	e[1] /= treble;
+	e[2] /= treble;
+	e[3] /= treble;
+	e[4] /= treble;
+	//and now it's neatly scaled, too
+	
+	treble = unbox*2.0; //averaging taps 1-8
+	gain = treble;
+	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;}
+	//there, now we have a neat little moving average with remainders
+	if (treble < 1.0) treble = 1.0;
+	f[0] /= treble;
+	f[1] /= treble;
+	f[2] /= treble;
+	f[3] /= treble;
+	f[4] /= treble;
+	f[5] /= treble;
+	f[6] /= treble;
+	f[7] /= treble;
+	f[8] /= treble;
+	f[9] /= treble;
+	//and now it's neatly scaled, too
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;}
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1];
+		aL[1] = aL[0]; aL[0] = inputSampleL;
+		inputSampleL *= e[0];
+		inputSampleL += (aL[1] * e[1]);
+		inputSampleL += (aL[2] * e[2]);
+		inputSampleL += (aL[3] * e[3]);
+		inputSampleL += (aL[4] * e[4]);
+		//this is now an average of inputSampleL
+		
+		aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1];
+		aR[1] = aR[0]; aR[0] = inputSampleR;
+		inputSampleR *= e[0];
+		inputSampleR += (aR[1] * e[1]);
+		inputSampleR += (aR[2] * e[2]);
+		inputSampleR += (aR[3] * e[3]);
+		inputSampleR += (aR[4] * e[4]);
+		//this is now an average of inputSampleR
+		
+		bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = inputSampleL;
+		inputSampleL *= e[0];
+		inputSampleL += (bL[1] * e[1]);
+		inputSampleL += (bL[2] * e[2]);
+		inputSampleL += (bL[3] * e[3]);
+		inputSampleL += (bL[4] * e[4]);
+		//this is now an average of an average of inputSampleL. Two poles
+		
+		bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = inputSampleR;
+		inputSampleR *= e[0];
+		inputSampleR += (bR[1] * e[1]);
+		inputSampleR += (bR[2] * e[2]);
+		inputSampleR += (bR[3] * e[3]);
+		inputSampleR += (bR[4] * e[4]);
+		//this is now an average of an average of inputSampleR. Two poles
+		
+		inputSampleL *= unbox;
+		inputSampleR *= unbox;
+		//clip to 1.2533141373155 to reach maximum output
+		if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
+		if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
+		inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
+		
+		if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
+		if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
+		inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
+		
+		inputSampleL /= unbox;	
+		inputSampleR /= unbox;	
+		//now we have a distorted inputSample at the correct volume relative to drySample
+		
+		long double accumulatorSampleL = (drySampleL - inputSampleL);
+		cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5];
+		cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1];
+		cL[1] = cL[0]; cL[0] = accumulatorSampleL;
+		accumulatorSampleL *= f[0];
+		accumulatorSampleL += (cL[1] * f[1]);
+		accumulatorSampleL += (cL[2] * f[2]);
+		accumulatorSampleL += (cL[3] * f[3]);
+		accumulatorSampleL += (cL[4] * f[4]);
+		accumulatorSampleL += (cL[5] * f[5]);
+		accumulatorSampleL += (cL[6] * f[6]);
+		accumulatorSampleL += (cL[7] * f[7]);
+		accumulatorSampleL += (cL[8] * f[8]);
+		accumulatorSampleL += (cL[9] * f[9]);
+		//this is now an average of all the recent variances from dry
+		
+		long double accumulatorSampleR = (drySampleR - inputSampleR);
+		cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5];
+		cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1];
+		cR[1] = cR[0]; cR[0] = accumulatorSampleR;
+		accumulatorSampleR *= f[0];
+		accumulatorSampleR += (cR[1] * f[1]);
+		accumulatorSampleR += (cR[2] * f[2]);
+		accumulatorSampleR += (cR[3] * f[3]);
+		accumulatorSampleR += (cR[4] * f[4]);
+		accumulatorSampleR += (cR[5] * f[5]);
+		accumulatorSampleR += (cR[6] * f[6]);
+		accumulatorSampleR += (cR[7] * f[7]);
+		accumulatorSampleR += (cR[8] * f[8]);
+		accumulatorSampleR += (cR[9] * f[9]);
+		//this is now an average of all the recent variances from dry
+		
+		iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
+		accumulatorSampleL -= iirSampleAL;
+		//two poles of IIR
+		
+		iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
+		accumulatorSampleR -= iirSampleAR;
+		//two poles of IIR
+		
+		iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
+		accumulatorSampleL -= iirSampleBL;
+		//highpass section
+		
+		iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
+		accumulatorSampleR -= iirSampleBR;
+		//highpass section
+		//this is now a highpassed average of all the recent variances from dry
+		
+		inputSampleL = drySampleL - accumulatorSampleL;
+		inputSampleR = drySampleR - accumulatorSampleR;
+		//we apply it as one operation, to get the result.
+		
+		if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;}
+		
+		//noise shaping to 64-bit floating point
+		double fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 64 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}