Pop

3 files changed, 850 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/Pop/Pop.cpp b/plugins/LinuxVST/src/Pop/Pop.cpp
new file mode 100755
index 0000000..049d9ed
--- /dev/null
+++ b/plugins/LinuxVST/src/Pop/Pop.cpp
@@ -0,0 +1,163 @@
+/* ========================================
+ *  Pop - Pop.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Pop_H
+#include "Pop.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Pop(audioMaster);}
+
+Pop::Pop(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.3;
+	B = 1.0;
+	C = 1.0;
+	fpNShapeL = 0.0;
+	fpNShapeR = 0.0;
+	
+	for(int count = 0; count < 10000; count++) {dL[count] = 0; dR[count] = 0;}
+	delay = 0;
+	flip = false;
+	
+	muSpeedAL = 10000;
+	muSpeedBL = 10000;
+	muCoefficientAL = 1;
+	muCoefficientBL = 1;
+	thickenL = 1;
+	muVaryL = 1;
+	previousL = 0.0;
+	previous2L = 0.0;
+	previous3L = 0.0;
+	previous4L = 0.0;
+	previous5L = 0.0;
+
+	muSpeedAR = 10000;
+	muSpeedBR = 10000;
+	muCoefficientAR = 1;
+	muCoefficientBR = 1;
+	thickenR = 1;
+	muVaryR = 1;
+	previousR = 0.0;
+	previous2R = 0.0;
+	previous3R = 0.0;
+	previous4R = 0.0;
+	previous5R = 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
+}
+
+Pop::~Pop() {}
+VstInt32 Pop::getVendorVersion () {return 1000;}
+void Pop::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Pop::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 Pop::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 Pop::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 Pop::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 Pop::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 Pop::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Intenst", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Output", kVstMaxParamStrLen); break;
+		case kParamC: 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 Pop::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string (C, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Pop::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 Pop::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Pop::getEffectName(char* name) {
+    vst_strncpy(name, "Pop", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Pop::getPlugCategory() {return kPlugCategEffect;}
+
+bool Pop::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Pop", kVstMaxProductStrLen); return true;
+}
+
+bool Pop::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Pop/Pop.h b/plugins/LinuxVST/src/Pop/Pop.h
new file mode 100755
index 0000000..cebafa6
--- /dev/null
+++ b/plugins/LinuxVST/src/Pop/Pop.h
@@ -0,0 +1,99 @@
+/* ========================================
+ *  Pop - Pop.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Pop_H
+#define __Pop_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 = 'popc';    //Change this to what the AU identity is!
+
+class Pop : 
+    public AudioEffectX 
+{
+public:
+    Pop(audioMasterCallback audioMaster);
+    ~Pop();
+    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
+	
+	double muVaryL;
+	double muAttackL;
+	double muNewSpeedL;
+	double muSpeedAL;
+	double muSpeedBL;
+	double muCoefficientAL;
+	double muCoefficientBL;
+	long double thickenL;
+	long double previousL;
+	long double previous2L;
+	long double previous3L;
+	long double previous4L;
+	long double previous5L;
+	double dL[10001];
+	
+	double muVaryR;
+	double muAttackR;
+	double muNewSpeedR;
+	double muSpeedAR;
+	double muSpeedBR;
+	double muCoefficientAR;
+	double muCoefficientBR;
+	long double thickenR;
+	long double previousR;
+	long double previous2R;
+	long double previous3R;
+	long double previous4R;
+	long double previous5R;
+	double dR[10001];
+	
+	int delay;
+	bool flip;	
+
+    float A;
+    float B;
+    float C;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Pop/PopProc.cpp b/plugins/LinuxVST/src/Pop/PopProc.cpp
new file mode 100755
index 0000000..ae7dd4d
--- /dev/null
+++ b/plugins/LinuxVST/src/Pop/PopProc.cpp
@@ -0,0 +1,588 @@
+/* ========================================
+ *  Pop - Pop.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Pop_H
+#include "Pop.h"
+#endif
+
+void Pop::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 highGainOffset = pow(A,2)*0.023;
+	double threshold = 1.001 - (1.0-pow(1.0-A,5));
+	double muMakeupGain = sqrt(1.0 / threshold);
+	//gain settings around threshold
+	double release = (A*100000.0) + 300000.0;
+	int maxdelay = (int)(1450.0 * overallscale);
+	if (maxdelay > 9999) maxdelay = 9999;
+	release /= overallscale;
+	double fastest = sqrt(release);
+	//speed settings around release
+	double output = B;
+	double wet = C;
+	// µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		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 aPop. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		dL[delay] = inputSampleL;
+		dR[delay] = inputSampleR;
+		delay--;
+		if (delay < 0 || delay > maxdelay) {delay = maxdelay;}
+		//yes this is a second bounds check. it's cheap, check EVERY time
+		inputSampleL = (inputSampleL * thickenL) + (dL[delay] * (1.0-thickenL));
+		inputSampleR = (inputSampleR * thickenR) + (dR[delay] * (1.0-thickenR));
+				
+		long double lowestSampleL = inputSampleL;
+		if (fabs(inputSampleL) > fabs(previousL)) lowestSampleL = previousL;
+		if (fabs(lowestSampleL) > fabs(previous2L)) lowestSampleL = (lowestSampleL + previous2L) / 1.99;
+		if (fabs(lowestSampleL) > fabs(previous3L)) lowestSampleL = (lowestSampleL + previous3L) / 1.98;
+		if (fabs(lowestSampleL) > fabs(previous4L)) lowestSampleL = (lowestSampleL + previous4L) / 1.97;
+		if (fabs(lowestSampleL) > fabs(previous5L)) lowestSampleL = (lowestSampleL + previous5L) / 1.96;
+		previous5L = previous4L;
+		previous4L = previous3L;
+		previous3L = previous2L;
+		previous2L = previousL;
+		previousL = inputSampleL;
+		inputSampleL *= muMakeupGain;
+		double punchinessL = 0.95-fabs(inputSampleL*0.08);
+		if (punchinessL < 0.65) punchinessL = 0.65;
+		
+		long double lowestSampleR = inputSampleR;
+		if (fabs(inputSampleR) > fabs(previousR)) lowestSampleR = previousR;
+		if (fabs(lowestSampleR) > fabs(previous2R)) lowestSampleR = (lowestSampleR + previous2R) / 1.99;
+		if (fabs(lowestSampleR) > fabs(previous3R)) lowestSampleR = (lowestSampleR + previous3R) / 1.98;
+		if (fabs(lowestSampleR) > fabs(previous4R)) lowestSampleR = (lowestSampleR + previous4R) / 1.97;
+		if (fabs(lowestSampleR) > fabs(previous5R)) lowestSampleR = (lowestSampleR + previous5R) / 1.96;
+		previous5R = previous4R;
+		previous4R = previous3R;
+		previous3R = previous2R;
+		previous2R = previousR;
+		previousR = inputSampleR;
+		inputSampleR *= muMakeupGain;
+		double punchinessR = 0.95-fabs(inputSampleR*0.08);
+		if (punchinessR < 0.65) punchinessR = 0.65;
+		
+		//adjust coefficients for L
+		if (flip)
+		{
+			if (fabs(lowestSampleL) > threshold)
+			{
+				muVaryL = threshold / fabs(lowestSampleL);
+				muAttackL = sqrt(fabs(muSpeedAL));
+				muCoefficientAL = muCoefficientAL * (muAttackL-1.0);
+				if (muVaryL < threshold)
+				{
+					muCoefficientAL = muCoefficientAL + threshold;
+				}
+				else
+				{
+					muCoefficientAL = muCoefficientAL + muVaryL;
+				}
+				muCoefficientAL = muCoefficientAL / muAttackL;
+			}
+			else
+			{
+				muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0);
+				muCoefficientAL = muCoefficientAL + 1.0;
+				muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL);
+			}
+			muNewSpeedL = muSpeedAL * (muSpeedAL-1);
+			muNewSpeedL = muNewSpeedL + fabs(lowestSampleL*release)+fastest;
+			muSpeedAL = muNewSpeedL / muSpeedAL;
+		}
+		else
+		{
+			if (fabs(lowestSampleL) > threshold)
+			{
+				muVaryL = threshold / fabs(lowestSampleL);
+				muAttackL = sqrt(fabs(muSpeedBL));
+				muCoefficientBL = muCoefficientBL * (muAttackL-1);
+				if (muVaryL < threshold)
+				{
+					muCoefficientBL = muCoefficientBL + threshold;
+				}
+				else
+				{
+					muCoefficientBL = muCoefficientBL + muVaryL;
+				}
+				muCoefficientBL = muCoefficientBL / muAttackL;
+			}
+			else
+			{
+				muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0);
+				muCoefficientBL = muCoefficientBL + 1.0;
+				muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL);
+			}
+			muNewSpeedL = muSpeedBL * (muSpeedBL-1);
+			muNewSpeedL = muNewSpeedL + fabs(lowestSampleL*release)+fastest;
+			muSpeedBL = muNewSpeedL / muSpeedBL;
+		}
+		//got coefficients, adjusted speeds for L
+		
+		//adjust coefficients for R
+		if (flip)
+		{
+			if (fabs(lowestSampleR) > threshold)
+			{
+				muVaryR = threshold / fabs(lowestSampleR);
+				muAttackR = sqrt(fabs(muSpeedAR));
+				muCoefficientAR = muCoefficientAR * (muAttackR-1.0);
+				if (muVaryR < threshold)
+				{
+					muCoefficientAR = muCoefficientAR + threshold;
+				}
+				else
+				{
+					muCoefficientAR = muCoefficientAR + muVaryR;
+				}
+				muCoefficientAR = muCoefficientAR / muAttackR;
+			}
+			else
+			{
+				muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0);
+				muCoefficientAR = muCoefficientAR + 1.0;
+				muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR);
+			}
+			muNewSpeedR = muSpeedAR * (muSpeedAR-1);
+			muNewSpeedR = muNewSpeedR + fabs(lowestSampleR*release)+fastest;
+			muSpeedAR = muNewSpeedR / muSpeedAR;
+		}
+		else
+		{
+			if (fabs(lowestSampleR) > threshold)
+			{
+				muVaryR = threshold / fabs(lowestSampleR);
+				muAttackR = sqrt(fabs(muSpeedBR));
+				muCoefficientBR = muCoefficientBR * (muAttackR-1);
+				if (muVaryR < threshold)
+				{
+					muCoefficientBR = muCoefficientBR + threshold;
+				}
+				else
+				{
+					muCoefficientBR = muCoefficientBR + muVaryR;
+				}
+				muCoefficientBR = muCoefficientBR / muAttackR;
+			}
+			else
+			{
+				muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0);
+				muCoefficientBR = muCoefficientBR + 1.0;
+				muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR);
+			}
+			muNewSpeedR = muSpeedBR * (muSpeedBR-1);
+			muNewSpeedR = muNewSpeedR + fabs(lowestSampleR*release)+fastest;
+			muSpeedBR = muNewSpeedR / muSpeedBR;
+		}
+		//got coefficients, adjusted speeds for R
+		
+		long double coefficientL = highGainOffset;
+		if (flip) coefficientL += pow(muCoefficientAL,2);
+		else coefficientL += pow(muCoefficientBL,2);
+		inputSampleL *= coefficientL;
+		thickenL = (coefficientL/5)+punchinessL;//0.80;
+		thickenL = (1.0-wet)+(wet*thickenL);
+
+		long double coefficientR = highGainOffset;
+		if (flip) coefficientR += pow(muCoefficientAR,2);
+		else coefficientR += pow(muCoefficientBR,2);
+		inputSampleR *= coefficientR;
+		thickenR = (coefficientR/5)+punchinessR;//0.80;
+		thickenR = (1.0-wet)+(wet*thickenR);
+		//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
+		//applied gain correction to control output level- tends to constrain sound rather than inflate it
+		
+		long double bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.2533141373155) bridgerectifier = 1.2533141373155;
+		bridgerectifier = sin(bridgerectifier * fabs(bridgerectifier)) / ((bridgerectifier == 0.0) ?1:fabs(bridgerectifier));
+		//using Spiral instead of Density algorithm
+		if (inputSampleL > 0) inputSampleL = (inputSampleL*coefficientL)+(bridgerectifier*(1-coefficientL));
+		else inputSampleL = (inputSampleL*coefficientL)-(bridgerectifier*(1-coefficientL));
+		//second stage of overdrive to prevent overs and allow bloody loud extremeness
+
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.2533141373155) bridgerectifier = 1.2533141373155;
+		bridgerectifier = sin(bridgerectifier * fabs(bridgerectifier)) / ((bridgerectifier == 0.0) ?1:fabs(bridgerectifier));
+		//using Spiral instead of Density algorithm
+		if (inputSampleR > 0) inputSampleR = (inputSampleR*coefficientR)+(bridgerectifier*(1-coefficientR));
+		else inputSampleR = (inputSampleR*coefficientR)-(bridgerectifier*(1-coefficientR));
+		//second stage of overdrive to prevent overs and allow bloody loud extremeness
+		
+		flip = !flip;
+		
+		if (output < 1.0) {inputSampleL *= output;inputSampleR *= output;}
+		if (wet<1.0) {
+			inputSampleL = (drySampleL*(1.0-wet))+(inputSampleL*wet);
+			inputSampleR = (drySampleR*(1.0-wet))+(inputSampleR*wet);
+		}
+				
+		//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 Pop::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 highGainOffset = pow(A,2)*0.023;
+	double threshold = 1.001 - (1.0-pow(1.0-A,5));
+	double muMakeupGain = sqrt(1.0 / threshold);
+	//gain settings around threshold
+	double release = (A*100000.0) + 300000.0;
+	int maxdelay = (int)(1450.0 * overallscale);
+	if (maxdelay > 9999) maxdelay = 9999;
+	release /= overallscale;
+	double fastest = sqrt(release);
+	//speed settings around release
+	double output = B;
+	double wet = C;
+	// µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		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 aPop. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		dL[delay] = inputSampleL;
+		dR[delay] = inputSampleR;
+		delay--;
+		if (delay < 0 || delay > maxdelay) {delay = maxdelay;}
+		//yes this is a second bounds check. it's cheap, check EVERY time
+		inputSampleL = (inputSampleL * thickenL) + (dL[delay] * (1.0-thickenL));
+		inputSampleR = (inputSampleR * thickenR) + (dR[delay] * (1.0-thickenR));
+		
+		long double lowestSampleL = inputSampleL;
+		if (fabs(inputSampleL) > fabs(previousL)) lowestSampleL = previousL;
+		if (fabs(lowestSampleL) > fabs(previous2L)) lowestSampleL = (lowestSampleL + previous2L) / 1.99;
+		if (fabs(lowestSampleL) > fabs(previous3L)) lowestSampleL = (lowestSampleL + previous3L) / 1.98;
+		if (fabs(lowestSampleL) > fabs(previous4L)) lowestSampleL = (lowestSampleL + previous4L) / 1.97;
+		if (fabs(lowestSampleL) > fabs(previous5L)) lowestSampleL = (lowestSampleL + previous5L) / 1.96;
+		previous5L = previous4L;
+		previous4L = previous3L;
+		previous3L = previous2L;
+		previous2L = previousL;
+		previousL = inputSampleL;
+		inputSampleL *= muMakeupGain;
+		double punchinessL = 0.95-fabs(inputSampleL*0.08);
+		if (punchinessL < 0.65) punchinessL = 0.65;
+		
+		long double lowestSampleR = inputSampleR;
+		if (fabs(inputSampleR) > fabs(previousR)) lowestSampleR = previousR;
+		if (fabs(lowestSampleR) > fabs(previous2R)) lowestSampleR = (lowestSampleR + previous2R) / 1.99;
+		if (fabs(lowestSampleR) > fabs(previous3R)) lowestSampleR = (lowestSampleR + previous3R) / 1.98;
+		if (fabs(lowestSampleR) > fabs(previous4R)) lowestSampleR = (lowestSampleR + previous4R) / 1.97;
+		if (fabs(lowestSampleR) > fabs(previous5R)) lowestSampleR = (lowestSampleR + previous5R) / 1.96;
+		previous5R = previous4R;
+		previous4R = previous3R;
+		previous3R = previous2R;
+		previous2R = previousR;
+		previousR = inputSampleR;
+		inputSampleR *= muMakeupGain;
+		double punchinessR = 0.95-fabs(inputSampleR*0.08);
+		if (punchinessR < 0.65) punchinessR = 0.65;
+		
+		//adjust coefficients for L
+		if (flip)
+		{
+			if (fabs(lowestSampleL) > threshold)
+			{
+				muVaryL = threshold / fabs(lowestSampleL);
+				muAttackL = sqrt(fabs(muSpeedAL));
+				muCoefficientAL = muCoefficientAL * (muAttackL-1.0);
+				if (muVaryL < threshold)
+				{
+					muCoefficientAL = muCoefficientAL + threshold;
+				}
+				else
+				{
+					muCoefficientAL = muCoefficientAL + muVaryL;
+				}
+				muCoefficientAL = muCoefficientAL / muAttackL;
+			}
+			else
+			{
+				muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0);
+				muCoefficientAL = muCoefficientAL + 1.0;
+				muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL);
+			}
+			muNewSpeedL = muSpeedAL * (muSpeedAL-1);
+			muNewSpeedL = muNewSpeedL + fabs(lowestSampleL*release)+fastest;
+			muSpeedAL = muNewSpeedL / muSpeedAL;
+		}
+		else
+		{
+			if (fabs(lowestSampleL) > threshold)
+			{
+				muVaryL = threshold / fabs(lowestSampleL);
+				muAttackL = sqrt(fabs(muSpeedBL));
+				muCoefficientBL = muCoefficientBL * (muAttackL-1);
+				if (muVaryL < threshold)
+				{
+					muCoefficientBL = muCoefficientBL + threshold;
+				}
+				else
+				{
+					muCoefficientBL = muCoefficientBL + muVaryL;
+				}
+				muCoefficientBL = muCoefficientBL / muAttackL;
+			}
+			else
+			{
+				muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0);
+				muCoefficientBL = muCoefficientBL + 1.0;
+				muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL);
+			}
+			muNewSpeedL = muSpeedBL * (muSpeedBL-1);
+			muNewSpeedL = muNewSpeedL + fabs(lowestSampleL*release)+fastest;
+			muSpeedBL = muNewSpeedL / muSpeedBL;
+		}
+		//got coefficients, adjusted speeds for L
+		
+		//adjust coefficients for R
+		if (flip)
+		{
+			if (fabs(lowestSampleR) > threshold)
+			{
+				muVaryR = threshold / fabs(lowestSampleR);
+				muAttackR = sqrt(fabs(muSpeedAR));
+				muCoefficientAR = muCoefficientAR * (muAttackR-1.0);
+				if (muVaryR < threshold)
+				{
+					muCoefficientAR = muCoefficientAR + threshold;
+				}
+				else
+				{
+					muCoefficientAR = muCoefficientAR + muVaryR;
+				}
+				muCoefficientAR = muCoefficientAR / muAttackR;
+			}
+			else
+			{
+				muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0);
+				muCoefficientAR = muCoefficientAR + 1.0;
+				muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR);
+			}
+			muNewSpeedR = muSpeedAR * (muSpeedAR-1);
+			muNewSpeedR = muNewSpeedR + fabs(lowestSampleR*release)+fastest;
+			muSpeedAR = muNewSpeedR / muSpeedAR;
+		}
+		else
+		{
+			if (fabs(lowestSampleR) > threshold)
+			{
+				muVaryR = threshold / fabs(lowestSampleR);
+				muAttackR = sqrt(fabs(muSpeedBR));
+				muCoefficientBR = muCoefficientBR * (muAttackR-1);
+				if (muVaryR < threshold)
+				{
+					muCoefficientBR = muCoefficientBR + threshold;
+				}
+				else
+				{
+					muCoefficientBR = muCoefficientBR + muVaryR;
+				}
+				muCoefficientBR = muCoefficientBR / muAttackR;
+			}
+			else
+			{
+				muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0);
+				muCoefficientBR = muCoefficientBR + 1.0;
+				muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR);
+			}
+			muNewSpeedR = muSpeedBR * (muSpeedBR-1);
+			muNewSpeedR = muNewSpeedR + fabs(lowestSampleR*release)+fastest;
+			muSpeedBR = muNewSpeedR / muSpeedBR;
+		}
+		//got coefficients, adjusted speeds for R
+		
+		long double coefficientL = highGainOffset;
+		if (flip) coefficientL += pow(muCoefficientAL,2);
+		else coefficientL += pow(muCoefficientBL,2);
+		inputSampleL *= coefficientL;
+		thickenL = (coefficientL/5)+punchinessL;//0.80;
+		thickenL = (1.0-wet)+(wet*thickenL);
+		
+		long double coefficientR = highGainOffset;
+		if (flip) coefficientR += pow(muCoefficientAR,2);
+		else coefficientR += pow(muCoefficientBR,2);
+		inputSampleR *= coefficientR;
+		thickenR = (coefficientR/5)+punchinessR;//0.80;
+		thickenR = (1.0-wet)+(wet*thickenR);
+		//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
+		//applied gain correction to control output level- tends to constrain sound rather than inflate it
+		
+		long double bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.2533141373155) bridgerectifier = 1.2533141373155;
+		bridgerectifier = sin(bridgerectifier * fabs(bridgerectifier)) / ((bridgerectifier == 0.0) ?1:fabs(bridgerectifier));
+		//using Spiral instead of Density algorithm
+		if (inputSampleL > 0) inputSampleL = (inputSampleL*coefficientL)+(bridgerectifier*(1-coefficientL));
+		else inputSampleL = (inputSampleL*coefficientL)-(bridgerectifier*(1-coefficientL));
+		//second stage of overdrive to prevent overs and allow bloody loud extremeness
+		
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.2533141373155) bridgerectifier = 1.2533141373155;
+		bridgerectifier = sin(bridgerectifier * fabs(bridgerectifier)) / ((bridgerectifier == 0.0) ?1:fabs(bridgerectifier));
+		//using Spiral instead of Density algorithm
+		if (inputSampleR > 0) inputSampleR = (inputSampleR*coefficientR)+(bridgerectifier*(1-coefficientR));
+		else inputSampleR = (inputSampleR*coefficientR)-(bridgerectifier*(1-coefficientR));
+		//second stage of overdrive to prevent overs and allow bloody loud extremeness
+		
+		flip = !flip;
+		
+		if (output < 1.0) {inputSampleL *= output;inputSampleR *= output;}
+		if (wet<1.0) {
+			inputSampleL = (drySampleL*(1.0-wet))+(inputSampleL*wet);
+			inputSampleR = (drySampleR*(1.0-wet))+(inputSampleR*wet);
+		}
+		
+		//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.	
+}