Updates (in case my plane crashes)

3 files changed, 1203 insertions, 0 deletions

diff --git a/plugins/MacVST/ADClip7/source/ADClip7.cpp b/plugins/MacVST/ADClip7/source/ADClip7.cpp
new file mode 100755
index 0000000..fb6d6c6
--- /dev/null
+++ b/plugins/MacVST/ADClip7/source/ADClip7.cpp
@@ -0,0 +1,165 @@
+/* ========================================
+ *  ADClip7 - ADClip7.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __ADClip7_H
+#include "ADClip7.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new ADClip7(audioMaster);}
+
+ADClip7::ADClip7(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 0.5;
+	C = 0.5;
+	D = 0.0;
+
+	lastSampleL = 0.0;
+	lastSampleR = 0.0;
+	for(int count = 0; count < 22199; count++) {bL[count] = 0; bR[count] = 0;}
+	gcount = 0;
+	lowsL = 0;
+	lowsR = 0;
+	refclipL = 0.99;
+	refclipR = 0.99;
+	iirLowsAL = 0.0;
+	iirLowsAR = 0.0;
+	iirLowsBL = 0.0;	
+	iirLowsBR = 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
+}
+
+ADClip7::~ADClip7() {}
+VstInt32 ADClip7::getVendorVersion () {return 1000;}
+void ADClip7::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void ADClip7::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 ADClip7::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	chunkData[2] = C;
+	chunkData[3] = D;
+	/* 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 ADClip7::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	C = pinParameter(chunkData[2]);
+	D = pinParameter(chunkData[3]);
+	/* 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 ADClip7::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        case kParamD: D = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float ADClip7::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        case kParamC: return C; break;
+        case kParamD: return D; 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 ADClip7::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Boost", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Soften", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Enhance", kVstMaxParamStrLen); break;
+		case kParamD: vst_strncpy (text, "Mode", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void ADClip7::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A*18.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string (C, text, kVstMaxParamStrLen); break;
+        case kParamD: switch((VstInt32)( D * 2.999 )) //0 to almost edge of # of params
+		{case 0: vst_strncpy (text, "Normal", kVstMaxParamStrLen); break;
+		 case 1: vst_strncpy (text, "Atten", kVstMaxParamStrLen); break;
+		 case 2: vst_strncpy (text, "Clips", kVstMaxParamStrLen); break;
+		 default: break; // unknown parameter, shouldn't happen!
+		} break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void ADClip7::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 ADClip7::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool ADClip7::getEffectName(char* name) {
+    vst_strncpy(name, "ADClip7", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory ADClip7::getPlugCategory() {return kPlugCategEffect;}
+
+bool ADClip7::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows ADClip7", kVstMaxProductStrLen); return true;
+}
+
+bool ADClip7::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/ADClip7/source/ADClip7.h b/plugins/MacVST/ADClip7/source/ADClip7.h
new file mode 100755
index 0000000..f20d3fb
--- /dev/null
+++ b/plugins/MacVST/ADClip7/source/ADClip7.h
@@ -0,0 +1,85 @@
+/* ========================================
+ *  ADClip7 - ADClip7.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __ADClip7_H
+#define __ADClip7_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+	kParamC = 2,
+	kParamD = 3,
+  kNumParameters = 4
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'adcr';    //Change this to what the AU identity is!
+
+class ADClip7 : 
+    public AudioEffectX 
+{
+public:
+    ADClip7(audioMasterCallback audioMaster);
+    ~ADClip7();
+    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;
+	long double lastSampleR;
+	float bL[22200];
+	float bR[22200];
+	int gcount;
+	double lowsL;
+	double lowsR;
+	double iirLowsAL;
+	double iirLowsAR;
+	double iirLowsBL;
+	double iirLowsBR;
+	long double refclipL;
+	long double refclipR;
+
+    float A;
+    float B;
+    float C;
+    float D;
+
+};
+
+#endif
diff --git a/plugins/MacVST/ADClip7/source/ADClip7Proc.cpp b/plugins/MacVST/ADClip7/source/ADClip7Proc.cpp
new file mode 100755
index 0000000..2705d61
--- /dev/null
+++ b/plugins/MacVST/ADClip7/source/ADClip7Proc.cpp
@@ -0,0 +1,953 @@
+/* ========================================
+ *  ADClip7 - ADClip7.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __ADClip7_H
+#include "ADClip7.h"
+#endif
+
+void ADClip7::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();
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double inputGain = pow(10.0,(A*18.0)/20.0);
+	double softness = B * fpNew;
+	double hardness = 1.0 - softness;
+	double highslift = 0.307 * C;
+	double adjust = pow(highslift,3) * 0.416;
+	double subslift = 0.796 * C;
+	double calibsubs = subslift/53;
+	double invcalibsubs = 1.0 - calibsubs;
+	double subs = 0.81 + (calibsubs*2);
+	long double bridgerectifier;
+	int mode = (int) floor(D*2.999)+1;
+	double overshootL;
+	double overshootR;
+	double offsetH1 = 1.84;
+	offsetH1 *= overallscale;
+	double offsetH2 = offsetH1 * 1.9;
+	double offsetH3 = offsetH1 * 2.7;
+	double offsetL1 = 612;
+	offsetL1 *= overallscale;
+	double offsetL2 = offsetL1 * 2.0;
+	int refH1 = (int)floor(offsetH1);
+	int refH2 = (int)floor(offsetH2);
+	int refH3 = (int)floor(offsetH3);
+	int refL1 = (int)floor(offsetL1);
+	int refL2 = (int)floor(offsetL2);
+	int temp;
+	double fractionH1 = offsetH1 - floor(offsetH1);
+	double fractionH2 = offsetH2 - floor(offsetH2);
+	double fractionH3 = offsetH3 - floor(offsetH3);
+	double minusH1 = 1.0 - fractionH1;
+	double minusH2 = 1.0 - fractionH2;
+	double minusH3 = 1.0 - fractionH3;
+	double highsL = 0.0;
+	double highsR = 0.0;
+	int count = 0;
+	
+	long double inputSampleL;
+	long double inputSampleR;
+	
+    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.
+		}
+		
+
+		
+		if (inputGain != 1.0) {
+			inputSampleL *= inputGain;
+			inputSampleR *= inputGain;
+		}
+		
+		overshootL = fabs(inputSampleL) - refclipL;
+		overshootR = fabs(inputSampleR) - refclipR;
+		if (overshootL < 0.0) overshootL = 0.0;
+		if (overshootR < 0.0) overshootR = 0.0;
+		
+		if (gcount < 0 || gcount > 11020) {gcount = 11020;}
+		count = gcount;
+		bL[count+11020] = bL[count] = overshootL;
+		bR[count+11020] = bR[count] = overshootR;
+		gcount--;
+		
+		if (highslift > 0.0)
+		{
+			//we have a big pile of b[] which is overshoots
+			temp = count+refH3;
+			highsL = -(bL[temp] * minusH3); //less as value moves away from .0
+			highsL -= bL[temp+1]; //we can assume always using this in one way or another?
+			highsL -= (bL[temp+2] * fractionH3); //greater as value moves away from .0
+			highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 3 is a negative add
+			highsR = -(bR[temp] * minusH3); //less as value moves away from .0
+			highsR -= bR[temp+1]; //we can assume always using this in one way or another?
+			highsR -= (bR[temp+2] * fractionH3); //greater as value moves away from .0
+			highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 3 is a negative add
+			temp = count+refH2;
+			highsL += (bL[temp] * minusH2); //less as value moves away from .0
+			highsL += bL[temp+1]; //we can assume always using this in one way or another?
+			highsL += (bL[temp+2] * fractionH2); //greater as value moves away from .0
+			highsL -= (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 2 is a positive feedback of the overshoot
+			highsR += (bR[temp] * minusH2); //less as value moves away from .0
+			highsR += bR[temp+1]; //we can assume always using this in one way or another?
+			highsR += (bR[temp+2] * fractionH2); //greater as value moves away from .0
+			highsR -= (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 2 is a positive feedback of the overshoot
+			temp = count+refH1;
+			highsL -= (bL[temp] * minusH1); //less as value moves away from .0
+			highsL -= bL[temp+1]; //we can assume always using this in one way or another?
+			highsL -= (bL[temp+2] * fractionH1); //greater as value moves away from .0
+			highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 1 is a negative feedback of the overshoot
+			highsR -= (bR[temp] * minusH1); //less as value moves away from .0
+			highsR -= bR[temp+1]; //we can assume always using this in one way or another?
+			highsR -= (bR[temp+2] * fractionH1); //greater as value moves away from .0
+			highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 1 is a negative feedback of the overshoot
+			//done with interpolated mostly negative feedback of the overshoot
+		}
+		
+		bridgerectifier = sin(fabs(highsL) * hardness);
+		//this will wrap around and is scaled back by softness
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (highsL > 0) highsL = bridgerectifier;
+		else highsL = -bridgerectifier;
+		
+		bridgerectifier = sin(fabs(highsR) * hardness);
+		//this will wrap around and is scaled back by softness
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (highsR > 0) highsR = bridgerectifier;
+		else highsR = -bridgerectifier;
+		
+		if (subslift > 0.0) 
+		{
+			lowsL *= subs;
+			lowsR *= subs;
+			//going in we'll reel back some of the swing
+			temp = count+refL1;
+			
+			lowsL -= bL[temp+127];
+			lowsL -= bL[temp+113];
+			lowsL -= bL[temp+109];
+			lowsL -= bL[temp+107];
+			lowsL -= bL[temp+103];
+			lowsL -= bL[temp+101];
+			lowsL -= bL[temp+97];
+			lowsL -= bL[temp+89];
+			lowsL -= bL[temp+83];
+			lowsL -= bL[temp+79];
+			lowsL -= bL[temp+73];
+			lowsL -= bL[temp+71];
+			lowsL -= bL[temp+67];
+			lowsL -= bL[temp+61];
+			lowsL -= bL[temp+59];
+			lowsL -= bL[temp+53];
+			lowsL -= bL[temp+47];
+			lowsL -= bL[temp+43];
+			lowsL -= bL[temp+41];
+			lowsL -= bL[temp+37];
+			lowsL -= bL[temp+31];
+			lowsL -= bL[temp+29];
+			lowsL -= bL[temp+23];
+			lowsL -= bL[temp+19];
+			lowsL -= bL[temp+17];
+			lowsL -= bL[temp+13];
+			lowsL -= bL[temp+11];
+			lowsL -= bL[temp+7];
+			lowsL -= bL[temp+5];
+			lowsL -= bL[temp+3];
+			lowsL -= bL[temp+2];
+			lowsL -= bL[temp+1];
+			//initial negative lobe
+			
+			lowsR -= bR[temp+127];
+			lowsR -= bR[temp+113];
+			lowsR -= bR[temp+109];
+			lowsR -= bR[temp+107];
+			lowsR -= bR[temp+103];
+			lowsR -= bR[temp+101];
+			lowsR -= bR[temp+97];
+			lowsR -= bR[temp+89];
+			lowsR -= bR[temp+83];
+			lowsR -= bR[temp+79];
+			lowsR -= bR[temp+73];
+			lowsR -= bR[temp+71];
+			lowsR -= bR[temp+67];
+			lowsR -= bR[temp+61];
+			lowsR -= bR[temp+59];
+			lowsR -= bR[temp+53];
+			lowsR -= bR[temp+47];
+			lowsR -= bR[temp+43];
+			lowsR -= bR[temp+41];
+			lowsR -= bR[temp+37];
+			lowsR -= bR[temp+31];
+			lowsR -= bR[temp+29];
+			lowsR -= bR[temp+23];
+			lowsR -= bR[temp+19];
+			lowsR -= bR[temp+17];
+			lowsR -= bR[temp+13];
+			lowsR -= bR[temp+11];
+			lowsR -= bR[temp+7];
+			lowsR -= bR[temp+5];
+			lowsR -= bR[temp+3];
+			lowsR -= bR[temp+2];
+			lowsR -= bR[temp+1];
+			//initial negative lobe
+			
+			lowsL *= subs;
+			lowsL *= subs;
+			lowsR *= subs;
+			lowsR *= subs;
+			//twice, to minimize the suckout in low boost situations
+			temp = count+refL2;
+			
+			lowsL += bL[temp+127];
+			lowsL += bL[temp+113];
+			lowsL += bL[temp+109];
+			lowsL += bL[temp+107];
+			lowsL += bL[temp+103];
+			lowsL += bL[temp+101];
+			lowsL += bL[temp+97];
+			lowsL += bL[temp+89];
+			lowsL += bL[temp+83];
+			lowsL += bL[temp+79];
+			lowsL += bL[temp+73];
+			lowsL += bL[temp+71];
+			lowsL += bL[temp+67];
+			lowsL += bL[temp+61];
+			lowsL += bL[temp+59];
+			lowsL += bL[temp+53];
+			lowsL += bL[temp+47];
+			lowsL += bL[temp+43];
+			lowsL += bL[temp+41];
+			lowsL += bL[temp+37];
+			lowsL += bL[temp+31];
+			lowsL += bL[temp+29];
+			lowsL += bL[temp+23];
+			lowsL += bL[temp+19];
+			lowsL += bL[temp+17];
+			lowsL += bL[temp+13];
+			lowsL += bL[temp+11];
+			lowsL += bL[temp+7];
+			lowsL += bL[temp+5];
+			lowsL += bL[temp+3];
+			lowsL += bL[temp+2];
+			lowsL += bL[temp+1];
+			//followup positive lobe
+
+			lowsR += bR[temp+127];
+			lowsR += bR[temp+113];
+			lowsR += bR[temp+109];
+			lowsR += bR[temp+107];
+			lowsR += bR[temp+103];
+			lowsR += bR[temp+101];
+			lowsR += bR[temp+97];
+			lowsR += bR[temp+89];
+			lowsR += bR[temp+83];
+			lowsR += bR[temp+79];
+			lowsR += bR[temp+73];
+			lowsR += bR[temp+71];
+			lowsR += bR[temp+67];
+			lowsR += bR[temp+61];
+			lowsR += bR[temp+59];
+			lowsR += bR[temp+53];
+			lowsR += bR[temp+47];
+			lowsR += bR[temp+43];
+			lowsR += bR[temp+41];
+			lowsR += bR[temp+37];
+			lowsR += bR[temp+31];
+			lowsR += bR[temp+29];
+			lowsR += bR[temp+23];
+			lowsR += bR[temp+19];
+			lowsR += bR[temp+17];
+			lowsR += bR[temp+13];
+			lowsR += bR[temp+11];
+			lowsR += bR[temp+7];
+			lowsR += bR[temp+5];
+			lowsR += bR[temp+3];
+			lowsR += bR[temp+2];
+			lowsR += bR[temp+1];
+			//followup positive lobe
+			
+			lowsL *= subs;
+			lowsR *= subs;
+			//now we have the lows content to use
+		}
+		
+		bridgerectifier = sin(fabs(lowsL) * softness);
+		//this will wrap around and is scaled back by hardness: hard = less bass push, more treble
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+
+		bridgerectifier = sin(fabs(lowsR) * softness);
+		//this will wrap around and is scaled back by hardness: hard = less bass push, more treble
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		iirLowsAL = (iirLowsAL * invcalibsubs) + (lowsL * calibsubs);
+		lowsL = iirLowsAL;
+		bridgerectifier = sin(fabs(lowsL));
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+
+		iirLowsAR = (iirLowsAR * invcalibsubs) + (lowsR * calibsubs);
+		lowsR = iirLowsAR;
+		bridgerectifier = sin(fabs(lowsR));
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		iirLowsBL = (iirLowsBL * invcalibsubs) + (lowsL * calibsubs);
+		lowsL = iirLowsBL;
+		bridgerectifier = sin(fabs(lowsL)) * 2.0;
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+		
+		iirLowsBR = (iirLowsBR * invcalibsubs) + (lowsR * calibsubs);
+		lowsR = iirLowsBR;
+		bridgerectifier = sin(fabs(lowsR)) * 2.0;
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		if (highslift > 0.0) inputSampleL += (highsL * (1.0-fabs(inputSampleL*hardness)));
+		if (subslift > 0.0) inputSampleL += (lowsL * (1.0-fabs(inputSampleL*softness)));
+
+		if (highslift > 0.0) inputSampleR += (highsR * (1.0-fabs(inputSampleR*hardness)));
+		if (subslift > 0.0) inputSampleR += (lowsR * (1.0-fabs(inputSampleR*softness)));
+
+		if (inputSampleL > refclipL && refclipL > 0.9) refclipL -= 0.01;
+		if (inputSampleL < -refclipL && refclipL > 0.9) refclipL -= 0.01;
+		if (refclipL < 0.99) refclipL += 0.00001;
+		//adjust clip level on the fly
+
+		if (inputSampleR > refclipR && refclipR > 0.9) refclipR -= 0.01;
+		if (inputSampleR < -refclipR && refclipR > 0.9) refclipR -= 0.01;
+		if (refclipR < 0.99) refclipR += 0.00001;
+		//adjust clip level on the fly
+		
+		if (lastSampleL >= refclipL)
+		{
+			if (inputSampleL < refclipL) lastSampleL = ((refclipL*hardness) + (inputSampleL * softness));
+			else lastSampleL = refclipL;
+		}
+
+		if (lastSampleR >= refclipR)
+		{
+			if (inputSampleR < refclipR) lastSampleR = ((refclipR*hardness) + (inputSampleR * softness));
+			else lastSampleR = refclipR;
+		}
+		
+		if (lastSampleL <= -refclipL)
+		{
+			if (inputSampleL > -refclipL) lastSampleL = ((-refclipL*hardness) + (inputSampleL * softness));
+			else lastSampleL = -refclipL;
+		}
+		
+		if (lastSampleR <= -refclipR)
+		{
+			if (inputSampleR > -refclipR) lastSampleR = ((-refclipR*hardness) + (inputSampleR * softness));
+			else lastSampleR = -refclipR;
+		}
+		
+		if (inputSampleL > refclipL)
+		{
+			if (lastSampleL < refclipL) inputSampleL = ((refclipL*hardness) + (lastSampleL * softness));
+			else inputSampleL = refclipL;
+		}
+		
+		if (inputSampleR > refclipR)
+		{
+			if (lastSampleR < refclipR) inputSampleR = ((refclipR*hardness) + (lastSampleR * softness));
+			else inputSampleR = refclipR;
+		}
+		
+		if (inputSampleL < -refclipL)
+		{
+			if (lastSampleL > -refclipL) inputSampleL = ((-refclipL*hardness) + (lastSampleL * softness));
+			else inputSampleL = -refclipL;
+		}
+		
+		if (inputSampleR < -refclipR)
+		{
+			if (lastSampleR > -refclipR) inputSampleR = ((-refclipR*hardness) + (lastSampleR * softness));
+			else inputSampleR = -refclipR;
+		}
+		lastSampleL = inputSampleL;
+		lastSampleR = inputSampleR;
+		
+		switch (mode)
+		{
+			case 1: break; //Normal
+			case 2: inputSampleL /= inputGain; inputSampleR /= inputGain; break; //Gain Match
+			case 3: inputSampleL = overshootL + highsL + lowsL; inputSampleR = overshootR + highsR + lowsR; break; //Clip Only
+		}
+		//this is our output mode switch, showing the effects
+		
+		if (inputSampleL > refclipL) inputSampleL = refclipL;
+		if (inputSampleL < -refclipL) inputSampleL = -refclipL;
+		if (inputSampleR > refclipR) inputSampleR = refclipR;
+		if (inputSampleR < -refclipR) inputSampleR = -refclipR;
+		//final iron bar
+		
+		//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 ADClip7::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 fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	
+	double inputGain = pow(10.0,(A*18.0)/20.0);
+	double softness = B * fpNew;
+	double hardness = 1.0 - softness;
+	double highslift = 0.307 * C;
+	double adjust = pow(highslift,3) * 0.416;
+	double subslift = 0.796 * C;
+	double calibsubs = subslift/53;
+	double invcalibsubs = 1.0 - calibsubs;
+	double subs = 0.81 + (calibsubs*2);
+	long double bridgerectifier;
+	int mode = (int) floor(D*2.999)+1;
+	double overshootL;
+	double overshootR;
+	double offsetH1 = 1.84;
+	offsetH1 *= overallscale;
+	double offsetH2 = offsetH1 * 1.9;
+	double offsetH3 = offsetH1 * 2.7;
+	double offsetL1 = 612;
+	offsetL1 *= overallscale;
+	double offsetL2 = offsetL1 * 2.0;
+	int refH1 = (int)floor(offsetH1);
+	int refH2 = (int)floor(offsetH2);
+	int refH3 = (int)floor(offsetH3);
+	int refL1 = (int)floor(offsetL1);
+	int refL2 = (int)floor(offsetL2);
+	int temp;
+	double fractionH1 = offsetH1 - floor(offsetH1);
+	double fractionH2 = offsetH2 - floor(offsetH2);
+	double fractionH3 = offsetH3 - floor(offsetH3);
+	double minusH1 = 1.0 - fractionH1;
+	double minusH2 = 1.0 - fractionH2;
+	double minusH3 = 1.0 - fractionH3;
+	double highsL = 0.0;
+	double highsR = 0.0;
+	int count = 0;
+	
+	long double inputSampleL;
+	long double inputSampleR;
+
+
+    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.
+		}
+		
+		
+		
+		if (inputGain != 1.0) {
+			inputSampleL *= inputGain;
+			inputSampleR *= inputGain;
+		}
+		
+		overshootL = fabs(inputSampleL) - refclipL;
+		overshootR = fabs(inputSampleR) - refclipR;
+		if (overshootL < 0.0) overshootL = 0.0;
+		if (overshootR < 0.0) overshootR = 0.0;
+		
+		if (gcount < 0 || gcount > 11020) {gcount = 11020;}
+		count = gcount;
+		bL[count+11020] = bL[count] = overshootL;
+		bR[count+11020] = bR[count] = overshootR;
+		gcount--;
+		
+		if (highslift > 0.0)
+		{
+			//we have a big pile of b[] which is overshoots
+			temp = count+refH3;
+			highsL = -(bL[temp] * minusH3); //less as value moves away from .0
+			highsL -= bL[temp+1]; //we can assume always using this in one way or another?
+			highsL -= (bL[temp+2] * fractionH3); //greater as value moves away from .0
+			highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 3 is a negative add
+			highsR = -(bR[temp] * minusH3); //less as value moves away from .0
+			highsR -= bR[temp+1]; //we can assume always using this in one way or another?
+			highsR -= (bR[temp+2] * fractionH3); //greater as value moves away from .0
+			highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 3 is a negative add
+			temp = count+refH2;
+			highsL += (bL[temp] * minusH2); //less as value moves away from .0
+			highsL += bL[temp+1]; //we can assume always using this in one way or another?
+			highsL += (bL[temp+2] * fractionH2); //greater as value moves away from .0
+			highsL -= (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 2 is a positive feedback of the overshoot
+			highsR += (bR[temp] * minusH2); //less as value moves away from .0
+			highsR += bR[temp+1]; //we can assume always using this in one way or another?
+			highsR += (bR[temp+2] * fractionH2); //greater as value moves away from .0
+			highsR -= (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 2 is a positive feedback of the overshoot
+			temp = count+refH1;
+			highsL -= (bL[temp] * minusH1); //less as value moves away from .0
+			highsL -= bL[temp+1]; //we can assume always using this in one way or another?
+			highsL -= (bL[temp+2] * fractionH1); //greater as value moves away from .0
+			highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us
+			highsL *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 1 is a negative feedback of the overshoot
+			highsR -= (bR[temp] * minusH1); //less as value moves away from .0
+			highsR -= bR[temp+1]; //we can assume always using this in one way or another?
+			highsR -= (bR[temp+2] * fractionH1); //greater as value moves away from .0
+			highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us
+			highsR *= adjust; //add in the kernel elements backwards saves multiplies
+			//stage 1 is a negative feedback of the overshoot
+			//done with interpolated mostly negative feedback of the overshoot
+		}
+		
+		bridgerectifier = sin(fabs(highsL) * hardness);
+		//this will wrap around and is scaled back by softness
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (highsL > 0) highsL = bridgerectifier;
+		else highsL = -bridgerectifier;
+		
+		bridgerectifier = sin(fabs(highsR) * hardness);
+		//this will wrap around and is scaled back by softness
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (highsR > 0) highsR = bridgerectifier;
+		else highsR = -bridgerectifier;
+		
+		if (subslift > 0.0) 
+		{
+			lowsL *= subs;
+			lowsR *= subs;
+			//going in we'll reel back some of the swing
+			temp = count+refL1;
+			
+			lowsL -= bL[temp+127];
+			lowsL -= bL[temp+113];
+			lowsL -= bL[temp+109];
+			lowsL -= bL[temp+107];
+			lowsL -= bL[temp+103];
+			lowsL -= bL[temp+101];
+			lowsL -= bL[temp+97];
+			lowsL -= bL[temp+89];
+			lowsL -= bL[temp+83];
+			lowsL -= bL[temp+79];
+			lowsL -= bL[temp+73];
+			lowsL -= bL[temp+71];
+			lowsL -= bL[temp+67];
+			lowsL -= bL[temp+61];
+			lowsL -= bL[temp+59];
+			lowsL -= bL[temp+53];
+			lowsL -= bL[temp+47];
+			lowsL -= bL[temp+43];
+			lowsL -= bL[temp+41];
+			lowsL -= bL[temp+37];
+			lowsL -= bL[temp+31];
+			lowsL -= bL[temp+29];
+			lowsL -= bL[temp+23];
+			lowsL -= bL[temp+19];
+			lowsL -= bL[temp+17];
+			lowsL -= bL[temp+13];
+			lowsL -= bL[temp+11];
+			lowsL -= bL[temp+7];
+			lowsL -= bL[temp+5];
+			lowsL -= bL[temp+3];
+			lowsL -= bL[temp+2];
+			lowsL -= bL[temp+1];
+			//initial negative lobe
+			
+			lowsR -= bR[temp+127];
+			lowsR -= bR[temp+113];
+			lowsR -= bR[temp+109];
+			lowsR -= bR[temp+107];
+			lowsR -= bR[temp+103];
+			lowsR -= bR[temp+101];
+			lowsR -= bR[temp+97];
+			lowsR -= bR[temp+89];
+			lowsR -= bR[temp+83];
+			lowsR -= bR[temp+79];
+			lowsR -= bR[temp+73];
+			lowsR -= bR[temp+71];
+			lowsR -= bR[temp+67];
+			lowsR -= bR[temp+61];
+			lowsR -= bR[temp+59];
+			lowsR -= bR[temp+53];
+			lowsR -= bR[temp+47];
+			lowsR -= bR[temp+43];
+			lowsR -= bR[temp+41];
+			lowsR -= bR[temp+37];
+			lowsR -= bR[temp+31];
+			lowsR -= bR[temp+29];
+			lowsR -= bR[temp+23];
+			lowsR -= bR[temp+19];
+			lowsR -= bR[temp+17];
+			lowsR -= bR[temp+13];
+			lowsR -= bR[temp+11];
+			lowsR -= bR[temp+7];
+			lowsR -= bR[temp+5];
+			lowsR -= bR[temp+3];
+			lowsR -= bR[temp+2];
+			lowsR -= bR[temp+1];
+			//initial negative lobe
+			
+			lowsL *= subs;
+			lowsL *= subs;
+			lowsR *= subs;
+			lowsR *= subs;
+			//twice, to minimize the suckout in low boost situations
+			temp = count+refL2;
+			
+			lowsL += bL[temp+127];
+			lowsL += bL[temp+113];
+			lowsL += bL[temp+109];
+			lowsL += bL[temp+107];
+			lowsL += bL[temp+103];
+			lowsL += bL[temp+101];
+			lowsL += bL[temp+97];
+			lowsL += bL[temp+89];
+			lowsL += bL[temp+83];
+			lowsL += bL[temp+79];
+			lowsL += bL[temp+73];
+			lowsL += bL[temp+71];
+			lowsL += bL[temp+67];
+			lowsL += bL[temp+61];
+			lowsL += bL[temp+59];
+			lowsL += bL[temp+53];
+			lowsL += bL[temp+47];
+			lowsL += bL[temp+43];
+			lowsL += bL[temp+41];
+			lowsL += bL[temp+37];
+			lowsL += bL[temp+31];
+			lowsL += bL[temp+29];
+			lowsL += bL[temp+23];
+			lowsL += bL[temp+19];
+			lowsL += bL[temp+17];
+			lowsL += bL[temp+13];
+			lowsL += bL[temp+11];
+			lowsL += bL[temp+7];
+			lowsL += bL[temp+5];
+			lowsL += bL[temp+3];
+			lowsL += bL[temp+2];
+			lowsL += bL[temp+1];
+			//followup positive lobe
+			
+			lowsR += bR[temp+127];
+			lowsR += bR[temp+113];
+			lowsR += bR[temp+109];
+			lowsR += bR[temp+107];
+			lowsR += bR[temp+103];
+			lowsR += bR[temp+101];
+			lowsR += bR[temp+97];
+			lowsR += bR[temp+89];
+			lowsR += bR[temp+83];
+			lowsR += bR[temp+79];
+			lowsR += bR[temp+73];
+			lowsR += bR[temp+71];
+			lowsR += bR[temp+67];
+			lowsR += bR[temp+61];
+			lowsR += bR[temp+59];
+			lowsR += bR[temp+53];
+			lowsR += bR[temp+47];
+			lowsR += bR[temp+43];
+			lowsR += bR[temp+41];
+			lowsR += bR[temp+37];
+			lowsR += bR[temp+31];
+			lowsR += bR[temp+29];
+			lowsR += bR[temp+23];
+			lowsR += bR[temp+19];
+			lowsR += bR[temp+17];
+			lowsR += bR[temp+13];
+			lowsR += bR[temp+11];
+			lowsR += bR[temp+7];
+			lowsR += bR[temp+5];
+			lowsR += bR[temp+3];
+			lowsR += bR[temp+2];
+			lowsR += bR[temp+1];
+			//followup positive lobe
+			
+			lowsL *= subs;
+			lowsR *= subs;
+			//now we have the lows content to use
+		}
+		
+		bridgerectifier = sin(fabs(lowsL) * softness);
+		//this will wrap around and is scaled back by hardness: hard = less bass push, more treble
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+		
+		bridgerectifier = sin(fabs(lowsR) * softness);
+		//this will wrap around and is scaled back by hardness: hard = less bass push, more treble
+		//wrap around is the same principle as Fracture: no top limit to sin()
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		iirLowsAL = (iirLowsAL * invcalibsubs) + (lowsL * calibsubs);
+		lowsL = iirLowsAL;
+		bridgerectifier = sin(fabs(lowsL));
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+		
+		iirLowsAR = (iirLowsAR * invcalibsubs) + (lowsR * calibsubs);
+		lowsR = iirLowsAR;
+		bridgerectifier = sin(fabs(lowsR));
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		iirLowsBL = (iirLowsBL * invcalibsubs) + (lowsL * calibsubs);
+		lowsL = iirLowsBL;
+		bridgerectifier = sin(fabs(lowsL)) * 2.0;
+		if (lowsL > 0) lowsL = bridgerectifier;
+		else lowsL = -bridgerectifier;
+		
+		iirLowsBR = (iirLowsBR * invcalibsubs) + (lowsR * calibsubs);
+		lowsR = iirLowsBR;
+		bridgerectifier = sin(fabs(lowsR)) * 2.0;
+		if (lowsR > 0) lowsR = bridgerectifier;
+		else lowsR = -bridgerectifier;
+		
+		if (highslift > 0.0) inputSampleL += (highsL * (1.0-fabs(inputSampleL*hardness)));
+		if (subslift > 0.0) inputSampleL += (lowsL * (1.0-fabs(inputSampleL*softness)));
+		
+		if (highslift > 0.0) inputSampleR += (highsR * (1.0-fabs(inputSampleR*hardness)));
+		if (subslift > 0.0) inputSampleR += (lowsR * (1.0-fabs(inputSampleR*softness)));
+		
+		if (inputSampleL > refclipL && refclipL > 0.9) refclipL -= 0.01;
+		if (inputSampleL < -refclipL && refclipL > 0.9) refclipL -= 0.01;
+		if (refclipL < 0.99) refclipL += 0.00001;
+		//adjust clip level on the fly
+		
+		if (inputSampleR > refclipR && refclipR > 0.9) refclipR -= 0.01;
+		if (inputSampleR < -refclipR && refclipR > 0.9) refclipR -= 0.01;
+		if (refclipR < 0.99) refclipR += 0.00001;
+		//adjust clip level on the fly
+		
+		if (lastSampleL >= refclipL)
+		{
+			if (inputSampleL < refclipL) lastSampleL = ((refclipL*hardness) + (inputSampleL * softness));
+			else lastSampleL = refclipL;
+		}
+		
+		if (lastSampleR >= refclipR)
+		{
+			if (inputSampleR < refclipR) lastSampleR = ((refclipR*hardness) + (inputSampleR * softness));
+			else lastSampleR = refclipR;
+		}
+		
+		if (lastSampleL <= -refclipL)
+		{
+			if (inputSampleL > -refclipL) lastSampleL = ((-refclipL*hardness) + (inputSampleL * softness));
+			else lastSampleL = -refclipL;
+		}
+		
+		if (lastSampleR <= -refclipR)
+		{
+			if (inputSampleR > -refclipR) lastSampleR = ((-refclipR*hardness) + (inputSampleR * softness));
+			else lastSampleR = -refclipR;
+		}
+		
+		if (inputSampleL > refclipL)
+		{
+			if (lastSampleL < refclipL) inputSampleL = ((refclipL*hardness) + (lastSampleL * softness));
+			else inputSampleL = refclipL;
+		}
+		
+		if (inputSampleR > refclipR)
+		{
+			if (lastSampleR < refclipR) inputSampleR = ((refclipR*hardness) + (lastSampleR * softness));
+			else inputSampleR = refclipR;
+		}
+		
+		if (inputSampleL < -refclipL)
+		{
+			if (lastSampleL > -refclipL) inputSampleL = ((-refclipL*hardness) + (lastSampleL * softness));
+			else inputSampleL = -refclipL;
+		}
+		
+		if (inputSampleR < -refclipR)
+		{
+			if (lastSampleR > -refclipR) inputSampleR = ((-refclipR*hardness) + (lastSampleR * softness));
+			else inputSampleR = -refclipR;
+		}
+		lastSampleL = inputSampleL;
+		lastSampleR = inputSampleR;
+		
+		switch (mode)
+		{
+			case 1: break; //Normal
+			case 2: inputSampleL /= inputGain; inputSampleR /= inputGain; break; //Gain Match
+			case 3: inputSampleL = overshootL + highsL + lowsL; inputSampleR = overshootR + highsR + lowsR; break; //Clip Only
+		}
+		//this is our output mode switch, showing the effects
+		
+		if (inputSampleL > refclipL) inputSampleL = refclipL;
+		if (inputSampleL < -refclipL) inputSampleL = -refclipL;
+		if (inputSampleR > refclipR) inputSampleR = refclipR;
+		if (inputSampleR < -refclipR) inputSampleR = -refclipR;
+		//final iron bar
+		
+		//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