Updates (in case my plane crashes)

3 files changed, 1194 insertions, 0 deletions

diff --git a/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.cpp b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.cpp
new file mode 100755
index 0000000..9b5ea81
--- /dev/null
+++ b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.cpp
@@ -0,0 +1,166 @@
+/* ========================================
+ *  AQuickVoiceClip - AQuickVoiceClip.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __AQuickVoiceClip_H
+#include "AQuickVoiceClip.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new AQuickVoiceClip(audioMaster);}
+
+AQuickVoiceClip::AQuickVoiceClip(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.42;
+	LataLast6Sample = LataLast5Sample = LataLast4Sample = 0.0;
+	LataLast3Sample = LataLast2Sample = LataLast1Sample = 0.0;
+	LataHalfwaySample = LataHalfDrySample = LataHalfDiffSample = 0.0;
+	LataDrySample = LataDiffSample = LataPrevDiffSample = 0.0;
+	
+	RataLast6Sample = RataLast5Sample = RataLast4Sample = 0.0;
+	RataLast3Sample = RataLast2Sample = RataLast1Sample = 0.0;
+	RataHalfwaySample = RataHalfDrySample = RataHalfDiffSample = 0.0;
+	RataDrySample = RataDiffSample = RataPrevDiffSample = 0.0;
+	
+	LlastSample = 0.0;
+	LlastOutSample = 0.0;
+	LlastOut2Sample = 0.0;
+	LlastOut3Sample = 0.0;
+	LlpDepth = 0.0;
+	Lovershoot = 0.0;
+	Loverall = 0;
+	LiirSampleA = 0.0;
+	LiirSampleB = 0.0;
+	LiirSampleC = 0.0;
+	LiirSampleD = 0.0;
+
+	RlastSample = 0.0;
+	RlastOutSample = 0.0;
+	RlastOut2Sample = 0.0;
+	RlastOut3Sample = 0.0;
+	RlpDepth = 0.0;
+	Rovershoot = 0.0;
+	Roverall = 0;
+	RiirSampleA = 0.0;
+	RiirSampleB = 0.0;
+	RiirSampleC = 0.0;
+	RiirSampleD = 0.0;
+	flip = true;
+	
+	ataK1 = -0.646; //first FIR shaping of interpolated sample, brightens
+	ataK2 = 0.311; //second FIR shaping of interpolated sample, thickens
+	ataK6 = -0.093; //third FIR shaping of interpolated sample, brings air
+	ataK7 = 0.057; //fourth FIR shaping of interpolated sample, thickens
+	ataK8 = -0.023; //fifth FIR shaping of interpolated sample, brings air
+	ataK3 = 0.114; //add raw to interpolated dry, toughens
+	ataK4 = 0.886; //remainder of interpolated dry, adds up to 1.0
+	ataK5 = 0.431; //subtract this much prev. diff sample, brightens.  0.431 becomes flat
+	
+	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
+}
+
+AQuickVoiceClip::~AQuickVoiceClip() {}
+VstInt32 AQuickVoiceClip::getVendorVersion () {return 1000;}
+void AQuickVoiceClip::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void AQuickVoiceClip::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 AQuickVoiceClip::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	/* 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 AQuickVoiceClip::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	/* 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 AQuickVoiceClip::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float AQuickVoiceClip::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; 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 AQuickVoiceClip::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Highpass", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void AQuickVoiceClip::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((pow(A,3)*2070)+30, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void AQuickVoiceClip::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "hz", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 AQuickVoiceClip::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool AQuickVoiceClip::getEffectName(char* name) {
+    vst_strncpy(name, "AQuickVoiceClip", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory AQuickVoiceClip::getPlugCategory() {return kPlugCategEffect;}
+
+bool AQuickVoiceClip::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows AQuickVoiceClip", kVstMaxProductStrLen); return true;
+}
+
+bool AQuickVoiceClip::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.h b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.h
new file mode 100755
index 0000000..dc8f5fa
--- /dev/null
+++ b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClip.h
@@ -0,0 +1,128 @@
+/* ========================================
+ *  AQuickVoiceClip - AQuickVoiceClip.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __AQuickVoiceClip_H
+#define __AQuickVoiceClip_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+  kNumParameters = 1
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'aqvc';    //Change this to what the AU identity is!
+
+class AQuickVoiceClip : 
+    public AudioEffectX 
+{
+public:
+    AQuickVoiceClip(audioMasterCallback audioMaster);
+    ~AQuickVoiceClip();
+    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;
+    
+	
+	double LataLast6Sample;
+	double LataLast5Sample;
+	double LataLast4Sample;
+	double LataLast3Sample;
+	double LataLast2Sample;
+	double LataLast1Sample;
+	double LataHalfwaySample;
+	double LataHalfDrySample;
+	double LataHalfDiffSample;
+	double LataLastDiffSample;
+	double LataDrySample;
+	double LataDiffSample;
+	double LataPrevDiffSample;
+	
+	double RataLast6Sample;
+	double RataLast5Sample;
+	double RataLast4Sample;
+	double RataLast3Sample;
+	double RataLast2Sample;
+	double RataLast1Sample;
+	double RataHalfwaySample;
+	double RataHalfDrySample;
+	double RataHalfDiffSample;
+	double RataLastDiffSample;
+	double RataDrySample;
+	double RataDiffSample;
+	double RataPrevDiffSample;
+	
+	double ataK1;
+	double ataK2;
+	double ataK3;
+	double ataK4;
+	double ataK5;
+	double ataK6;
+	double ataK7;
+	double ataK8; //end antialiasing variables
+
+	double LlastSample;
+	double LlastOutSample;
+	double LlastOut2Sample;
+	double LlastOut3Sample;
+	double LlpDepth;
+	double Lovershoot;
+	double Loverall;
+	double LiirSampleA;
+	double LiirSampleB;
+	double LiirSampleC;
+	double LiirSampleD;
+
+	double RlastSample;
+	double RlastOutSample;
+	double RlastOut2Sample;
+	double RlastOut3Sample;
+	double RlpDepth;
+	double Rovershoot;
+	double Roverall;
+	double RiirSampleA;
+	double RiirSampleB;
+	double RiirSampleC;
+	double RiirSampleD;
+	bool flip;
+		
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool fpFlip;
+	//default stuff
+
+    float A;
+};
+
+#endif
diff --git a/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClipProc.cpp b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClipProc.cpp
new file mode 100755
index 0000000..69dc283
--- /dev/null
+++ b/plugins/MacVST/AQuickVoiceClip/source/AQuickVoiceClipProc.cpp
@@ -0,0 +1,900 @@
+/* ========================================
+ *  AQuickVoiceClip - AQuickVoiceClip.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __AQuickVoiceClip_H
+#include "AQuickVoiceClip.h"
+#endif
+
+void AQuickVoiceClip::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 softness = 0.484416;
+	double hardness = 1.0 - softness;
+	double iirAmount = ((pow(A,3)*2070)+30)/8000.0;
+	iirAmount /= overallscale;
+	double altAmount = (1.0 - iirAmount);
+	double cancelnew = 0.0682276;
+	double cancelold = 1.0 - cancelnew;
+	double lpSpeed = 0.0009;
+	double cliplevel = 0.98;
+	double refclip = 0.5; //preset to cut out gain quite a lot. 91%? no touchy unless clip
+	
+	double LmaxRecent;
+	bool LclipOnset;
+	double LpassThrough;
+	double LoutputSample;
+	double LdrySample;
+	
+	double RmaxRecent;
+	bool RclipOnset;
+	double RpassThrough;
+	double RoutputSample;
+	double RdrySample;
+	
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	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.
+		}
+		LpassThrough = LataDrySample = inputSampleL;
+		RpassThrough = RataDrySample = inputSampleR;
+		
+		LataHalfDrySample = LataHalfwaySample = (inputSampleL + LataLast1Sample + (LataLast2Sample*ataK1) + (LataLast3Sample*ataK2) + (LataLast4Sample*ataK6) + (LataLast5Sample*ataK7) + (LataLast6Sample*ataK8)) / 2.0;
+		LataLast6Sample = LataLast5Sample; LataLast5Sample = LataLast4Sample; LataLast4Sample = LataLast3Sample; LataLast3Sample = LataLast2Sample; LataLast2Sample = LataLast1Sample; LataLast1Sample = inputSampleL;
+		//setting up oversampled special antialiasing
+		RataHalfDrySample = RataHalfwaySample = (inputSampleR + RataLast1Sample + (RataLast2Sample*ataK1) + (RataLast3Sample*ataK2) + (RataLast4Sample*ataK6) + (RataLast5Sample*ataK7) + (RataLast6Sample*ataK8)) / 2.0;
+		RataLast6Sample = RataLast5Sample; RataLast5Sample = RataLast4Sample; RataLast4Sample = RataLast3Sample; RataLast3Sample = RataLast2Sample; RataLast2Sample = RataLast1Sample; RataLast1Sample = inputSampleR;
+		//setting up oversampled special antialiasing
+		LclipOnset = false;
+		RclipOnset = false;
+		
+		
+		LmaxRecent = fabs( LataLast6Sample );
+		if (fabs( LataLast5Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast5Sample );
+		if (fabs( LataLast4Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast4Sample );
+		if (fabs( LataLast3Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast3Sample );
+		if (fabs( LataLast2Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast2Sample );
+		if (fabs( LataLast1Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast1Sample );
+		if (fabs( inputSampleL ) > LmaxRecent ) LmaxRecent = fabs( inputSampleL );
+		//this gives us something that won't cut out in zero crossings, to interpolate with
+		
+		RmaxRecent = fabs( RataLast6Sample );
+		if (fabs( RataLast5Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast5Sample );
+		if (fabs( RataLast4Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast4Sample );
+		if (fabs( RataLast3Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast3Sample );
+		if (fabs( RataLast2Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast2Sample );
+		if (fabs( RataLast1Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast1Sample );
+		if (fabs( inputSampleR ) > RmaxRecent ) RmaxRecent = fabs( inputSampleR );
+		//this gives us something that won't cut out in zero crossings, to interpolate with
+		
+		LmaxRecent *= 2.0;
+		RmaxRecent *= 2.0;
+		//by refclip this is 1.0 and fully into the antialiasing
+		if (LmaxRecent > 1.0) LmaxRecent = 1.0;
+		if (RmaxRecent > 1.0) RmaxRecent = 1.0;
+		//and it tops out at 1. Higher means more antialiasing, lower blends into passThrough without antialiasing
+		
+		LataHalfwaySample -= Loverall;
+		RataHalfwaySample -= Roverall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		LdrySample = LataHalfwaySample;
+		RdrySample = RataHalfwaySample;
+		
+		
+		//begin L channel for the clipper
+		if (LlastSample >= refclip)
+		{
+			LlpDepth += 0.1;
+			if (LataHalfwaySample < refclip)
+			{
+				LlastSample = ((refclip*hardness) + (LataHalfwaySample * softness));
+			}
+			else LlastSample = refclip;
+		}
+		
+		if (LlastSample <= -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LataHalfwaySample > -refclip)
+			{
+				LlastSample = ((-refclip*hardness) + (LataHalfwaySample * softness));
+			}
+			else LlastSample = -refclip;
+		}
+		
+		if (LataHalfwaySample > refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample < refclip)
+			{
+				LataHalfwaySample = ((refclip*hardness) + (LlastSample * softness));
+			}
+			else LataHalfwaySample = refclip;
+		}
+		
+		if (LataHalfwaySample < -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample > -refclip)
+			{
+				LataHalfwaySample = ((-refclip*hardness) + (LlastSample * softness));
+			}
+			else LataHalfwaySample = -refclip;
+		}
+		///end L channel for the clipper
+		
+		//begin R channel for the clipper
+		if (RlastSample >= refclip)
+		{
+			RlpDepth += 0.1;
+			if (RataHalfwaySample < refclip)
+			{
+				RlastSample = ((refclip*hardness) + (RataHalfwaySample * softness));
+			}
+			else RlastSample = refclip;
+		}
+		
+		if (RlastSample <= -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RataHalfwaySample > -refclip)
+			{
+				RlastSample = ((-refclip*hardness) + (RataHalfwaySample * softness));
+			}
+			else RlastSample = -refclip;
+		}
+		
+		if (RataHalfwaySample > refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample < refclip)
+			{
+				RataHalfwaySample = ((refclip*hardness) + (RlastSample * softness));
+			}
+			else RataHalfwaySample = refclip;
+		}
+		
+		if (RataHalfwaySample < -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample > -refclip)
+			{
+				RataHalfwaySample = ((-refclip*hardness) + (RlastSample * softness));
+			}
+			else RataHalfwaySample = -refclip;
+		}
+		///end R channel for the clipper
+		
+        LoutputSample = LlastSample;
+        RoutputSample = RlastSample;
+		
+		LlastSample = LataHalfwaySample;
+		RlastSample = RataHalfwaySample;
+
+		LataHalfwaySample = LoutputSample;
+		RataHalfwaySample = RoutputSample;
+		//swap around in a circle for one final ADClip,
+		//this time not tracking overshoot anymore
+		//end interpolated sample		
+		//begin raw sample- inputSample and ataDrySample handled separately here
+		
+		inputSampleL -= Loverall;
+		inputSampleR -= Roverall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		LdrySample = inputSampleL;
+		RdrySample = inputSampleR;
+		
+		//begin second L clip
+		if (LlastSample >= refclip)
+		{
+			LlpDepth += 0.1;
+			if (inputSampleL < refclip)
+			{
+				LlastSample = ((refclip*hardness) + (inputSampleL * softness));
+			}
+			else LlastSample = refclip;
+		}
+		
+		if (LlastSample <= -refclip)
+		{
+			LlpDepth += 0.1;
+			if (inputSampleL > -refclip)
+			{
+				LlastSample = ((-refclip*hardness) + (inputSampleL * softness));
+			}
+			else LlastSample = -refclip;
+		}
+		
+		if (inputSampleL > refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample < refclip)
+			{
+				inputSampleL = ((refclip*hardness) + (LlastSample * softness));
+			}
+			else inputSampleL = refclip;
+		}
+		
+		if (inputSampleL < -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample > -refclip)
+			{
+				inputSampleL = ((-refclip*hardness) + (LlastSample * softness));
+			}
+			else inputSampleL = -refclip;
+		}		
+		//end second L clip
+		
+		//begin second R clip
+		if (RlastSample >= refclip)
+		{
+			RlpDepth += 0.1;
+			if (inputSampleR < refclip)
+			{
+				RlastSample = ((refclip*hardness) + (inputSampleR * softness));
+			}
+			else RlastSample = refclip;
+		}
+		
+		if (RlastSample <= -refclip)
+		{
+			RlpDepth += 0.1;
+			if (inputSampleR > -refclip)
+			{
+				RlastSample = ((-refclip*hardness) + (inputSampleR * softness));
+			}
+			else RlastSample = -refclip;
+		}
+		
+		if (inputSampleR > refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample < refclip)
+			{
+				inputSampleR = ((refclip*hardness) + (RlastSample * softness));
+			}
+			else inputSampleR = refclip;
+		}
+		
+		if (inputSampleR < -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample > -refclip)
+			{
+				inputSampleR = ((-refclip*hardness) + (RlastSample * softness));
+			}
+			else inputSampleR = -refclip;
+		}		
+		//end second R clip
+		
+		LoutputSample = LlastSample;
+		RoutputSample = RlastSample;
+		LlastSample = inputSampleL;
+		RlastSample = inputSampleR;
+		inputSampleL = LoutputSample;
+		inputSampleR = RoutputSample;
+		
+		LataHalfDrySample = (LataDrySample*ataK3)+(LataHalfDrySample*ataK4);
+		LataHalfDiffSample = (LataHalfwaySample - LataHalfDrySample)/2.0;
+		LataLastDiffSample = LataDiffSample*ataK5;
+		LataDiffSample = (inputSampleL - LataDrySample)/2.0;
+		LataDiffSample += LataHalfDiffSample;
+		LataDiffSample -= LataLastDiffSample;
+		inputSampleL = LataDrySample;
+		inputSampleL += LataDiffSample;
+		
+		RataHalfDrySample = (RataDrySample*ataK3)+(RataHalfDrySample*ataK4);
+		RataHalfDiffSample = (RataHalfwaySample - RataHalfDrySample)/2.0;
+		RataLastDiffSample = RataDiffSample*ataK5;
+		RataDiffSample = (inputSampleR - RataDrySample)/2.0;
+		RataDiffSample += RataHalfDiffSample;
+		RataDiffSample -= RataLastDiffSample;
+		inputSampleR = RataDrySample;
+		inputSampleR += RataDiffSample;
+		
+		Loverall = (Loverall * cancelold) + (LataDiffSample * cancelnew);
+		Roverall = (Roverall * cancelold) + (RataDiffSample * cancelnew);
+		//apply all the diffs to a lowpassed IIR
+
+		
+		if (flip)
+		{
+			LiirSampleA = (LiirSampleA * altAmount) + (inputSampleL * iirAmount);
+			inputSampleL -= LiirSampleA;
+			LiirSampleC = (LiirSampleC * altAmount) + (LpassThrough * iirAmount);
+			LpassThrough -= LiirSampleC;
+			
+			RiirSampleA = (RiirSampleA * altAmount) + (inputSampleR * iirAmount);
+			inputSampleR -= RiirSampleA;
+			RiirSampleC = (RiirSampleC * altAmount) + (RpassThrough * iirAmount);
+			RpassThrough -= RiirSampleC;
+		}
+		else
+		{
+			LiirSampleB = (LiirSampleB * altAmount) + (inputSampleL * iirAmount);
+			inputSampleL -= LiirSampleB;
+			LiirSampleD = (LiirSampleD * altAmount) + (LpassThrough * iirAmount);
+			LpassThrough -= LiirSampleD;
+			
+			RiirSampleB = (RiirSampleB * altAmount) + (inputSampleR * iirAmount);
+			inputSampleR -= RiirSampleB;
+			RiirSampleD = (RiirSampleD * altAmount) + (RpassThrough * iirAmount);
+			RpassThrough -= RiirSampleD;
+		}
+		flip = !flip;
+		//highpass section 
+		
+		LlastOut3Sample = LlastOut2Sample;
+		LlastOut2Sample = LlastOutSample;
+		LlastOutSample = inputSampleL;
+		
+		RlastOut3Sample = RlastOut2Sample;
+		RlastOut2Sample = RlastOutSample;
+		RlastOutSample = inputSampleR;
+		
+		
+		LlpDepth -= lpSpeed;
+		RlpDepth -= lpSpeed;
+
+		if (LlpDepth > 0.0)
+		{
+			if (LlpDepth > 1.0) LlpDepth = 1.0;
+			inputSampleL *= (1.0-LlpDepth);
+			inputSampleL += (((LlastOutSample + LlastOut2Sample + LlastOut3Sample) / 3.6)*LlpDepth);
+		}
+		
+		if (RlpDepth > 0.0)
+		{
+			if (RlpDepth > 1.0) RlpDepth = 1.0;
+			inputSampleR *= (1.0-RlpDepth);
+			inputSampleR += (((RlastOutSample + RlastOut2Sample + RlastOut3Sample) / 3.6)*RlpDepth);
+		}
+		
+		if (LlpDepth < 0.0) LlpDepth = 0.0;
+		if (RlpDepth < 0.0) RlpDepth = 0.0;
+				
+		//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
+		
+		inputSampleL *= (1.0-LmaxRecent);
+		inputSampleR *= (1.0-RmaxRecent);
+		inputSampleL += (LpassThrough * LmaxRecent);
+		inputSampleR += (RpassThrough * RmaxRecent);
+		//there's our raw signal, without antialiasing. Brings up low level stuff and softens more when hot
+		
+		if (inputSampleL > cliplevel) inputSampleL = cliplevel;
+		if (inputSampleL < -cliplevel) inputSampleL = -cliplevel;
+		if (inputSampleR > cliplevel) inputSampleR = cliplevel;
+		if (inputSampleR < -cliplevel) inputSampleR = -cliplevel;
+		//final iron bar
+		
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void AQuickVoiceClip::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 softness = 0.484416;
+	double hardness = 1.0 - softness;
+	double iirAmount = ((pow(A,3)*2070)+30)/8000.0;
+	iirAmount /= overallscale;
+	double altAmount = (1.0 - iirAmount);
+	double cancelnew = 0.0682276;
+	double cancelold = 1.0 - cancelnew;
+	double lpSpeed = 0.0009;
+	double cliplevel = 0.98;
+	double refclip = 0.5; //preset to cut out gain quite a lot. 91%? no touchy unless clip
+	
+	double LmaxRecent;
+	bool LclipOnset;
+	double LpassThrough;
+	double LoutputSample;
+	double LdrySample;
+	
+	double RmaxRecent;
+	bool RclipOnset;
+	double RpassThrough;
+	double RoutputSample;
+	double RdrySample;
+	
+	double fpTemp; //this is different from singlereplacing
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	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.
+		}
+		LpassThrough = LataDrySample = inputSampleL;
+		RpassThrough = RataDrySample = inputSampleR;
+		
+		LataHalfDrySample = LataHalfwaySample = (inputSampleL + LataLast1Sample + (LataLast2Sample*ataK1) + (LataLast3Sample*ataK2) + (LataLast4Sample*ataK6) + (LataLast5Sample*ataK7) + (LataLast6Sample*ataK8)) / 2.0;
+		LataLast6Sample = LataLast5Sample; LataLast5Sample = LataLast4Sample; LataLast4Sample = LataLast3Sample; LataLast3Sample = LataLast2Sample; LataLast2Sample = LataLast1Sample; LataLast1Sample = inputSampleL;
+		//setting up oversampled special antialiasing
+		RataHalfDrySample = RataHalfwaySample = (inputSampleR + RataLast1Sample + (RataLast2Sample*ataK1) + (RataLast3Sample*ataK2) + (RataLast4Sample*ataK6) + (RataLast5Sample*ataK7) + (RataLast6Sample*ataK8)) / 2.0;
+		RataLast6Sample = RataLast5Sample; RataLast5Sample = RataLast4Sample; RataLast4Sample = RataLast3Sample; RataLast3Sample = RataLast2Sample; RataLast2Sample = RataLast1Sample; RataLast1Sample = inputSampleR;
+		//setting up oversampled special antialiasing
+		LclipOnset = false;
+		RclipOnset = false;
+		
+		
+		LmaxRecent = fabs( LataLast6Sample );
+		if (fabs( LataLast5Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast5Sample );
+		if (fabs( LataLast4Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast4Sample );
+		if (fabs( LataLast3Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast3Sample );
+		if (fabs( LataLast2Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast2Sample );
+		if (fabs( LataLast1Sample ) > LmaxRecent ) LmaxRecent = fabs( LataLast1Sample );
+		if (fabs( inputSampleL ) > LmaxRecent ) LmaxRecent = fabs( inputSampleL );
+		//this gives us something that won't cut out in zero crossings, to interpolate with
+		
+		RmaxRecent = fabs( RataLast6Sample );
+		if (fabs( RataLast5Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast5Sample );
+		if (fabs( RataLast4Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast4Sample );
+		if (fabs( RataLast3Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast3Sample );
+		if (fabs( RataLast2Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast2Sample );
+		if (fabs( RataLast1Sample ) > RmaxRecent ) RmaxRecent = fabs( RataLast1Sample );
+		if (fabs( inputSampleR ) > RmaxRecent ) RmaxRecent = fabs( inputSampleR );
+		//this gives us something that won't cut out in zero crossings, to interpolate with
+		
+		LmaxRecent *= 2.0;
+		RmaxRecent *= 2.0;
+		//by refclip this is 1.0 and fully into the antialiasing
+		if (LmaxRecent > 1.0) LmaxRecent = 1.0;
+		if (RmaxRecent > 1.0) RmaxRecent = 1.0;
+		//and it tops out at 1. Higher means more antialiasing, lower blends into passThrough without antialiasing
+		
+		LataHalfwaySample -= Loverall;
+		RataHalfwaySample -= Roverall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		LdrySample = LataHalfwaySample;
+		RdrySample = RataHalfwaySample;
+		
+		
+		//begin L channel for the clipper
+		if (LlastSample >= refclip)
+		{
+			LlpDepth += 0.1;
+			if (LataHalfwaySample < refclip)
+			{
+				LlastSample = ((refclip*hardness) + (LataHalfwaySample * softness));
+			}
+			else LlastSample = refclip;
+		}
+		
+		if (LlastSample <= -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LataHalfwaySample > -refclip)
+			{
+				LlastSample = ((-refclip*hardness) + (LataHalfwaySample * softness));
+			}
+			else LlastSample = -refclip;
+		}
+		
+		if (LataHalfwaySample > refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample < refclip)
+			{
+				LataHalfwaySample = ((refclip*hardness) + (LlastSample * softness));
+			}
+			else LataHalfwaySample = refclip;
+		}
+		
+		if (LataHalfwaySample < -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample > -refclip)
+			{
+				LataHalfwaySample = ((-refclip*hardness) + (LlastSample * softness));
+			}
+			else LataHalfwaySample = -refclip;
+		}
+		///end L channel for the clipper
+		
+		//begin R channel for the clipper
+		if (RlastSample >= refclip)
+		{
+			RlpDepth += 0.1;
+			if (RataHalfwaySample < refclip)
+			{
+				RlastSample = ((refclip*hardness) + (RataHalfwaySample * softness));
+			}
+			else RlastSample = refclip;
+		}
+		
+		if (RlastSample <= -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RataHalfwaySample > -refclip)
+			{
+				RlastSample = ((-refclip*hardness) + (RataHalfwaySample * softness));
+			}
+			else RlastSample = -refclip;
+		}
+		
+		if (RataHalfwaySample > refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample < refclip)
+			{
+				RataHalfwaySample = ((refclip*hardness) + (RlastSample * softness));
+			}
+			else RataHalfwaySample = refclip;
+		}
+		
+		if (RataHalfwaySample < -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample > -refclip)
+			{
+				RataHalfwaySample = ((-refclip*hardness) + (RlastSample * softness));
+			}
+			else RataHalfwaySample = -refclip;
+		}
+		///end R channel for the clipper
+		
+        LoutputSample = LlastSample;
+        RoutputSample = RlastSample;
+		
+		LlastSample = LataHalfwaySample;
+		RlastSample = RataHalfwaySample;
+		
+		LataHalfwaySample = LoutputSample;
+		RataHalfwaySample = RoutputSample;
+		//swap around in a circle for one final ADClip,
+		//this time not tracking overshoot anymore
+		//end interpolated sample		
+		//begin raw sample- inputSample and ataDrySample handled separately here
+		
+		inputSampleL -= Loverall;
+		inputSampleR -= Roverall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		LdrySample = inputSampleL;
+		RdrySample = inputSampleR;
+		
+		//begin second L clip
+		if (LlastSample >= refclip)
+		{
+			LlpDepth += 0.1;
+			if (inputSampleL < refclip)
+			{
+				LlastSample = ((refclip*hardness) + (inputSampleL * softness));
+			}
+			else LlastSample = refclip;
+		}
+		
+		if (LlastSample <= -refclip)
+		{
+			LlpDepth += 0.1;
+			if (inputSampleL > -refclip)
+			{
+				LlastSample = ((-refclip*hardness) + (inputSampleL * softness));
+			}
+			else LlastSample = -refclip;
+		}
+		
+		if (inputSampleL > refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample < refclip)
+			{
+				inputSampleL = ((refclip*hardness) + (LlastSample * softness));
+			}
+			else inputSampleL = refclip;
+		}
+		
+		if (inputSampleL < -refclip)
+		{
+			LlpDepth += 0.1;
+			if (LlastSample > -refclip)
+			{
+				inputSampleL = ((-refclip*hardness) + (LlastSample * softness));
+			}
+			else inputSampleL = -refclip;
+		}		
+		//end second L clip
+		
+		//begin second R clip
+		if (RlastSample >= refclip)
+		{
+			RlpDepth += 0.1;
+			if (inputSampleR < refclip)
+			{
+				RlastSample = ((refclip*hardness) + (inputSampleR * softness));
+			}
+			else RlastSample = refclip;
+		}
+		
+		if (RlastSample <= -refclip)
+		{
+			RlpDepth += 0.1;
+			if (inputSampleR > -refclip)
+			{
+				RlastSample = ((-refclip*hardness) + (inputSampleR * softness));
+			}
+			else RlastSample = -refclip;
+		}
+		
+		if (inputSampleR > refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample < refclip)
+			{
+				inputSampleR = ((refclip*hardness) + (RlastSample * softness));
+			}
+			else inputSampleR = refclip;
+		}
+		
+		if (inputSampleR < -refclip)
+		{
+			RlpDepth += 0.1;
+			if (RlastSample > -refclip)
+			{
+				inputSampleR = ((-refclip*hardness) + (RlastSample * softness));
+			}
+			else inputSampleR = -refclip;
+		}		
+		//end second R clip
+		
+		LoutputSample = LlastSample;
+		RoutputSample = RlastSample;
+		LlastSample = inputSampleL;
+		RlastSample = inputSampleR;
+		inputSampleL = LoutputSample;
+		inputSampleR = RoutputSample;
+		
+		LataHalfDrySample = (LataDrySample*ataK3)+(LataHalfDrySample*ataK4);
+		LataHalfDiffSample = (LataHalfwaySample - LataHalfDrySample)/2.0;
+		LataLastDiffSample = LataDiffSample*ataK5;
+		LataDiffSample = (inputSampleL - LataDrySample)/2.0;
+		LataDiffSample += LataHalfDiffSample;
+		LataDiffSample -= LataLastDiffSample;
+		inputSampleL = LataDrySample;
+		inputSampleL += LataDiffSample;
+		
+		RataHalfDrySample = (RataDrySample*ataK3)+(RataHalfDrySample*ataK4);
+		RataHalfDiffSample = (RataHalfwaySample - RataHalfDrySample)/2.0;
+		RataLastDiffSample = RataDiffSample*ataK5;
+		RataDiffSample = (inputSampleR - RataDrySample)/2.0;
+		RataDiffSample += RataHalfDiffSample;
+		RataDiffSample -= RataLastDiffSample;
+		inputSampleR = RataDrySample;
+		inputSampleR += RataDiffSample;
+		
+		Loverall = (Loverall * cancelold) + (LataDiffSample * cancelnew);
+		Roverall = (Roverall * cancelold) + (RataDiffSample * cancelnew);
+		//apply all the diffs to a lowpassed IIR
+		
+		
+		if (flip)
+		{
+			LiirSampleA = (LiirSampleA * altAmount) + (inputSampleL * iirAmount);
+			inputSampleL -= LiirSampleA;
+			LiirSampleC = (LiirSampleC * altAmount) + (LpassThrough * iirAmount);
+			LpassThrough -= LiirSampleC;
+			
+			RiirSampleA = (RiirSampleA * altAmount) + (inputSampleR * iirAmount);
+			inputSampleR -= RiirSampleA;
+			RiirSampleC = (RiirSampleC * altAmount) + (RpassThrough * iirAmount);
+			RpassThrough -= RiirSampleC;
+		}
+		else
+		{
+			LiirSampleB = (LiirSampleB * altAmount) + (inputSampleL * iirAmount);
+			inputSampleL -= LiirSampleB;
+			LiirSampleD = (LiirSampleD * altAmount) + (LpassThrough * iirAmount);
+			LpassThrough -= LiirSampleD;
+			
+			RiirSampleB = (RiirSampleB * altAmount) + (inputSampleR * iirAmount);
+			inputSampleR -= RiirSampleB;
+			RiirSampleD = (RiirSampleD * altAmount) + (RpassThrough * iirAmount);
+			RpassThrough -= RiirSampleD;
+		}
+		flip = !flip;
+		//highpass section 
+		
+		LlastOut3Sample = LlastOut2Sample;
+		LlastOut2Sample = LlastOutSample;
+		LlastOutSample = inputSampleL;
+		
+		RlastOut3Sample = RlastOut2Sample;
+		RlastOut2Sample = RlastOutSample;
+		RlastOutSample = inputSampleR;
+		
+		
+		LlpDepth -= lpSpeed;
+		RlpDepth -= lpSpeed;
+		
+		if (LlpDepth > 0.0)
+		{
+			if (LlpDepth > 1.0) LlpDepth = 1.0;
+			inputSampleL *= (1.0-LlpDepth);
+			inputSampleL += (((LlastOutSample + LlastOut2Sample + LlastOut3Sample) / 3.6)*LlpDepth);
+		}
+		
+		if (RlpDepth > 0.0)
+		{
+			if (RlpDepth > 1.0) RlpDepth = 1.0;
+			inputSampleR *= (1.0-RlpDepth);
+			inputSampleR += (((RlastOutSample + RlastOut2Sample + RlastOut3Sample) / 3.6)*RlpDepth);
+		}
+		
+		if (LlpDepth < 0.0) LlpDepth = 0.0;
+		if (RlpDepth < 0.0) RlpDepth = 0.0;
+		
+		//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
+		
+		inputSampleL *= (1.0-LmaxRecent);
+		inputSampleR *= (1.0-RmaxRecent);
+		inputSampleL += (LpassThrough * LmaxRecent);
+		inputSampleR += (RpassThrough * RmaxRecent);
+		//there's our raw signal, without antialiasing. Brings up low level stuff and softens more when hot
+		
+		if (inputSampleL > cliplevel) inputSampleL = cliplevel;
+		if (inputSampleL < -cliplevel) inputSampleL = -cliplevel;
+		if (inputSampleR > cliplevel) inputSampleR = cliplevel;
+		if (inputSampleR < -cliplevel) inputSampleR = -cliplevel;
+		//final iron bar
+				
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
\ No newline at end of file