Biquad (and DeHiss)

7 files changed, 1017 insertions, 0 deletions

diff --git a/plugins/LinuxVST/CMakeLists.txt b/plugins/LinuxVST/CMakeLists.txt
index 1eff1ae..19de5da 100755
--- a/plugins/LinuxVST/CMakeLists.txt
+++ b/plugins/LinuxVST/CMakeLists.txt
@@ -17,6 +17,7 @@ add_airwindows_plugin(AtmosphereChannel)
 add_airwindows_plugin(Aura)
 add_airwindows_plugin(Average)
 add_airwindows_plugin(BassKit)
+add_airwindows_plugin(Biquad)
 add_airwindows_plugin(Bite)
 add_airwindows_plugin(BitGlitter)
 add_airwindows_plugin(BitShiftGain)
@@ -51,6 +52,7 @@ add_airwindows_plugin(curve)
 add_airwindows_plugin(DCVoltage)
 add_airwindows_plugin(Deckwrecka)
 add_airwindows_plugin(DeEss)
+add_airwindows_plugin(DeHiss)
 add_airwindows_plugin(Density)
 add_airwindows_plugin(DeRez)
 add_airwindows_plugin(DeRez2)
diff --git a/plugins/LinuxVST/src/Biquad/Biquad.cpp b/plugins/LinuxVST/src/Biquad/Biquad.cpp
new file mode 100755
index 0000000..b619691
--- /dev/null
+++ b/plugins/LinuxVST/src/Biquad/Biquad.cpp
@@ -0,0 +1,143 @@
+/* ========================================
+ *  Biquad - Biquad.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Biquad_H
+#include "Biquad.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Biquad(audioMaster);}
+
+Biquad::Biquad(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	for (int x = 0; x < 11; x++) {biquad[x] = 0.0;}
+	A = 1.0;
+	B = 0.5;
+	C = 0.5;
+	D = 1.0;
+	fpd = 17;
+	//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
+}
+
+Biquad::~Biquad() {}
+VstInt32 Biquad::getVendorVersion () {return 1000;}
+void Biquad::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Biquad::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 Biquad::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 Biquad::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 Biquad::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 Biquad::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 Biquad::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Type", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Freq", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Q", kVstMaxParamStrLen); break;
+		case kParamD: vst_strncpy (text, "Inv/Wet", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void Biquad::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((float)ceil((A*3.999)+0.00001), text, kVstMaxParamStrLen); break;
+        case kParamB: float2string ((B*B*B*0.9999)+0.0001, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string ((C*C*C*29.99)+0.01, text, kVstMaxParamStrLen); break;
+        case kParamD: float2string ((D*2.0)-1.0, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Biquad::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;
+        case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Biquad::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Biquad::getEffectName(char* name) {
+    vst_strncpy(name, "Biquad", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Biquad::getPlugCategory() {return kPlugCategEffect;}
+
+bool Biquad::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Biquad", kVstMaxProductStrLen); return true;
+}
+
+bool Biquad::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Biquad/Biquad.h b/plugins/LinuxVST/src/Biquad/Biquad.h
new file mode 100755
index 0000000..17993cd
--- /dev/null
+++ b/plugins/LinuxVST/src/Biquad/Biquad.h
@@ -0,0 +1,71 @@
+/* ========================================
+ *  Biquad - Biquad.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Biquad_H
+#define __Biquad_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 = 'biqd';    //Change this to what the AU identity is!
+
+class Biquad : 
+    public AudioEffectX 
+{
+public:
+    Biquad(audioMasterCallback audioMaster);
+    ~Biquad();
+    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 biquad[11]; //note that this stereo form doesn't require L and R forms!
+	//This is because so much of it is coefficients etc. that are the same on both channels.
+	//So the stored samples are in 7-8 and 9-10, and freq/res/coefficients serve both.
+	
+	uint32_t fpd;
+	//default stuff
+
+    float A;
+    float B;
+    float C;
+    float D;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Biquad/BiquadProc.cpp b/plugins/LinuxVST/src/Biquad/BiquadProc.cpp
new file mode 100755
index 0000000..0dfbfe4
--- /dev/null
+++ b/plugins/LinuxVST/src/Biquad/BiquadProc.cpp
@@ -0,0 +1,322 @@
+/* ========================================
+ *  Biquad - Biquad.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Biquad_H
+#include "Biquad.h"
+#endif
+
+void Biquad::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();
+
+	int type = ceil((A*3.999)+0.00001);
+	
+	biquad[0] = ((B*B*B*0.9999)+0.0001)*0.499;
+	if (biquad[0] < 0.0001) biquad[0] = 0.0001;
+	
+    biquad[1] = (C*C*C*29.99)+0.01;
+	if (biquad[1] < 0.0001) biquad[1] = 0.0001;
+	
+	double wet = (D*2.0)-1.0;
+	
+	//biquad contains these values:
+	//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
+	//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
+	//[2] is a0 but you need distinct ones for additional biquad instances so it's here
+	//[3] is a1 but you need distinct ones for additional biquad instances so it's here
+	//[4] is a2 but you need distinct ones for additional biquad instances so it's here
+	//[5] is b1 but you need distinct ones for additional biquad instances so it's here
+	//[6] is b2 but you need distinct ones for additional biquad instances so it's here
+	//[7] is LEFT stored delayed sample (freq and res are stored so you can move them sample by sample)
+	//[8] is LEFT stored delayed sample (you have to include the coefficient making code if you do that)
+	//[9] is RIGHT stored delayed sample (freq and res are stored so you can move them sample by sample)
+	//[10] is RIGHT stored delayed sample (you have to include the coefficient making code if you do that)
+	
+	//to build a dedicated filter, rename 'biquad' to whatever the new filter is, then
+	//put this code either within the sample buffer (for smoothly modulating freq or res)
+	//or in this 'read the controls' area (for letting you change freq and res with controls)
+	//or in 'reset' if the freq and res are absolutely fixed (use GetSampleRate to define freq)
+	
+	if (type == 1) { //lowpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = K * K * norm;
+		biquad[3] = 2.0 * biquad[2];
+		biquad[4] = biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 2) { //highpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = norm;
+		biquad[3] = -2.0 * biquad[2];
+		biquad[4] = biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 3) { //bandpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = K / biquad[1] * norm;
+		biquad[3] = 0.0; //bandpass can simplify the biquad kernel: leave out this multiply
+		biquad[4] = -biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 4) { //notch
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = (1.0 + K * K) * norm;
+		biquad[3] = 2.0 * (K * K - 1) * norm;
+		biquad[4] = biquad[2];
+		biquad[5] = biquad[3];
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
+		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		
+		inputSampleL = sin(inputSampleL);
+		inputSampleR = sin(inputSampleR);
+		//encode Console5: good cleanness
+		
+		/*
+		 long double mid = inputSampleL + inputSampleR;
+		 long double side = inputSampleL - inputSampleR;
+		 //assign mid and side.Between these sections, you can do mid/side processing
+		 
+		 long double tempSampleM = (mid * biquad[2]) + biquad[7];
+		 biquad[7] = (mid * biquad[3]) - (tempSampleM * biquad[5]) + biquad[8];
+		 biquad[8] = (mid * biquad[4]) - (tempSampleM * biquad[6]);
+		 mid = tempSampleM; //like mono AU, 7 and 8 store mid channel
+		 
+		 long double tempSampleS = (side * biquad[2]) + biquad[9];
+		 biquad[9] = (side * biquad[3]) - (tempSampleS * biquad[5]) + biquad[10];
+		 biquad[10] = (side * biquad[4]) - (tempSampleS * biquad[6]);
+		 inputSampleR = tempSampleS; //note: 9 and 10 store the side channel		
+		 
+		 inputSampleL = (mid+side)/2.0;
+		 inputSampleR = (mid-side)/2.0;
+		 //unassign mid and side
+		 */
+		
+		long double tempSampleL = (inputSampleL * biquad[2]) + biquad[7];
+		biquad[7] = (inputSampleL * biquad[3]) - (tempSampleL * biquad[5]) + biquad[8];
+		biquad[8] = (inputSampleL * biquad[4]) - (tempSampleL * biquad[6]);
+		inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel
+		
+		long double tempSampleR = (inputSampleR * biquad[2]) + biquad[9];
+		biquad[9] = (inputSampleR * biquad[3]) - (tempSampleR * biquad[5]) + biquad[10];
+		biquad[10] = (inputSampleR * biquad[4]) - (tempSampleR * biquad[6]);
+		inputSampleR = tempSampleR; //note: 9 and 10 store the R channel
+		
+		if (inputSampleL > 1.0) inputSampleL = 1.0;
+		if (inputSampleL < -1.0) inputSampleL = -1.0;
+		if (inputSampleR > 1.0) inputSampleR = 1.0;
+		if (inputSampleR < -1.0) inputSampleR = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleL = asin(inputSampleL);
+		inputSampleR = asin(inputSampleR);
+		//amplitude aspect
+		
+		if (wet < 1.0) {
+			inputSampleL = (inputSampleL*wet) + (drySampleL*(1.0-fabs(wet)));
+			inputSampleR = (inputSampleR*wet) + (drySampleR*(1.0-fabs(wet)));
+			//inv/dry/wet lets us turn LP into HP and band into notch
+		}
+		
+		//begin 32 bit stereo floating point dither
+		int expon; frexpf((float)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		frexpf((float)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		//end 32 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void Biquad::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();
+	
+	int type = ceil((A*3.999)+0.00001);
+	
+	biquad[0] = ((B*B*B*0.9999)+0.0001)*0.499;
+	if (biquad[0] < 0.0001) biquad[0] = 0.0001;
+	
+    biquad[1] = (C*C*C*29.99)+0.01;
+	if (biquad[1] < 0.0001) biquad[1] = 0.0001;
+	
+	double wet = (D*2.0)-1.0;
+	
+	//biquad contains these values:
+	//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
+	//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
+	//[2] is a0 but you need distinct ones for additional biquad instances so it's here
+	//[3] is a1 but you need distinct ones for additional biquad instances so it's here
+	//[4] is a2 but you need distinct ones for additional biquad instances so it's here
+	//[5] is b1 but you need distinct ones for additional biquad instances so it's here
+	//[6] is b2 but you need distinct ones for additional biquad instances so it's here
+	//[7] is LEFT stored delayed sample (freq and res are stored so you can move them sample by sample)
+	//[8] is LEFT stored delayed sample (you have to include the coefficient making code if you do that)
+	//[9] is RIGHT stored delayed sample (freq and res are stored so you can move them sample by sample)
+	//[10] is RIGHT stored delayed sample (you have to include the coefficient making code if you do that)
+	
+	//to build a dedicated filter, rename 'biquad' to whatever the new filter is, then
+	//put this code either within the sample buffer (for smoothly modulating freq or res)
+	//or in this 'read the controls' area (for letting you change freq and res with controls)
+	//or in 'reset' if the freq and res are absolutely fixed (use GetSampleRate to define freq)
+	
+	if (type == 1) { //lowpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = K * K * norm;
+		biquad[3] = 2.0 * biquad[2];
+		biquad[4] = biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 2) { //highpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = norm;
+		biquad[3] = -2.0 * biquad[2];
+		biquad[4] = biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 3) { //bandpass
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = K / biquad[1] * norm;
+		biquad[3] = 0.0; //bandpass can simplify the biquad kernel: leave out this multiply
+		biquad[4] = -biquad[2];
+		biquad[5] = 2.0 * (K * K - 1.0) * norm;
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+	if (type == 4) { //notch
+		double K = tan(M_PI * biquad[0]);
+		double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
+		biquad[2] = (1.0 + K * K) * norm;
+		biquad[3] = 2.0 * (K * K - 1) * norm;
+		biquad[4] = biquad[2];
+		biquad[5] = biquad[3];
+		biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
+	}
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
+		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		
+		inputSampleL = sin(inputSampleL);
+		inputSampleR = sin(inputSampleR);
+		//encode Console5: good cleanness
+		
+		/*
+		 long double mid = inputSampleL + inputSampleR;
+		 long double side = inputSampleL - inputSampleR;
+		 //assign mid and side.Between these sections, you can do mid/side processing
+		 
+		 long double tempSampleM = (mid * biquad[2]) + biquad[7];
+		 biquad[7] = (mid * biquad[3]) - (tempSampleM * biquad[5]) + biquad[8];
+		 biquad[8] = (mid * biquad[4]) - (tempSampleM * biquad[6]);
+		 mid = tempSampleM; //like mono AU, 7 and 8 store mid channel
+		 
+		 long double tempSampleS = (side * biquad[2]) + biquad[9];
+		 biquad[9] = (side * biquad[3]) - (tempSampleS * biquad[5]) + biquad[10];
+		 biquad[10] = (side * biquad[4]) - (tempSampleS * biquad[6]);
+		 inputSampleR = tempSampleS; //note: 9 and 10 store the side channel		
+		 
+		 inputSampleL = (mid+side)/2.0;
+		 inputSampleR = (mid-side)/2.0;
+		 //unassign mid and side
+		 */
+		
+		long double tempSampleL = (inputSampleL * biquad[2]) + biquad[7];
+		biquad[7] = (inputSampleL * biquad[3]) - (tempSampleL * biquad[5]) + biquad[8];
+		biquad[8] = (inputSampleL * biquad[4]) - (tempSampleL * biquad[6]);
+		inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel
+		
+		long double tempSampleR = (inputSampleR * biquad[2]) + biquad[9];
+		biquad[9] = (inputSampleR * biquad[3]) - (tempSampleR * biquad[5]) + biquad[10];
+		biquad[10] = (inputSampleR * biquad[4]) - (tempSampleR * biquad[6]);
+		inputSampleR = tempSampleR; //note: 9 and 10 store the R channel
+		
+		if (inputSampleL > 1.0) inputSampleL = 1.0;
+		if (inputSampleL < -1.0) inputSampleL = -1.0;
+		if (inputSampleR > 1.0) inputSampleR = 1.0;
+		if (inputSampleR < -1.0) inputSampleR = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleL = asin(inputSampleL);
+		inputSampleR = asin(inputSampleR);
+		//amplitude aspect
+		
+		if (wet < 1.0) {
+			inputSampleL = (inputSampleL*wet) + (drySampleL*(1.0-fabs(wet)));
+			inputSampleR = (inputSampleR*wet) + (drySampleR*(1.0-fabs(wet)));
+			//inv/dry/wet lets us turn LP into HP and band into notch
+		}
+		
+		//begin 64 bit stereo floating point dither
+		int expon; frexp((double)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		frexp((double)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		//end 64 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
diff --git a/plugins/LinuxVST/src/DeHiss/DeHiss.cpp b/plugins/LinuxVST/src/DeHiss/DeHiss.cpp
new file mode 100755
index 0000000..1f2d563
--- /dev/null
+++ b/plugins/LinuxVST/src/DeHiss/DeHiss.cpp
@@ -0,0 +1,137 @@
+/* ========================================
+ *  DeHiss - DeHiss.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __DeHiss_H
+#include "DeHiss.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new DeHiss(audioMaster);}
+
+DeHiss::DeHiss(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 1.0;
+	
+	storedL[0] = storedL[1] = 0.0;
+	diffL[0] = diffL[1] = diffL[2] = diffL[3] = diffL[4] = diffL[5] = 0.0;
+	gateL = 1.0;
+	rawL = 2.0;
+
+	storedR[0] = storedR[1] = 0.0;
+	diffR[0] = diffR[1] = diffR[2] = diffR[3] = diffR[4] = diffR[5] = 0.0;
+	gateR = 1.0;
+	rawR = 2.0;	
+	
+	fpd = 17;
+	//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
+}
+
+DeHiss::~DeHiss() {}
+VstInt32 DeHiss::getVendorVersion () {return 1000;}
+void DeHiss::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void DeHiss::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 DeHiss::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	/* Note: The way this is set up, it will break if you manage to save settings on an Intel
+	 machine and load them on a PPC Mac. However, it's fine if you stick to the machine you 
+	 started with. */
+	
+	*data = chunkData;
+	return kNumParameters * sizeof(float);
+}
+
+VstInt32 DeHiss::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	/* We're ignoring byteSize as we found it to be a filthy liar */
+	
+	/* calculate any other fields you need here - you could copy in 
+	 code from setParameter() here. */
+	return 0;
+}
+
+void DeHiss::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float DeHiss::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        default: break; // unknown parameter, shouldn't happen!
+    } return 0.0; //we only need to update the relevant name, this is simple to manage
+}
+
+void DeHiss::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Thresh", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Dry/Wet", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void DeHiss::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void DeHiss::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 DeHiss::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool DeHiss::getEffectName(char* name) {
+    vst_strncpy(name, "DeHiss", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory DeHiss::getPlugCategory() {return kPlugCategEffect;}
+
+bool DeHiss::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows DeHiss", kVstMaxProductStrLen); return true;
+}
+
+bool DeHiss::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/DeHiss/DeHiss.h b/plugins/LinuxVST/src/DeHiss/DeHiss.h
new file mode 100755
index 0000000..f8fb0a7
--- /dev/null
+++ b/plugins/LinuxVST/src/DeHiss/DeHiss.h
@@ -0,0 +1,72 @@
+/* ========================================
+ *  DeHiss - DeHiss.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __DeHiss_H
+#define __DeHiss_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+  kNumParameters = 2
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'Dehs';    //Change this to what the AU identity is!
+
+class DeHiss : 
+    public AudioEffectX 
+{
+public:
+    DeHiss(audioMasterCallback audioMaster);
+    ~DeHiss();
+    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;
+    
+	uint32_t fpd;
+	//default stuff
+	double storedL[2];
+	double diffL[6];
+	double gateL;
+	double rawL;
+
+	double storedR[2];
+	double diffR[6];
+	double gateR;
+	double rawR;
+
+    float A;
+    float B;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/DeHiss/DeHissProc.cpp b/plugins/LinuxVST/src/DeHiss/DeHissProc.cpp
new file mode 100755
index 0000000..405566e
--- /dev/null
+++ b/plugins/LinuxVST/src/DeHiss/DeHissProc.cpp
@@ -0,0 +1,270 @@
+/* ========================================
+ *  DeHiss - DeHiss.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __DeHiss_H
+#include "DeHiss.h"
+#endif
+
+void DeHiss::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 meanAL;
+	double meanBL;
+	double meanOutL = 0;
+	double meanLastL;
+	double averageL[5];
+	
+	double meanAR;
+	double meanBR;
+	double meanOutR = 0;
+	double meanLastR;
+	double averageR[5];
+	
+	double threshold = pow(A,9);
+	double wet = B;
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
+		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		//begin L
+		storedL[1] = storedL[0];
+		storedL[0] = inputSampleL;
+		diffL[5] = diffL[4];
+		diffL[4] = diffL[3];
+		diffL[3] = diffL[2];
+		diffL[2] = diffL[1];
+		diffL[1] = diffL[0];
+		diffL[0] = storedL[0] - storedL[1];
+		
+		averageL[4] = (diffL[0] + diffL[1] + diffL[2] + diffL[3] + diffL[4] + diffL[5])/6.0;
+		averageL[3] = (diffL[0] + diffL[1] + diffL[2] + diffL[3] + diffL[4])/5.0;
+		averageL[2] = (diffL[0] + diffL[1] + diffL[2] + diffL[3])/4.0;
+		averageL[1] = (diffL[0] + diffL[1] + diffL[2])/3.0;
+		averageL[0] = (diffL[0] + diffL[1])/2.0;
+		
+		meanAL = diffL[0];
+		meanBL = diffL[0];
+		if (fabs(averageL[4]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[4];}
+		if (fabs(averageL[3]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[3];}
+		if (fabs(averageL[2]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[2];}
+		if (fabs(averageL[1]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[1];}
+		if (fabs(averageL[0]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[0];}
+		meanLastL = meanOutL;
+		meanOutL = ((meanAL+meanBL)/2.0);
+		
+		if (rawL > 0) rawL -= 0.001;
+		if (fabs(inputSampleL) > threshold) {gateL = 1.0; rawL = 2.0;}
+		else {gateL = (gateL * 0.999); if (threshold > 0) gateL += ((fabs(inputSampleL)/threshold) * 0.001);}
+		
+		if ((fabs(meanOutL) > threshold) || (fabs(meanLastL) > threshold)){}
+		else storedL[0] = storedL[1] + (meanOutL * gateL);
+		
+		if (rawL < 1) inputSampleL = (inputSampleL * rawL) + (storedL[0] * (1-rawL));
+		//end L
+		
+		//begin R
+		storedR[1] = storedR[0];
+		storedR[0] = inputSampleR;
+		diffR[5] = diffR[4];
+		diffR[4] = diffR[3];
+		diffR[3] = diffR[2];
+		diffR[2] = diffR[1];
+		diffR[1] = diffR[0];
+		diffR[0] = storedR[0] - storedR[1];
+		
+		averageR[4] = (diffR[0] + diffR[1] + diffR[2] + diffR[3] + diffR[4] + diffR[5])/6.0;
+		averageR[3] = (diffR[0] + diffR[1] + diffR[2] + diffR[3] + diffR[4])/5.0;
+		averageR[2] = (diffR[0] + diffR[1] + diffR[2] + diffR[3])/4.0;
+		averageR[1] = (diffR[0] + diffR[1] + diffR[2])/3.0;
+		averageR[0] = (diffR[0] + diffR[1])/2.0;
+		
+		meanAR = diffR[0];
+		meanBR = diffR[0];
+		if (fabs(averageR[4]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[4];}
+		if (fabs(averageR[3]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[3];}
+		if (fabs(averageR[2]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[2];}
+		if (fabs(averageR[1]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[1];}
+		if (fabs(averageR[0]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[0];}
+		meanLastR = meanOutR;
+		meanOutR = ((meanAR+meanBR)/2.0);
+		
+		if (rawR > 0) rawR -= 0.001;
+		if (fabs(inputSampleR) > threshold) {gateR = 1.0; rawR = 2.0;}
+		else {gateR = (gateR * 0.999); if (threshold > 0) gateR += ((fabs(inputSampleR)/threshold) * 0.001);}
+		
+		if ((fabs(meanOutR) > threshold) || (fabs(meanLastR) > threshold)){}
+		else storedR[0] = storedR[1] + (meanOutR * gateR);
+		
+		if (rawR < 1) inputSampleR = (inputSampleR * rawR) + (storedR[0] * (1-rawR));
+		//end R
+		
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
+			inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
+		}
+		
+		//begin 32 bit stereo floating point dither
+		int expon; frexpf((float)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		frexpf((float)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		//end 32 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void DeHiss::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 meanAL;
+	double meanBL;
+	double meanOutL = 0;
+	double meanLastL;
+	double averageL[5];
+	
+	double meanAR;
+	double meanBR;
+	double meanOutR = 0;
+	double meanLastR;
+	double averageR[5];
+	
+	double threshold = pow(A,9);
+	double wet = B;
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
+		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		//begin L
+		storedL[1] = storedL[0];
+		storedL[0] = inputSampleL;
+		diffL[5] = diffL[4];
+		diffL[4] = diffL[3];
+		diffL[3] = diffL[2];
+		diffL[2] = diffL[1];
+		diffL[1] = diffL[0];
+		diffL[0] = storedL[0] - storedL[1];
+		
+		averageL[4] = (diffL[0] + diffL[1] + diffL[2] + diffL[3] + diffL[4] + diffL[5])/6.0;
+		averageL[3] = (diffL[0] + diffL[1] + diffL[2] + diffL[3] + diffL[4])/5.0;
+		averageL[2] = (diffL[0] + diffL[1] + diffL[2] + diffL[3])/4.0;
+		averageL[1] = (diffL[0] + diffL[1] + diffL[2])/3.0;
+		averageL[0] = (diffL[0] + diffL[1])/2.0;
+		
+		meanAL = diffL[0];
+		meanBL = diffL[0];
+		if (fabs(averageL[4]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[4];}
+		if (fabs(averageL[3]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[3];}
+		if (fabs(averageL[2]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[2];}
+		if (fabs(averageL[1]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[1];}
+		if (fabs(averageL[0]) < fabs(meanBL)) {meanAL = meanBL; meanBL = averageL[0];}
+		meanLastL = meanOutL;
+		meanOutL = ((meanAL+meanBL)/2.0);
+		
+		if (rawL > 0) rawL -= 0.001;
+		if (fabs(inputSampleL) > threshold) {gateL = 1.0; rawL = 2.0;}
+		else {gateL = (gateL * 0.999); if (threshold > 0) gateL += ((fabs(inputSampleL)/threshold) * 0.001);}
+		
+		if ((fabs(meanOutL) > threshold) || (fabs(meanLastL) > threshold)){}
+		else storedL[0] = storedL[1] + (meanOutL * gateL);
+		
+		if (rawL < 1) inputSampleL = (inputSampleL * rawL) + (storedL[0] * (1-rawL));
+		//end L
+		
+		//begin R
+		storedR[1] = storedR[0];
+		storedR[0] = inputSampleR;
+		diffR[5] = diffR[4];
+		diffR[4] = diffR[3];
+		diffR[3] = diffR[2];
+		diffR[2] = diffR[1];
+		diffR[1] = diffR[0];
+		diffR[0] = storedR[0] - storedR[1];
+		
+		averageR[4] = (diffR[0] + diffR[1] + diffR[2] + diffR[3] + diffR[4] + diffR[5])/6.0;
+		averageR[3] = (diffR[0] + diffR[1] + diffR[2] + diffR[3] + diffR[4])/5.0;
+		averageR[2] = (diffR[0] + diffR[1] + diffR[2] + diffR[3])/4.0;
+		averageR[1] = (diffR[0] + diffR[1] + diffR[2])/3.0;
+		averageR[0] = (diffR[0] + diffR[1])/2.0;
+		
+		meanAR = diffR[0];
+		meanBR = diffR[0];
+		if (fabs(averageR[4]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[4];}
+		if (fabs(averageR[3]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[3];}
+		if (fabs(averageR[2]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[2];}
+		if (fabs(averageR[1]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[1];}
+		if (fabs(averageR[0]) < fabs(meanBR)) {meanAR = meanBR; meanBR = averageR[0];}
+		meanLastR = meanOutR;
+		meanOutR = ((meanAR+meanBR)/2.0);
+		
+		if (rawR > 0) rawR -= 0.001;
+		if (fabs(inputSampleR) > threshold) {gateR = 1.0; rawR = 2.0;}
+		else {gateR = (gateR * 0.999); if (threshold > 0) gateR += ((fabs(inputSampleR)/threshold) * 0.001);}
+		
+		if ((fabs(meanOutR) > threshold) || (fabs(meanLastR) > threshold)){}
+		else storedR[0] = storedR[1] + (meanOutR * gateR);
+		
+		if (rawR < 1) inputSampleR = (inputSampleR * rawR) + (storedR[0] * (1-rawR));
+		//end R
+		
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
+			inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
+		}
+		
+		//begin 64 bit stereo floating point dither
+		int expon; frexp((double)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		frexp((double)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		//end 64 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}