AverMatrix and Dark

7 files changed, 828 insertions, 0 deletions

diff --git a/plugins/LinuxVST/CMakeLists.txt b/plugins/LinuxVST/CMakeLists.txt
index f1197f4..0be4c44 100755
--- a/plugins/LinuxVST/CMakeLists.txt
+++ b/plugins/LinuxVST/CMakeLists.txt
@@ -17,6 +17,7 @@ add_airwindows_plugin(AtmosphereBuss)
 add_airwindows_plugin(AtmosphereChannel)
 add_airwindows_plugin(Aura)
 add_airwindows_plugin(Average)
+add_airwindows_plugin(AverMatrix)
 add_airwindows_plugin(BassDrive)
 add_airwindows_plugin(BassKit)
 add_airwindows_plugin(Baxandall)
@@ -59,6 +60,7 @@ add_airwindows_plugin(CrunchyGrooveWear)
 add_airwindows_plugin(Crystal)
 add_airwindows_plugin(CStrip)
 add_airwindows_plugin(curve)
+add_airwindows_plugin(Dark)
 add_airwindows_plugin(DCVoltage)
 add_airwindows_plugin(Deckwrecka)
 add_airwindows_plugin(DeBess)
diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp b/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp
new file mode 100755
index 0000000..585bd85
--- /dev/null
+++ b/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp
@@ -0,0 +1,141 @@
+/* ========================================
+ *  AverMatrix - AverMatrix.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __AverMatrix_H
+#include "AverMatrix.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new AverMatrix(audioMaster);}
+
+AverMatrix::AverMatrix(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 0.0;
+	C = 1.0;
+	for(int x = 0; x < 11; x++) {
+		f[x] = 0.0;
+		for (int y = 0; y < 11; y++) {
+			bL[x][y] = 0.0; bR[x][y] = 0.0;
+		}
+	}
+
+	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
+}
+
+AverMatrix::~AverMatrix() {}
+VstInt32 AverMatrix::getVendorVersion () {return 1000;}
+void AverMatrix::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void AverMatrix::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 AverMatrix::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	chunkData[2] = C;
+	/* Note: The way this is set up, it will break if you manage to save settings on an Intel
+	 machine and load them on a PPC Mac. However, it's fine if you stick to the machine you 
+	 started with. */
+	
+	*data = chunkData;
+	return kNumParameters * sizeof(float);
+}
+
+VstInt32 AverMatrix::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	C = pinParameter(chunkData[2]);
+	/* We're ignoring byteSize as we found it to be a filthy liar */
+	
+	/* calculate any other fields you need here - you could copy in 
+	 code from setParameter() here. */
+	return 0;
+}
+
+void AverMatrix::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float AverMatrix::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        case kParamC: return C; break;
+        default: break; // unknown parameter, shouldn't happen!
+    } return 0.0; //we only need to update the relevant name, this is simple to manage
+}
+
+void AverMatrix::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Average", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Depth", kVstMaxParamStrLen); break;
+		case kParamC: 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 AverMatrix::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((A * 9.0)+1.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string ((B * 9.0)+1.0, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string ((C * 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 AverMatrix::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "taps", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "poles", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 AverMatrix::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool AverMatrix::getEffectName(char* name) {
+    vst_strncpy(name, "AverMatrix", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory AverMatrix::getPlugCategory() {return kPlugCategEffect;}
+
+bool AverMatrix::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows AverMatrix", kVstMaxProductStrLen); return true;
+}
+
+bool AverMatrix::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrix.h b/plugins/LinuxVST/src/AverMatrix/AverMatrix.h
new file mode 100755
index 0000000..b5f2012
--- /dev/null
+++ b/plugins/LinuxVST/src/AverMatrix/AverMatrix.h
@@ -0,0 +1,68 @@
+/* ========================================
+ *  AverMatrix - AverMatrix.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __AverMatrix_H
+#define __AverMatrix_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+	kParamC = 2,
+  kNumParameters = 3
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'avrm';    //Change this to what the AU identity is!
+
+class AverMatrix : 
+    public AudioEffectX 
+{
+public:
+    AverMatrix(audioMasterCallback audioMaster);
+    ~AverMatrix();
+    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 bL[11][11];
+	double bR[11][11];
+	double f[11];		
+	uint32_t fpd;
+	//default stuff
+
+    float A;
+    float B;
+    float C;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp b/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp
new file mode 100755
index 0000000..381ea9c
--- /dev/null
+++ b/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp
@@ -0,0 +1,191 @@
+/* ========================================
+ *  AverMatrix - AverMatrix.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __AverMatrix_H
+#include "AverMatrix.h"
+#endif
+
+void AverMatrix::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	double overalltaps = (A * 9.0)+1.0;
+	double taps = overalltaps;
+	//this is our averaging, which is not integer but continuous
+	
+	double overallpoles = (B * 9.0)+1.0;
+	//this is the poles of the filter, also not integer but continuous
+	int yLimit = floor(overallpoles)+1;
+	double yPartial = overallpoles - floor(overallpoles);
+	//now we can do a for loop, and also apply the final pole continuously
+	
+	double wet = (C * 2.0)-1.0;
+	double dry = (1.0-wet);
+	if (dry > 1.0) dry = 1.0;
+	
+	int xLimit = 1;
+	for(int x = 0; x < 11; x++) {
+		if (taps > 1.0) {
+			f[x] = 1.0;
+			taps -= 1.0;
+			xLimit++;
+		} else {
+			f[x] = taps;
+			taps = 0.0;
+		}
+	} //there, now we have a neat little moving average with remainders
+	if (xLimit > 9) xLimit = 9;
+	
+	if (overalltaps < 1.0) overalltaps = 1.0;
+	for(int x = 0; x < xLimit; x++) {
+		f[x] /= overalltaps;
+	} //and now it's neatly scaled, too	
+    
+    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;
+		
+		long double previousPoleL = 0;		
+		long double previousPoleR = 0;		
+		for (int y = 0; y < yLimit; y++) {
+			for (int x = xLimit; x >= 0; x--) {
+				bL[x+1][y] = bL[x][y];
+				bR[x+1][y] = bR[x][y];
+			}
+			bL[0][y] = previousPoleL = inputSampleL;
+			bR[0][y] = previousPoleR = inputSampleR;
+			inputSampleL = 0.0;
+			inputSampleR = 0.0;
+			for (int x = 0; x < xLimit; x++) {
+				inputSampleL += (bL[x][y] * f[x]);
+				inputSampleR += (bR[x][y] * f[x]);
+			}
+		}
+		inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial);
+		inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial);
+		//in this way we can blend in the final pole
+		
+		inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+		inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		//wet can be negative, in which case dry is always full volume and they cancel
+		
+		//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 AverMatrix::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	
+	double overalltaps = (A * 9.0)+1.0;
+	double taps = overalltaps;
+	//this is our averaging, which is not integer but continuous
+	
+	double overallpoles = (B * 9.0)+1.0;
+	//this is the poles of the filter, also not integer but continuous
+	int yLimit = floor(overallpoles)+1;
+	double yPartial = overallpoles - floor(overallpoles);
+	//now we can do a for loop, and also apply the final pole continuously
+	
+	double wet = (C * 2.0)-1.0;
+	double dry = (1.0-wet);
+	if (dry > 1.0) dry = 1.0;
+	
+	int xLimit = 1;
+	for(int x = 0; x < 11; x++) {
+		if (taps > 1.0) {
+			f[x] = 1.0;
+			taps -= 1.0;
+			xLimit++;
+		} else {
+			f[x] = taps;
+			taps = 0.0;
+		}
+	} //there, now we have a neat little moving average with remainders
+	if (xLimit > 9) xLimit = 9;
+	
+	if (overalltaps < 1.0) overalltaps = 1.0;
+	for(int x = 0; x < xLimit; x++) {
+		f[x] /= overalltaps;
+	} //and now it's neatly scaled, too	
+	
+    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;
+
+		long double previousPoleL = 0;		
+		long double previousPoleR = 0;		
+		for (int y = 0; y < yLimit; y++) {
+			for (int x = xLimit; x >= 0; x--) {
+				bL[x+1][y] = bL[x][y];
+				bR[x+1][y] = bR[x][y];
+			}
+			bL[0][y] = previousPoleL = inputSampleL;
+			bR[0][y] = previousPoleR = inputSampleR;
+			inputSampleL = 0.0;
+			inputSampleR = 0.0;
+			for (int x = 0; x < xLimit; x++) {
+				inputSampleL += (bL[x][y] * f[x]);
+				inputSampleR += (bR[x][y] * f[x]);
+			}
+		}
+		inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial);
+		inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial);
+		//in this way we can blend in the final pole
+		
+		inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+		inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		//wet can be negative, in which case dry is always full volume and they cancel
+		
+		//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/Dark/Dark.cpp b/plugins/LinuxVST/src/Dark/Dark.cpp
new file mode 100755
index 0000000..0f9721c
--- /dev/null
+++ b/plugins/LinuxVST/src/Dark/Dark.cpp
@@ -0,0 +1,126 @@
+/* ========================================
+ *  Dark - Dark.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Dark_H
+#include "Dark.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Dark(audioMaster);}
+
+Dark::Dark(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 1.0;
+	for(int count = 0; count < 99; count++) {
+		lastSampleL[count] = 0;
+		lastSampleR[count] = 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
+}
+
+Dark::~Dark() {}
+VstInt32 Dark::getVendorVersion () {return 1000;}
+void Dark::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Dark::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 Dark::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 Dark::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 Dark::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+		default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float Dark::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 Dark::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Quant", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void Dark::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: switch((VstInt32)( A * 1.999 )) //0 to almost edge of # of params
+		{	case 0: vst_strncpy (text, "CD 16", kVstMaxParamStrLen); break;
+			case 1: vst_strncpy (text, "HD 24", kVstMaxParamStrLen); break;
+			default: break; // unknown parameter, shouldn't happen!
+		} break; //completed consoletype 'popup' parameter, exit
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Dark::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Dark::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Dark::getEffectName(char* name) {
+    vst_strncpy(name, "Dark", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Dark::getPlugCategory() {return kPlugCategEffect;}
+
+bool Dark::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Dark", kVstMaxProductStrLen); return true;
+}
+
+bool Dark::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Dark/Dark.h b/plugins/LinuxVST/src/Dark/Dark.h
new file mode 100755
index 0000000..a54d31b
--- /dev/null
+++ b/plugins/LinuxVST/src/Dark/Dark.h
@@ -0,0 +1,63 @@
+/* ========================================
+ *  Dark - Dark.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Dark_H
+#define __Dark_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 = 'dark';    //Change this to what the AU identity is!
+
+class Dark : 
+    public AudioEffectX 
+{
+public:
+    Dark(audioMasterCallback audioMaster);
+    ~Dark();
+    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;
+    
+	float lastSampleL[100];
+	float lastSampleR[100];
+	uint32_t fpd;
+	//default stuff
+
+    float A;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Dark/DarkProc.cpp b/plugins/LinuxVST/src/Dark/DarkProc.cpp
new file mode 100755
index 0000000..672d8e0
--- /dev/null
+++ b/plugins/LinuxVST/src/Dark/DarkProc.cpp
@@ -0,0 +1,237 @@
+/* ========================================
+ *  Dark - Dark.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Dark_H
+#include "Dark.h"
+#endif
+
+void Dark::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];	
+	int processing = (VstInt32)( A * 1.999 );
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	int depth = (int)(17.0*overallscale);
+	if (depth < 3) depth = 3;
+	if (depth > 98) depth = 98;
+	bool highres = false;
+	if (processing == 1) highres = true;
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		
+		if (highres) {
+			inputSampleL *= 8388608.0;
+			inputSampleR *= 8388608.0;
+		} else {
+			inputSampleL *= 32768.0;
+			inputSampleR *= 32768.0;
+		}
+		//0-1 is now one bit, now we dither
+		//We are doing it first Left, then Right, because the loops may run faster if
+		//they aren't too jammed full of variables. This means re-running code.
+		
+		//begin left
+		int quantA = floor(inputSampleL);
+		int quantB = floor(inputSampleL+1.0);
+		//to do this style of dither, we quantize in either direction and then
+		//do a reconstruction of what the result will be for each choice.
+		//We then evaluate which one we like, and keep a history of what we previously had
+		
+		float expectedSlew = 0;
+		for(int x = 0; x < depth; x++) {
+			expectedSlew += (lastSampleL[x+1] - lastSampleL[x]);
+		}
+		expectedSlew /= depth; //we have an average of all recent slews
+		//we are doing that to voice the thing down into the upper mids a bit
+		//it mustn't just soften the brightest treble, it must smooth high mids too
+		
+		float testA = fabs((lastSampleL[0] - quantA) - expectedSlew);
+		float testB = fabs((lastSampleL[0] - quantB) - expectedSlew);
+		
+		if (testA < testB) inputSampleL = quantA;
+		else inputSampleL = quantB;
+		//select whichever one departs LEAST from the vector of averaged
+		//reconstructed previous final samples. This will force a kind of dithering
+		//as it'll make the output end up as smooth as possible
+		
+		for(int x = depth; x >=0; x--) {
+			lastSampleL[x+1] = lastSampleL[x];
+		}
+		lastSampleL[0] = inputSampleL;
+		//end left
+		
+		//begin right
+		quantA = floor(inputSampleR);
+		quantB = floor(inputSampleR+1.0);
+		//to do this style of dither, we quantize in either direction and then
+		//do a reconstruction of what the result will be for each choice.
+		//We then evaluate which one we like, and keep a history of what we previously had
+		
+		expectedSlew = 0;
+		for(int x = 0; x < depth; x++) {
+			expectedSlew += (lastSampleR[x+1] - lastSampleR[x]);
+		}
+		expectedSlew /= depth; //we have an average of all recent slews
+		//we are doing that to voice the thing down into the upper mids a bit
+		//it mustn't just soften the brightest treble, it must smooth high mids too
+		
+		testA = fabs((lastSampleR[0] - quantA) - expectedSlew);
+		testB = fabs((lastSampleR[0] - quantB) - expectedSlew);
+		
+		if (testA < testB) inputSampleR = quantA;
+		else inputSampleR = quantB;
+		//select whichever one departs LEAST from the vector of averaged
+		//reconstructed previous final samples. This will force a kind of dithering
+		//as it'll make the output end up as smooth as possible
+		
+		for(int x = depth; x >=0; x--) {
+			lastSampleR[x+1] = lastSampleR[x];
+		}
+		lastSampleR[0] = inputSampleR;
+		//end right
+		
+		if (highres) {
+			inputSampleL /= 8388608.0;
+			inputSampleR /= 8388608.0;
+		} else {
+			inputSampleL /= 32768.0;
+			inputSampleR /= 32768.0;
+		}
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void Dark::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+	
+	int processing = (VstInt32)( A * 1.999 );
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	int depth = (int)(17.0*overallscale);
+	if (depth < 3) depth = 3;
+	if (depth > 98) depth = 98;
+	bool highres = false;
+	if (processing == 1) highres = true;
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		
+		if (highres) {
+			inputSampleL *= 8388608.0;
+			inputSampleR *= 8388608.0;
+		} else {
+			inputSampleL *= 32768.0;
+			inputSampleR *= 32768.0;
+		}
+		//0-1 is now one bit, now we dither
+		//We are doing it first Left, then Right, because the loops may run faster if
+		//they aren't too jammed full of variables. This means re-running code.
+		
+		//begin left
+		int quantA = floor(inputSampleL);
+		int quantB = floor(inputSampleL+1.0);
+		//to do this style of dither, we quantize in either direction and then
+		//do a reconstruction of what the result will be for each choice.
+		//We then evaluate which one we like, and keep a history of what we previously had
+		
+		float expectedSlew = 0;
+		for(int x = 0; x < depth; x++) {
+			expectedSlew += (lastSampleL[x+1] - lastSampleL[x]);
+		}
+		expectedSlew /= depth; //we have an average of all recent slews
+		//we are doing that to voice the thing down into the upper mids a bit
+		//it mustn't just soften the brightest treble, it must smooth high mids too
+		
+		float testA = fabs((lastSampleL[0] - quantA) - expectedSlew);
+		float testB = fabs((lastSampleL[0] - quantB) - expectedSlew);
+		
+		if (testA < testB) inputSampleL = quantA;
+		else inputSampleL = quantB;
+		//select whichever one departs LEAST from the vector of averaged
+		//reconstructed previous final samples. This will force a kind of dithering
+		//as it'll make the output end up as smooth as possible
+		
+		for(int x = depth; x >=0; x--) {
+			lastSampleL[x+1] = lastSampleL[x];
+		}
+		lastSampleL[0] = inputSampleL;
+		//end left
+		
+		//begin right
+		quantA = floor(inputSampleR);
+		quantB = floor(inputSampleR+1.0);
+		//to do this style of dither, we quantize in either direction and then
+		//do a reconstruction of what the result will be for each choice.
+		//We then evaluate which one we like, and keep a history of what we previously had
+		
+		expectedSlew = 0;
+		for(int x = 0; x < depth; x++) {
+			expectedSlew += (lastSampleR[x+1] - lastSampleR[x]);
+		}
+		expectedSlew /= depth; //we have an average of all recent slews
+		//we are doing that to voice the thing down into the upper mids a bit
+		//it mustn't just soften the brightest treble, it must smooth high mids too
+		
+		testA = fabs((lastSampleR[0] - quantA) - expectedSlew);
+		testB = fabs((lastSampleR[0] - quantB) - expectedSlew);
+		
+		if (testA < testB) inputSampleR = quantA;
+		else inputSampleR = quantB;
+		//select whichever one departs LEAST from the vector of averaged
+		//reconstructed previous final samples. This will force a kind of dithering
+		//as it'll make the output end up as smooth as possible
+		
+		for(int x = depth; x >=0; x--) {
+			lastSampleR[x+1] = lastSampleR[x];
+		}
+		lastSampleR[0] = inputSampleR;
+		//end right
+		
+		if (highres) {
+			inputSampleL /= 8388608.0;
+			inputSampleR /= 8388608.0;
+		} else {
+			inputSampleL /= 32768.0;
+			inputSampleR /= 32768.0;
+		}
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}