StereoFX

3 files changed, 541 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/StereoFX/StereoFX.cpp b/plugins/LinuxVST/src/StereoFX/StereoFX.cpp
new file mode 100755
index 0000000..63c23db
--- /dev/null
+++ b/plugins/LinuxVST/src/StereoFX/StereoFX.cpp
@@ -0,0 +1,140 @@
+/* ========================================
+ *  StereoFX - StereoFX.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __StereoFX_H
+#include "StereoFX.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new StereoFX(audioMaster);}
+
+StereoFX::StereoFX(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 0.0;
+	C = 0.0;
+	iirSampleA = 0.0;
+	iirSampleB = 0.0;
+	fpNShapeLA = 0.0;
+	fpNShapeLB = 0.0;
+	fpNShapeRA = 0.0;
+	fpNShapeRB = 0.0;
+	flip = false;
+	//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
+}
+
+StereoFX::~StereoFX() {}
+VstInt32 StereoFX::getVendorVersion () {return 1000;}
+void StereoFX::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Wide", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "MonoBs", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "CSquish", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void StereoFX::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string (C, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void StereoFX::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 StereoFX::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool StereoFX::getEffectName(char* name) {
+    vst_strncpy(name, "StereoFX", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory StereoFX::getPlugCategory() {return kPlugCategEffect;}
+
+bool StereoFX::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows StereoFX", kVstMaxProductStrLen); return true;
+}
+
+bool StereoFX::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/StereoFX/StereoFX.h b/plugins/LinuxVST/src/StereoFX/StereoFX.h
new file mode 100755
index 0000000..6748d1e
--- /dev/null
+++ b/plugins/LinuxVST/src/StereoFX/StereoFX.h
@@ -0,0 +1,71 @@
+/* ========================================
+ *  StereoFX - StereoFX.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __StereoFX_H
+#define __StereoFX_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 = 'stfx';    //Change this to what the AU identity is!
+
+class StereoFX : 
+    public AudioEffectX 
+{
+public:
+    StereoFX(audioMasterCallback audioMaster);
+    ~StereoFX();
+    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 iirSampleA;
+	double iirSampleB;
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool flip;
+	//default stuff
+
+    float A;
+    float B;
+    float C;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp b/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp
new file mode 100755
index 0000000..49be44f
--- /dev/null
+++ b/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp
@@ -0,0 +1,330 @@
+/* ========================================
+ *  StereoFX - StereoFX.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __StereoFX_H
+#include "StereoFX.h"
+#endif
+
+void StereoFX::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	long double inputSampleL;
+	long double inputSampleR;
+	long double mid;
+	long double side;
+	//High Impact section
+	double stereowide = A;
+	double centersquish = C;
+	double density = stereowide * 2.4;
+	double sustain = 1.0 - (1.0/(1.0 + (density/7.0)));
+	//this way, enhance increases up to 50% and then mid falls off beyond that
+	double bridgerectifier;
+	double count;
+	//Highpass section
+	double iirAmount = pow(B,3)/overallscale;
+	double tight = -0.33333333333333;
+	double offset;
+	//we are setting it up so that to either extreme we can get an audible sound,
+	//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
+	
+    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.
+		}
+		//assign working variables		
+		mid = inputSampleL + inputSampleR;
+		side = inputSampleL - inputSampleR;
+		//assign mid and side. Now, High Impact code
+		count = density;
+		while (count > 1.0)
+		{
+			bridgerectifier = fabs(side)*1.57079633;
+			if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+			//max value for sine function
+			bridgerectifier = sin(bridgerectifier);
+			if (side > 0.0) side = bridgerectifier;
+			else side = -bridgerectifier;
+			count = count - 1.0;
+		}
+		//we have now accounted for any really high density settings.
+		bridgerectifier = fabs(side)*1.57079633;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (side > 0) side = (side*(1-count))+(bridgerectifier*count);
+		else side = (side*(1-count))-(bridgerectifier*count);
+		//blend according to density control
+		//done first density. Next, sustain-reducer
+		bridgerectifier = fabs(side)*1.57079633;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = (1-cos(bridgerectifier))*3.141592653589793;
+		if (side > 0) side = (side*(1-sustain))+(bridgerectifier*sustain);
+		else side = (side*(1-sustain))-(bridgerectifier*sustain);
+		//done with High Impact code
+		
+		//now, Highpass code
+		offset = 0.666666666666666 + ((1-fabs(side))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (flip)
+		{
+			iirSampleA = (iirSampleA * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
+			side = side - iirSampleA;
+		}
+		else
+		{
+			iirSampleB = (iirSampleB * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
+			side = side - iirSampleB;
+		}
+		//done with Highpass code
+		
+		bridgerectifier = fabs(mid)/1.273239544735162;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier)*1.273239544735162;
+		if (mid > 0) mid = (mid*(1-centersquish))+(bridgerectifier*centersquish);
+		else mid = (mid*(1-centersquish))-(bridgerectifier*centersquish);
+		//done with the mid saturating section.
+		
+		inputSampleL = (mid+side)/2.0;
+		inputSampleR = (mid-side)/2.0;
+		
+		//noise shaping to 32-bit floating point
+		if (flip) {
+			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;
+		}
+		flip = !flip;
+		//end noise shaping on 32 bit output
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void StereoFX::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	double fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	long double inputSampleL;
+	long double inputSampleR;
+	long double mid;
+	long double side;
+	//High Impact section
+	double stereowide = A;
+	double centersquish = C;
+	double density = stereowide * 2.4;
+	double sustain = 1.0 - (1.0/(1.0 + (density/7.0)));
+	//this way, enhance increases up to 50% and then mid falls off beyond that
+	double bridgerectifier;
+	double count;
+	//Highpass section
+	double iirAmount = pow(B,3)/overallscale;
+	double tight = -0.33333333333333;
+	double offset;
+	//we are setting it up so that to either extreme we can get an audible sound,
+	//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
+	
+    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.
+		}
+		//assign working variables		
+		mid = inputSampleL + inputSampleR;
+		side = inputSampleL - inputSampleR;
+		//assign mid and side. Now, High Impact code
+		count = density;
+		while (count > 1.0)
+		{
+			bridgerectifier = fabs(side)*1.57079633;
+			if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+			//max value for sine function
+			bridgerectifier = sin(bridgerectifier);
+			if (side > 0.0) side = bridgerectifier;
+			else side = -bridgerectifier;
+			count = count - 1.0;
+		}
+		//we have now accounted for any really high density settings.
+		bridgerectifier = fabs(side)*1.57079633;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (side > 0) side = (side*(1-count))+(bridgerectifier*count);
+		else side = (side*(1-count))-(bridgerectifier*count);
+		//blend according to density control
+		//done first density. Next, sustain-reducer
+		bridgerectifier = fabs(side)*1.57079633;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = (1-cos(bridgerectifier))*3.141592653589793;
+		if (side > 0) side = (side*(1-sustain))+(bridgerectifier*sustain);
+		else side = (side*(1-sustain))-(bridgerectifier*sustain);
+		//done with High Impact code
+		
+		//now, Highpass code
+		offset = 0.666666666666666 + ((1-fabs(side))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (flip)
+		{
+			iirSampleA = (iirSampleA * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
+			side = side - iirSampleA;
+		}
+		else
+		{
+			iirSampleB = (iirSampleB * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
+			side = side - iirSampleB;
+		}
+		//done with Highpass code
+		
+		bridgerectifier = fabs(mid)/1.273239544735162;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier)*1.273239544735162;
+		if (mid > 0) mid = (mid*(1-centersquish))+(bridgerectifier*centersquish);
+		else mid = (mid*(1-centersquish))-(bridgerectifier*centersquish);
+		//done with the mid saturating section.
+		
+		inputSampleL = (mid+side)/2.0;
+		inputSampleR = (mid-side)/2.0;
+		
+		//noise shaping to 64-bit floating point
+		if (flip) {
+			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;
+		}
+		flip = !flip;
+		//end noise shaping on 64 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
\ No newline at end of file