Updates (in case my plane crashes)

3 files changed, 536 insertions, 0 deletions

diff --git a/plugins/MacVST/Pyewacket/source/Pyewacket.cpp b/plugins/MacVST/Pyewacket/source/Pyewacket.cpp
new file mode 100755
index 0000000..ee2db40
--- /dev/null
+++ b/plugins/MacVST/Pyewacket/source/Pyewacket.cpp
@@ -0,0 +1,141 @@
+/* ========================================
+ *  Pyewacket - Pyewacket.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Pyewacket_H
+#include "Pyewacket.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Pyewacket(audioMaster);}
+
+Pyewacket::Pyewacket(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.5;
+	C = 0.5;
+	chase = 1.0;
+	lastrectifierL = 0.0;
+	lastrectifierR = 0.0;
+	fpNShapeLA = 0.0;
+	fpNShapeLB = 0.0;
+	fpNShapeRA = 0.0;
+	fpNShapeRB = 0.0;
+	fpFlip = true;
+	//this is reset: values being initialized only once. Startup values, whatever they are.
+	
+    _canDo.insert("plugAsChannelInsert"); // plug-in can be used as a channel insert effect.
+    _canDo.insert("plugAsSend"); // plug-in can be used as a send effect.
+    _canDo.insert("x2in2out"); 
+    setNumInputs(kNumInputs);
+    setNumOutputs(kNumOutputs);
+    setUniqueID(kUniqueId);
+    canProcessReplacing();     // supports output replacing
+    canDoubleReplacing();      // supports double precision processing
+	programsAreChunks(true);
+    vst_strncpy (_programName, "Default", kVstMaxProgNameLen); // default program name
+}
+
+Pyewacket::~Pyewacket() {}
+VstInt32 Pyewacket::getVendorVersion () {return 1000;}
+void Pyewacket::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Pyewacket::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 Pyewacket::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 Pyewacket::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 Pyewacket::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 Pyewacket::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 Pyewacket::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Input Gain", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Release", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Output Gain", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void Pyewacket::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((A*24.0)-12.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string ((C*24.0)-12.0, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Pyewacket::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break; //the percent
+        case kParamC: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Pyewacket::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Pyewacket::getEffectName(char* name) {
+    vst_strncpy(name, "Pyewacket", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Pyewacket::getPlugCategory() {return kPlugCategEffect;}
+
+bool Pyewacket::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Pyewacket", kVstMaxProductStrLen); return true;
+}
+
+bool Pyewacket::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/Pyewacket/source/Pyewacket.h b/plugins/MacVST/Pyewacket/source/Pyewacket.h
new file mode 100755
index 0000000..593001f
--- /dev/null
+++ b/plugins/MacVST/Pyewacket/source/Pyewacket.h
@@ -0,0 +1,73 @@
+/* ========================================
+ *  Pyewacket - Pyewacket.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Pyewacket_H
+#define __Pyewacket_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 = 'pyew';    //Change this to what the AU identity is!
+
+class Pyewacket : 
+    public AudioEffectX 
+{
+public:
+    Pyewacket(audioMasterCallback audioMaster);
+    ~Pyewacket();
+    virtual bool getEffectName(char* name);                       // The plug-in name
+    virtual VstPlugCategory getPlugCategory();                    // The general category for the plug-in
+    virtual bool getProductString(char* text);                    // This is a unique plug-in string provided by Steinberg
+    virtual bool getVendorString(char* text);                     // Vendor info
+    virtual VstInt32 getVendorVersion();                          // Version number
+    virtual void processReplacing (float** inputs, float** outputs, VstInt32 sampleFrames);
+    virtual void processDoubleReplacing (double** inputs, double** outputs, VstInt32 sampleFrames);
+    virtual void getProgramName(char *name);                      // read the name from the host
+    virtual void setProgramName(char *name);                      // changes the name of the preset displayed in the host
+	virtual VstInt32 getChunk (void** data, bool isPreset);
+	virtual VstInt32 setChunk (void* data, VstInt32 byteSize, bool isPreset);
+    virtual float getParameter(VstInt32 index);                   // get the parameter value at the specified index
+    virtual void setParameter(VstInt32 index, float value);       // set the parameter at index to value
+    virtual void getParameterLabel(VstInt32 index, char *text);  // label for the parameter (eg dB)
+    virtual void getParameterName(VstInt32 index, char *text);    // name of the parameter
+    virtual void getParameterDisplay(VstInt32 index, char *text); // text description of the current value    
+    virtual VstInt32 canDo(char *text);
+private:
+    char _programName[kVstMaxProgNameLen + 1];
+    std::set< std::string > _canDo;
+    
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool fpFlip;
+	//default stuff
+	double chase;
+	double lastrectifierL;
+	double lastrectifierR;
+	
+    float A;
+    float B;
+    float C;
+
+};
+
+#endif
diff --git a/plugins/MacVST/Pyewacket/source/PyewacketProc.cpp b/plugins/MacVST/Pyewacket/source/PyewacketProc.cpp
new file mode 100755
index 0000000..83112a2
--- /dev/null
+++ b/plugins/MacVST/Pyewacket/source/PyewacketProc.cpp
@@ -0,0 +1,322 @@
+/* ========================================
+ *  Pyewacket - Pyewacket.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Pyewacket_H
+#include "Pyewacket.h"
+#endif
+
+void Pyewacket::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 drySampleL;
+	long double drySampleR;
+	double bridgerectifier;
+	double temprectifier;
+	double inputSense;
+	
+	double inputGain = pow(10.0,((A*24.0)-12.0)/20.0);
+	double attack = ((B+0.5)*0.006)/overallscale;
+	double decay = ((B+0.01)*0.0004)/overallscale;
+	double outputGain = pow(10.0,((C*24.0)-12.0)/20.0);
+	double wet;
+	double maxblur;
+	double blurdry;
+	double out;
+	double dry;
+	
+    while (--sampleFrames >= 0)
+    {
+		inputSampleL = *in1;
+		inputSampleR = *in2;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			static int noisesource = 0;
+			//this declares a variable before anything else is compiled. It won't keep assigning
+			//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
+			//but it lets me add this denormalization fix in a single place rather than updating
+			//it in three different locations. The variable isn't thread-safe but this is only
+			//a random seed and we can share it with whatever.
+			noisesource = noisesource % 1700021; noisesource++;
+			int residue = noisesource * noisesource;
+			residue = residue % 170003; residue *= residue;
+			residue = residue % 17011; residue *= residue;
+			residue = residue % 1709; residue *= residue;
+			residue = residue % 173; residue *= residue;
+			residue = residue % 17;
+			double applyresidue = residue;
+			applyresidue *= 0.00000001;
+			applyresidue *= 0.00000001;
+			inputSampleL = applyresidue;
+		}
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			static int noisesource = 0;
+			noisesource = noisesource % 1700021; noisesource++;
+			int residue = noisesource * noisesource;
+			residue = residue % 170003; residue *= residue;
+			residue = residue % 17011; residue *= residue;
+			residue = residue % 1709; residue *= residue;
+			residue = residue % 173; residue *= residue;
+			residue = residue % 17;
+			double applyresidue = residue;
+			applyresidue *= 0.00000001;
+			applyresidue *= 0.00000001;
+			inputSampleR = applyresidue;
+			//this denormalization routine produces a white noise at -300 dB which the noise
+			//shaping will interact with to produce a bipolar output, but the noise is actually
+			//all positive. That should stop any variables from going denormal, and the routine
+			//only kicks in if digital black is input. As a final touch, if you save to 24-bit
+			//the silence will return to being digital black again.
+		}
+		
+		if (inputGain != 1.0) {
+			inputSampleL *= inputGain;
+			inputSampleR *= inputGain;
+		}
+		drySampleL = inputSampleL;
+		drySampleR = inputSampleR;
+		inputSense = fabs(inputSampleL);
+		if (fabs(inputSampleR) > inputSense)
+			inputSense = fabs(inputSampleR);
+		//we will take the greater of either channel and just use that, then apply the result
+		//to both stereo channels.
+		if (chase < inputSense) chase += attack;
+		if (chase > 1.0) chase = 1.0;
+		if (chase > inputSense) chase -= decay;
+		if (chase < 0.0) chase = 0.0;
+		//chase will be between 0 and ? (if input is super hot)
+		out = wet = chase;
+		if (wet > 1.0) wet = 1.0;
+		maxblur = wet * fpNew;
+		blurdry = 1.0 - maxblur;
+		//scaled back so that blur remains balance of both
+		if (out > fpOld) out = fpOld - (out - fpOld);
+		if (out < 0.0) out = 0.0;
+		dry = 1.0 - wet;
+		
+		if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
+		if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
+		if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
+		if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
+		
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		temprectifier = 1-cos(bridgerectifier);
+		bridgerectifier = ((lastrectifierL*maxblur) + (temprectifier*blurdry));
+		lastrectifierL = temprectifier;
+		//starved version is also blurred by one sample
+		if (inputSampleL > 0) inputSampleL = (inputSampleL*dry)+(bridgerectifier*out);
+		else inputSampleL = (inputSampleL*dry)-(bridgerectifier*out);
+		
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		temprectifier = 1-cos(bridgerectifier);
+		bridgerectifier = ((lastrectifierR*maxblur) + (temprectifier*blurdry));
+		lastrectifierR = temprectifier;
+		//starved version is also blurred by one sample
+		if (inputSampleR > 0) inputSampleR = (inputSampleR*dry)+(bridgerectifier*out);
+		else inputSampleR = (inputSampleR*dry)-(bridgerectifier*out);
+		
+		if (outputGain != 1.0) {
+			inputSampleL *= outputGain;
+			inputSampleR *= outputGain;
+		}
+		
+		//noise shaping to 32-bit floating point
+		if (fpFlip) {
+			fpTemp = inputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLA;
+			fpTemp = inputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = inputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLB;
+			fpTemp = inputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 32 bit output
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void Pyewacket::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; //this is different from singlereplacing
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	long double inputSampleL;
+	long double inputSampleR;
+	long double drySampleL;
+	long double drySampleR;
+	double bridgerectifier;
+	double temprectifier;
+	double inputSense;
+	
+	double inputGain = pow(10.0,((A*24.0)-12.0)/20.0);
+	double attack = ((B+0.5)*0.006)/overallscale;
+	double decay = ((B+0.01)*0.0004)/overallscale;
+	double outputGain = pow(10.0,((C*24.0)-12.0)/20.0);
+	double wet;
+	double maxblur;
+	double blurdry;
+	double out;
+	double dry;	
+
+    while (--sampleFrames >= 0)
+    {
+		inputSampleL = *in1;
+		inputSampleR = *in2;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			static int noisesource = 0;
+			//this declares a variable before anything else is compiled. It won't keep assigning
+			//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
+			//but it lets me add this denormalization fix in a single place rather than updating
+			//it in three different locations. The variable isn't thread-safe but this is only
+			//a random seed and we can share it with whatever.
+			noisesource = noisesource % 1700021; noisesource++;
+			int residue = noisesource * noisesource;
+			residue = residue % 170003; residue *= residue;
+			residue = residue % 17011; residue *= residue;
+			residue = residue % 1709; residue *= residue;
+			residue = residue % 173; residue *= residue;
+			residue = residue % 17;
+			double applyresidue = residue;
+			applyresidue *= 0.00000001;
+			applyresidue *= 0.00000001;
+			inputSampleL = applyresidue;
+		}
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			static int noisesource = 0;
+			noisesource = noisesource % 1700021; noisesource++;
+			int residue = noisesource * noisesource;
+			residue = residue % 170003; residue *= residue;
+			residue = residue % 17011; residue *= residue;
+			residue = residue % 1709; residue *= residue;
+			residue = residue % 173; residue *= residue;
+			residue = residue % 17;
+			double applyresidue = residue;
+			applyresidue *= 0.00000001;
+			applyresidue *= 0.00000001;
+			inputSampleR = applyresidue;
+			//this denormalization routine produces a white noise at -300 dB which the noise
+			//shaping will interact with to produce a bipolar output, but the noise is actually
+			//all positive. That should stop any variables from going denormal, and the routine
+			//only kicks in if digital black is input. As a final touch, if you save to 24-bit
+			//the silence will return to being digital black again.
+		}
+		
+		if (inputGain != 1.0) {
+			inputSampleL *= inputGain;
+			inputSampleR *= inputGain;
+		}
+		drySampleL = inputSampleL;
+		drySampleR = inputSampleR;
+		inputSense = fabs(inputSampleL);
+		if (fabs(inputSampleR) > inputSense)
+			inputSense = fabs(inputSampleR);
+		//we will take the greater of either channel and just use that, then apply the result
+		//to both stereo channels.
+		if (chase < inputSense) chase += attack;
+		if (chase > 1.0) chase = 1.0;
+		if (chase > inputSense) chase -= decay;
+		if (chase < 0.0) chase = 0.0;
+		//chase will be between 0 and ? (if input is super hot)
+		out = wet = chase;
+		if (wet > 1.0) wet = 1.0;
+		maxblur = wet * fpNew;
+		blurdry = 1.0 - maxblur;
+		//scaled back so that blur remains balance of both
+		if (out > fpOld) out = fpOld - (out - fpOld);
+		if (out < 0.0) out = 0.0;
+		dry = 1.0 - wet;
+		
+		if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
+		if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
+		if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
+		if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
+		
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		temprectifier = 1-cos(bridgerectifier);
+		bridgerectifier = ((lastrectifierL*maxblur) + (temprectifier*blurdry));
+		lastrectifierL = temprectifier;
+		//starved version is also blurred by one sample
+		if (inputSampleL > 0) inputSampleL = (inputSampleL*dry)+(bridgerectifier*out);
+		else inputSampleL = (inputSampleL*dry)-(bridgerectifier*out);
+		
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		temprectifier = 1-cos(bridgerectifier);
+		bridgerectifier = ((lastrectifierR*maxblur) + (temprectifier*blurdry));
+		lastrectifierR = temprectifier;
+		//starved version is also blurred by one sample
+		if (inputSampleR > 0) inputSampleR = (inputSampleR*dry)+(bridgerectifier*out);
+		else inputSampleR = (inputSampleR*dry)-(bridgerectifier*out);
+		
+		if (outputGain != 1.0) {
+			inputSampleL *= outputGain;
+			inputSampleR *= outputGain;
+		}
+		
+		//noise shaping to 64-bit floating point
+		if (fpFlip) {
+			fpTemp = inputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLA;
+			fpTemp = inputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = inputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
+			inputSampleL += fpNShapeLB;
+			fpTemp = inputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
+			inputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 64 bit output
+
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
\ No newline at end of file