Updates (in case my plane crashes)

3 files changed, 473 insertions, 0 deletions

diff --git a/plugins/MacVST/ToneSlant/source/ToneSlant.cpp b/plugins/MacVST/ToneSlant/source/ToneSlant.cpp
new file mode 100755
index 0000000..0ed3618
--- /dev/null
+++ b/plugins/MacVST/ToneSlant/source/ToneSlant.cpp
@@ -0,0 +1,131 @@
+/* ========================================
+ *  ToneSlant - ToneSlant.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __ToneSlant_H
+#include "ToneSlant.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new ToneSlant(audioMaster);}
+
+ToneSlant::ToneSlant(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 0.0;
+	for(int count = 0; count < 102; count++) {bL[count] = 0.0; bR[count] = 0.0; f[count] = 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
+}
+
+ToneSlant::~ToneSlant() {}
+VstInt32 ToneSlant::getVendorVersion () {return 1000;}
+void ToneSlant::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void ToneSlant::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 ToneSlant::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 ToneSlant::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 ToneSlant::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 ToneSlant::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 ToneSlant::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Voicing", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Highs", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void ToneSlant::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((A*99.0)+1.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string ((B*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 ToneSlant::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "taps", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break; //the percent
+        default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 ToneSlant::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool ToneSlant::getEffectName(char* name) {
+    vst_strncpy(name, "ToneSlant", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory ToneSlant::getPlugCategory() {return kPlugCategEffect;}
+
+bool ToneSlant::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows ToneSlant", kVstMaxProductStrLen); return true;
+}
+
+bool ToneSlant::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/ToneSlant/source/ToneSlant.h b/plugins/MacVST/ToneSlant/source/ToneSlant.h
new file mode 100755
index 0000000..655cc55
--- /dev/null
+++ b/plugins/MacVST/ToneSlant/source/ToneSlant.h
@@ -0,0 +1,71 @@
+/* ========================================
+ *  ToneSlant - ToneSlant.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __ToneSlant_H
+#define __ToneSlant_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 = 'tosl';    //Change this to what the AU identity is!
+
+class ToneSlant : 
+    public AudioEffectX 
+{
+public:
+    ToneSlant(audioMasterCallback audioMaster);
+    ~ToneSlant();
+    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
+
+    float A;
+    float B;
+	double bL[102];
+	double bR[102];
+	double f[102];		
+
+};
+
+#endif
diff --git a/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp b/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp
new file mode 100755
index 0000000..b594b05
--- /dev/null
+++ b/plugins/MacVST/ToneSlant/source/ToneSlantProc.cpp
@@ -0,0 +1,271 @@
+/* ========================================
+ *  ToneSlant - ToneSlant.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __ToneSlant_H
+#include "ToneSlant.h"
+#endif
+
+void ToneSlant::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double inputSampleL;
+	double inputSampleR;
+	double correctionSampleL;
+	double correctionSampleR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double drySampleL;
+	double drySampleR;
+	double overallscale = (A*99.0)+1.0;
+	double applySlant = (B*2.0)-1.0;
+	
+	
+	f[0] = 1.0 / overallscale;
+	//count to f(gain) which will be 0. f(0) is x1
+	for (int count = 1; count < 102; count++) {
+		if (count <= overallscale) {
+			f[count] = (1.0 - (count / overallscale)) / overallscale;
+			//recalc the filter and don't change the buffer it'll apply to
+		} else {
+			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+		}
+	}
+    
+    while (--sampleFrames >= 0)
+    {
+		for (int count = overallscale; count >= 0; count--) {
+			bL[count+1] = bL[count];
+			bR[count+1] = bR[count];
+		}
+		
+		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.
+		}
+		
+		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
+		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleR *= f[0];
+
+		for (int count = 1; count < overallscale; count++) {
+			accumulatorSampleL += (bL[count] * f[count]);
+			accumulatorSampleR += (bR[count] * f[count]);
+		}
+		
+		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
+		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
+		//we're gonna apply the total effect of all these calculations as a single subtract
+		
+		inputSampleL += (correctionSampleL * applySlant);
+		inputSampleR += (correctionSampleR * applySlant);
+		//our one math operation on the input data coming in
+
+		//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 ToneSlant::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	double fpTemp; //this is different from singlereplacing
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double inputSampleL;
+	double inputSampleR;
+	double correctionSampleL;
+	double correctionSampleR;
+	double accumulatorSampleL;
+	double accumulatorSampleR;
+	double drySampleL;
+	double drySampleR;
+	double overallscale = (A*99.0)+1.0;
+	double applySlant = (B*2.0)-1.0;
+	
+	f[0] = 1.0 / overallscale;
+	//count to f(gain) which will be 0. f(0) is x1
+	for (int count = 1; count < 102; count++) {
+		if (count <= overallscale) {
+			f[count] = (1.0 - (count / overallscale)) / overallscale;
+			//recalc the filter and don't change the buffer it'll apply to
+		} else {
+			bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+			bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
+		}
+	}
+	
+    while (--sampleFrames >= 0)
+    {
+		for (int count = overallscale; count >= 0; count--) {
+			bL[count+1] = bL[count];
+			bR[count+1] = bR[count];
+		}
+		
+		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.
+		}
+		
+		bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
+		bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
+		
+		accumulatorSampleL *= f[0];
+		accumulatorSampleR *= f[0];
+		
+		for (int count = 1; count < overallscale; count++) {
+			accumulatorSampleL += (bL[count] * f[count]);
+			accumulatorSampleR += (bR[count] * f[count]);
+		}
+		
+		correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
+		correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
+		//we're gonna apply the total effect of all these calculations as a single subtract
+		
+		inputSampleL += (correctionSampleL * applySlant);
+		inputSampleR += (correctionSampleR * applySlant);
+		//our one math operation on the input data coming in
+		
+		//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