Updates (in case my plane crashes)

3 files changed, 518 insertions, 0 deletions

diff --git a/plugins/MacVST/Lowpass/source/Lowpass.cpp b/plugins/MacVST/Lowpass/source/Lowpass.cpp
new file mode 100755
index 0000000..8178799
--- /dev/null
+++ b/plugins/MacVST/Lowpass/source/Lowpass.cpp
@@ -0,0 +1,142 @@
+/* ========================================
+ *  Lowpass - Lowpass.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Lowpass_H
+#include "Lowpass.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Lowpass(audioMaster);}
+
+Lowpass::Lowpass(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 1.0;
+	B = 0.5;
+	C = 1.0;
+	iirSampleAL = 0.0;
+	iirSampleBL = 0.0;
+	iirSampleAR = 0.0;
+	iirSampleBR = 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
+}
+
+Lowpass::~Lowpass() {}
+VstInt32 Lowpass::getVendorVersion () {return 1000;}
+void Lowpass::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Lowpass::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 Lowpass::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 Lowpass::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 Lowpass::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break; //percent. Using this value, it'll be 0-100 everywhere
+        case kParamC: C = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float Lowpass::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 Lowpass::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Lowpass", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Soft/Hard", kVstMaxParamStrLen); break;
+		case kParamC: 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 Lowpass::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string ((B * 2.0)-1.0, text, kVstMaxParamStrLen); break; //also display 0-1 as percent
+        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 Lowpass::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, " ", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break; //the percent
+        case kParamC: vst_strncpy (text, " ", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Lowpass::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Lowpass::getEffectName(char* name) {
+    vst_strncpy(name, "Lowpass", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Lowpass::getPlugCategory() {return kPlugCategEffect;}
+
+bool Lowpass::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Lowpass", kVstMaxProductStrLen); return true;
+}
+
+bool Lowpass::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/Lowpass/source/Lowpass.h b/plugins/MacVST/Lowpass/source/Lowpass.h
new file mode 100755
index 0000000..1b99c55
--- /dev/null
+++ b/plugins/MacVST/Lowpass/source/Lowpass.h
@@ -0,0 +1,74 @@
+/* ========================================
+ *  Lowpass - Lowpass.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Lowpass_H
+#define __Lowpass_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 = 'lops';    //Change this to what the AU identity is!
+
+class Lowpass : 
+    public AudioEffectX 
+{
+public:
+    Lowpass(audioMasterCallback audioMaster);
+    ~Lowpass();
+    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;
+    float C;
+	double iirSampleAL;
+	double iirSampleBL;
+	double iirSampleAR;
+	double iirSampleBR;
+
+};
+
+#endif
diff --git a/plugins/MacVST/Lowpass/source/LowpassProc.cpp b/plugins/MacVST/Lowpass/source/LowpassProc.cpp
new file mode 100755
index 0000000..bcce714
--- /dev/null
+++ b/plugins/MacVST/Lowpass/source/LowpassProc.cpp
@@ -0,0 +1,302 @@
+/* ========================================
+ *  Lowpass - Lowpass.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Lowpass_H
+#include "Lowpass.h"
+#endif
+
+void Lowpass::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 iirAmount = (pow(A,2)+A)/2.0;
+	iirAmount /= overallscale;
+	double tight = (B*2.0)-1.0;
+	double wet = C;
+	double dry = 1.0 - wet;
+	double offset;
+	double inputSampleL;
+	double inputSampleR;
+	double outputSampleL;
+	double outputSampleR;
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;
+	
+	iirAmount += (iirAmount * tight * tight);
+	if (tight > 0) tight /= 1.5;
+	else tight /= 3.0;
+	//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.
+	if (iirAmount <= 0.0) iirAmount = 0.0;
+	if (iirAmount > 1.0) iirAmount = 1.0;
+	//handle the change in cutoff frequency
+    
+    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.
+		}
+		
+		outputSampleL = inputSampleL;
+		outputSampleR = inputSampleR;
+		
+		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
+		else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (fpFlip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
+			outputSampleL = iirSampleAL;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
+			outputSampleL = iirSampleBL;
+		}
+
+		
+		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
+		else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (fpFlip)
+		{
+			iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
+			outputSampleR = iirSampleAR;
+		}
+		else
+		{
+			iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
+			outputSampleR = iirSampleBR;
+		}
+
+		
+		
+		if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
+		if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
+
+		//noise shaping to 32-bit floating point
+		if (fpFlip) {
+			fpTemp = outputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((outputSampleL-fpTemp)*fpNew);
+			outputSampleL += fpNShapeLA;
+			
+			fpTemp = outputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((outputSampleR-fpTemp)*fpNew);
+			outputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = outputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((outputSampleL-fpTemp)*fpNew);
+			outputSampleL += fpNShapeLB;
+			
+			fpTemp = outputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((outputSampleR-fpTemp)*fpNew);
+			outputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 32 bit output
+
+		*out1 = outputSampleL;
+		*out2 = outputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void Lowpass::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 iirAmount = (pow(A,2)+A)/2.0;
+	iirAmount /= overallscale;
+	double tight = (B*2.0)-1.0;
+	double wet = C;
+	double dry = 1.0 - wet;
+	double offset;
+	double inputSampleL;
+	double inputSampleR;
+	double outputSampleL;
+	double outputSampleR;
+	double fpTemp;
+	double fpOld = 0.618033988749894848204586; //golden ratio!
+	double fpNew = 1.0 - fpOld;
+	
+	iirAmount += (iirAmount * tight * tight);
+	if (tight > 0) tight /= 1.5;
+	else tight /= 3.0;
+	//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.
+	if (iirAmount <= 0.0) iirAmount = 0.0;
+	if (iirAmount > 1.0) iirAmount = 1.0;
+	//handle the change in cutoff frequency
+	
+    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.
+		}
+		
+		outputSampleL = inputSampleL;
+		outputSampleR = inputSampleR;
+		
+		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight);
+		else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (fpFlip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
+			outputSampleL = iirSampleAL;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount));
+			outputSampleL = iirSampleBL;
+		}
+		
+		
+		if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight);
+		else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight);
+		if (offset < 0) offset = 0;
+		if (offset > 1) offset = 1;
+		if (fpFlip)
+		{
+			iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
+			outputSampleR = iirSampleAR;
+		}
+		else
+		{
+			iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount));
+			outputSampleR = iirSampleBR;
+		}
+		
+		
+		
+		if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry);
+		if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry);
+		
+		//noise shaping to 64-bit floating point
+		if (fpFlip) {
+			fpTemp = outputSampleL;
+			fpNShapeLA = (fpNShapeLA*fpOld)+((outputSampleL-fpTemp)*fpNew);
+			outputSampleL += fpNShapeLA;
+			
+			fpTemp = outputSampleR;
+			fpNShapeRA = (fpNShapeRA*fpOld)+((outputSampleR-fpTemp)*fpNew);
+			outputSampleR += fpNShapeRA;
+		}
+		else {
+			fpTemp = outputSampleL;
+			fpNShapeLB = (fpNShapeLB*fpOld)+((outputSampleL-fpTemp)*fpNew);
+			outputSampleL += fpNShapeLB;
+			
+			fpTemp = outputSampleR;
+			fpNShapeRB = (fpNShapeRB*fpOld)+((outputSampleR-fpTemp)*fpNew);
+			outputSampleR += fpNShapeRB;
+		}
+		fpFlip = !fpFlip;
+		//end noise shaping on 32 bit output
+		
+		*out1 = outputSampleL;
+		*out2 = outputSampleR;
+		
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
\ No newline at end of file