Updates (in case my plane crashes)

3 files changed, 773 insertions, 0 deletions

diff --git a/plugins/MacVST/BassKit/source/BassKit.cpp b/plugins/MacVST/BassKit/source/BassKit.cpp
new file mode 100755
index 0000000..008f045
--- /dev/null
+++ b/plugins/MacVST/BassKit/source/BassKit.cpp
@@ -0,0 +1,195 @@
+/* ========================================
+ *  BassKit - BassKit.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __BassKit_H
+#include "BassKit.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new BassKit(audioMaster);}
+
+BassKit::BassKit(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.5;
+	C = 0.5;
+	D = 0.5;
+	
+	WasNegative = false;
+	SubOctave = false;
+	flip = false;
+	bflip = 0;
+	iirDriveSampleA = 0.0;
+	iirDriveSampleB = 0.0;
+	iirDriveSampleC = 0.0;
+	iirDriveSampleD = 0.0;
+	iirDriveSampleE = 0.0;
+	iirDriveSampleF = 0.0;
+	
+	iirHeadBumpA = 0.0;
+	iirHeadBumpB = 0.0;
+	iirHeadBumpC = 0.0;
+	
+	iirSubBumpA = 0.0;
+	iirSubBumpB = 0.0;
+	iirSubBumpC = 0.0;
+	
+	lastHeadBump = 0.0;
+	lastSubBump = 0.0;
+	
+	iirSampleA = 0.0;
+	iirSampleB = 0.0;
+	iirSampleC = 0.0;
+	iirSampleD = 0.0;
+	iirSampleE = 0.0;
+	iirSampleF = 0.0;
+	iirSampleG = 0.0;
+	iirSampleH = 0.0;
+	iirSampleI = 0.0;
+	iirSampleJ = 0.0;
+	iirSampleK = 0.0;
+	iirSampleL = 0.0;
+	iirSampleM = 0.0;
+	iirSampleN = 0.0;
+	iirSampleO = 0.0;
+	iirSampleP = 0.0;
+	iirSampleQ = 0.0;
+	iirSampleR = 0.0;
+	iirSampleS = 0.0;
+	iirSampleT = 0.0;
+	iirSampleU = 0.0;
+	iirSampleV = 0.0;
+	iirSampleW = 0.0;
+	iirSampleX = 0.0;
+	iirSampleY = 0.0;
+	iirSampleZ = 0.0;	
+	
+	oscGate = 1.0;
+		
+	fpNShapeL = 0.0;
+	fpNShapeR = 0.0;
+	//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
+}
+
+BassKit::~BassKit() {}
+VstInt32 BassKit::getVendorVersion () {return 1000;}
+void BassKit::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void BassKit::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 BassKit::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	chunkData[2] = C;
+	chunkData[3] = D;
+	/* 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 BassKit::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	C = pinParameter(chunkData[2]);
+	D = pinParameter(chunkData[3]);
+	/* 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 BassKit::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        case kParamD: D = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float BassKit::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        case kParamC: return C; break;
+        case kParamD: return D; 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 BassKit::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Drive", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Voicing", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "BassOut", kVstMaxParamStrLen); break;
+		case kParamD: vst_strncpy (text, "SubOut", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void BassKit::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*2.0)-1.0, text, kVstMaxParamStrLen); break;
+        case kParamD: float2string ((D*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 BassKit::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;
+        case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 BassKit::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool BassKit::getEffectName(char* name) {
+    vst_strncpy(name, "BassKit", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory BassKit::getPlugCategory() {return kPlugCategEffect;}
+
+bool BassKit::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows BassKit", kVstMaxProductStrLen); return true;
+}
+
+bool BassKit::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/BassKit/source/BassKit.h b/plugins/MacVST/BassKit/source/BassKit.h
new file mode 100755
index 0000000..f421d3d
--- /dev/null
+++ b/plugins/MacVST/BassKit/source/BassKit.h
@@ -0,0 +1,118 @@
+/* ========================================
+ *  BassKit - BassKit.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __BassKit_H
+#define __BassKit_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+	kParamC = 2,
+	kParamD = 3,
+  kNumParameters = 4
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'bskt';    //Change this to what the AU identity is!
+
+class BassKit : 
+    public AudioEffectX 
+{
+public:
+    BassKit(audioMasterCallback audioMaster);
+    ~BassKit();
+    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 iirDriveSampleA;
+	double iirDriveSampleB;
+	double iirDriveSampleC;
+	double iirDriveSampleD;
+	double iirDriveSampleE;
+	double iirDriveSampleF;
+	bool flip; //drive things
+	
+	int bflip;
+	bool WasNegative;
+	bool SubOctave;
+	double iirHeadBumpA;
+	double iirHeadBumpB;
+	double iirHeadBumpC;
+	
+	double iirSubBumpA;
+	double iirSubBumpB;
+	double iirSubBumpC;
+	
+	double lastHeadBump;
+	double lastSubBump;
+	
+	double iirSampleA;
+	double iirSampleB;
+	double iirSampleC;
+	double iirSampleD;
+	double iirSampleE;
+	double iirSampleF;
+	double iirSampleG;
+	double iirSampleH;
+	double iirSampleI;
+	double iirSampleJ;
+	double iirSampleK;
+	double iirSampleL;
+	double iirSampleM;
+	double iirSampleN;
+	double iirSampleO;
+	double iirSampleP;
+	double iirSampleQ;
+	double iirSampleR;
+	double iirSampleS;
+	double iirSampleT;
+	double iirSampleU;
+	double iirSampleV;
+	double iirSampleW;
+	double iirSampleX;
+	double iirSampleY;
+	double iirSampleZ;		
+	double oscGate;	
+	
+	long double fpNShapeL;
+	long double fpNShapeR;
+	//default stuff
+
+    float A;
+    float B;
+    float C;
+    float D;
+};
+
+#endif
diff --git a/plugins/MacVST/BassKit/source/BassKitProc.cpp b/plugins/MacVST/BassKit/source/BassKitProc.cpp
new file mode 100755
index 0000000..df3310e
--- /dev/null
+++ b/plugins/MacVST/BassKit/source/BassKitProc.cpp
@@ -0,0 +1,460 @@
+/* ========================================
+ *  BassKit - BassKit.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __BassKit_H
+#include "BassKit.h"
+#endif
+
+void BassKit::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 ataLowpass;
+	double randy;
+	double invrandy;
+	double HeadBump = 0.0;
+	double BassGain = A * 0.1;
+	double HeadBumpFreq = ((B*0.1)+0.02)/overallscale;
+	double iirAmount = HeadBumpFreq/44.1;
+	double BassOutGain = ((C*2.0)-1.0)*fabs(((C*2.0)-1.0));
+	double SubBump = 0.0;
+	double SubOutGain = ((D*2.0)-1.0)*fabs(((D*2.0)-1.0))*4.0;
+	double clamp = 0.0;
+	double fuzz = 0.111;
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it again. We want a 'air' hiss
+		
+		ataLowpass = (inputSampleL + inputSampleR) / 2.0;
+		iirDriveSampleA = (iirDriveSampleA * (1.0 - HeadBumpFreq)) + (ataLowpass * HeadBumpFreq); ataLowpass = iirDriveSampleA;
+		iirDriveSampleB = (iirDriveSampleB * (1.0 - HeadBumpFreq)) + (ataLowpass * HeadBumpFreq); ataLowpass = iirDriveSampleB;
+		
+		
+		oscGate += fabs(ataLowpass * 10.0);
+		oscGate -= 0.001;
+		if (oscGate > 1.0) oscGate = 1.0;
+		if (oscGate < 0) oscGate = 0;
+		//got a value that only goes down low when there's silence or near silence on input
+		clamp = 1.0-oscGate;
+		clamp *= 0.00001;
+		//set up the thing to choke off oscillations- belt and suspenders affair
+		
+		if (ataLowpass > 0)
+		{if (WasNegative){SubOctave = !SubOctave;} WasNegative = false;}
+		else {WasNegative = true;}
+		//set up polarities for sub-bass version
+		randy = (rand()/(double)RAND_MAX)*fuzz; //0 to 1 the noise, may not be needed
+		invrandy = (1.0-randy);
+		randy /= 2.0;
+		//set up the noise
+		
+		iirSampleA = (iirSampleA * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleA;
+		iirSampleB = (iirSampleB * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleB;
+		iirSampleC = (iirSampleC * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleC;
+		iirSampleD = (iirSampleD * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleD;
+		iirSampleE = (iirSampleE * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleE;
+		iirSampleF = (iirSampleF * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleF;
+		iirSampleG = (iirSampleG * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleG;
+		iirSampleH = (iirSampleH * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleH;
+		iirSampleI = (iirSampleI * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleI;
+		iirSampleJ = (iirSampleJ * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleJ;
+		iirSampleK = (iirSampleK * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleK;
+		iirSampleL = (iirSampleL * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleL;
+		iirSampleM = (iirSampleM * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleM;
+		iirSampleN = (iirSampleN * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleN;
+		iirSampleO = (iirSampleO * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleO;
+		iirSampleP = (iirSampleP * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleP;
+		iirSampleQ = (iirSampleQ * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleQ;
+		iirSampleR = (iirSampleR * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleR;
+		iirSampleS = (iirSampleS * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleS;
+		iirSampleT = (iirSampleT * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleT;
+		iirSampleU = (iirSampleU * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleU;
+		iirSampleV = (iirSampleV * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleV;
+		
+		switch (bflip)
+		{
+			case 1:				
+				iirHeadBumpA += (ataLowpass * BassGain);
+				iirHeadBumpA -= (iirHeadBumpA * iirHeadBumpA * iirHeadBumpA * HeadBumpFreq);
+				iirHeadBumpA = (invrandy * iirHeadBumpA) + (randy * iirHeadBumpB) + (randy * iirHeadBumpC);
+				if (iirHeadBumpA > 0) iirHeadBumpA -= clamp;
+				if (iirHeadBumpA < 0) iirHeadBumpA += clamp;
+				HeadBump = iirHeadBumpA;
+				break;
+			case 2:
+				iirHeadBumpB += (ataLowpass * BassGain);
+				iirHeadBumpB -= (iirHeadBumpB * iirHeadBumpB * iirHeadBumpB * HeadBumpFreq);
+				iirHeadBumpB = (randy * iirHeadBumpA) + (invrandy * iirHeadBumpB) + (randy * iirHeadBumpC);
+				if (iirHeadBumpB > 0) iirHeadBumpB -= clamp;
+				if (iirHeadBumpB < 0) iirHeadBumpB += clamp;
+				HeadBump = iirHeadBumpB;
+				break;
+			case 3:
+				iirHeadBumpC += (ataLowpass * BassGain);
+				iirHeadBumpC -= (iirHeadBumpC * iirHeadBumpC * iirHeadBumpC * HeadBumpFreq);
+				iirHeadBumpC = (randy * iirHeadBumpA) + (randy * iirHeadBumpB) + (invrandy * iirHeadBumpC);
+				if (iirHeadBumpC > 0) iirHeadBumpC -= clamp;
+				if (iirHeadBumpC < 0) iirHeadBumpC += clamp;
+				HeadBump = iirHeadBumpC;
+				break;
+		}
+		
+		iirSampleW = (iirSampleW * (1.0 - iirAmount)) + (HeadBump * iirAmount); HeadBump -= iirSampleW;
+		iirSampleX = (iirSampleX * (1.0 - iirAmount)) + (HeadBump * iirAmount); HeadBump -= iirSampleX;
+		
+		SubBump = HeadBump;
+		iirSampleY = (iirSampleY * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump -= iirSampleY;
+		
+		iirDriveSampleC = (iirDriveSampleC * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirDriveSampleC;
+		iirDriveSampleD = (iirDriveSampleD * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirDriveSampleD;
+		
+		
+		SubBump = fabs(SubBump);
+		if (SubOctave == false) {SubBump = -SubBump;}
+		
+		switch (bflip)
+		{
+			case 1:				
+				iirSubBumpA += SubBump;// * BassGain);
+				iirSubBumpA -= (iirSubBumpA * iirSubBumpA * iirSubBumpA * HeadBumpFreq);
+				iirSubBumpA = (invrandy * iirSubBumpA) + (randy * iirSubBumpB) + (randy * iirSubBumpC);
+				if (iirSubBumpA > 0) iirSubBumpA -= clamp;
+				if (iirSubBumpA < 0) iirSubBumpA += clamp;
+				SubBump = iirSubBumpA;
+				break;
+			case 2:
+				iirSubBumpB += SubBump;// * BassGain);
+				iirSubBumpB -= (iirSubBumpB * iirSubBumpB * iirSubBumpB * HeadBumpFreq);
+				iirSubBumpB = (randy * iirSubBumpA) + (invrandy * iirSubBumpB) + (randy * iirSubBumpC);
+				if (iirSubBumpB > 0) iirSubBumpB -= clamp;
+				if (iirSubBumpB < 0) iirSubBumpB += clamp;
+				SubBump = iirSubBumpB;
+				break;
+			case 3:
+				iirSubBumpC += SubBump;// * BassGain);
+				iirSubBumpC -= (iirSubBumpC * iirSubBumpC * iirSubBumpC * HeadBumpFreq);
+				iirSubBumpC = (randy * iirSubBumpA) + (randy * iirSubBumpB) + (invrandy * iirSubBumpC);
+				if (iirSubBumpC > 0) iirSubBumpC -= clamp;
+				if (iirSubBumpC < 0) iirSubBumpC += clamp;
+				SubBump = iirSubBumpC;
+				break;
+		}
+		
+		iirSampleZ = (iirSampleZ * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirSampleZ;
+		iirDriveSampleE = (iirDriveSampleE * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump = iirDriveSampleE;
+		iirDriveSampleF = (iirDriveSampleF * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump = iirDriveSampleF;
+		
+		
+		inputSampleL += (HeadBump * BassOutGain);
+		inputSampleL += (SubBump * SubOutGain);
+		
+		inputSampleR += (HeadBump * BassOutGain);
+		inputSampleR += (SubBump * SubOutGain);
+		
+		
+		flip = !flip;
+		bflip++;
+		if (bflip < 1 || bflip > 3) bflip = 1;
+		
+		//noise shaping to 32-bit floating point
+		float fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 32 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}
+
+void BassKit::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 ataLowpass;
+	double randy;
+	double invrandy;
+	double HeadBump = 0.0;
+	double BassGain = A * 0.1;
+	double HeadBumpFreq = ((B*0.1)+0.02)/overallscale;
+	double iirAmount = HeadBumpFreq/44.1;
+	double BassOutGain = ((C*2.0)-1.0)*fabs(((C*2.0)-1.0));
+	double SubBump = 0.0;
+	double SubOutGain = ((D*2.0)-1.0)*fabs(((D*2.0)-1.0))*4.0;
+	double clamp = 0.0;
+	double fuzz = 0.111;
+		
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it again. We want a 'air' hiss
+		
+		ataLowpass = (inputSampleL + inputSampleR) / 2.0;
+		iirDriveSampleA = (iirDriveSampleA * (1.0 - HeadBumpFreq)) + (ataLowpass * HeadBumpFreq); ataLowpass = iirDriveSampleA;
+		iirDriveSampleB = (iirDriveSampleB * (1.0 - HeadBumpFreq)) + (ataLowpass * HeadBumpFreq); ataLowpass = iirDriveSampleB;
+		
+		
+		oscGate += fabs(ataLowpass * 10.0);
+		oscGate -= 0.001;
+		if (oscGate > 1.0) oscGate = 1.0;
+		if (oscGate < 0) oscGate = 0;
+		//got a value that only goes down low when there's silence or near silence on input
+		clamp = 1.0-oscGate;
+		clamp *= 0.00001;
+		//set up the thing to choke off oscillations- belt and suspenders affair
+		
+		if (ataLowpass > 0)
+		{if (WasNegative){SubOctave = !SubOctave;} WasNegative = false;}
+		else {WasNegative = true;}
+		//set up polarities for sub-bass version
+		randy = (rand()/(double)RAND_MAX)*fuzz; //0 to 1 the noise, may not be needed
+		invrandy = (1.0-randy);
+		randy /= 2.0;
+		//set up the noise
+		
+		iirSampleA = (iirSampleA * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleA;
+		iirSampleB = (iirSampleB * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleB;
+		iirSampleC = (iirSampleC * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleC;
+		iirSampleD = (iirSampleD * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleD;
+		iirSampleE = (iirSampleE * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleE;
+		iirSampleF = (iirSampleF * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleF;
+		iirSampleG = (iirSampleG * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleG;
+		iirSampleH = (iirSampleH * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleH;
+		iirSampleI = (iirSampleI * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleI;
+		iirSampleJ = (iirSampleJ * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleJ;
+		iirSampleK = (iirSampleK * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleK;
+		iirSampleL = (iirSampleL * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleL;
+		iirSampleM = (iirSampleM * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleM;
+		iirSampleN = (iirSampleN * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleN;
+		iirSampleO = (iirSampleO * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleO;
+		iirSampleP = (iirSampleP * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleP;
+		iirSampleQ = (iirSampleQ * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleQ;
+		iirSampleR = (iirSampleR * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleR;
+		iirSampleS = (iirSampleS * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleS;
+		iirSampleT = (iirSampleT * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleT;
+		iirSampleU = (iirSampleU * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleU;
+		iirSampleV = (iirSampleV * (1.0 - iirAmount)) + (ataLowpass * iirAmount); ataLowpass -= iirSampleV;
+		
+		switch (bflip)
+		{
+			case 1:				
+				iirHeadBumpA += (ataLowpass * BassGain);
+				iirHeadBumpA -= (iirHeadBumpA * iirHeadBumpA * iirHeadBumpA * HeadBumpFreq);
+				iirHeadBumpA = (invrandy * iirHeadBumpA) + (randy * iirHeadBumpB) + (randy * iirHeadBumpC);
+				if (iirHeadBumpA > 0) iirHeadBumpA -= clamp;
+				if (iirHeadBumpA < 0) iirHeadBumpA += clamp;
+				HeadBump = iirHeadBumpA;
+				break;
+			case 2:
+				iirHeadBumpB += (ataLowpass * BassGain);
+				iirHeadBumpB -= (iirHeadBumpB * iirHeadBumpB * iirHeadBumpB * HeadBumpFreq);
+				iirHeadBumpB = (randy * iirHeadBumpA) + (invrandy * iirHeadBumpB) + (randy * iirHeadBumpC);
+				if (iirHeadBumpB > 0) iirHeadBumpB -= clamp;
+				if (iirHeadBumpB < 0) iirHeadBumpB += clamp;
+				HeadBump = iirHeadBumpB;
+				break;
+			case 3:
+				iirHeadBumpC += (ataLowpass * BassGain);
+				iirHeadBumpC -= (iirHeadBumpC * iirHeadBumpC * iirHeadBumpC * HeadBumpFreq);
+				iirHeadBumpC = (randy * iirHeadBumpA) + (randy * iirHeadBumpB) + (invrandy * iirHeadBumpC);
+				if (iirHeadBumpC > 0) iirHeadBumpC -= clamp;
+				if (iirHeadBumpC < 0) iirHeadBumpC += clamp;
+				HeadBump = iirHeadBumpC;
+				break;
+		}
+		
+		iirSampleW = (iirSampleW * (1.0 - iirAmount)) + (HeadBump * iirAmount); HeadBump -= iirSampleW;
+		iirSampleX = (iirSampleX * (1.0 - iirAmount)) + (HeadBump * iirAmount); HeadBump -= iirSampleX;
+		
+		SubBump = HeadBump;
+		iirSampleY = (iirSampleY * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump -= iirSampleY;
+		
+		iirDriveSampleC = (iirDriveSampleC * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirDriveSampleC;
+		iirDriveSampleD = (iirDriveSampleD * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirDriveSampleD;
+		
+		
+		SubBump = fabs(SubBump);
+		if (SubOctave == false) {SubBump = -SubBump;}
+		
+		switch (bflip)
+		{
+			case 1:				
+				iirSubBumpA += SubBump;// * BassGain);
+				iirSubBumpA -= (iirSubBumpA * iirSubBumpA * iirSubBumpA * HeadBumpFreq);
+				iirSubBumpA = (invrandy * iirSubBumpA) + (randy * iirSubBumpB) + (randy * iirSubBumpC);
+				if (iirSubBumpA > 0) iirSubBumpA -= clamp;
+				if (iirSubBumpA < 0) iirSubBumpA += clamp;
+				SubBump = iirSubBumpA;
+				break;
+			case 2:
+				iirSubBumpB += SubBump;// * BassGain);
+				iirSubBumpB -= (iirSubBumpB * iirSubBumpB * iirSubBumpB * HeadBumpFreq);
+				iirSubBumpB = (randy * iirSubBumpA) + (invrandy * iirSubBumpB) + (randy * iirSubBumpC);
+				if (iirSubBumpB > 0) iirSubBumpB -= clamp;
+				if (iirSubBumpB < 0) iirSubBumpB += clamp;
+				SubBump = iirSubBumpB;
+				break;
+			case 3:
+				iirSubBumpC += SubBump;// * BassGain);
+				iirSubBumpC -= (iirSubBumpC * iirSubBumpC * iirSubBumpC * HeadBumpFreq);
+				iirSubBumpC = (randy * iirSubBumpA) + (randy * iirSubBumpB) + (invrandy * iirSubBumpC);
+				if (iirSubBumpC > 0) iirSubBumpC -= clamp;
+				if (iirSubBumpC < 0) iirSubBumpC += clamp;
+				SubBump = iirSubBumpC;
+				break;
+		}
+		
+		iirSampleZ = (iirSampleZ * (1.0 - HeadBumpFreq)) + (SubBump * HeadBumpFreq); SubBump = iirSampleZ;
+		iirDriveSampleE = (iirDriveSampleE * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump = iirDriveSampleE;
+		iirDriveSampleF = (iirDriveSampleF * (1.0 - iirAmount)) + (SubBump * iirAmount); SubBump = iirDriveSampleF;
+		
+		
+		inputSampleL += (HeadBump * BassOutGain);
+		inputSampleL += (SubBump * SubOutGain);
+		
+		inputSampleR += (HeadBump * BassOutGain);
+		inputSampleR += (SubBump * SubOutGain);
+		
+		
+		flip = !flip;
+		bflip++;
+		if (bflip < 1 || bflip > 3) bflip = 1;
+		
+		//noise shaping to 64-bit floating point
+		double fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 64 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}