Updates (in case my plane crashes)

3 files changed, 978 insertions, 0 deletions

diff --git a/plugins/MacVST/NCSeventeen/source/NCSeventeen.cpp b/plugins/MacVST/NCSeventeen/source/NCSeventeen.cpp
new file mode 100755
index 0000000..9bc129a
--- /dev/null
+++ b/plugins/MacVST/NCSeventeen/source/NCSeventeen.cpp
@@ -0,0 +1,147 @@
+/* ========================================
+ *  NCSeventeen - NCSeventeen.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __NCSeventeen_H
+#include "NCSeventeen.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new NCSeventeen(audioMaster);}
+
+NCSeventeen::NCSeventeen(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 1.0;
+	
+	lastSampleL = 0.0;
+	iirSampleAL = 0.0;
+	iirSampleBL = 0.0;
+	basslevL = 0.0;
+	treblevL = 0.0;
+	cheblevL = 0.0;
+	
+	lastSampleR = 0.0;
+	iirSampleAR = 0.0;
+	iirSampleBR = 0.0;
+	basslevR = 0.0;
+	treblevR = 0.0;
+	cheblevR = 0.0;
+	
+	flip = false;
+
+	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
+}
+
+NCSeventeen::~NCSeventeen() {}
+VstInt32 NCSeventeen::getVendorVersion () {return 1000;}
+void NCSeventeen::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void NCSeventeen::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 NCSeventeen::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 NCSeventeen::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 NCSeventeen::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 NCSeventeen::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 NCSeventeen::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "LOUDER", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "Output", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void NCSeventeen::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A*24.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void NCSeventeen::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 NCSeventeen::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool NCSeventeen::getEffectName(char* name) {
+    vst_strncpy(name, "NC-17", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory NCSeventeen::getPlugCategory() {return kPlugCategEffect;}
+
+bool NCSeventeen::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows NC-17", kVstMaxProductStrLen); return true;
+}
+
+bool NCSeventeen::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/MacVST/NCSeventeen/source/NCSeventeen.h b/plugins/MacVST/NCSeventeen/source/NCSeventeen.h
new file mode 100755
index 0000000..01179e9
--- /dev/null
+++ b/plugins/MacVST/NCSeventeen/source/NCSeventeen.h
@@ -0,0 +1,84 @@
+/* ========================================
+ *  NCSeventeen - NCSeventeen.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __NCSeventeen_H
+#define __NCSeventeen_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 = 'ncse';    //Change this to what the AU identity is!
+
+class NCSeventeen : 
+    public AudioEffectX 
+{
+public:
+    NCSeventeen(audioMasterCallback audioMaster);
+    ~NCSeventeen();
+    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;
+    
+private: 
+	double lastSampleL;
+	double iirSampleAL;
+	double iirSampleBL;
+	double basslevL;
+	double treblevL;
+	double cheblevL;
+
+	double lastSampleR;
+	double iirSampleAR;
+	double iirSampleBR;
+	double basslevR;
+	double treblevR;
+	double cheblevR;
+	
+	bool flip;
+	
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool fpFlip;
+	//default stuff
+
+    float A;
+    float B;
+};
+
+#endif
diff --git a/plugins/MacVST/NCSeventeen/source/NCSeventeenProc.cpp b/plugins/MacVST/NCSeventeen/source/NCSeventeenProc.cpp
new file mode 100755
index 0000000..b2f2aa1
--- /dev/null
+++ b/plugins/MacVST/NCSeventeen/source/NCSeventeenProc.cpp
@@ -0,0 +1,747 @@
+/* ========================================
+ *  NCSeventeen - NCSeventeen.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __NCSeventeen_H
+#include "NCSeventeen.h"
+#endif
+
+void NCSeventeen::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	double inP2;
+	double chebyshev;
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	float fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+	double IIRscaleback = 0.0004716;
+	double bassScaleback = 0.0002364;
+	double trebleScaleback = 0.0005484;
+	double addBassBuss = 0.000243;
+	double addTrebBuss = 0.000407;
+	double addShortBuss = 0.000326;
+	IIRscaleback /= overallscale;
+	bassScaleback /= overallscale;
+	trebleScaleback /= overallscale;
+	addBassBuss /= overallscale;
+	addTrebBuss /= overallscale;
+	addShortBuss /= overallscale;
+	double limitingBass = 0.39;
+	double limitingTreb = 0.6;
+	double limiting = 0.36;
+	double maxfeedBass = 0.972;
+	double maxfeedTreb = 0.972;
+	double maxfeed = 0.975;
+	double bridgerectifier;
+	long double inputSampleL;
+	double lowSampleL = 0.0;
+	double highSampleL;
+	double distSampleL;
+	double minusSampleL;
+	double plusSampleL;
+	long double inputSampleR;
+	double lowSampleR = 0.0;
+	double highSampleR;
+	double distSampleR;
+	double minusSampleR;
+	double plusSampleR;
+	double gain = pow(10.0,(A*24.0)/20);
+	double outlevel = B;
+	
+    
+    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.
+		}
+
+		inputSampleL *= gain;
+		inputSampleR *= gain;
+		
+		if (flip)
+		{
+			iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1);
+			lowSampleL = iirSampleAL;
+			iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1);
+			lowSampleR = iirSampleAR;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1);
+			lowSampleL = iirSampleBL;
+			iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1);
+			lowSampleR = iirSampleBR;
+		}
+		highSampleL = inputSampleL - lowSampleL;
+		highSampleR = inputSampleR - lowSampleR;
+		flip = !flip;
+		//we now have two bands and the original source
+		
+		inP2 = lowSampleL * lowSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= basslevL;
+		//second harmonic max +1
+		if (basslevL > 0) basslevL -= bassScaleback;
+		if (basslevL < 0) basslevL += bassScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(lowSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = lowSampleL - distSampleL;
+		plusSampleL = lowSampleL + distSampleL;
+		if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass;
+		if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass;
+		if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass;
+		if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass;
+		if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss);
+		if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss);
+		if (basslevL > 1.0)  basslevL = 1.0;
+		if (basslevL < -1.0) basslevL = -1.0;
+		bridgerectifier = fabs(lowSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleL > 0.0) lowSampleL = bridgerectifier;
+		else lowSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		lowSampleL /= (1.0+fabs(basslevL*limitingBass));
+		lowSampleL += chebyshev;
+		//apply the correction measuresL
+		
+		inP2 = lowSampleR * lowSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= basslevR;
+		//second harmonic max +1
+		if (basslevR > 0) basslevR -= bassScaleback;
+		if (basslevR < 0) basslevR += bassScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(lowSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = lowSampleR - distSampleR;
+		plusSampleR = lowSampleR + distSampleR;
+		if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass;
+		if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass;
+		if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass;
+		if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass;
+		if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss);
+		if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss);
+		if (basslevR > 1.0)  basslevR = 1.0;
+		if (basslevR < -1.0) basslevR = -1.0;
+		bridgerectifier = fabs(lowSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleR > 0.0) lowSampleR = bridgerectifier;
+		else lowSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		lowSampleR /= (1.0+fabs(basslevR*limitingBass));
+		lowSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		inP2 = highSampleL * highSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= treblevL;
+		//second harmonic max +1
+		if (treblevL > 0) treblevL -= trebleScaleback;
+		if (treblevL < 0) treblevL += trebleScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(highSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = highSampleL - distSampleL;
+		plusSampleL = highSampleL + distSampleL;
+		if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb;
+		if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb;
+		if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb;
+		if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb;
+		if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss);
+		if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss);
+		if (treblevL > 1.0)  treblevL = 1.0;
+		if (treblevL < -1.0) treblevL = -1.0;
+		bridgerectifier = fabs(highSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleL > 0.0) highSampleL = bridgerectifier;
+		else highSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		highSampleL /= (1.0+fabs(treblevL*limitingTreb));
+		highSampleL += chebyshev;
+		//apply the correction measuresL
+		
+		inP2 = highSampleR * highSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= treblevR;
+		//second harmonic max +1
+		if (treblevR > 0) treblevR -= trebleScaleback;
+		if (treblevR < 0) treblevR += trebleScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(highSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = highSampleR - distSampleR;
+		plusSampleR = highSampleR + distSampleR;
+		if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb;
+		if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb;
+		if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb;
+		if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb;
+		if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss);
+		if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss);
+		if (treblevR > 1.0)  treblevR = 1.0;
+		if (treblevR < -1.0) treblevR = -1.0;
+		bridgerectifier = fabs(highSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleR > 0.0) highSampleR = bridgerectifier;
+		else highSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		highSampleR /= (1.0+fabs(treblevR*limitingTreb));
+		highSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		inputSampleL = lowSampleL + highSampleL;
+		inputSampleR = lowSampleR + highSampleR;
+		
+		inP2 = inputSampleL * inputSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= cheblevL;
+		//third harmonic max -1
+		if (cheblevL > 0) cheblevL -= (IIRscaleback);
+		if (cheblevL < 0) cheblevL += (IIRscaleback);
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = inputSampleL - distSampleL;
+		plusSampleL = inputSampleL + distSampleL;
+		if (minusSampleL > maxfeed) minusSampleL = maxfeed;
+		if (plusSampleL > maxfeed) plusSampleL = maxfeed;
+		if (plusSampleL < -maxfeed) plusSampleL = -maxfeed;
+		if (minusSampleL < -maxfeed) minusSampleL = -maxfeed;
+		if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss);
+		if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss);
+		if (cheblevL > 1.0)  cheblevL = 1.0;
+		if (cheblevL < -1.0) cheblevL = -1.0;
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleL > 0.0) inputSampleL = bridgerectifier;
+		else inputSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		inputSampleL /= (1.0+fabs(cheblevL*limiting));
+		inputSampleL += chebyshev;
+		//apply the correction measuresL
+
+		inP2 = inputSampleR * inputSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= cheblevR;
+		//third harmonic max -1
+		if (cheblevR > 0) cheblevR -= IIRscaleback;
+		if (cheblevR < 0) cheblevR += IIRscaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = inputSampleR - distSampleR;
+		plusSampleR = inputSampleR + distSampleR;
+		if (minusSampleR > maxfeed) minusSampleR = maxfeed;
+		if (plusSampleR > maxfeed) plusSampleR = maxfeed;
+		if (plusSampleR < -maxfeed) plusSampleR = -maxfeed;
+		if (minusSampleR < -maxfeed) minusSampleR = -maxfeed;
+		if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss);
+		if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss);
+		if (cheblevR > 1.0)  cheblevR = 1.0;
+		if (cheblevR < -1.0) cheblevR = -1.0;
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleR > 0.0) inputSampleR = bridgerectifier;
+		else inputSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		inputSampleR /= (1.0+fabs(cheblevR*limiting));
+		inputSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		if (outlevel < 1.0) {
+			inputSampleL *= outlevel;
+			inputSampleR *= outlevel;
+		}
+		
+		if (inputSampleL > 0.95) inputSampleL = 0.95;
+		if (inputSampleL < -0.95) inputSampleL = -0.95;
+		if (inputSampleR > 0.95) inputSampleR = 0.95;
+		if (inputSampleR < -0.95) inputSampleR = -0.95;
+		//iron bar
+
+		//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 NCSeventeen::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	double inP2;
+	double chebyshev;
+	double overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= getSampleRate();
+	double fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	
+	double IIRscaleback = 0.0004716;
+	double bassScaleback = 0.0002364;
+	double trebleScaleback = 0.0005484;
+	double addBassBuss = 0.000243;
+	double addTrebBuss = 0.000407;
+	double addShortBuss = 0.000326;
+	IIRscaleback /= overallscale;
+	bassScaleback /= overallscale;
+	trebleScaleback /= overallscale;
+	addBassBuss /= overallscale;
+	addTrebBuss /= overallscale;
+	addShortBuss /= overallscale;
+	double limitingBass = 0.39;
+	double limitingTreb = 0.6;
+	double limiting = 0.36;
+	double maxfeedBass = 0.972;
+	double maxfeedTreb = 0.972;
+	double maxfeed = 0.975;
+	double bridgerectifier;
+	long double inputSampleL;
+	double lowSampleL = 0.0;
+	double highSampleL;
+	double distSampleL;
+	double minusSampleL;
+	double plusSampleL;
+	long double inputSampleR;
+	double lowSampleR = 0.0;
+	double highSampleR;
+	double distSampleR;
+	double minusSampleR;
+	double plusSampleR;
+	double gain = pow(10.0,(A*24.0)/20);
+	double outlevel = B;
+	
+    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.
+		}
+		
+		inputSampleL *= gain;
+		inputSampleR *= gain;
+		
+		if (flip)
+		{
+			iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1);
+			lowSampleL = iirSampleAL;
+			iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1);
+			lowSampleR = iirSampleAR;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1);
+			lowSampleL = iirSampleBL;
+			iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1);
+			lowSampleR = iirSampleBR;
+		}
+		highSampleL = inputSampleL - lowSampleL;
+		highSampleR = inputSampleR - lowSampleR;
+		flip = !flip;
+		//we now have two bands and the original source
+		
+		inP2 = lowSampleL * lowSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= basslevL;
+		//second harmonic max +1
+		if (basslevL > 0) basslevL -= bassScaleback;
+		if (basslevL < 0) basslevL += bassScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(lowSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = lowSampleL - distSampleL;
+		plusSampleL = lowSampleL + distSampleL;
+		if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass;
+		if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass;
+		if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass;
+		if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass;
+		if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss);
+		if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss);
+		if (basslevL > 1.0)  basslevL = 1.0;
+		if (basslevL < -1.0) basslevL = -1.0;
+		bridgerectifier = fabs(lowSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleL > 0.0) lowSampleL = bridgerectifier;
+		else lowSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		lowSampleL /= (1.0+fabs(basslevL*limitingBass));
+		lowSampleL += chebyshev;
+		//apply the correction measuresL
+		
+		inP2 = lowSampleR * lowSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= basslevR;
+		//second harmonic max +1
+		if (basslevR > 0) basslevR -= bassScaleback;
+		if (basslevR < 0) basslevR += bassScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(lowSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = lowSampleR - distSampleR;
+		plusSampleR = lowSampleR + distSampleR;
+		if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass;
+		if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass;
+		if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass;
+		if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass;
+		if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss);
+		if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss);
+		if (basslevR > 1.0)  basslevR = 1.0;
+		if (basslevR < -1.0) basslevR = -1.0;
+		bridgerectifier = fabs(lowSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (lowSampleR > 0.0) lowSampleR = bridgerectifier;
+		else lowSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		lowSampleR /= (1.0+fabs(basslevR*limitingBass));
+		lowSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		inP2 = highSampleL * highSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= treblevL;
+		//second harmonic max +1
+		if (treblevL > 0) treblevL -= trebleScaleback;
+		if (treblevL < 0) treblevL += trebleScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(highSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = highSampleL - distSampleL;
+		plusSampleL = highSampleL + distSampleL;
+		if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb;
+		if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb;
+		if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb;
+		if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb;
+		if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss);
+		if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss);
+		if (treblevL > 1.0)  treblevL = 1.0;
+		if (treblevL < -1.0) treblevL = -1.0;
+		bridgerectifier = fabs(highSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleL > 0.0) highSampleL = bridgerectifier;
+		else highSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		highSampleL /= (1.0+fabs(treblevL*limitingTreb));
+		highSampleL += chebyshev;
+		//apply the correction measuresL
+		
+		inP2 = highSampleR * highSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= treblevR;
+		//second harmonic max +1
+		if (treblevR > 0) treblevR -= trebleScaleback;
+		if (treblevR < 0) treblevR += trebleScaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(highSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = highSampleR - distSampleR;
+		plusSampleR = highSampleR + distSampleR;
+		if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb;
+		if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb;
+		if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb;
+		if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb;
+		if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss);
+		if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss);
+		if (treblevR > 1.0)  treblevR = 1.0;
+		if (treblevR < -1.0) treblevR = -1.0;
+		bridgerectifier = fabs(highSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (highSampleR > 0.0) highSampleR = bridgerectifier;
+		else highSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		highSampleR /= (1.0+fabs(treblevR*limitingTreb));
+		highSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		inputSampleL = lowSampleL + highSampleL;
+		inputSampleR = lowSampleR + highSampleR;
+		
+		inP2 = inputSampleL * inputSampleL;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= cheblevL;
+		//third harmonic max -1
+		if (cheblevL > 0) cheblevL -= (IIRscaleback);
+		if (cheblevL < 0) cheblevL += (IIRscaleback);
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleL > 0.0) distSampleL = bridgerectifier;
+		else distSampleL = -bridgerectifier;
+		minusSampleL = inputSampleL - distSampleL;
+		plusSampleL = inputSampleL + distSampleL;
+		if (minusSampleL > maxfeed) minusSampleL = maxfeed;
+		if (plusSampleL > maxfeed) plusSampleL = maxfeed;
+		if (plusSampleL < -maxfeed) plusSampleL = -maxfeed;
+		if (minusSampleL < -maxfeed) minusSampleL = -maxfeed;
+		if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss);
+		if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss);
+		if (cheblevL > 1.0)  cheblevL = 1.0;
+		if (cheblevL < -1.0) cheblevL = -1.0;
+		bridgerectifier = fabs(inputSampleL);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleL > 0.0) inputSampleL = bridgerectifier;
+		else inputSampleL = -bridgerectifier;
+		//apply the distortion transform for reals
+		inputSampleL /= (1.0+fabs(cheblevL*limiting));
+		inputSampleL += chebyshev;
+		//apply the correction measuresL
+		
+		inP2 = inputSampleR * inputSampleR;
+		if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
+		chebyshev = (2 * inP2);
+		chebyshev *= cheblevR;
+		//third harmonic max -1
+		if (cheblevR > 0) cheblevR -= IIRscaleback;
+		if (cheblevR < 0) cheblevR += IIRscaleback;
+		//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleR > 0.0) distSampleR = bridgerectifier;
+		else distSampleR = -bridgerectifier;
+		minusSampleR = inputSampleR - distSampleR;
+		plusSampleR = inputSampleR + distSampleR;
+		if (minusSampleR > maxfeed) minusSampleR = maxfeed;
+		if (plusSampleR > maxfeed) plusSampleR = maxfeed;
+		if (plusSampleR < -maxfeed) plusSampleR = -maxfeed;
+		if (minusSampleR < -maxfeed) minusSampleR = -maxfeed;
+		if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss);
+		if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss);
+		if (cheblevR > 1.0)  cheblevR = 1.0;
+		if (cheblevR < -1.0) cheblevR = -1.0;
+		bridgerectifier = fabs(inputSampleR);
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		//max value for sine function
+		bridgerectifier = sin(bridgerectifier);
+		if (inputSampleR > 0.0) inputSampleR = bridgerectifier;
+		else inputSampleR = -bridgerectifier;
+		//apply the distortion transform for reals
+		inputSampleR /= (1.0+fabs(cheblevR*limiting));
+		inputSampleR += chebyshev;
+		//apply the correction measuresR
+		
+		if (outlevel < 1.0) {
+			inputSampleL *= outlevel;
+			inputSampleR *= outlevel;
+		}		
+		
+		if (inputSampleL > 0.95) inputSampleL = 0.95;
+		if (inputSampleL < -0.95) inputSampleL = -0.95;
+		if (inputSampleR > 0.95) inputSampleR = 0.95;
+		if (inputSampleR < -0.95) inputSampleR = -0.95;
+		//iron bar
+		
+		//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