DrumSlam

3 files changed, 745 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/DrumSlam/DrumSlam.cpp b/plugins/LinuxVST/src/DrumSlam/DrumSlam.cpp
new file mode 100755
index 0000000..8a6ae7b
--- /dev/null
+++ b/plugins/LinuxVST/src/DrumSlam/DrumSlam.cpp
@@ -0,0 +1,159 @@
+/* ========================================
+ *  DrumSlam - DrumSlam.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __DrumSlam_H
+#include "DrumSlam.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new DrumSlam(audioMaster);}
+
+DrumSlam::DrumSlam(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.0;
+	B = 1.0;
+	C = 1.0;
+
+	iirSampleAL = 0.0;
+	iirSampleBL = 0.0;
+	iirSampleCL = 0.0;
+	iirSampleDL = 0.0;
+	iirSampleEL = 0.0;
+	iirSampleFL = 0.0;
+	iirSampleGL = 0.0;
+	iirSampleHL = 0.0;
+	lastSampleL = 0.0;
+
+	iirSampleAR = 0.0;
+	iirSampleBR = 0.0;
+	iirSampleCR = 0.0;
+	iirSampleDR = 0.0;
+	iirSampleER = 0.0;
+	iirSampleFR = 0.0;
+	iirSampleGR = 0.0;
+	iirSampleHR = 0.0;
+	lastSampleR = 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
+}
+
+DrumSlam::~DrumSlam() {}
+VstInt32 DrumSlam::getVendorVersion () {return 1000;}
+void DrumSlam::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void DrumSlam::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 DrumSlam::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 DrumSlam::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 DrumSlam::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float DrumSlam::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 DrumSlam::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Drive", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Output", 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 DrumSlam::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((A*3.0)+1.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        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 DrumSlam::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;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 DrumSlam::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool DrumSlam::getEffectName(char* name) {
+    vst_strncpy(name, "DrumSlam", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory DrumSlam::getPlugCategory() {return kPlugCategEffect;}
+
+bool DrumSlam::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows DrumSlam", kVstMaxProductStrLen); return true;
+}
+
+bool DrumSlam::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/DrumSlam/DrumSlam.h b/plugins/LinuxVST/src/DrumSlam/DrumSlam.h
new file mode 100755
index 0000000..5ec2fda
--- /dev/null
+++ b/plugins/LinuxVST/src/DrumSlam/DrumSlam.h
@@ -0,0 +1,92 @@
+/* ========================================
+ *  DrumSlam - DrumSlam.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __DrumSlam_H
+#define __DrumSlam_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 = 'drsl';    //Change this to what the AU identity is!
+
+class DrumSlam : 
+    public AudioEffectX 
+{
+public:
+    DrumSlam(audioMasterCallback audioMaster);
+    ~DrumSlam();
+    virtual bool getEffectName(char* name);                       // The plug-in name
+    virtual VstPlugCategory getPlugCategory();                    // The general category for the plug-in
+    virtual bool getProductString(char* text);                    // This is a unique plug-in string provided by Steinberg
+    virtual bool getVendorString(char* text);                     // Vendor info
+    virtual VstInt32 getVendorVersion();                          // Version number
+    virtual void processReplacing (float** inputs, float** outputs, VstInt32 sampleFrames);
+    virtual void processDoubleReplacing (double** inputs, double** outputs, VstInt32 sampleFrames);
+    virtual void getProgramName(char *name);                      // read the name from the host
+    virtual void setProgramName(char *name);                      // changes the name of the preset displayed in the host
+	virtual VstInt32 getChunk (void** data, bool isPreset);
+	virtual VstInt32 setChunk (void* data, VstInt32 byteSize, bool isPreset);
+    virtual float getParameter(VstInt32 index);                   // get the parameter value at the specified index
+    virtual void setParameter(VstInt32 index, float value);       // set the parameter at index to value
+    virtual void getParameterLabel(VstInt32 index, char *text);  // label for the parameter (eg dB)
+    virtual void getParameterName(VstInt32 index, char *text);    // name of the parameter
+    virtual void getParameterDisplay(VstInt32 index, char *text); // text description of the current value    
+    virtual VstInt32 canDo(char *text);
+private:
+    char _programName[kVstMaxProgNameLen + 1];
+    std::set< std::string > _canDo;
+    
+	long double fpNShapeLA;
+	long double fpNShapeLB;
+	long double fpNShapeRA;
+	long double fpNShapeRB;
+	bool fpFlip;
+	//default stuff
+	
+	double iirSampleAL;
+	double iirSampleBL;
+	double iirSampleCL;
+	double iirSampleDL;
+	double iirSampleEL;
+	double iirSampleFL;
+	double iirSampleGL;
+	double iirSampleHL;
+	double lastSampleL;
+	
+	double iirSampleAR;
+	double iirSampleBR;
+	double iirSampleCR;
+	double iirSampleDR;
+	double iirSampleER;
+	double iirSampleFR;
+	double iirSampleGR;
+	double iirSampleHR;
+	double lastSampleR;
+		
+    float A;
+    float B;
+    float C;
+    float D;
+    float E; //parameters. Always 0-1, and we scale/alter them elsewhere.
+
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/DrumSlam/DrumSlamProc.cpp b/plugins/LinuxVST/src/DrumSlam/DrumSlamProc.cpp
new file mode 100755
index 0000000..171b353
--- /dev/null
+++ b/plugins/LinuxVST/src/DrumSlam/DrumSlamProc.cpp
@@ -0,0 +1,494 @@
+/* ========================================
+ *  DrumSlam - DrumSlam.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __DrumSlam_H
+#include "DrumSlam.h"
+#endif
+
+void DrumSlam::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 iirAmountL = 0.0819;
+	iirAmountL /= overallscale;
+	double iirAmountH = 0.377933067;
+	iirAmountH /= overallscale;
+	double drive = (A*3.0)+1.0;
+	double out = B;
+	double wet = C;
+	double dry = 1.0 - wet;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double 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.
+		}
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		long double lowSampleL;
+		long double lowSampleR;
+		long double midSampleL;
+		long double midSampleR;
+		long double highSampleL;
+		long double highSampleR;
+		
+
+		inputSampleL *= drive;
+		inputSampleR *= drive;
+		
+		if (fpFlip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
+			iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL);
+			lowSampleL = iirSampleBL;
+			
+			iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
+			iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL);
+			lowSampleR = iirSampleBR;
+
+			iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
+			iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH);
+			midSampleL = iirSampleFL - iirSampleBL;
+
+			iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
+			iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH);
+			midSampleR = iirSampleFR - iirSampleBR;
+
+			highSampleL = inputSampleL - iirSampleFL;
+			highSampleR = inputSampleR - iirSampleFR;
+		}
+		else
+		{
+			iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
+			iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL);
+			lowSampleL = iirSampleDL;
+
+			iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
+			iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL);
+			lowSampleR = iirSampleDR;
+
+			iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
+			iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH);
+			midSampleL = iirSampleHL - iirSampleDL;
+
+			iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
+			iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH);
+			midSampleR = iirSampleHR - iirSampleDR;
+
+			highSampleL = inputSampleL - iirSampleHL;
+			highSampleR = inputSampleR - iirSampleHR;
+		}
+		//generate the tone bands we're using
+		if (lowSampleL > 1.0) {lowSampleL = 1.0;}
+		if (lowSampleL < -1.0) {lowSampleL = -1.0;}
+		if (lowSampleR > 1.0) {lowSampleR = 1.0;}
+		if (lowSampleR < -1.0) {lowSampleR = -1.0;}
+		lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) );
+		lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) );
+		lowSampleL *= drive;
+		lowSampleR *= drive;
+		
+		if (highSampleL > 1.0) {highSampleL = 1.0;}
+		if (highSampleL < -1.0) {highSampleL = -1.0;}
+		if (highSampleR > 1.0) {highSampleR = 1.0;}
+		if (highSampleR < -1.0) {highSampleR = -1.0;}
+		highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) );
+		highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) );
+		highSampleL *= drive;
+		highSampleR *= drive;
+		
+		midSampleL = midSampleL * drive;
+		midSampleR = midSampleR * drive;
+		
+		long double skew = (midSampleL - lastSampleL);
+		lastSampleL = midSampleL;
+		//skew will be direction/angle
+		long double bridgerectifier = fabs(skew);
+		if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
+		//for skew we want it to go to zero effect again, so we use full range of the sine
+		bridgerectifier = sin(bridgerectifier);
+		if (skew > 0) skew = bridgerectifier*3.1415926;
+		else skew = -bridgerectifier*3.1415926;
+		//skew is now sined and clamped and then re-amplified again
+		skew *= midSampleL;
+		//cools off sparkliness and crossover distortion
+		skew *= 1.557079633;
+		//crank up the gain on this so we can make it sing
+		bridgerectifier = fabs(midSampleL);
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		bridgerectifier *= drive;
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		if (midSampleL > 0)
+		{
+			midSampleL = bridgerectifier;
+		}
+		else
+		{
+			midSampleL = -bridgerectifier;
+		}
+		//blend according to positive and negative controls, left
+		
+		skew = (midSampleR - lastSampleR);
+		lastSampleR = midSampleR;
+		//skew will be direction/angle
+		bridgerectifier = fabs(skew);
+		if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
+		//for skew we want it to go to zero effect again, so we use full range of the sine
+		bridgerectifier = sin(bridgerectifier);
+		if (skew > 0) skew = bridgerectifier*3.1415926;
+		else skew = -bridgerectifier*3.1415926;
+		//skew is now sined and clamped and then re-amplified again
+		skew *= midSampleR;
+		//cools off sparkliness and crossover distortion
+		skew *= 1.557079633;
+		//crank up the gain on this so we can make it sing
+		bridgerectifier = fabs(midSampleR);
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		bridgerectifier *= drive;
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		if (midSampleR > 0)
+		{
+			midSampleR = bridgerectifier;
+		}
+		else
+		{
+			midSampleR = -bridgerectifier;
+		}
+		//blend according to positive and negative controls, right
+		
+		inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out;
+		inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out;
+		
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		}
+		
+		//noise shaping to 32-bit floating point
+		float fpTemp;
+		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 DrumSlam::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 iirAmountL = 0.0819;
+	iirAmountL /= overallscale;
+	double iirAmountH = 0.377933067;
+	iirAmountH /= overallscale;
+	double drive = (A*3.0)+1.0;
+	double out = B;
+	double wet = C;
+	double dry = 1.0 - wet;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double 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.
+		}
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		long double lowSampleL;
+		long double lowSampleR;
+		long double midSampleL;
+		long double midSampleR;
+		long double highSampleL;
+		long double highSampleR;
+		
+		
+		inputSampleL *= drive;
+		inputSampleR *= drive;
+		
+		if (fpFlip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
+			iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL);
+			lowSampleL = iirSampleBL;
+			
+			iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
+			iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL);
+			lowSampleR = iirSampleBR;
+			
+			iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
+			iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH);
+			midSampleL = iirSampleFL - iirSampleBL;
+			
+			iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
+			iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH);
+			midSampleR = iirSampleFR - iirSampleBR;
+			
+			highSampleL = inputSampleL - iirSampleFL;
+			highSampleR = inputSampleR - iirSampleFR;
+		}
+		else
+		{
+			iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
+			iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL);
+			lowSampleL = iirSampleDL;
+			
+			iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
+			iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL);
+			lowSampleR = iirSampleDR;
+			
+			iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
+			iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH);
+			midSampleL = iirSampleHL - iirSampleDL;
+			
+			iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
+			iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH);
+			midSampleR = iirSampleHR - iirSampleDR;
+			
+			highSampleL = inputSampleL - iirSampleHL;
+			highSampleR = inputSampleR - iirSampleHR;
+		}
+		//generate the tone bands we're using
+		if (lowSampleL > 1.0) {lowSampleL = 1.0;}
+		if (lowSampleL < -1.0) {lowSampleL = -1.0;}
+		if (lowSampleR > 1.0) {lowSampleR = 1.0;}
+		if (lowSampleR < -1.0) {lowSampleR = -1.0;}
+		lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) );
+		lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) );
+		lowSampleL *= drive;
+		lowSampleR *= drive;
+		
+		if (highSampleL > 1.0) {highSampleL = 1.0;}
+		if (highSampleL < -1.0) {highSampleL = -1.0;}
+		if (highSampleR > 1.0) {highSampleR = 1.0;}
+		if (highSampleR < -1.0) {highSampleR = -1.0;}
+		highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) );
+		highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) );
+		highSampleL *= drive;
+		highSampleR *= drive;
+		
+		midSampleL = midSampleL * drive;
+		midSampleR = midSampleR * drive;
+		
+		long double skew = (midSampleL - lastSampleL);
+		lastSampleL = midSampleL;
+		//skew will be direction/angle
+		long double bridgerectifier = fabs(skew);
+		if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
+		//for skew we want it to go to zero effect again, so we use full range of the sine
+		bridgerectifier = sin(bridgerectifier);
+		if (skew > 0) skew = bridgerectifier*3.1415926;
+		else skew = -bridgerectifier*3.1415926;
+		//skew is now sined and clamped and then re-amplified again
+		skew *= midSampleL;
+		//cools off sparkliness and crossover distortion
+		skew *= 1.557079633;
+		//crank up the gain on this so we can make it sing
+		bridgerectifier = fabs(midSampleL);
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		bridgerectifier *= drive;
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		if (midSampleL > 0)
+		{
+			midSampleL = bridgerectifier;
+		}
+		else
+		{
+			midSampleL = -bridgerectifier;
+		}
+		//blend according to positive and negative controls, left
+		
+		skew = (midSampleR - lastSampleR);
+		lastSampleR = midSampleR;
+		//skew will be direction/angle
+		bridgerectifier = fabs(skew);
+		if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
+		//for skew we want it to go to zero effect again, so we use full range of the sine
+		bridgerectifier = sin(bridgerectifier);
+		if (skew > 0) skew = bridgerectifier*3.1415926;
+		else skew = -bridgerectifier*3.1415926;
+		//skew is now sined and clamped and then re-amplified again
+		skew *= midSampleR;
+		//cools off sparkliness and crossover distortion
+		skew *= 1.557079633;
+		//crank up the gain on this so we can make it sing
+		bridgerectifier = fabs(midSampleR);
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		bridgerectifier *= drive;
+		bridgerectifier += skew;
+		if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
+		bridgerectifier = sin(bridgerectifier);
+		if (midSampleR > 0)
+		{
+			midSampleR = bridgerectifier;
+		}
+		else
+		{
+			midSampleR = -bridgerectifier;
+		}
+		//blend according to positive and negative controls, right
+		
+		inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out;
+		inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out;
+		
+		if (wet !=1.0) {
+			inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
+			inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
+		}
+		
+		//noise shaping to 64-bit floating point
+		double fpTemp;
+		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