VoiceOfTheStarship

3 files changed, 646 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.cpp b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.cpp
new file mode 100644
index 0000000..c30bd47
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.cpp
@@ -0,0 +1,144 @@
+/* ========================================
+ *  VoiceOfTheStarship - VoiceOfTheStarship.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceOfTheStarship_H
+#include "VoiceOfTheStarship.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new VoiceOfTheStarship(audioMaster);}
+
+VoiceOfTheStarship::VoiceOfTheStarship(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.0;
+	position = 99999999;
+	quadratic = 0;
+	noiseAL = 0.0;
+	noiseBL = 0.0;
+	noiseCL = 0.0;
+	flipL = false;
+	noiseAR = 0.0;
+	noiseBR = 0.0;
+	noiseCR = 0.0;
+	flipR = false;
+	filterflip = false;
+	for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0; f[count] = 0.0;}
+	lastAlgorithm = 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
+}
+
+VoiceOfTheStarship::~VoiceOfTheStarship() {}
+VstInt32 VoiceOfTheStarship::getVendorVersion () {return 1000;}
+void VoiceOfTheStarship::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void VoiceOfTheStarship::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 VoiceOfTheStarship::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 VoiceOfTheStarship::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 VoiceOfTheStarship::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 VoiceOfTheStarship::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 VoiceOfTheStarship::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Filter", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Algrthm", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void VoiceOfTheStarship::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (floor(B*16.9), text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void VoiceOfTheStarship::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 VoiceOfTheStarship::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool VoiceOfTheStarship::getEffectName(char* name) {
+    vst_strncpy(name, "VoiceOfTheStarship", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory VoiceOfTheStarship::getPlugCategory() {return kPlugCategEffect;}
+
+bool VoiceOfTheStarship::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows VoiceOfTheStarship", kVstMaxProductStrLen); return true;
+}
+
+bool VoiceOfTheStarship::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.h b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.h
new file mode 100644
index 0000000..febb344
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarship.h
@@ -0,0 +1,87 @@
+/* ========================================
+ *  VoiceOfTheStarship - VoiceOfTheStarship.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceOfTheStarship_H
+#define __VoiceOfTheStarship_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 = 'vots';    //Change this to what the AU identity is!
+
+class VoiceOfTheStarship : 
+    public AudioEffectX 
+{
+public:
+    VoiceOfTheStarship(audioMasterCallback audioMaster);
+    ~VoiceOfTheStarship();
+    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 noiseAL;
+	double noiseBL;
+	double noiseCL;
+	double noiseAR;
+	double noiseBR;
+	double noiseCR;
+	
+	int position;
+	int quadratic;
+	bool flipL;
+	bool flipR;
+	bool filterflip;
+	
+	double bL[11];
+	double bR[11];
+	
+	double f[11];
+	
+	int lastAlgorithm;
+    
+	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/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarshipProc.cpp b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarshipProc.cpp
new file mode 100644
index 0000000..32979d0
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceOfTheStarship/VoiceOfTheStarshipProc.cpp
@@ -0,0 +1,415 @@
+/* ========================================
+ *  VoiceOfTheStarship - VoiceOfTheStarship.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceOfTheStarship_H
+#include "VoiceOfTheStarship.h"
+#endif
+
+void VoiceOfTheStarship::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+	double cutoff = pow((A*0.89)+0.1,3);
+	if (cutoff > 1.0) cutoff = 1.0;
+	double invcutoff = 1.0 - cutoff;
+	//this is the lowpass
+	
+	double overallscale = ((1.0-A)*9.0)+1.0;
+	double gain = overallscale;
+	if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
+	//this is the moving average with remainders
+	if (overallscale < 1.0) overallscale = 1.0;
+	f[0] /= overallscale;
+	f[1] /= overallscale;
+	f[2] /= overallscale;
+	f[3] /= overallscale;
+	f[4] /= overallscale;
+	f[5] /= overallscale;
+	f[6] /= overallscale;
+	f[7] /= overallscale;
+	f[8] /= overallscale;
+	f[9] /= overallscale;
+	//and now it's neatly scaled, too
+	
+	int lowcut = floor(B*16.9);
+	if (lastAlgorithm != lowcut) {
+		noiseAL = 0.0; noiseBL = 0.0; noiseCL = 0.0;
+		noiseAR = 0.0; noiseBR = 0.0; noiseCR = 0.0;
+		for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0;}
+		lastAlgorithm = lowcut;
+	}
+	//cuts the noise back to 0 if we are changing algorithms,
+	//because that also changes gains and can make loud pops.
+	//We still get pops, but they'd be even worse
+	int dcut;
+	if (lowcut > 15) {lowcut = 1151; dcut= 11517;}
+	if (lowcut == 15) {lowcut = 113; dcut= 1151;}
+	if (lowcut == 14) {lowcut = 71; dcut= 719;}
+	if (lowcut == 13) {lowcut = 53; dcut= 541;}
+	if (lowcut == 12) {lowcut = 31; dcut= 311;}
+	if (lowcut == 11) {lowcut = 23; dcut= 233;}
+	if (lowcut == 10) {lowcut = 19; dcut= 191;}
+	if (lowcut == 9) {lowcut = 17; dcut= 173;}
+	if (lowcut == 8) {lowcut = 13; dcut= 131;}
+	if (lowcut == 7) {lowcut = 11; dcut= 113;}
+	if (lowcut == 6) {lowcut = 7; dcut= 79;}
+	if (lowcut == 5) {lowcut = 6; dcut= 67;}
+	if (lowcut == 4) {lowcut = 5; dcut= 59;}
+	if (lowcut == 3) {lowcut = 4; dcut= 43;}
+	if (lowcut == 2) {lowcut = 3; dcut= 37;}
+	if (lowcut == 1) {lowcut = 2; dcut= 23;}
+	if (lowcut < 1) {lowcut = 1; dcut= 11;}
+	//this is the mechanism for cutting back subs without filtering
+	
+	double rumbletrim = sqrt(lowcut);
+	//this among other things is just to give volume compensation
+	double inputSampleL;
+	double inputSampleR;
+	
+	float fpTemp;
+	double fpOld = 0.618033988749894848204586; //golden ratio!
+	double fpNew = 1.0 - fpOld;
+	
+    while (--sampleFrames >= 0)
+    {
+		inputSampleL = *in1;
+		inputSampleR = *in2;
+		//we then ignore this!
+
+		quadratic -= 1;
+		if (quadratic < 0)
+		{
+			position += 1;		
+			quadratic = position * position;
+			quadratic = quadratic % 170003; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % 17011; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % 1709; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % dcut; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % lowcut;
+			//sets density of the centering force
+			if (noiseAL < 0) {flipL = true;}
+			else {flipL = false;}
+			if (noiseAR < 0) {flipR = true;}
+			else {flipR = false;}
+			//every time we come here, we force the random walk to be
+			//toward the center of the waveform. Without this,
+			//it's a pure random walk that will generate DC.
+		}
+		
+		if (flipL) noiseAL += (rand()/(double)RAND_MAX);
+		else noiseAL -= (rand()/(double)RAND_MAX);
+		if (flipR) noiseAR += (rand()/(double)RAND_MAX);
+		else noiseAR -= (rand()/(double)RAND_MAX);
+		//here's the guts of the random walk
+		
+		if (filterflip)
+		{
+			noiseBL *= invcutoff; noiseBL += (noiseAL*cutoff);
+			inputSampleL = noiseBL;
+			noiseBR *= invcutoff; noiseBR += (noiseAR*cutoff);
+			inputSampleR = noiseBR;
+		}
+		else 
+		{
+			noiseCL *= invcutoff; noiseCL += (noiseAL*cutoff);
+			inputSampleL = noiseCL;
+			noiseCR *= invcutoff; noiseCR += (noiseAR*cutoff);
+			inputSampleR = noiseCR;
+		}
+		//now we have the output of the filter as inputSample.
+		//this filter is shallower than a straight IIR: it's interleaved
+		
+		
+		
+		
+		bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
+		bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = inputSampleL;
+		
+		bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
+		bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = inputSampleR;
+		
+		inputSampleL *= f[0];
+		inputSampleL += (bL[1] * f[1]);
+		inputSampleL += (bL[2] * f[2]);
+		inputSampleL += (bL[3] * f[3]);
+		inputSampleL += (bL[4] * f[4]);
+		inputSampleL += (bL[5] * f[5]);
+		inputSampleL += (bL[6] * f[6]);
+		inputSampleL += (bL[7] * f[7]);
+		inputSampleL += (bL[8] * f[8]);
+		inputSampleL += (bL[9] * f[9]);
+
+		inputSampleR *= f[0];
+		inputSampleR += (bR[1] * f[1]);
+		inputSampleR += (bR[2] * f[2]);
+		inputSampleR += (bR[3] * f[3]);
+		inputSampleR += (bR[4] * f[4]);
+		inputSampleR += (bR[5] * f[5]);
+		inputSampleR += (bR[6] * f[6]);
+		inputSampleR += (bR[7] * f[7]);
+		inputSampleR += (bR[8] * f[8]);
+		inputSampleR += (bR[9] * f[9]);
+
+		inputSampleL *= 0.1;
+		inputSampleR *= 0.1;
+		inputSampleL *= invcutoff;
+		inputSampleR *= invcutoff;
+		inputSampleL /= rumbletrim;
+		inputSampleR /= rumbletrim;
+		
+		flipL = !flipL;
+		flipR = !flipR;
+		filterflip = !filterflip;
+		
+		
+		//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 VoiceOfTheStarship::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+	double cutoff = pow((A*0.89)+0.1,3);
+	if (cutoff > 1.0) cutoff = 1.0;
+	double invcutoff = 1.0 - cutoff;
+	//this is the lowpass
+	
+	double overallscale = ((1.0-A)*9.0)+1.0;
+	double gain = overallscale;
+	if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
+	if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
+	//this is the moving average with remainders
+	if (overallscale < 1.0) overallscale = 1.0;
+	f[0] /= overallscale;
+	f[1] /= overallscale;
+	f[2] /= overallscale;
+	f[3] /= overallscale;
+	f[4] /= overallscale;
+	f[5] /= overallscale;
+	f[6] /= overallscale;
+	f[7] /= overallscale;
+	f[8] /= overallscale;
+	f[9] /= overallscale;
+	//and now it's neatly scaled, too
+	
+	int lowcut = floor(B*16.9);
+	if (lastAlgorithm != lowcut) {
+		noiseAL = 0.0; noiseBL = 0.0; noiseCL = 0.0;
+		noiseAR = 0.0; noiseBR = 0.0; noiseCR = 0.0;
+		for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0;}
+		lastAlgorithm = lowcut;
+	}
+	//cuts the noise back to 0 if we are changing algorithms,
+	//because that also changes gains and can make loud pops.
+	//We still get pops, but they'd be even worse
+	int dcut;
+	if (lowcut > 15) {lowcut = 1151; dcut= 11517;}
+	if (lowcut == 15) {lowcut = 113; dcut= 1151;}
+	if (lowcut == 14) {lowcut = 71; dcut= 719;}
+	if (lowcut == 13) {lowcut = 53; dcut= 541;}
+	if (lowcut == 12) {lowcut = 31; dcut= 311;}
+	if (lowcut == 11) {lowcut = 23; dcut= 233;}
+	if (lowcut == 10) {lowcut = 19; dcut= 191;}
+	if (lowcut == 9) {lowcut = 17; dcut= 173;}
+	if (lowcut == 8) {lowcut = 13; dcut= 131;}
+	if (lowcut == 7) {lowcut = 11; dcut= 113;}
+	if (lowcut == 6) {lowcut = 7; dcut= 79;}
+	if (lowcut == 5) {lowcut = 6; dcut= 67;}
+	if (lowcut == 4) {lowcut = 5; dcut= 59;}
+	if (lowcut == 3) {lowcut = 4; dcut= 43;}
+	if (lowcut == 2) {lowcut = 3; dcut= 37;}
+	if (lowcut == 1) {lowcut = 2; dcut= 23;}
+	if (lowcut < 1) {lowcut = 1; dcut= 11;}
+	//this is the mechanism for cutting back subs without filtering
+	
+	double rumbletrim = sqrt(lowcut);
+	//this among other things is just to give volume compensation
+	double inputSampleL;
+	double inputSampleR;
+	
+	double fpTemp;
+	long double fpOld = 0.618033988749894848204586; //golden ratio!
+	long double fpNew = 1.0 - fpOld;	
+
+    while (--sampleFrames >= 0)
+    {
+		inputSampleL = *in1;
+		inputSampleR = *in2;
+		//we then ignore this!
+		
+		quadratic -= 1;
+		if (quadratic < 0)
+		{
+			position += 1;		
+			quadratic = position * position;
+			quadratic = quadratic % 170003; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % 17011; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % 1709; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % dcut; //% is C++ mod operator
+			quadratic *= quadratic;
+			quadratic = quadratic % lowcut;
+			//sets density of the centering force
+			if (noiseAL < 0) {flipL = true;}
+			else {flipL = false;}
+			if (noiseAR < 0) {flipR = true;}
+			else {flipR = false;}
+			//every time we come here, we force the random walk to be
+			//toward the center of the waveform. Without this,
+			//it's a pure random walk that will generate DC.
+		}
+		
+		if (flipL) noiseAL += (rand()/(double)RAND_MAX);
+		else noiseAL -= (rand()/(double)RAND_MAX);
+		if (flipR) noiseAR += (rand()/(double)RAND_MAX);
+		else noiseAR -= (rand()/(double)RAND_MAX);
+		//here's the guts of the random walk
+		
+		if (filterflip)
+		{
+			noiseBL *= invcutoff; noiseBL += (noiseAL*cutoff);
+			inputSampleL = noiseBL;
+			noiseBR *= invcutoff; noiseBR += (noiseAR*cutoff);
+			inputSampleR = noiseBR;
+		}
+		else 
+		{
+			noiseCL *= invcutoff; noiseCL += (noiseAL*cutoff);
+			inputSampleL = noiseCL;
+			noiseCR *= invcutoff; noiseCR += (noiseAR*cutoff);
+			inputSampleR = noiseCR;
+		}
+		//now we have the output of the filter as inputSample.
+		//this filter is shallower than a straight IIR: it's interleaved
+		
+		
+		
+		
+		bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
+		bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
+		bL[1] = bL[0]; bL[0] = inputSampleL;
+		
+		bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
+		bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
+		bR[1] = bR[0]; bR[0] = inputSampleR;
+		
+		inputSampleL *= f[0];
+		inputSampleL += (bL[1] * f[1]);
+		inputSampleL += (bL[2] * f[2]);
+		inputSampleL += (bL[3] * f[3]);
+		inputSampleL += (bL[4] * f[4]);
+		inputSampleL += (bL[5] * f[5]);
+		inputSampleL += (bL[6] * f[6]);
+		inputSampleL += (bL[7] * f[7]);
+		inputSampleL += (bL[8] * f[8]);
+		inputSampleL += (bL[9] * f[9]);
+		
+		inputSampleR *= f[0];
+		inputSampleR += (bR[1] * f[1]);
+		inputSampleR += (bR[2] * f[2]);
+		inputSampleR += (bR[3] * f[3]);
+		inputSampleR += (bR[4] * f[4]);
+		inputSampleR += (bR[5] * f[5]);
+		inputSampleR += (bR[6] * f[6]);
+		inputSampleR += (bR[7] * f[7]);
+		inputSampleR += (bR[8] * f[8]);
+		inputSampleR += (bR[9] * f[9]);
+		
+		inputSampleL *= 0.1;
+		inputSampleR *= 0.1;
+		inputSampleL *= invcutoff;
+		inputSampleR *= invcutoff;
+		inputSampleL /= rumbletrim;
+		inputSampleR /= rumbletrim;
+		
+		flipL = !flipL;
+		flipR = !flipR;
+		filterflip = !filterflip;
+		
+		//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