Baxandall

4 files changed, 503 insertions, 0 deletions

diff --git a/plugins/LinuxVST/CMakeLists.txt b/plugins/LinuxVST/CMakeLists.txt
index 6e23160..69d0cf3 100755
--- a/plugins/LinuxVST/CMakeLists.txt
+++ b/plugins/LinuxVST/CMakeLists.txt
@@ -18,6 +18,7 @@ add_airwindows_plugin(Aura)
 add_airwindows_plugin(Average)
 add_airwindows_plugin(BassDrive)
 add_airwindows_plugin(BassKit)
+add_airwindows_plugin(Baxandall)
 add_airwindows_plugin(Biquad)
 add_airwindows_plugin(Biquad2)
 add_airwindows_plugin(BiquadOneHalf)
diff --git a/plugins/LinuxVST/src/Baxandall/Baxandall.cpp b/plugins/LinuxVST/src/Baxandall/Baxandall.cpp
new file mode 100755
index 0000000..d27affa
--- /dev/null
+++ b/plugins/LinuxVST/src/Baxandall/Baxandall.cpp
@@ -0,0 +1,145 @@
+/* ========================================
+ *  Baxandall - Baxandall.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Baxandall_H
+#include "Baxandall.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Baxandall(audioMaster);}
+
+Baxandall::Baxandall(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.5;
+	C = 0.5;
+	for (int x = 0; x < 9; x++) {
+		trebleAL[x] = 0.0;
+		trebleBL[x] = 0.0;
+		bassAL[x] = 0.0;
+		bassBL[x] = 0.0;
+		trebleAR[x] = 0.0;
+		trebleBR[x] = 0.0;
+		bassAR[x] = 0.0;
+		bassBR[x] = 0.0;
+	}
+	flip = false;
+	fpd = 17;
+	//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
+}
+
+Baxandall::~Baxandall() {}
+VstInt32 Baxandall::getVendorVersion () {return 1000;}
+void Baxandall::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Baxandall::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 Baxandall::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 Baxandall::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 Baxandall::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 Baxandall::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 Baxandall::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Treble", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Bass", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Output", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void Baxandall::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string ((A*30.0)-15.0, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string ((B*30.0)-15.0, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string ((C*30.0)-15.0, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Baxandall::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "dB", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Baxandall::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Baxandall::getEffectName(char* name) {
+    vst_strncpy(name, "Baxandall", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Baxandall::getPlugCategory() {return kPlugCategEffect;}
+
+bool Baxandall::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Baxandall", kVstMaxProductStrLen); return true;
+}
+
+bool Baxandall::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Baxandall/Baxandall.h b/plugins/LinuxVST/src/Baxandall/Baxandall.h
new file mode 100755
index 0000000..380e3d0
--- /dev/null
+++ b/plugins/LinuxVST/src/Baxandall/Baxandall.h
@@ -0,0 +1,75 @@
+/* ========================================
+ *  Baxandall - Baxandall.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Baxandall_H
+#define __Baxandall_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 = 'baxa';    //Change this to what the AU identity is!
+
+class Baxandall : 
+    public AudioEffectX 
+{
+public:
+    Baxandall(audioMasterCallback audioMaster);
+    ~Baxandall();
+    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;
+    
+	uint32_t fpd;
+	//default stuff
+	long double trebleAL[9];
+	long double trebleBL[9];
+	long double bassAL[9];
+	long double bassBL[9];
+	
+	long double trebleAR[9];
+	long double trebleBR[9];
+	long double bassAR[9];
+	long double bassBR[9];
+	bool flip;
+	
+    float A;
+    float B;
+    float C;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp b/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp
new file mode 100755
index 0000000..123d328
--- /dev/null
+++ b/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp
@@ -0,0 +1,282 @@
+/* ========================================
+ *  Baxandall - Baxandall.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Baxandall_H
+#include "Baxandall.h"
+#endif
+
+void Baxandall::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+	
+	double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0);
+	double trebleFreq = (4410.0*trebleGain)/getSampleRate();
+	if (trebleFreq > 0.45) trebleFreq = 0.45;
+	trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq;
+	double bassGain = pow(10.0,((B*30.0)-15.0)/20.0);
+	double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0);
+	bassFreq = (4410.0*bassFreq)/getSampleRate();
+	if (bassFreq > 0.45) bassFreq = 0.45;
+	bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq;
+    trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4;
+    bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2;
+	double output = pow(10.0,((C*30.0)-15.0)/20.0);
+	
+	double K = tan(M_PI * trebleAL[0]);
+	double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K);
+	trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm;
+	trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2];
+	trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2];
+	trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm;
+	trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm;
+	
+	K = tan(M_PI * bassAL[0]);
+	norm = 1.0 / (1.0 + K / bassAL[1] + K * K);
+	bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm;
+	bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2];
+	bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2];
+	bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm;
+	bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[1] + K * K) * norm;
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
+		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
+		
+		if (output != 1.0) {
+			inputSampleL *= output;
+			inputSampleR *= output;
+		}//gain trim in front of plugin, in case Console stage clips
+		
+		inputSampleL = sin(inputSampleL);
+		inputSampleR = sin(inputSampleR);
+		//encode Console5: good cleanness
+		
+		long double trebleSampleL;
+		long double bassSampleL;
+		long double trebleSampleR;
+		long double bassSampleR;
+		
+		if (flip)
+		{
+			trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7];
+			trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8];
+			trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]);
+			trebleSampleL = inputSampleL - trebleSampleL;
+			
+			bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7];
+			bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8];
+			bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]);
+
+			trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7];
+			trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8];
+			trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]);
+			trebleSampleR = inputSampleR - trebleSampleR;
+			
+			bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7];
+			bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8];
+			bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]);
+		}
+		else
+		{
+			trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7];
+			trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8];
+			trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]);
+			trebleSampleL = inputSampleL - trebleSampleL;
+			
+			bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7];
+			bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8];
+			bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]);
+
+			trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7];
+			trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8];
+			trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]);
+			trebleSampleR = inputSampleR - trebleSampleR;
+			
+			bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7];
+			bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8];
+			bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]);
+		}
+		flip = !flip;
+		
+		trebleSampleL *= trebleGain;
+		bassSampleL *= bassGain;
+		inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
+		trebleSampleR *= trebleGain;
+		bassSampleR *= bassGain;
+		inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
+		
+		if (inputSampleL > 1.0) inputSampleL = 1.0;
+		if (inputSampleL < -1.0) inputSampleL = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleL = asin(inputSampleL);
+		//amplitude aspect
+
+		if (inputSampleR > 1.0) inputSampleR = 1.0;
+		if (inputSampleR < -1.0) inputSampleR = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleR = asin(inputSampleR);
+		//amplitude aspect
+		
+		//begin 32 bit stereo floating point dither
+		int expon; frexpf((float)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		frexpf((float)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
+		//end 32 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}
+
+void Baxandall::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+	
+	double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0);
+	double trebleFreq = (4410.0*trebleGain)/getSampleRate();
+	if (trebleFreq > 0.45) trebleFreq = 0.45;
+	trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq;
+	double bassGain = pow(10.0,((B*30.0)-15.0)/20.0);
+	double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0);
+	bassFreq = (4410.0*bassFreq)/getSampleRate();
+	if (bassFreq > 0.45) bassFreq = 0.45;
+	bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq;
+    trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4;
+    bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2;
+	double output = pow(10.0,((C*30.0)-15.0)/20.0);
+	
+	double K = tan(M_PI * trebleAL[0]);
+	double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K);
+	trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm;
+	trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2];
+	trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2];
+	trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm;
+	trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm;
+	
+	K = tan(M_PI * bassAL[0]);
+	norm = 1.0 / (1.0 + K / bassAL[1] + K * K);
+	bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm;
+	bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2];
+	bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2];
+	bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm;
+	bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[1] + K * K) * norm;
+    
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
+		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
+		
+		if (output != 1.0) {
+			inputSampleL *= output;
+			inputSampleR *= output;
+		}//gain trim in front of plugin, in case Console stage clips
+		
+		inputSampleL = sin(inputSampleL);
+		inputSampleR = sin(inputSampleR);
+		//encode Console5: good cleanness
+		
+		long double trebleSampleL;
+		long double bassSampleL;
+		long double trebleSampleR;
+		long double bassSampleR;
+		
+		if (flip)
+		{
+			trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7];
+			trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8];
+			trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]);
+			trebleSampleL = inputSampleL - trebleSampleL;
+			
+			bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7];
+			bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8];
+			bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]);
+			
+			trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7];
+			trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8];
+			trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]);
+			trebleSampleR = inputSampleR - trebleSampleR;
+			
+			bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7];
+			bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8];
+			bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]);
+		}
+		else
+		{
+			trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7];
+			trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8];
+			trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]);
+			trebleSampleL = inputSampleL - trebleSampleL;
+			
+			bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7];
+			bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8];
+			bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]);
+			
+			trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7];
+			trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8];
+			trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]);
+			trebleSampleR = inputSampleR - trebleSampleR;
+			
+			bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7];
+			bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8];
+			bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]);
+		}
+		flip = !flip;
+		
+		trebleSampleL *= trebleGain;
+		bassSampleL *= bassGain;
+		inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
+		trebleSampleR *= trebleGain;
+		bassSampleR *= bassGain;
+		inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
+		
+		if (inputSampleL > 1.0) inputSampleL = 1.0;
+		if (inputSampleL < -1.0) inputSampleL = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleL = asin(inputSampleL);
+		//amplitude aspect
+		
+		if (inputSampleR > 1.0) inputSampleR = 1.0;
+		if (inputSampleR < -1.0) inputSampleR = -1.0;
+		//without this, you can get a NaN condition where it spits out DC offset at full blast!
+		inputSampleR = asin(inputSampleR);
+		//amplitude aspect
+		
+		//begin 64 bit stereo floating point dither
+		int expon; frexp((double)inputSampleL, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		frexp((double)inputSampleR, &expon);
+		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
+		//end 64 bit stereo floating point dither
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+}