Spiral2

3 files changed, 527 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/Spiral2/Spiral2.cpp b/plugins/LinuxVST/src/Spiral2/Spiral2.cpp
new file mode 100755
index 0000000..29b3c61
--- /dev/null
+++ b/plugins/LinuxVST/src/Spiral2/Spiral2.cpp
@@ -0,0 +1,159 @@
+/* ========================================
+ *  Spiral2 - Spiral2.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Spiral2_H
+#include "Spiral2.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Spiral2(audioMaster);}
+
+Spiral2::Spiral2(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 0.5;
+	B = 0.0;
+	C = 0.5;
+	D = 1.0;
+	E = 1.0;
+	iirSampleAL = 0.0;
+	iirSampleBL = 0.0;
+	prevSampleL = 0.0;
+	fpNShapeL = 0.0;
+
+	iirSampleAR = 0.0;
+	iirSampleBR = 0.0;
+	prevSampleR = 0.0;
+	fpNShapeR = 0.0;
+	flip = 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
+}
+
+Spiral2::~Spiral2() {}
+VstInt32 Spiral2::getVendorVersion () {return 1000;}
+void Spiral2::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Spiral2::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 Spiral2::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	chunkData[1] = B;
+	chunkData[2] = C;
+	chunkData[3] = D;
+	chunkData[4] = E;
+	/* 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 Spiral2::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	B = pinParameter(chunkData[1]);
+	C = pinParameter(chunkData[2]);
+	D = pinParameter(chunkData[3]);
+	E = pinParameter(chunkData[4]);
+	/* 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 Spiral2::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        case kParamB: B = value; break;
+        case kParamC: C = value; break;
+        case kParamD: D = value; break;
+        case kParamE: E = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float Spiral2::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; break;
+        case kParamB: return B; break;
+        case kParamC: return C; break;
+        case kParamD: return D; break;
+        case kParamE: return E; 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 Spiral2::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Input", kVstMaxParamStrLen); break;
+		case kParamB: vst_strncpy (text, "Highpass", kVstMaxParamStrLen); break;
+		case kParamC: vst_strncpy (text, "Presence", kVstMaxParamStrLen); break;
+		case kParamD: vst_strncpy (text, "Output", kVstMaxParamStrLen); break;
+		case kParamE: 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 Spiral2::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        case kParamB: float2string (B, text, kVstMaxParamStrLen); break;
+        case kParamC: float2string (C, text, kVstMaxParamStrLen); break;
+        case kParamD: float2string (D, text, kVstMaxParamStrLen); break;
+        case kParamE: float2string (E, text, kVstMaxParamStrLen); break;			
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Spiral2::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+        case kParamE: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 Spiral2::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Spiral2::getEffectName(char* name) {
+    vst_strncpy(name, "Spiral2", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Spiral2::getPlugCategory() {return kPlugCategEffect;}
+
+bool Spiral2::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows Spiral2", kVstMaxProductStrLen); return true;
+}
+
+bool Spiral2::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Spiral2/Spiral2.h b/plugins/LinuxVST/src/Spiral2/Spiral2.h
new file mode 100755
index 0000000..e918870
--- /dev/null
+++ b/plugins/LinuxVST/src/Spiral2/Spiral2.h
@@ -0,0 +1,79 @@
+/* ========================================
+ *  Spiral2 - Spiral2.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Spiral2_H
+#define __Spiral2_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+	kParamB = 1,
+	kParamC = 2,
+	kParamD = 3,
+	kParamE = 4,
+  kNumParameters = 5
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'spis';    //Change this to what the AU identity is!
+
+class Spiral2 : 
+    public AudioEffectX 
+{
+public:
+    Spiral2(audioMasterCallback audioMaster);
+    ~Spiral2();
+    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 iirSampleAL;
+	long double iirSampleBL;
+	long double prevSampleL;
+	long double fpNShapeL;
+
+	long double iirSampleAR;
+	long double iirSampleBR;
+	long double prevSampleR;
+	long double fpNShapeR;
+	bool flip;
+	//default stuff
+
+    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/Spiral2/Spiral2Proc.cpp b/plugins/LinuxVST/src/Spiral2/Spiral2Proc.cpp
new file mode 100755
index 0000000..d309448
--- /dev/null
+++ b/plugins/LinuxVST/src/Spiral2/Spiral2Proc.cpp
@@ -0,0 +1,289 @@
+/* ========================================
+ *  Spiral2 - Spiral2.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Spiral2_H
+#include "Spiral2.h"
+#endif
+
+void Spiral2::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 gain = pow(A*2.0,2.0);
+	double iirAmount = pow(B,3.0)/overallscale;
+	double presence = C;
+	double output = D;
+	double wet = E;
+
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it aSpiral2. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		if (gain != 1.0) {
+			inputSampleL *= gain;
+			inputSampleR *= gain;
+			prevSampleL *= gain;
+			prevSampleR *= gain;
+		}
+		
+		if (flip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (inputSampleL * iirAmount);
+			iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (inputSampleR * iirAmount);
+			inputSampleL -= iirSampleAL;
+			inputSampleR -= iirSampleAR;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (inputSampleL * iirAmount);
+			iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (inputSampleR * iirAmount);
+			inputSampleL -= iirSampleBL;
+			inputSampleR -= iirSampleBR;
+		}
+		//highpass section
+		
+		long double presenceSampleL = sin(inputSampleL * fabs(prevSampleL)) / ((prevSampleL == 0.0) ?1:fabs(prevSampleL));
+		long double presenceSampleR = sin(inputSampleR * fabs(prevSampleR)) / ((prevSampleR == 0.0) ?1:fabs(prevSampleR));
+		//change from first Spiral: delay of one sample on the scaling factor.
+		inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
+		inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
+		
+		if (output < 1.0) {
+			inputSampleL *= output;
+			inputSampleR *= output;
+			presenceSampleL *= output;
+			presenceSampleR *= output;
+		}
+		if (presence > 0.0) {
+			inputSampleL = (inputSampleL * (1.0-presence)) + (presenceSampleL * presence);
+			inputSampleR = (inputSampleR * (1.0-presence)) + (presenceSampleR * presence);
+		}
+		if (wet < 1.0) {
+			inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
+			inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
+		}
+		//nice little output stage template: if we have another scale of floating point
+		//number, we really don't want to meaninglessly multiply that by 1.0.
+		
+		prevSampleL = drySampleL;
+		prevSampleR = drySampleR;
+		flip = !flip;
+		
+		//noise shaping to 32-bit floating point
+		float fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 32 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}
+
+void Spiral2::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 gain = pow(A*2.0,2.0);
+	double iirAmount = pow(B,3.0)/overallscale;
+	double presence = C;
+	double output = D;
+	double wet = E;
+	
+	
+    while (--sampleFrames >= 0)
+    {
+		long double inputSampleL = *in1;
+		long double inputSampleR = *in2;
+
+		static int noisesourceL = 0;
+		static int noisesourceR = 850010;
+		int residue;
+		double applyresidue;
+		
+		noisesourceL = noisesourceL % 1700021; noisesourceL++;
+		residue = noisesourceL * noisesourceL;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleL += applyresidue;
+		if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
+			inputSampleL -= applyresidue;
+		}
+		
+		noisesourceR = noisesourceR % 1700021; noisesourceR++;
+		residue = noisesourceR * noisesourceR;
+		residue = residue % 170003; residue *= residue;
+		residue = residue % 17011; residue *= residue;
+		residue = residue % 1709; residue *= residue;
+		residue = residue % 173; residue *= residue;
+		residue = residue % 17;
+		applyresidue = residue;
+		applyresidue *= 0.00000001;
+		applyresidue *= 0.00000001;
+		inputSampleR += applyresidue;
+		if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
+			inputSampleR -= applyresidue;
+		}
+		//for live air, we always apply the dither noise. Then, if our result is 
+		//effectively digital black, we'll subtract it aSpiral2. We want a 'air' hiss
+		long double drySampleL = inputSampleL;
+		long double drySampleR = inputSampleR;
+		
+		if (gain != 1.0) {
+			inputSampleL *= gain;
+			inputSampleR *= gain;
+			prevSampleL *= gain;
+			prevSampleR *= gain;
+		}
+		
+		if (flip)
+		{
+			iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (inputSampleL * iirAmount);
+			iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (inputSampleR * iirAmount);
+			inputSampleL -= iirSampleAL;
+			inputSampleR -= iirSampleAR;
+		}
+		else
+		{
+			iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (inputSampleL * iirAmount);
+			iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (inputSampleR * iirAmount);
+			inputSampleL -= iirSampleBL;
+			inputSampleR -= iirSampleBR;
+		}
+		//highpass section
+		
+		long double presenceSampleL = sin(inputSampleL * fabs(prevSampleL)) / ((prevSampleL == 0.0) ?1:fabs(prevSampleL));
+		long double presenceSampleR = sin(inputSampleR * fabs(prevSampleR)) / ((prevSampleR == 0.0) ?1:fabs(prevSampleR));
+		//change from first Spiral: delay of one sample on the scaling factor.
+		inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
+		inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
+		
+		if (output < 1.0) {
+			inputSampleL *= output;
+			inputSampleR *= output;
+			presenceSampleL *= output;
+			presenceSampleR *= output;
+		}
+		if (presence > 0.0) {
+			inputSampleL = (inputSampleL * (1.0-presence)) + (presenceSampleL * presence);
+			inputSampleR = (inputSampleR * (1.0-presence)) + (presenceSampleR * presence);
+		}
+		if (wet < 1.0) {
+			inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
+			inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
+		}
+		//nice little output stage template: if we have another scale of floating point
+		//number, we really don't want to meaninglessly multiply that by 1.0.
+		
+		prevSampleL = drySampleL;
+		prevSampleR = drySampleR;
+		flip = !flip;
+		
+		//noise shaping to 64-bit floating point
+		double fpTemp = inputSampleL;
+		fpNShapeL += (inputSampleL-fpTemp);
+		inputSampleL += fpNShapeL;
+		//if this confuses you look at the wordlength for fpTemp :)
+		fpTemp = inputSampleR;
+		fpNShapeR += (inputSampleR-fpTemp);
+		inputSampleR += fpNShapeR;
+		//for deeper space and warmth, we try a non-oscillating noise shaping
+		//that is kind of ruthless: it will forever retain the rounding errors
+		//except we'll dial it back a hair at the end of every buffer processed
+		//end noise shaping on 64 bit output
+		
+		*out1 = inputSampleL;
+		*out2 = inputSampleR;
+
+		*in1++;
+		*in2++;
+		*out1++;
+		*out2++;
+    }
+	fpNShapeL *= 0.999999;
+	fpNShapeR *= 0.999999;
+	//we will just delicately dial back the FP noise shaping, not even every sample
+	//this is a good place to put subtle 'no runaway' calculations, though bear in mind
+	//that it will be called more often when you use shorter sample buffers in the DAW.
+	//So, very low latency operation will call these calculations more often.	
+}