VoiceTrick

3 files changed, 412 insertions, 0 deletions

diff --git a/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.cpp b/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.cpp
new file mode 100755
index 0000000..ee066c8
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.cpp
@@ -0,0 +1,128 @@
+/* ========================================
+ *  VoiceTrick - VoiceTrick.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceTrick_H
+#include "VoiceTrick.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new VoiceTrick(audioMaster);}
+
+VoiceTrick::VoiceTrick(audioMasterCallback audioMaster) :
+    AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+	A = 1.0;
+	iirLowpassA = 0.0;
+	iirLowpassB = 0.0;
+	iirLowpassC = 0.0;
+	iirLowpassD = 0.0;
+	iirLowpassE = 0.0;
+	iirLowpassF = 0.0;
+	count = 0;
+	lowpassChase = 0.0;
+	lowpassAmount = 1.0;
+	lastLowpass = 1000.0;
+	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
+}
+
+VoiceTrick::~VoiceTrick() {}
+VstInt32 VoiceTrick::getVendorVersion () {return 1000;}
+void VoiceTrick::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void VoiceTrick::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 VoiceTrick::getChunk (void** data, bool isPreset)
+{
+	float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
+	chunkData[0] = A;
+	/* 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 VoiceTrick::setChunk (void* data, VstInt32 byteSize, bool isPreset)
+{	
+	float *chunkData = (float *)data;
+	A = pinParameter(chunkData[0]);
+	/* 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 VoiceTrick::setParameter(VstInt32 index, float value) {
+    switch (index) {
+        case kParamA: A = value; break;
+        default: throw; // unknown parameter, shouldn't happen!
+    }
+}
+
+float VoiceTrick::getParameter(VstInt32 index) {
+    switch (index) {
+        case kParamA: return A; 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 VoiceTrick::getParameterName(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "Tone", kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+    } //this is our labels for displaying in the VST host
+}
+
+void VoiceTrick::getParameterDisplay(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: float2string (A, text, kVstMaxParamStrLen); break;
+        default: break; // unknown parameter, shouldn't happen!
+	} //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void VoiceTrick::getParameterLabel(VstInt32 index, char *text) {
+    switch (index) {
+        case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break;
+		default: break; // unknown parameter, shouldn't happen!
+    }
+}
+
+VstInt32 VoiceTrick::canDo(char *text) 
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool VoiceTrick::getEffectName(char* name) {
+    vst_strncpy(name, "VoiceTrick", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory VoiceTrick::getPlugCategory() {return kPlugCategEffect;}
+
+bool VoiceTrick::getProductString(char* text) {
+  	vst_strncpy (text, "airwindows VoiceTrick", kVstMaxProductStrLen); return true;
+}
+
+bool VoiceTrick::getVendorString(char* text) {
+  	vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.h b/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.h
new file mode 100755
index 0000000..64c4b43
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceTrick/VoiceTrick.h
@@ -0,0 +1,70 @@
+/* ========================================
+ *  VoiceTrick - VoiceTrick.h
+ *  Created 8/12/11 by SPIAdmin 
+ *  Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceTrick_H
+#define __VoiceTrick_H
+
+#ifndef __audioeffect__
+#include "audioeffectx.h"
+#endif
+
+#include <set>
+#include <string>
+#include <math.h>
+
+enum {
+	kParamA = 0,
+  kNumParameters = 1
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'vtrk';    //Change this to what the AU identity is!
+
+class VoiceTrick : 
+    public AudioEffectX 
+{
+public:
+    VoiceTrick(audioMasterCallback audioMaster);
+    ~VoiceTrick();
+    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
+	double iirLowpassA;
+	double iirLowpassB;
+	double iirLowpassC;
+	double iirLowpassD;
+	double iirLowpassE;
+	double iirLowpassF;
+	int count;
+	double lowpassChase;
+	double lowpassAmount;
+	double lastLowpass;	
+    float A;
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/VoiceTrick/VoiceTrickProc.cpp b/plugins/LinuxVST/src/VoiceTrick/VoiceTrickProc.cpp
new file mode 100755
index 0000000..b57b695
--- /dev/null
+++ b/plugins/LinuxVST/src/VoiceTrick/VoiceTrickProc.cpp
@@ -0,0 +1,214 @@
+/* ========================================
+ *  VoiceTrick - VoiceTrick.h
+ *  Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __VoiceTrick_H
+#include "VoiceTrick.h"
+#endif
+
+void VoiceTrick::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
+{
+    float* in1  =  inputs[0];
+    float* in2  =  inputs[1];
+    float* out1 = outputs[0];
+    float* out2 = outputs[1];
+
+	lowpassChase = pow(A,2);
+	//should not scale with sample rate, because values reaching 1 are important
+	//to its ability to bypass when set to max
+	double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
+	lastLowpass = lowpassChase;	
+	double invLowpass;
+    
+    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;
+		
+		lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
+		//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
+		//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
+		
+		long double inputSample = (inputSampleL + inputSampleR) * 0.5;
+		//this is now our mono audio
+		
+		count++; if (count > 5) count = 0; switch (count)
+		{
+			case 0:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
+				break;
+			case 1:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
+				break;
+			case 2:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
+				break;
+			case 3:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
+				break;
+			case 4:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
+				break;
+			case 5:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
+				break;
+		}
+		//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
+		//steepens the filter after minimizing artifacts.
+		
+		
+		inputSampleL = -inputSample;
+		inputSampleR = inputSample;
+		
+		//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
+		//The purpose of all this is to allow for recording of lead vocals without use of headphones:
+		//or at least sealed headphones. You should be able to use this to record vocals with either
+		//open-back headphones, or literally speakers in the room so long as the mic is exactly
+		//equidistant from each speaker/headphone side.
+		
+		//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
+		//only reverb and echo for vibe. Direct sound is the singer's direct sound.
+		
+		//The filtering is because, if you use open-back headphones and move your head, highs will
+		//bleed through first like a through-zero flange coming out of cancellation (which it is).
+		//Therefore, you can filter off highs until the bleed isn't annoying.
+		//Or just run with it, it shouldn't be that loud.
+		
+		//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
+		
+		//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 VoiceTrick::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
+{
+    double* in1  =  inputs[0];
+    double* in2  =  inputs[1];
+    double* out1 = outputs[0];
+    double* out2 = outputs[1];
+
+	lowpassChase = pow(A,2);
+	//should not scale with sample rate, because values reaching 1 are important
+	//to its ability to bypass when set to max
+	double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
+	lastLowpass = lowpassChase;	
+	double invLowpass;
+	
+    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;
+		
+		lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
+		//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
+		//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
+		
+		long double inputSample = (inputSampleL + inputSampleR) * 0.5;
+		//this is now our mono audio
+		
+		count++; if (count > 5) count = 0; switch (count)
+		{
+			case 0:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
+				break;
+			case 1:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
+				break;
+			case 2:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
+				break;
+			case 3:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
+				break;
+			case 4:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
+				iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
+				break;
+			case 5:
+				iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
+				iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
+				iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
+				break;
+		}
+		//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
+		//steepens the filter after minimizing artifacts.
+		
+		
+		inputSampleL = -inputSample;
+		inputSampleR = inputSample;
+		
+		//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
+		//The purpose of all this is to allow for recording of lead vocals without use of headphones:
+		//or at least sealed headphones. You should be able to use this to record vocals with either
+		//open-back headphones, or literally speakers in the room so long as the mic is exactly
+		//equidistant from each speaker/headphone side.
+		
+		//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
+		//only reverb and echo for vibe. Direct sound is the singer's direct sound.
+		
+		//The filtering is because, if you use open-back headphones and move your head, highs will
+		//bleed through first like a through-zero flange coming out of cancellation (which it is).
+		//Therefore, you can filter off highs until the bleed isn't annoying.
+		//Or just run with it, it shouldn't be that loud.
+		
+		//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
+		
+		//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++;
+    }
+}