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Stonefire

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This commit is contained in:
Christopher Johnson 2024-04-06 19:54:41 -04:00
parent ca7a8d222b
commit d8efeb53eb
230 changed files with 30100 additions and 19389 deletions
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plugins/WinVST/Stonefire/.vs/Console4Channel64/v14/.suo Executable file
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plugins/WinVST/Stonefire/.vs/VSTProject/v14/.suo Executable file
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plugins/WinVST/Stonefire/Stonefire.cpp Executable file
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/* ========================================
* Stonefire - Stonefire.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Stonefire_H
#include "Stonefire.h"
#endif
AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Stonefire(audioMaster);}
Stonefire::Stonefire(audioMasterCallback audioMaster) :
AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
{
A = 0.5;
B = 0.5;
C = 0.5;
D = 0.5;
for (int x = 0; x < air_total; x++) air[x] = 0.0;
for (int x = 0; x < kal_total; x++) kal[x] = 0.0;
trebleGainA = 1.0; trebleGainB = 1.0;
midGainA = 1.0; midGainB = 1.0;
bassGainA = 1.0; bassGainB = 1.0;
fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
//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
}
Stonefire::~Stonefire() {}
VstInt32 Stonefire::getVendorVersion () {return 1000;}
void Stonefire::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
void Stonefire::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 Stonefire::getChunk (void** data, bool isPreset)
{
float *chunkData = (float *)calloc(kNumParameters, sizeof(float));
chunkData[0] = A;
chunkData[1] = B;
chunkData[2] = C;
chunkData[3] = D;
/* 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 Stonefire::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]);
/* 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 Stonefire::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;
default: throw; // unknown parameter, shouldn't happen!
}
}
float Stonefire::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;
default: break; // unknown parameter, shouldn't happen!
} return 0.0; //we only need to update the relevant name, this is simple to manage
}
void Stonefire::getParameterName(VstInt32 index, char *text) {
switch (index) {
case kParamA: vst_strncpy (text, "Air", kVstMaxParamStrLen); break;
case kParamB: vst_strncpy (text, "Fire", kVstMaxParamStrLen); break;
case kParamC: vst_strncpy (text, "Stone", kVstMaxParamStrLen); break;
case kParamD: vst_strncpy (text, "Range", kVstMaxParamStrLen); break;
default: break; // unknown parameter, shouldn't happen!
} //this is our labels for displaying in the VST host
}
void Stonefire::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;
default: break; // unknown parameter, shouldn't happen!
} //this displays the values and handles 'popups' where it's discrete choices
}
void Stonefire::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;
default: break; // unknown parameter, shouldn't happen!
}
}
VstInt32 Stonefire::canDo(char *text)
{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
bool Stonefire::getEffectName(char* name) {
vst_strncpy(name, "Stonefire", kVstMaxProductStrLen); return true;
}
VstPlugCategory Stonefire::getPlugCategory() {return kPlugCategEffect;}
bool Stonefire::getProductString(char* text) {
vst_strncpy (text, "airwindows Stonefire", kVstMaxProductStrLen); return true;
}
bool Stonefire::getVendorString(char* text) {
vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
}

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plugins/WinVST/Stonefire/Stonefire.h Executable file
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/* ========================================
* Stonefire - Stonefire.h
* Created 8/12/11 by SPIAdmin
* Copyright (c) Airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Stonefire_H
#define __Stonefire_H
#ifndef __audioeffect__
#include "audioeffectx.h"
#endif
#include <set>
#include <string>
#include <math.h>
enum {
kParamA = 0,
kParamB = 1,
kParamC = 2,
kParamD = 3,
kNumParameters = 4
}; //
const int kNumPrograms = 0;
const int kNumInputs = 2;
const int kNumOutputs = 2;
const unsigned long kUniqueId = 'stfr'; //Change this to what the AU identity is!
class Stonefire :
public AudioEffectX
{
public:
Stonefire(audioMasterCallback audioMaster);
~Stonefire();
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 fpdL;
uint32_t fpdR;
//default stuff
enum {
pvAL1,
pvSL1,
accSL1,
acc2SL1,
pvAL2,
pvSL2,
accSL2,
acc2SL2,
pvAL3,
pvSL3,
accSL3,
pvAL4,
pvSL4,
gndavgL,
outAL,
gainAL,
pvAR1,
pvSR1,
accSR1,
acc2SR1,
pvAR2,
pvSR2,
accSR2,
acc2SR2,
pvAR3,
pvSR3,
accSR3,
pvAR4,
pvSR4,
gndavgR,
outAR,
gainAR,
air_total
};
double air[air_total];
enum {
prevSampL1,
prevSlewL1,
accSlewL1,
prevSampL2,
prevSlewL2,
accSlewL2,
prevSampL3,
prevSlewL3,
accSlewL3,
kalGainL,
kalOutL,
prevSampR1,
prevSlewR1,
accSlewR1,
prevSampR2,
prevSlewR2,
accSlewR2,
prevSampR3,
prevSlewR3,
accSlewR3,
kalGainR,
kalOutR,
kal_total
};
double kal[kal_total];
double trebleGainA;
double trebleGainB;
double midGainA;
double midGainB;
double bassGainA;
double bassGainB;
float A;
float B;
float C;
float D;
};
#endif

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plugins/WinVST/Stonefire/StonefireProc.cpp Executable file
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/* ========================================
* Stonefire - Stonefire.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Stonefire_H
#include "Stonefire.h"
#endif
void Stonefire::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
trebleGainA = trebleGainB; trebleGainB = A*2.0;
midGainA = midGainB; midGainB = B*2.0;
bassGainA = bassGainB; bassGainB = C*2.0;
//simple three band to adjust
double kalman = 1.0-pow(D,2);
//crossover frequency between mid/bass
while (--sampleFrames >= 0)
{
double inputSampleL = *in1;
double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
double drySampleL = inputSampleL;
double drySampleR = inputSampleR;
double temp = (double)sampleFrames/inFramesToProcess;
double trebleGain = (trebleGainA*temp)+(trebleGainB*(1.0-temp));
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
if (trebleGain < 1.0) trebleGain = 1.0-pow(1.0-trebleGain,2);
double midGain = (midGainA*temp)+(midGainB*(1.0-temp));
if (midGain > 1.0) midGain *= midGain;
if (midGain < 1.0) midGain = 1.0-pow(1.0-midGain,2);
double bassGain = (bassGainA*temp)+(bassGainB*(1.0-temp));
if (bassGain > 1.0) bassGain *= bassGain;
if (bassGain < 1.0) bassGain = 1.0-pow(1.0-bassGain,2);
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
double trebleL = drySampleL-midL;
inputSampleL = midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
double trebleR = drySampleR-midR;
inputSampleR = midR;
//end Air3R
//begin KalmanL
temp = inputSampleL = inputSampleL*(1.0-kalman)*0.777;
inputSampleL *= (1.0-kalman);
//set up gain levels to control the beast
kal[prevSlewL3] += kal[prevSampL3] - kal[prevSampL2]; kal[prevSlewL3] *= 0.5;
kal[prevSlewL2] += kal[prevSampL2] - kal[prevSampL1]; kal[prevSlewL2] *= 0.5;
kal[prevSlewL1] += kal[prevSampL1] - inputSampleL; kal[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kal[accSlewL2] += kal[prevSlewL3] - kal[prevSlewL2]; kal[accSlewL2] *= 0.5;
kal[accSlewL1] += kal[prevSlewL2] - kal[prevSlewL1]; kal[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kal[accSlewL3] += (kal[accSlewL2] - kal[accSlewL1]); kal[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kal[kalOutL] += kal[prevSampL1] + kal[prevSlewL2] + kal[accSlewL3]; kal[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kal[kalGainL] += fabs(temp-kal[kalOutL])*kalman*8.0; kal[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kal[kalGainL] > kalman*0.5) kal[kalGainL] = kalman*0.5;
//attempts to avoid explosions
kal[kalOutL] += (temp*(1.0-(0.68+(kalman*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kal[prevSampL3] = kal[prevSampL2]; kal[prevSampL2] = kal[prevSampL1];
kal[prevSampL1] = (kal[kalGainL] * kal[kalOutL]) + ((1.0-kal[kalGainL])*temp);
//feed the chain of previous samples
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0; if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
double bassL = kal[kalOutL]*0.777;
midL -= bassL;
//end KalmanL
//begin KalmanR
temp = inputSampleR = inputSampleR*(1.0-kalman)*0.777;
inputSampleR *= (1.0-kalman);
//set up gain levels to control the beast
kal[prevSlewR3] += kal[prevSampR3] - kal[prevSampR2]; kal[prevSlewR3] *= 0.5;
kal[prevSlewR2] += kal[prevSampR2] - kal[prevSampR1]; kal[prevSlewR2] *= 0.5;
kal[prevSlewR1] += kal[prevSampR1] - inputSampleR; kal[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kal[accSlewR2] += kal[prevSlewR3] - kal[prevSlewR2]; kal[accSlewR2] *= 0.5;
kal[accSlewR1] += kal[prevSlewR2] - kal[prevSlewR1]; kal[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kal[accSlewR3] += (kal[accSlewR2] - kal[accSlewR1]); kal[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kal[kalOutR] += kal[prevSampR1] + kal[prevSlewR2] + kal[accSlewR3]; kal[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kal[kalGainR] += fabs(temp-kal[kalOutR])*kalman*8.0; kal[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kal[kalGainR] > kalman*0.5) kal[kalGainR] = kalman*0.5;
//attempts to avoid explosions
kal[kalOutR] += (temp*(1.0-(0.68+(kalman*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kal[prevSampR3] = kal[prevSampR2]; kal[prevSampR2] = kal[prevSampR1];
kal[prevSampR1] = (kal[kalGainR] * kal[kalOutR]) + ((1.0-kal[kalGainR])*temp);
//feed the chain of previous samples
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0; if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
double bassR = kal[kalOutR]*0.777;
midR -= bassR;
//end KalmanR
inputSampleL = (bassL*bassGain) + (midL*midGain) + (trebleL*trebleGain);
inputSampleR = (bassR*bassGain) + (midR*midGain) + (trebleR*trebleGain);
//applies pan section, and smoothed fader gain
//begin 32 bit stereo floating point dither
int expon; frexpf((float)inputSampleL, &expon);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
frexpf((float)inputSampleR, &expon);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
inputSampleR += ((double(fpdR)-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 Stonefire::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
trebleGainA = trebleGainB; trebleGainB = A*2.0;
midGainA = midGainB; midGainB = B*2.0;
bassGainA = bassGainB; bassGainB = C*2.0;
//simple three band to adjust
double kalman = 1.0-pow(D,2);
//crossover frequency between mid/bass
while (--sampleFrames >= 0)
{
double inputSampleL = *in1;
double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
double drySampleL = inputSampleL;
double drySampleR = inputSampleR;
double temp = (double)sampleFrames/inFramesToProcess;
double trebleGain = (trebleGainA*temp)+(trebleGainB*(1.0-temp));
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
if (trebleGain < 1.0) trebleGain = 1.0-pow(1.0-trebleGain,2);
double midGain = (midGainA*temp)+(midGainB*(1.0-temp));
if (midGain > 1.0) midGain *= midGain;
if (midGain < 1.0) midGain = 1.0-pow(1.0-midGain,2);
double bassGain = (bassGainA*temp)+(bassGainB*(1.0-temp));
if (bassGain > 1.0) bassGain *= bassGain;
if (bassGain < 1.0) bassGain = 1.0-pow(1.0-bassGain,2);
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
double trebleL = drySampleL-midL;
inputSampleL = midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
double trebleR = drySampleR-midR;
inputSampleR = midR;
//end Air3R
//begin KalmanL
temp = inputSampleL = inputSampleL*(1.0-kalman)*0.777;
inputSampleL *= (1.0-kalman);
//set up gain levels to control the beast
kal[prevSlewL3] += kal[prevSampL3] - kal[prevSampL2]; kal[prevSlewL3] *= 0.5;
kal[prevSlewL2] += kal[prevSampL2] - kal[prevSampL1]; kal[prevSlewL2] *= 0.5;
kal[prevSlewL1] += kal[prevSampL1] - inputSampleL; kal[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kal[accSlewL2] += kal[prevSlewL3] - kal[prevSlewL2]; kal[accSlewL2] *= 0.5;
kal[accSlewL1] += kal[prevSlewL2] - kal[prevSlewL1]; kal[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kal[accSlewL3] += (kal[accSlewL2] - kal[accSlewL1]); kal[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kal[kalOutL] += kal[prevSampL1] + kal[prevSlewL2] + kal[accSlewL3]; kal[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kal[kalGainL] += fabs(temp-kal[kalOutL])*kalman*8.0; kal[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kal[kalGainL] > kalman*0.5) kal[kalGainL] = kalman*0.5;
//attempts to avoid explosions
kal[kalOutL] += (temp*(1.0-(0.68+(kalman*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kal[prevSampL3] = kal[prevSampL2]; kal[prevSampL2] = kal[prevSampL1];
kal[prevSampL1] = (kal[kalGainL] * kal[kalOutL]) + ((1.0-kal[kalGainL])*temp);
//feed the chain of previous samples
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0; if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
double bassL = kal[kalOutL]*0.777;
midL -= bassL;
//end KalmanL
//begin KalmanR
temp = inputSampleR = inputSampleR*(1.0-kalman)*0.777;
inputSampleR *= (1.0-kalman);
//set up gain levels to control the beast
kal[prevSlewR3] += kal[prevSampR3] - kal[prevSampR2]; kal[prevSlewR3] *= 0.5;
kal[prevSlewR2] += kal[prevSampR2] - kal[prevSampR1]; kal[prevSlewR2] *= 0.5;
kal[prevSlewR1] += kal[prevSampR1] - inputSampleR; kal[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kal[accSlewR2] += kal[prevSlewR3] - kal[prevSlewR2]; kal[accSlewR2] *= 0.5;
kal[accSlewR1] += kal[prevSlewR2] - kal[prevSlewR1]; kal[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kal[accSlewR3] += (kal[accSlewR2] - kal[accSlewR1]); kal[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kal[kalOutR] += kal[prevSampR1] + kal[prevSlewR2] + kal[accSlewR3]; kal[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kal[kalGainR] += fabs(temp-kal[kalOutR])*kalman*8.0; kal[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kal[kalGainR] > kalman*0.5) kal[kalGainR] = kalman*0.5;
//attempts to avoid explosions
kal[kalOutR] += (temp*(1.0-(0.68+(kalman*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kal[prevSampR3] = kal[prevSampR2]; kal[prevSampR2] = kal[prevSampR1];
kal[prevSampR1] = (kal[kalGainR] * kal[kalOutR]) + ((1.0-kal[kalGainR])*temp);
//feed the chain of previous samples
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0; if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
double bassR = kal[kalOutR]*0.777;
midR -= bassR;
//end KalmanR
inputSampleL = (bassL*bassGain) + (midL*midGain) + (trebleL*trebleGain);
inputSampleR = (bassR*bassGain) + (midR*midGain) + (trebleR*trebleGain);
//applies pan section, and smoothed fader gain
//begin 64 bit stereo floating point dither
//int expon; frexp((double)inputSampleL, &expon);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
//frexp((double)inputSampleR, &expon);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
//inputSampleR += ((double(fpdR)-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++;
}
}

28
plugins/WinVST/Stonefire/VSTProject.sln Executable file
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Microsoft Visual Studio Solution File, Format Version 12.00
# Visual Studio 14
VisualStudioVersion = 14.0.25420.1
MinimumVisualStudioVersion = 10.0.40219.1
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "VSTProject", "VSTProject.vcxproj", "{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}"
EndProject
Global
GlobalSection(SolutionConfigurationPlatforms) = preSolution
Debug|x64 = Debug|x64
Debug|x86 = Debug|x86
Release|x64 = Release|x64
Release|x86 = Release|x86
EndGlobalSection
GlobalSection(ProjectConfigurationPlatforms) = postSolution
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x64.ActiveCfg = Debug|x64
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x64.Build.0 = Debug|x64
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x86.ActiveCfg = Debug|Win32
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x86.Build.0 = Debug|Win32
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x64.ActiveCfg = Release|x64
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x64.Build.0 = Release|x64
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x86.ActiveCfg = Release|Win32
{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x86.Build.0 = Release|Win32
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE
EndGlobalSection
EndGlobal

183
plugins/WinVST/Stonefire/VSTProject.vcxproj Executable file
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<?xml version="1.0" encoding="utf-8"?>
<Project DefaultTargets="Build" ToolsVersion="14.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
<ItemGroup Label="ProjectConfigurations">
<ProjectConfiguration Include="Debug|Win32">
<Configuration>Debug</Configuration>
<Platform>Win32</Platform>
</ProjectConfiguration>
<ProjectConfiguration Include="Release|Win32">
<Configuration>Release</Configuration>
<Platform>Win32</Platform>
</ProjectConfiguration>
<ProjectConfiguration Include="Debug|x64">
<Configuration>Debug</Configuration>
<Platform>x64</Platform>
</ProjectConfiguration>
<ProjectConfiguration Include="Release|x64">
<Configuration>Release</Configuration>
<Platform>x64</Platform>
</ProjectConfiguration>
</ItemGroup>
<ItemGroup>
<ClCompile Include="..\..\..\vstsdk2.4\public.sdk\source\vst2.x\audioeffect.cpp" />
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<ClCompile Include="Stonefire.cpp" />
<ClCompile Include="StonefireProc.cpp" />
</ItemGroup>
<ItemGroup>
<ClInclude Include="..\..\..\vstsdk2.4\public.sdk\source\vst2.x\aeffeditor.h" />
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<ClInclude Include="..\..\..\vstsdk2.4\public.sdk\source\vst2.x\audioeffectx.h" />
<ClInclude Include="Stonefire.h" />
</ItemGroup>
<PropertyGroup Label="Globals">
<ProjectGuid>{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}</ProjectGuid>
<RootNamespace>VSTProject</RootNamespace>
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<ProjectName>Stonefire64</ProjectName>
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<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
<ClCompile>
<WarningLevel>Level3</WarningLevel>
<Optimization>MaxSpeed</Optimization>
<SDLCheck>true</SDLCheck>
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<RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
<FavorSizeOrSpeed>Speed</FavorSizeOrSpeed>
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<BasicRuntimeChecks>Default</BasicRuntimeChecks>
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<DebugInformationFormat>None</DebugInformationFormat>
</ClCompile>
<Link>
<ModuleDefinitionFile>vstplug.def</ModuleDefinitionFile>
<IgnoreSpecificDefaultLibraries>libcmt.dll;libcmtd.dll;msvcrt.lib;%(IgnoreSpecificDefaultLibraries)</IgnoreSpecificDefaultLibraries>
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48
plugins/WinVST/Stonefire/VSTProject.vcxproj.filters Executable file
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@ -0,0 +1,48 @@
<?xml version="1.0" encoding="utf-8"?>
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<Extensions>h;hh;hpp;hxx;hm;inl;inc;xsd</Extensions>
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<Filter>Source Files</Filter>
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<ItemGroup>
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<Filter>Header Files</Filter>
</ClInclude>
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<Filter>Header Files</Filter>
</ClInclude>
<ClInclude Include="..\..\..\vstsdk2.4\public.sdk\source\vst2.x\audioeffectx.h">
<Filter>Header Files</Filter>
</ClInclude>
<ClInclude Include="Stonefire.h">
<Filter>Header Files</Filter>
</ClInclude>
</ItemGroup>
</Project>

19
plugins/WinVST/Stonefire/VSTProject.vcxproj.user Executable file
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@ -0,0 +1,19 @@
<?xml version="1.0" encoding="utf-8"?>
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3
plugins/WinVST/Stonefire/vstplug.def Executable file
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EXPORTS
VSTPluginMain
main=VSTPluginMain