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author | Chris Johnson <jinx6568@sover.net> | 2018-09-09 22:59:35 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-09-09 22:59:35 -0400 |
commit | 9ecd9a8c01bea3c02f6e94cf67dd7c4ad12d1e34 (patch) | |
tree | a93e71197b6adf49d6abc74ae055d7463c3c0597 /plugins/WinVST/UnBox | |
parent | 3dd30d920a374ba31994f784fadd5ee130752247 (diff) | |
download | airwindows-lv2-port-9ecd9a8c01bea3c02f6e94cf67dd7c4ad12d1e34.tar.gz airwindows-lv2-port-9ecd9a8c01bea3c02f6e94cf67dd7c4ad12d1e34.tar.bz2 airwindows-lv2-port-9ecd9a8c01bea3c02f6e94cf67dd7c4ad12d1e34.zip |
UnBox, and build folder cleanups
Note that I am compelled to break VST builds on Windows because my build folders have had some files in 'em: you'll have to find those yourself. Here's hoping anyone trying to develop for Windows VST (or any VST) is able to find what they need, but I cannot help you set up a build environment, I can only give you my code for the audio part.
Diffstat (limited to 'plugins/WinVST/UnBox')
-rwxr-xr-x | plugins/WinVST/UnBox/.vs/Console4Channel64/v14/.suo | bin | 0 -> 32768 bytes | |||
-rwxr-xr-x | plugins/WinVST/UnBox/.vs/VSTProject/v14/.suo | bin | 0 -> 22528 bytes | |||
-rwxr-xr-x | plugins/WinVST/UnBox/UnBox.cpp | 143 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/UnBox.h | 79 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/UnBoxProc.cpp | 488 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/VSTProject.sln | 28 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/VSTProject.vcxproj | 183 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/VSTProject.vcxproj.filters | 48 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/VSTProject.vcxproj.user | 19 | ||||
-rwxr-xr-x | plugins/WinVST/UnBox/vstplug.def | 3 |
10 files changed, 991 insertions, 0 deletions
diff --git a/plugins/WinVST/UnBox/.vs/Console4Channel64/v14/.suo b/plugins/WinVST/UnBox/.vs/Console4Channel64/v14/.suo Binary files differnew file mode 100755 index 0000000..777b846 --- /dev/null +++ b/plugins/WinVST/UnBox/.vs/Console4Channel64/v14/.suo diff --git a/plugins/WinVST/UnBox/.vs/VSTProject/v14/.suo b/plugins/WinVST/UnBox/.vs/VSTProject/v14/.suo Binary files differnew file mode 100755 index 0000000..b96b3ed --- /dev/null +++ b/plugins/WinVST/UnBox/.vs/VSTProject/v14/.suo diff --git a/plugins/WinVST/UnBox/UnBox.cpp b/plugins/WinVST/UnBox/UnBox.cpp new file mode 100755 index 0000000..bb2c7d4 --- /dev/null +++ b/plugins/WinVST/UnBox/UnBox.cpp @@ -0,0 +1,143 @@ +/* ======================================== + * UnBox - UnBox.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __UnBox_H +#include "UnBox.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new UnBox(audioMaster);} + +UnBox::UnBox(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.5; + B = 0.0; + C = 0.5; + + for(int count = 0; count < 5; count++) {aL[count] = 0.0; bL[count] = 0.0; aR[count] = 0.0; bR[count] = 0.0; e[count] = 0.0;} + for(int count = 0; count < 11; count++) {cL[count] = 0.0; cR[count] = 0.0; f[count] = 0.0;} + iirSampleAL = 0.0; + iirSampleBL = 0.0; + iirSampleAR = 0.0; + iirSampleBR = 0.0; + + fpNShapeL = 0.0; + fpNShapeR = 0.0; + //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 +} + +UnBox::~UnBox() {} +VstInt32 UnBox::getVendorVersion () {return 1000;} +void UnBox::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void UnBox::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 UnBox::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 UnBox::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 UnBox::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 UnBox::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 UnBox::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Input", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "UnBox", 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 UnBox::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string (A*2.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string (B, text, kVstMaxParamStrLen); break; + case kParamC: float2string (C*2.0, text, kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void UnBox::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; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 UnBox::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool UnBox::getEffectName(char* name) { + vst_strncpy(name, "UnBox", kVstMaxProductStrLen); return true; +} + +VstPlugCategory UnBox::getPlugCategory() {return kPlugCategEffect;} + +bool UnBox::getProductString(char* text) { + vst_strncpy (text, "airwindows UnBox", kVstMaxProductStrLen); return true; +} + +bool UnBox::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/WinVST/UnBox/UnBox.h b/plugins/WinVST/UnBox/UnBox.h new file mode 100755 index 0000000..61db142 --- /dev/null +++ b/plugins/WinVST/UnBox/UnBox.h @@ -0,0 +1,79 @@ +/* ======================================== + * UnBox - UnBox.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __UnBox_H +#define __UnBox_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 = 'unbx'; //Change this to what the AU identity is! + +class UnBox : + public AudioEffectX +{ +public: + UnBox(audioMasterCallback audioMaster); + ~UnBox(); + 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 fpNShapeL; + long double fpNShapeR; + //default stuff + long double aL[5]; + long double bL[5]; + long double cL[11]; + long double aR[5]; + long double bR[5]; + long double cR[11]; + long double e[5]; + long double f[11]; + long double iirSampleAL; + long double iirSampleBL; + long double iirSampleAR; + long double iirSampleBR; + + + float A; + float B; + float C; +}; + +#endif diff --git a/plugins/WinVST/UnBox/UnBoxProc.cpp b/plugins/WinVST/UnBox/UnBoxProc.cpp new file mode 100755 index 0000000..12133dc --- /dev/null +++ b/plugins/WinVST/UnBox/UnBoxProc.cpp @@ -0,0 +1,488 @@ +/* ======================================== + * UnBox - UnBox.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __UnBox_H +#include "UnBox.h" +#endif + +void UnBox::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 input = A*2.0; + double unbox = B+1.0; + unbox *= unbox; //let's get some more gain into this + double iirAmount = (unbox*0.00052)/overallscale; + double output = C*2.0; + + double treble = unbox; //averaging taps 1-4 + double gain = treble; + if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;} + if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;} + if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;} + if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;} + if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + e[0] /= treble; + e[1] /= treble; + e[2] /= treble; + e[3] /= treble; + e[4] /= treble; + //and now it's neatly scaled, too + + treble = unbox*2.0; //averaging taps 1-8 + gain = treble; + 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;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + f[0] /= treble; + f[1] /= treble; + f[2] /= treble; + f[3] /= treble; + f[4] /= treble; + f[5] /= treble; + f[6] /= treble; + f[7] /= treble; + f[8] /= treble; + f[9] /= treble; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;} + + 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 aUnBox. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1]; + aL[1] = aL[0]; aL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (aL[1] * e[1]); + inputSampleL += (aL[2] * e[2]); + inputSampleL += (aL[3] * e[3]); + inputSampleL += (aL[4] * e[4]); + //this is now an average of inputSampleL + + aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1]; + aR[1] = aR[0]; aR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (aR[1] * e[1]); + inputSampleR += (aR[2] * e[2]); + inputSampleR += (aR[3] * e[3]); + inputSampleR += (aR[4] * e[4]); + //this is now an average of inputSampleR + + bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (bL[1] * e[1]); + inputSampleL += (bL[2] * e[2]); + inputSampleL += (bL[3] * e[3]); + inputSampleL += (bL[4] * e[4]); + //this is now an average of an average of inputSampleL. Two poles + + bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (bR[1] * e[1]); + inputSampleR += (bR[2] * e[2]); + inputSampleR += (bR[3] * e[3]); + inputSampleR += (bR[4] * e[4]); + //this is now an average of an average of inputSampleR. Two poles + + inputSampleL *= unbox; + inputSampleR *= unbox; + //clip to 1.2533141373155 to reach maximum output + if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155; + if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155; + inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + + if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155; + if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155; + inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + + inputSampleL /= unbox; + inputSampleR /= unbox; + //now we have a distorted inputSample at the correct volume relative to drySample + + long double accumulatorSampleL = (drySampleL - inputSampleL); + cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5]; + cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1]; + cL[1] = cL[0]; cL[0] = accumulatorSampleL; + accumulatorSampleL *= f[0]; + accumulatorSampleL += (cL[1] * f[1]); + accumulatorSampleL += (cL[2] * f[2]); + accumulatorSampleL += (cL[3] * f[3]); + accumulatorSampleL += (cL[4] * f[4]); + accumulatorSampleL += (cL[5] * f[5]); + accumulatorSampleL += (cL[6] * f[6]); + accumulatorSampleL += (cL[7] * f[7]); + accumulatorSampleL += (cL[8] * f[8]); + accumulatorSampleL += (cL[9] * f[9]); + //this is now an average of all the recent variances from dry + + long double accumulatorSampleR = (drySampleR - inputSampleR); + cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5]; + cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1]; + cR[1] = cR[0]; cR[0] = accumulatorSampleR; + accumulatorSampleR *= f[0]; + accumulatorSampleR += (cR[1] * f[1]); + accumulatorSampleR += (cR[2] * f[2]); + accumulatorSampleR += (cR[3] * f[3]); + accumulatorSampleR += (cR[4] * f[4]); + accumulatorSampleR += (cR[5] * f[5]); + accumulatorSampleR += (cR[6] * f[6]); + accumulatorSampleR += (cR[7] * f[7]); + accumulatorSampleR += (cR[8] * f[8]); + accumulatorSampleR += (cR[9] * f[9]); + //this is now an average of all the recent variances from dry + + iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleAL; + //two poles of IIR + + iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleAR; + //two poles of IIR + + iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleBL; + //highpass section + + iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleBR; + //highpass section + //this is now a highpassed average of all the recent variances from dry + + inputSampleL = drySampleL - accumulatorSampleL; + inputSampleR = drySampleR - accumulatorSampleR; + //we apply it as one operation, to get the result. + + if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;} + + //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 UnBox::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 input = A*2.0; + double unbox = B+1.0; + unbox *= unbox; //let's get some more gain into this + double iirAmount = (unbox*0.00052)/overallscale; + double output = C*2.0; + + double treble = unbox; //averaging taps 1-4 + double gain = treble; + if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;} + if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;} + if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;} + if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;} + if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + e[0] /= treble; + e[1] /= treble; + e[2] /= treble; + e[3] /= treble; + e[4] /= treble; + //and now it's neatly scaled, too + + treble = unbox*2.0; //averaging taps 1-8 + gain = treble; + 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;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + f[0] /= treble; + f[1] /= treble; + f[2] /= treble; + f[3] /= treble; + f[4] /= treble; + f[5] /= treble; + f[6] /= treble; + f[7] /= treble; + f[8] /= treble; + f[9] /= treble; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;} + + 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 aUnBox. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1]; + aL[1] = aL[0]; aL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (aL[1] * e[1]); + inputSampleL += (aL[2] * e[2]); + inputSampleL += (aL[3] * e[3]); + inputSampleL += (aL[4] * e[4]); + //this is now an average of inputSampleL + + aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1]; + aR[1] = aR[0]; aR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (aR[1] * e[1]); + inputSampleR += (aR[2] * e[2]); + inputSampleR += (aR[3] * e[3]); + inputSampleR += (aR[4] * e[4]); + //this is now an average of inputSampleR + + bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (bL[1] * e[1]); + inputSampleL += (bL[2] * e[2]); + inputSampleL += (bL[3] * e[3]); + inputSampleL += (bL[4] * e[4]); + //this is now an average of an average of inputSampleL. Two poles + + bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (bR[1] * e[1]); + inputSampleR += (bR[2] * e[2]); + inputSampleR += (bR[3] * e[3]); + inputSampleR += (bR[4] * e[4]); + //this is now an average of an average of inputSampleR. Two poles + + inputSampleL *= unbox; + inputSampleR *= unbox; + //clip to 1.2533141373155 to reach maximum output + if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155; + if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155; + inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + + if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155; + if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155; + inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + + inputSampleL /= unbox; + inputSampleR /= unbox; + //now we have a distorted inputSample at the correct volume relative to drySample + + long double accumulatorSampleL = (drySampleL - inputSampleL); + cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5]; + cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1]; + cL[1] = cL[0]; cL[0] = accumulatorSampleL; + accumulatorSampleL *= f[0]; + accumulatorSampleL += (cL[1] * f[1]); + accumulatorSampleL += (cL[2] * f[2]); + accumulatorSampleL += (cL[3] * f[3]); + accumulatorSampleL += (cL[4] * f[4]); + accumulatorSampleL += (cL[5] * f[5]); + accumulatorSampleL += (cL[6] * f[6]); + accumulatorSampleL += (cL[7] * f[7]); + accumulatorSampleL += (cL[8] * f[8]); + accumulatorSampleL += (cL[9] * f[9]); + //this is now an average of all the recent variances from dry + + long double accumulatorSampleR = (drySampleR - inputSampleR); + cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5]; + cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1]; + cR[1] = cR[0]; cR[0] = accumulatorSampleR; + accumulatorSampleR *= f[0]; + accumulatorSampleR += (cR[1] * f[1]); + accumulatorSampleR += (cR[2] * f[2]); + accumulatorSampleR += (cR[3] * f[3]); + accumulatorSampleR += (cR[4] * f[4]); + accumulatorSampleR += (cR[5] * f[5]); + accumulatorSampleR += (cR[6] * f[6]); + accumulatorSampleR += (cR[7] * f[7]); + accumulatorSampleR += (cR[8] * f[8]); + accumulatorSampleR += (cR[9] * f[9]); + //this is now an average of all the recent variances from dry + + iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleAL; + //two poles of IIR + + iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleAR; + //two poles of IIR + + iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleBL; + //highpass section + + iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleBR; + //highpass section + //this is now a highpassed average of all the recent variances from dry + + inputSampleL = drySampleL - accumulatorSampleL; + inputSampleR = drySampleR - accumulatorSampleR; + //we apply it as one operation, to get the result. + + if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;} + + //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. +} diff --git a/plugins/WinVST/UnBox/VSTProject.sln b/plugins/WinVST/UnBox/VSTProject.sln new file mode 100755 index 0000000..694b424 --- /dev/null +++ b/plugins/WinVST/UnBox/VSTProject.sln @@ -0,0 +1,28 @@ +
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\ No newline at end of file diff --git a/plugins/WinVST/UnBox/VSTProject.vcxproj.filters b/plugins/WinVST/UnBox/VSTProject.vcxproj.filters new file mode 100755 index 0000000..1038112 --- /dev/null +++ b/plugins/WinVST/UnBox/VSTProject.vcxproj.filters @@ -0,0 +1,48 @@ +<?xml version="1.0" encoding="utf-8"?>
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\ No newline at end of file diff --git a/plugins/WinVST/UnBox/VSTProject.vcxproj.user b/plugins/WinVST/UnBox/VSTProject.vcxproj.user new file mode 100755 index 0000000..2216267 --- /dev/null +++ b/plugins/WinVST/UnBox/VSTProject.vcxproj.user @@ -0,0 +1,19 @@ +<?xml version="1.0" encoding="utf-8"?>
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\ No newline at end of file diff --git a/plugins/WinVST/UnBox/vstplug.def b/plugins/WinVST/UnBox/vstplug.def new file mode 100755 index 0000000..5bf499a --- /dev/null +++ b/plugins/WinVST/UnBox/vstplug.def @@ -0,0 +1,3 @@ +EXPORTS
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\ No newline at end of file |