From 633be2e22c6648c901f08f3b4cd4e8e14ea86443 Mon Sep 17 00:00:00 2001 From: Chris Johnson Date: Mon, 22 Oct 2018 18:04:06 -0400 Subject: Updates (in case my plane crashes) --- .../WinVST/Average/.vs/Console4Channel64/v14/.suo | Bin 0 -> 32768 bytes plugins/WinVST/Average/.vs/VSTProject/v14/.suo | Bin 0 -> 22528 bytes plugins/WinVST/Average/Average.cpp | 133 ++++++++ plugins/WinVST/Average/Average.h | 72 ++++ plugins/WinVST/Average/AverageProc.cpp | 361 +++++++++++++++++++++ plugins/WinVST/Average/VSTProject.sln | 28 ++ plugins/WinVST/Average/VSTProject.vcxproj | 183 +++++++++++ plugins/WinVST/Average/VSTProject.vcxproj.filters | 48 +++ plugins/WinVST/Average/VSTProject.vcxproj.user | 19 ++ plugins/WinVST/Average/vstplug.def | 3 + 10 files changed, 847 insertions(+) create mode 100755 plugins/WinVST/Average/.vs/Console4Channel64/v14/.suo create mode 100755 plugins/WinVST/Average/.vs/VSTProject/v14/.suo create mode 100755 plugins/WinVST/Average/Average.cpp create mode 100755 plugins/WinVST/Average/Average.h create mode 100755 plugins/WinVST/Average/AverageProc.cpp create mode 100755 plugins/WinVST/Average/VSTProject.sln create mode 100755 plugins/WinVST/Average/VSTProject.vcxproj create mode 100755 plugins/WinVST/Average/VSTProject.vcxproj.filters create mode 100755 plugins/WinVST/Average/VSTProject.vcxproj.user create mode 100755 plugins/WinVST/Average/vstplug.def (limited to 'plugins/WinVST/Average') diff --git a/plugins/WinVST/Average/.vs/Console4Channel64/v14/.suo b/plugins/WinVST/Average/.vs/Console4Channel64/v14/.suo new file mode 100755 index 0000000..777b846 Binary files /dev/null and b/plugins/WinVST/Average/.vs/Console4Channel64/v14/.suo differ diff --git a/plugins/WinVST/Average/.vs/VSTProject/v14/.suo b/plugins/WinVST/Average/.vs/VSTProject/v14/.suo new file mode 100755 index 0000000..24b500e Binary files /dev/null and b/plugins/WinVST/Average/.vs/VSTProject/v14/.suo differ diff --git a/plugins/WinVST/Average/Average.cpp b/plugins/WinVST/Average/Average.cpp new file mode 100755 index 0000000..945c76c --- /dev/null +++ b/plugins/WinVST/Average/Average.cpp @@ -0,0 +1,133 @@ +/* ======================================== + * Average - Average.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Average_H +#include "Average.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Average(audioMaster);} + +Average::Average(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 1.0; + + for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0; f[count] = 0.0;} + + fpNShapeLA = 0.0; + fpNShapeLB = 0.0; + fpNShapeRA = 0.0; + fpNShapeRB = 0.0; + fpFlip = 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 +} + +Average::~Average() {} +VstInt32 Average::getVendorVersion () {return 1000;} +void Average::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Average::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 Average::getChunk (void** data, bool isPreset) +{ + float *chunkData = (float *)calloc(kNumParameters, sizeof(float)); + chunkData[0] = A; + chunkData[1] = B; + /* 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 Average::setChunk (void* data, VstInt32 byteSize, bool isPreset) +{ + float *chunkData = (float *)data; + A = pinParameter(chunkData[0]); + B = pinParameter(chunkData[1]); + /* 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 Average::setParameter(VstInt32 index, float value) { + switch (index) { + case kParamA: A = value; break; + case kParamB: B = value; break; //percent. Using this value, it'll be 0-100 everywhere + default: throw; // unknown parameter, shouldn't happen! + } +} + +float Average::getParameter(VstInt32 index) { + switch (index) { + case kParamA: return A; break; + case kParamB: return B; 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 Average::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Average", kVstMaxParamStrLen); break; + case kParamB: 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 Average::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string ((A * 9.0)+1.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string (B, text, kVstMaxParamStrLen); break; //also display 0-1 as percent + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void Average::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "taps", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break; //the percent + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 Average::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Average::getEffectName(char* name) { + vst_strncpy(name, "Average", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Average::getPlugCategory() {return kPlugCategEffect;} + +bool Average::getProductString(char* text) { + vst_strncpy (text, "airwindows Average", kVstMaxProductStrLen); return true; +} + +bool Average::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/WinVST/Average/Average.h b/plugins/WinVST/Average/Average.h new file mode 100755 index 0000000..7e4b25e --- /dev/null +++ b/plugins/WinVST/Average/Average.h @@ -0,0 +1,72 @@ +/* ======================================== + * Average - Average.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Average_H +#define __Average_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include +#include +#include + +enum { + kParamA = 0, + kParamB = 1, + kNumParameters = 2 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'aver'; //Change this to what the AU identity is! + +class Average : + public AudioEffectX +{ +public: + Average(audioMasterCallback audioMaster); + ~Average(); + 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; + + double bL[11]; + double f[11]; + double bR[11]; + + long double fpNShapeLA; + long double fpNShapeLB; + long double fpNShapeRA; + long double fpNShapeRB; + bool fpFlip; + //default stuff + + float A; + float B; + +}; + +#endif diff --git a/plugins/WinVST/Average/AverageProc.cpp b/plugins/WinVST/Average/AverageProc.cpp new file mode 100755 index 0000000..2b1c355 --- /dev/null +++ b/plugins/WinVST/Average/AverageProc.cpp @@ -0,0 +1,361 @@ +/* ======================================== + * Average - Average.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Average_H +#include "Average.h" +#endif + +void Average::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double inputSampleL; + double inputSampleR; + + double overallscale = (A * 9.0)+1.0; + double wet = B; + double dry = 1.0 - wet; + double gain = overallscale; + + 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 (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5]; + bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL; + + bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5]; + bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR; + //primitive way of doing this: for larger batches of samples, you might + //try using a circular buffer like in a reverb. If you add the new sample + //and subtract the one on the end you can keep a running tally of the samples + //between. Beware of tiny floating-point math errors eventually screwing up + //your system, though! + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + inputSampleL -= correctionSample; + + correctionSample = inputSampleR - accumulatorSampleR; + inputSampleR -= correctionSample; + //our one math operation on the input data coming in + + if (wet < 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + //dry/wet control only applies if you're using it. We don't do a multiply by 1.0 + //if it 'won't change anything' but our sample might be at a very different scaling + //in the floating point system. + + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Average::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double inputSampleL; + double inputSampleR; + + double overallscale = (A * 9.0)+1.0; + double wet = B; + double dry = 1.0 - wet; + double gain = overallscale; + + 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 (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5]; + bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL; + + bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5]; + bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR; + //primitive way of doing this: for larger batches of samples, you might + //try using a circular buffer like in a reverb. If you add the new sample + //and subtract the one on the end you can keep a running tally of the samples + //between. Beware of tiny floating-point math errors eventually screwing up + //your system, though! + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + inputSampleL -= correctionSample; + + correctionSample = inputSampleR - accumulatorSampleR; + inputSampleR -= correctionSample; + //our one math operation on the input data coming in + + if (wet < 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + //dry/wet control only applies if you're using it. We don't do a multiply by 1.0 + //if it 'won't change anything' but our sample might be at a very different scaling + //in the floating point system. + + //noise shaping to 64-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} \ No newline at end of file diff --git a/plugins/WinVST/Average/VSTProject.sln b/plugins/WinVST/Average/VSTProject.sln new file mode 100755 index 0000000..694b424 --- /dev/null +++ b/plugins/WinVST/Average/VSTProject.sln @@ -0,0 +1,28 @@ + +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 diff --git a/plugins/WinVST/Average/VSTProject.vcxproj b/plugins/WinVST/Average/VSTProject.vcxproj new file mode 100755 index 0000000..16bde27 --- /dev/null +++ b/plugins/WinVST/Average/VSTProject.vcxproj @@ -0,0 +1,183 @@ + + + + + Debug + Win32 + + + Release + Win32 + + + Debug + x64 + + + Release + x64 + + + + + + + + + + + + + + + + + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C} + VSTProject + 8.1 + Average64 + + + + DynamicLibrary + true + v140 + NotSet + + + DynamicLibrary + false + v140 + false + NotSet + + + DynamicLibrary + true + v140 + NotSet + + + DynamicLibrary + false + v140 + false + NotSet + + + + + + + + + + + + + + + + + + + + + .dll + + + $(SolutionDir)$(Configuration)\ + $(Configuration)\ + $(VC_ExecutablePath_x64);$(WindowsSDK_ExecutablePath);$(VS_ExecutablePath);$(MSBuild_ExecutablePath);$(SystemRoot)\SysWow64;$(FxCopDir);$(PATH) + + + $(SolutionDir)$(Configuration)\ + $(Configuration)\ + $(VC_ExecutablePath_x64);$(WindowsSDK_ExecutablePath);$(VS_ExecutablePath);$(MSBuild_ExecutablePath);$(SystemRoot)\SysWow64;$(FxCopDir);$(PATH) + + + + Level3 + MaxSpeed + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + MultiThreadedDebug + Speed + false + Default + false + None + + + vstplug.def + libcmt.dll;libcmtd.dll;msvcrt.lib;%(IgnoreSpecificDefaultLibraries) + kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + + + + + Level3 + MaxSpeed + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + false + MultiThreadedDebug + Default + false + None + + + kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + libcmt.dll;libcmtd.dll;msvcrt.lib;%(IgnoreSpecificDefaultLibraries) + vstplug.def + + + + + Level3 + MaxSpeed + false + false + true + MultiThreaded + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + None + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + + + true + true + libcmt.dll;libcmtd.dll;msvcrt.lib;libc.lib;libcd.lib;libcmt.lib;msvcrtd.lib;%(IgnoreSpecificDefaultLibraries) + libcmt.lib;uuid.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + vstplug.def + + + + + Level3 + MaxSpeed + false + false + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + None + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + MultiThreaded + + + true + true + libcmt.dll;libcmtd.dll;msvcrt.lib;libc.lib;libcd.lib;libcmt.lib;msvcrtd.lib;%(IgnoreSpecificDefaultLibraries) + libcmt.lib;uuid.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + vstplug.def + + + + + + \ No newline at end of file diff --git a/plugins/WinVST/Average/VSTProject.vcxproj.filters b/plugins/WinVST/Average/VSTProject.vcxproj.filters new file mode 100755 index 0000000..b6c15ea --- /dev/null +++ b/plugins/WinVST/Average/VSTProject.vcxproj.filters @@ -0,0 +1,48 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hh;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + \ No newline at end of file diff --git a/plugins/WinVST/Average/VSTProject.vcxproj.user b/plugins/WinVST/Average/VSTProject.vcxproj.user new file mode 100755 index 0000000..2216267 --- /dev/null +++ b/plugins/WinVST/Average/VSTProject.vcxproj.user @@ -0,0 +1,19 @@ + + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + \ No newline at end of file diff --git a/plugins/WinVST/Average/vstplug.def b/plugins/WinVST/Average/vstplug.def new file mode 100755 index 0000000..5bf499a --- /dev/null +++ b/plugins/WinVST/Average/vstplug.def @@ -0,0 +1,3 @@ +EXPORTS + VSTPluginMain + main=VSTPluginMain \ No newline at end of file -- cgit v1.2.3