From 169631d08c44b5a46391e5ab90284ef07de46853 Mon Sep 17 00:00:00 2001 From: Chris Johnson Date: Mon, 8 Jun 2020 12:56:05 -0400 Subject: AverMatrix and Dark --- plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp | 141 ++++++++++++ plugins/LinuxVST/src/AverMatrix/AverMatrix.h | 68 ++++++ plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp | 191 +++++++++++++++++ plugins/LinuxVST/src/Dark/Dark.cpp | 126 +++++++++++ plugins/LinuxVST/src/Dark/Dark.h | 63 ++++++ plugins/LinuxVST/src/Dark/DarkProc.cpp | 237 +++++++++++++++++++++ 6 files changed, 826 insertions(+) create mode 100755 plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp create mode 100755 plugins/LinuxVST/src/AverMatrix/AverMatrix.h create mode 100755 plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp create mode 100755 plugins/LinuxVST/src/Dark/Dark.cpp create mode 100755 plugins/LinuxVST/src/Dark/Dark.h create mode 100755 plugins/LinuxVST/src/Dark/DarkProc.cpp (limited to 'plugins/LinuxVST/src') diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp b/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp new file mode 100755 index 0000000..585bd85 --- /dev/null +++ b/plugins/LinuxVST/src/AverMatrix/AverMatrix.cpp @@ -0,0 +1,141 @@ +/* ======================================== + * AverMatrix - AverMatrix.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __AverMatrix_H +#include "AverMatrix.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new AverMatrix(audioMaster);} + +AverMatrix::AverMatrix(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 0.0; + C = 1.0; + for(int x = 0; x < 11; x++) { + f[x] = 0.0; + for (int y = 0; y < 11; y++) { + bL[x][y] = 0.0; bR[x][y] = 0.0; + } + } + + 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 +} + +AverMatrix::~AverMatrix() {} +VstInt32 AverMatrix::getVendorVersion () {return 1000;} +void AverMatrix::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void AverMatrix::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 AverMatrix::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 AverMatrix::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 AverMatrix::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 AverMatrix::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 AverMatrix::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Average", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Depth", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "Inv/Wet", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void AverMatrix::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string ((A * 9.0)+1.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string ((B * 9.0)+1.0, text, kVstMaxParamStrLen); break; + case kParamC: float2string ((C * 2.0)-1.0, text, kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void AverMatrix::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "taps", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "poles", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 AverMatrix::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool AverMatrix::getEffectName(char* name) { + vst_strncpy(name, "AverMatrix", kVstMaxProductStrLen); return true; +} + +VstPlugCategory AverMatrix::getPlugCategory() {return kPlugCategEffect;} + +bool AverMatrix::getProductString(char* text) { + vst_strncpy (text, "airwindows AverMatrix", kVstMaxProductStrLen); return true; +} + +bool AverMatrix::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrix.h b/plugins/LinuxVST/src/AverMatrix/AverMatrix.h new file mode 100755 index 0000000..b5f2012 --- /dev/null +++ b/plugins/LinuxVST/src/AverMatrix/AverMatrix.h @@ -0,0 +1,68 @@ +/* ======================================== + * AverMatrix - AverMatrix.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __AverMatrix_H +#define __AverMatrix_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include +#include +#include + +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 = 'avrm'; //Change this to what the AU identity is! + +class AverMatrix : + public AudioEffectX +{ +public: + AverMatrix(audioMasterCallback audioMaster); + ~AverMatrix(); + 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][11]; + double bR[11][11]; + double f[11]; + uint32_t fpd; + //default stuff + + float A; + float B; + float C; +}; + +#endif diff --git a/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp b/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp new file mode 100755 index 0000000..381ea9c --- /dev/null +++ b/plugins/LinuxVST/src/AverMatrix/AverMatrixProc.cpp @@ -0,0 +1,191 @@ +/* ======================================== + * AverMatrix - AverMatrix.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __AverMatrix_H +#include "AverMatrix.h" +#endif + +void AverMatrix::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double overalltaps = (A * 9.0)+1.0; + double taps = overalltaps; + //this is our averaging, which is not integer but continuous + + double overallpoles = (B * 9.0)+1.0; + //this is the poles of the filter, also not integer but continuous + int yLimit = floor(overallpoles)+1; + double yPartial = overallpoles - floor(overallpoles); + //now we can do a for loop, and also apply the final pole continuously + + double wet = (C * 2.0)-1.0; + double dry = (1.0-wet); + if (dry > 1.0) dry = 1.0; + + int xLimit = 1; + for(int x = 0; x < 11; x++) { + if (taps > 1.0) { + f[x] = 1.0; + taps -= 1.0; + xLimit++; + } else { + f[x] = taps; + taps = 0.0; + } + } //there, now we have a neat little moving average with remainders + if (xLimit > 9) xLimit = 9; + + if (overalltaps < 1.0) overalltaps = 1.0; + for(int x = 0; x < xLimit; x++) { + f[x] /= overalltaps; + } //and now it's neatly scaled, too + + 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; + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + long double previousPoleL = 0; + long double previousPoleR = 0; + for (int y = 0; y < yLimit; y++) { + for (int x = xLimit; x >= 0; x--) { + bL[x+1][y] = bL[x][y]; + bR[x+1][y] = bR[x][y]; + } + bL[0][y] = previousPoleL = inputSampleL; + bR[0][y] = previousPoleR = inputSampleR; + inputSampleL = 0.0; + inputSampleR = 0.0; + for (int x = 0; x < xLimit; x++) { + inputSampleL += (bL[x][y] * f[x]); + inputSampleR += (bR[x][y] * f[x]); + } + } + inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial); + inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial); + //in this way we can blend in the final pole + + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + //wet can be negative, in which case dry is always full volume and they cancel + + //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 AverMatrix::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + + double overalltaps = (A * 9.0)+1.0; + double taps = overalltaps; + //this is our averaging, which is not integer but continuous + + double overallpoles = (B * 9.0)+1.0; + //this is the poles of the filter, also not integer but continuous + int yLimit = floor(overallpoles)+1; + double yPartial = overallpoles - floor(overallpoles); + //now we can do a for loop, and also apply the final pole continuously + + double wet = (C * 2.0)-1.0; + double dry = (1.0-wet); + if (dry > 1.0) dry = 1.0; + + int xLimit = 1; + for(int x = 0; x < 11; x++) { + if (taps > 1.0) { + f[x] = 1.0; + taps -= 1.0; + xLimit++; + } else { + f[x] = taps; + taps = 0.0; + } + } //there, now we have a neat little moving average with remainders + if (xLimit > 9) xLimit = 9; + + if (overalltaps < 1.0) overalltaps = 1.0; + for(int x = 0; x < xLimit; x++) { + f[x] /= overalltaps; + } //and now it's neatly scaled, too + + 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; + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + long double previousPoleL = 0; + long double previousPoleR = 0; + for (int y = 0; y < yLimit; y++) { + for (int x = xLimit; x >= 0; x--) { + bL[x+1][y] = bL[x][y]; + bR[x+1][y] = bR[x][y]; + } + bL[0][y] = previousPoleL = inputSampleL; + bR[0][y] = previousPoleR = inputSampleR; + inputSampleL = 0.0; + inputSampleR = 0.0; + for (int x = 0; x < xLimit; x++) { + inputSampleL += (bL[x][y] * f[x]); + inputSampleR += (bR[x][y] * f[x]); + } + } + inputSampleL = (previousPoleL * (1.0-yPartial)) + (inputSampleL * yPartial); + inputSampleR = (previousPoleR * (1.0-yPartial)) + (inputSampleR * yPartial); + //in this way we can blend in the final pole + + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + //wet can be negative, in which case dry is always full volume and they cancel + + //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++; + } +} diff --git a/plugins/LinuxVST/src/Dark/Dark.cpp b/plugins/LinuxVST/src/Dark/Dark.cpp new file mode 100755 index 0000000..0f9721c --- /dev/null +++ b/plugins/LinuxVST/src/Dark/Dark.cpp @@ -0,0 +1,126 @@ +/* ======================================== + * Dark - Dark.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Dark_H +#include "Dark.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Dark(audioMaster);} + +Dark::Dark(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 1.0; + for(int count = 0; count < 99; count++) { + lastSampleL[count] = 0; + lastSampleR[count] = 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 +} + +Dark::~Dark() {} +VstInt32 Dark::getVendorVersion () {return 1000;} +void Dark::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Dark::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 Dark::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 Dark::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 Dark::setParameter(VstInt32 index, float value) { + switch (index) { + case kParamA: A = value; break; + default: throw; // unknown parameter, shouldn't happen! + } +} + +float Dark::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 Dark::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Quant", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void Dark::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: switch((VstInt32)( A * 1.999 )) //0 to almost edge of # of params + { case 0: vst_strncpy (text, "CD 16", kVstMaxParamStrLen); break; + case 1: vst_strncpy (text, "HD 24", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } break; //completed consoletype 'popup' parameter, exit + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void Dark::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 Dark::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Dark::getEffectName(char* name) { + vst_strncpy(name, "Dark", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Dark::getPlugCategory() {return kPlugCategEffect;} + +bool Dark::getProductString(char* text) { + vst_strncpy (text, "airwindows Dark", kVstMaxProductStrLen); return true; +} + +bool Dark::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/Dark/Dark.h b/plugins/LinuxVST/src/Dark/Dark.h new file mode 100755 index 0000000..a54d31b --- /dev/null +++ b/plugins/LinuxVST/src/Dark/Dark.h @@ -0,0 +1,63 @@ +/* ======================================== + * Dark - Dark.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Dark_H +#define __Dark_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include +#include +#include + +enum { + kParamA = 0, + kNumParameters = 1 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'dark'; //Change this to what the AU identity is! + +class Dark : + public AudioEffectX +{ +public: + Dark(audioMasterCallback audioMaster); + ~Dark(); + 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; + + float lastSampleL[100]; + float lastSampleR[100]; + uint32_t fpd; + //default stuff + + float A; +}; + +#endif diff --git a/plugins/LinuxVST/src/Dark/DarkProc.cpp b/plugins/LinuxVST/src/Dark/DarkProc.cpp new file mode 100755 index 0000000..672d8e0 --- /dev/null +++ b/plugins/LinuxVST/src/Dark/DarkProc.cpp @@ -0,0 +1,237 @@ +/* ======================================== + * Dark - Dark.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Dark_H +#include "Dark.h" +#endif + +void Dark::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + int processing = (VstInt32)( A * 1.999 ); + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + int depth = (int)(17.0*overallscale); + if (depth < 3) depth = 3; + if (depth > 98) depth = 98; + bool highres = false; + if (processing == 1) highres = true; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37; + fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; + if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37; + fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; + + if (highres) { + inputSampleL *= 8388608.0; + inputSampleR *= 8388608.0; + } else { + inputSampleL *= 32768.0; + inputSampleR *= 32768.0; + } + //0-1 is now one bit, now we dither + //We are doing it first Left, then Right, because the loops may run faster if + //they aren't too jammed full of variables. This means re-running code. + + //begin left + int quantA = floor(inputSampleL); + int quantB = floor(inputSampleL+1.0); + //to do this style of dither, we quantize in either direction and then + //do a reconstruction of what the result will be for each choice. + //We then evaluate which one we like, and keep a history of what we previously had + + float expectedSlew = 0; + for(int x = 0; x < depth; x++) { + expectedSlew += (lastSampleL[x+1] - lastSampleL[x]); + } + expectedSlew /= depth; //we have an average of all recent slews + //we are doing that to voice the thing down into the upper mids a bit + //it mustn't just soften the brightest treble, it must smooth high mids too + + float testA = fabs((lastSampleL[0] - quantA) - expectedSlew); + float testB = fabs((lastSampleL[0] - quantB) - expectedSlew); + + if (testA < testB) inputSampleL = quantA; + else inputSampleL = quantB; + //select whichever one departs LEAST from the vector of averaged + //reconstructed previous final samples. This will force a kind of dithering + //as it'll make the output end up as smooth as possible + + for(int x = depth; x >=0; x--) { + lastSampleL[x+1] = lastSampleL[x]; + } + lastSampleL[0] = inputSampleL; + //end left + + //begin right + quantA = floor(inputSampleR); + quantB = floor(inputSampleR+1.0); + //to do this style of dither, we quantize in either direction and then + //do a reconstruction of what the result will be for each choice. + //We then evaluate which one we like, and keep a history of what we previously had + + expectedSlew = 0; + for(int x = 0; x < depth; x++) { + expectedSlew += (lastSampleR[x+1] - lastSampleR[x]); + } + expectedSlew /= depth; //we have an average of all recent slews + //we are doing that to voice the thing down into the upper mids a bit + //it mustn't just soften the brightest treble, it must smooth high mids too + + testA = fabs((lastSampleR[0] - quantA) - expectedSlew); + testB = fabs((lastSampleR[0] - quantB) - expectedSlew); + + if (testA < testB) inputSampleR = quantA; + else inputSampleR = quantB; + //select whichever one departs LEAST from the vector of averaged + //reconstructed previous final samples. This will force a kind of dithering + //as it'll make the output end up as smooth as possible + + for(int x = depth; x >=0; x--) { + lastSampleR[x+1] = lastSampleR[x]; + } + lastSampleR[0] = inputSampleR; + //end right + + if (highres) { + inputSampleL /= 8388608.0; + inputSampleR /= 8388608.0; + } else { + inputSampleL /= 32768.0; + inputSampleR /= 32768.0; + } + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Dark::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + int processing = (VstInt32)( A * 1.999 ); + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + int depth = (int)(17.0*overallscale); + if (depth < 3) depth = 3; + if (depth > 98) depth = 98; + bool highres = false; + if (processing == 1) highres = true; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43; + fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; + if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43; + fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; + + if (highres) { + inputSampleL *= 8388608.0; + inputSampleR *= 8388608.0; + } else { + inputSampleL *= 32768.0; + inputSampleR *= 32768.0; + } + //0-1 is now one bit, now we dither + //We are doing it first Left, then Right, because the loops may run faster if + //they aren't too jammed full of variables. This means re-running code. + + //begin left + int quantA = floor(inputSampleL); + int quantB = floor(inputSampleL+1.0); + //to do this style of dither, we quantize in either direction and then + //do a reconstruction of what the result will be for each choice. + //We then evaluate which one we like, and keep a history of what we previously had + + float expectedSlew = 0; + for(int x = 0; x < depth; x++) { + expectedSlew += (lastSampleL[x+1] - lastSampleL[x]); + } + expectedSlew /= depth; //we have an average of all recent slews + //we are doing that to voice the thing down into the upper mids a bit + //it mustn't just soften the brightest treble, it must smooth high mids too + + float testA = fabs((lastSampleL[0] - quantA) - expectedSlew); + float testB = fabs((lastSampleL[0] - quantB) - expectedSlew); + + if (testA < testB) inputSampleL = quantA; + else inputSampleL = quantB; + //select whichever one departs LEAST from the vector of averaged + //reconstructed previous final samples. This will force a kind of dithering + //as it'll make the output end up as smooth as possible + + for(int x = depth; x >=0; x--) { + lastSampleL[x+1] = lastSampleL[x]; + } + lastSampleL[0] = inputSampleL; + //end left + + //begin right + quantA = floor(inputSampleR); + quantB = floor(inputSampleR+1.0); + //to do this style of dither, we quantize in either direction and then + //do a reconstruction of what the result will be for each choice. + //We then evaluate which one we like, and keep a history of what we previously had + + expectedSlew = 0; + for(int x = 0; x < depth; x++) { + expectedSlew += (lastSampleR[x+1] - lastSampleR[x]); + } + expectedSlew /= depth; //we have an average of all recent slews + //we are doing that to voice the thing down into the upper mids a bit + //it mustn't just soften the brightest treble, it must smooth high mids too + + testA = fabs((lastSampleR[0] - quantA) - expectedSlew); + testB = fabs((lastSampleR[0] - quantB) - expectedSlew); + + if (testA < testB) inputSampleR = quantA; + else inputSampleR = quantB; + //select whichever one departs LEAST from the vector of averaged + //reconstructed previous final samples. This will force a kind of dithering + //as it'll make the output end up as smooth as possible + + for(int x = depth; x >=0; x--) { + lastSampleR[x+1] = lastSampleR[x]; + } + lastSampleR[0] = inputSampleR; + //end right + + if (highres) { + inputSampleL /= 8388608.0; + inputSampleR /= 8388608.0; + } else { + inputSampleL /= 32768.0; + inputSampleR /= 32768.0; + } + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} -- cgit v1.2.3