diff options
author | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
---|---|---|
committer | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
commit | 633be2e22c6648c901f08f3b4cd4e8e14ea86443 (patch) | |
tree | 1e272c3d2b5bd29636b9f9f521af62734e4df012 /plugins/MacVST/Desk/source | |
parent | 057757aa8eb0a463caf0cdfdb5894ac5f723ff3f (diff) | |
download | airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.gz airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.bz2 airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.zip |
Updates (in case my plane crashes)
Diffstat (limited to 'plugins/MacVST/Desk/source')
-rwxr-xr-x | plugins/MacVST/Desk/source/Desk.cpp | 88 | ||||
-rwxr-xr-x | plugins/MacVST/Desk/source/Desk.h | 70 | ||||
-rwxr-xr-x | plugins/MacVST/Desk/source/DeskProc.cpp | 356 |
3 files changed, 514 insertions, 0 deletions
diff --git a/plugins/MacVST/Desk/source/Desk.cpp b/plugins/MacVST/Desk/source/Desk.cpp new file mode 100755 index 0000000..7a30943 --- /dev/null +++ b/plugins/MacVST/Desk/source/Desk.cpp @@ -0,0 +1,88 @@ +/* ======================================== + * Desk - Desk.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Desk_H +#include "Desk.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Desk(audioMaster);} + +Desk::Desk(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + lastSampleL = 0.0; + lastOutSampleL = 0.0; + lastSlewL = 0.0; + lastSampleR = 0.0; + lastOutSampleR = 0.0; + lastSlewR = 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 +} + +Desk::~Desk() {} +VstInt32 Desk::getVendorVersion () {return 1000;} +void Desk::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Desk::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! + +VstInt32 Desk::getChunk (void** data, bool isPreset) +{ + return kNumParameters * sizeof(float); +} + +VstInt32 Desk::setChunk (void* data, VstInt32 byteSize, bool isPreset) +{ + return 0; +} + +void Desk::setParameter(VstInt32 index, float value) { +} + +float Desk::getParameter(VstInt32 index) { + return 0.0; //we only need to update the relevant name, this is simple to manage +} + +void Desk::getParameterName(VstInt32 index, char *text) { +} + +void Desk::getParameterDisplay(VstInt32 index, char *text) { +} + +void Desk::getParameterLabel(VstInt32 index, char *text) { +} + +VstInt32 Desk::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Desk::getEffectName(char* name) { + vst_strncpy(name, "Desk", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Desk::getPlugCategory() {return kPlugCategEffect;} + +bool Desk::getProductString(char* text) { + vst_strncpy (text, "airwindows Desk", kVstMaxProductStrLen); return true; +} + +bool Desk::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/MacVST/Desk/source/Desk.h b/plugins/MacVST/Desk/source/Desk.h new file mode 100755 index 0000000..8c5c398 --- /dev/null +++ b/plugins/MacVST/Desk/source/Desk.h @@ -0,0 +1,70 @@ +/* ======================================== + * Desk - Desk.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Desk_H +#define __Desk_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include <set> +#include <string> +#include <math.h> + +enum { + kNumParameters = 0 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'desk'; //Change this to what the AU identity is! + +class Desk : + public AudioEffectX +{ +public: + Desk(audioMasterCallback audioMaster); + ~Desk(); + 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 fpNShapeLA; + long double fpNShapeLB; + long double fpNShapeRA; + long double fpNShapeRB; + bool fpFlip; + //default stuff + + long double lastSampleL; + long double lastOutSampleL; + double lastSlewL; + long double lastSampleR; + long double lastOutSampleR; + double lastSlewR; + +}; + +#endif diff --git a/plugins/MacVST/Desk/source/DeskProc.cpp b/plugins/MacVST/Desk/source/DeskProc.cpp new file mode 100755 index 0000000..a2bc92c --- /dev/null +++ b/plugins/MacVST/Desk/source/DeskProc.cpp @@ -0,0 +1,356 @@ +/* ======================================== + * Desk - Desk.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Desk_H +#include "Desk.h" +#endif + +void Desk::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double gain = 0.135; + double slewgain = 0.208; + double prevslew = 0.333; + double balanceB = 0.0001; + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + slewgain *= overallscale; + prevslew *= overallscale; + balanceB /= overallscale; + double balanceA = 1.0 - balanceB; + double slew; + double bridgerectifier; + double combsample; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + long double drySampleL; + long double drySampleR; + + 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; + + //begin L + slew = inputSampleL - lastSampleL; + lastSampleL = inputSampleL; + //Set up direct reference for slew + + bridgerectifier = fabs(slew*slewgain); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (slew > 0) slew = bridgerectifier/slewgain; + else slew = -(bridgerectifier/slewgain); + + inputSampleL = (lastOutSampleL*balanceA) + (lastSampleL*balanceB) + slew; + //go from last slewed, but include some raw values + lastOutSampleL = inputSampleL; + //Set up slewed reference + + combsample = fabs(drySampleL*lastSampleL); + if (combsample > 1.0) combsample = 1.0; + //bailout for very high input gains + inputSampleL -= (lastSlewL * combsample * prevslew); + lastSlewL = slew; + //slew interaction with previous slew + + inputSampleL *= gain; + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + + if (inputSampleL > 0) inputSampleL = bridgerectifier; + else inputSampleL = -bridgerectifier; + //drive section + inputSampleL /= gain; + //end L + + //begin R + slew = inputSampleR - lastSampleR; + lastSampleR = inputSampleR; + //Set up direct reference for slew + + bridgerectifier = fabs(slew*slewgain); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (slew > 0) slew = bridgerectifier/slewgain; + else slew = -(bridgerectifier/slewgain); + + inputSampleR = (lastOutSampleR*balanceA) + (lastSampleR*balanceB) + slew; + //go from last slewed, but include some raw values + lastOutSampleR = inputSampleR; + //Set up slewed reference + + combsample = fabs(drySampleR*lastSampleR); + if (combsample > 1.0) combsample = 1.0; + //bailout for very high input gains + inputSampleR -= (lastSlewR * combsample * prevslew); + lastSlewR = slew; + //slew interaction with previous slew + + inputSampleR *= gain; + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + + if (inputSampleR > 0) inputSampleR = bridgerectifier; + else inputSampleR = -bridgerectifier; + //drive section + inputSampleR /= gain; + //end R + + //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 Desk::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double gain = 0.135; + double slewgain = 0.208; + double prevslew = 0.333; + double balanceB = 0.0001; + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + slewgain *= overallscale; + prevslew *= overallscale; + balanceB /= overallscale; + double balanceA = 1.0 - balanceB; + double slew; + double bridgerectifier; + double combsample; + + double fpTemp; //this is different from singlereplacing + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + long double drySampleL; + long double drySampleR; + + 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; + + //begin L + slew = inputSampleL - lastSampleL; + lastSampleL = inputSampleL; + //Set up direct reference for slew + + bridgerectifier = fabs(slew*slewgain); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (slew > 0) slew = bridgerectifier/slewgain; + else slew = -(bridgerectifier/slewgain); + + inputSampleL = (lastOutSampleL*balanceA) + (lastSampleL*balanceB) + slew; + //go from last slewed, but include some raw values + lastOutSampleL = inputSampleL; + //Set up slewed reference + + combsample = fabs(drySampleL*lastSampleL); + if (combsample > 1.0) combsample = 1.0; + //bailout for very high input gains + inputSampleL -= (lastSlewL * combsample * prevslew); + lastSlewL = slew; + //slew interaction with previous slew + + inputSampleL *= gain; + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + + if (inputSampleL > 0) inputSampleL = bridgerectifier; + else inputSampleL = -bridgerectifier; + //drive section + inputSampleL /= gain; + //end L + + //begin R + slew = inputSampleR - lastSampleR; + lastSampleR = inputSampleR; + //Set up direct reference for slew + + bridgerectifier = fabs(slew*slewgain); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + if (slew > 0) slew = bridgerectifier/slewgain; + else slew = -(bridgerectifier/slewgain); + + inputSampleR = (lastOutSampleR*balanceA) + (lastSampleR*balanceB) + slew; + //go from last slewed, but include some raw values + lastOutSampleR = inputSampleR; + //Set up slewed reference + + combsample = fabs(drySampleR*lastSampleR); + if (combsample > 1.0) combsample = 1.0; + //bailout for very high input gains + inputSampleR -= (lastSlewR * combsample * prevslew); + lastSlewR = slew; + //slew interaction with previous slew + + inputSampleR *= gain; + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.0; + else bridgerectifier = sin(bridgerectifier); + + if (inputSampleR > 0) inputSampleR = bridgerectifier; + else inputSampleR = -bridgerectifier; + //drive section + inputSampleR /= gain; + //end R + + //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 |