diff options
Diffstat (limited to 'plugins/LinuxVST/src/Point')
-rwxr-xr-x | plugins/LinuxVST/src/Point/Point.cpp | 147 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Point/Point.h | 77 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Point/PointProc.cpp | 312 |
3 files changed, 536 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/Point/Point.cpp b/plugins/LinuxVST/src/Point/Point.cpp new file mode 100755 index 0000000..22885b5 --- /dev/null +++ b/plugins/LinuxVST/src/Point/Point.cpp @@ -0,0 +1,147 @@ +/* ======================================== + * Point - Point.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Point_H +#include "Point.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Point(audioMaster);} + +Point::Point(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.5; + B = 0.5; + C = 0.5; + nibAL = 0.0; + nobAL = 0.0; + nibBL = 0.0; + nobBL = 0.0; + nibAR = 0.0; + nobAR = 0.0; + nibBR = 0.0; + nobBR = 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 +} + +Point::~Point() {} +VstInt32 Point::getVendorVersion () {return 1000;} +void Point::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Point::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 Point::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 Point::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 Point::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 + case kParamC: C = value; break; + default: throw; // unknown parameter, shouldn't happen! + } +} + +float Point::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 Point::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Input Trim", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Point", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "Reaction Speed", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void Point::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string ((A*24.0)-12.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string ((B*2.0)-1.0, text, kVstMaxParamStrLen); break; + case kParamC: float2string (C, text, kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void Point::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, " ", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, " ", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 Point::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Point::getEffectName(char* name) { + vst_strncpy(name, "Point", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Point::getPlugCategory() {return kPlugCategEffect;} + +bool Point::getProductString(char* text) { + vst_strncpy (text, "airwindows Point", kVstMaxProductStrLen); return true; +} + +bool Point::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/Point/Point.h b/plugins/LinuxVST/src/Point/Point.h new file mode 100755 index 0000000..12c8001 --- /dev/null +++ b/plugins/LinuxVST/src/Point/Point.h @@ -0,0 +1,77 @@ +/* ======================================== + * Point - Point.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Point_H +#define __Point_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 = 'poit'; //Change this to what the AU identity is! + +class Point : + public AudioEffectX +{ +public: + Point(audioMasterCallback audioMaster); + ~Point(); + 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 + double nibAL; + double nobAL; + double nibBL; + double nobBL; + double nibAR; + double nobAR; + double nibBR; + double nobBR; + + float A; + float B; + float C; +}; + +#endif diff --git a/plugins/LinuxVST/src/Point/PointProc.cpp b/plugins/LinuxVST/src/Point/PointProc.cpp new file mode 100755 index 0000000..a00263d --- /dev/null +++ b/plugins/LinuxVST/src/Point/PointProc.cpp @@ -0,0 +1,312 @@ +/* ======================================== + * Point - Point.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Point_H +#include "Point.h" +#endif + +void Point::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 gaintrim = pow(10.0,((A*24.0)-12.0)/20); + double nibDiv = 1 / pow(C+0.2,7); + nibDiv /= overallscale; + double nobDiv; + if (((B*2.0)-1.0) > 0) nobDiv = nibDiv / (1.001-((B*2.0)-1.0)); + else nobDiv = nibDiv * (1.001-pow(((B*2.0)-1.0)*0.75,2)); + double nibnobFactor = 0.0; //start with the fallthrough value, why not + double absolute; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + long double inputSampleL; + long double inputSampleR; + + 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. + } + + inputSampleL *= gaintrim; + absolute = fabs(inputSampleL); + if (fpFlip) + { + nibAL = nibAL + (absolute / nibDiv); + nibAL = nibAL / (1 + (1/nibDiv)); + nobAL = nobAL + (absolute / nobDiv); + nobAL = nobAL / (1 + (1/nobDiv)); + if (nobAL > 0) + { + nibnobFactor = nibAL / nobAL; + } + } + else + { + nibBL = nibBL + (absolute / nibDiv); + nibBL = nibBL / (1 + (1/nibDiv)); + nobBL = nobBL + (absolute / nobDiv); + nobBL = nobBL / (1 + (1/nobDiv)); + if (nobBL > 0) + { + nibnobFactor = nibBL / nobBL; + } + } + inputSampleL *= nibnobFactor; + + + inputSampleR *= gaintrim; + absolute = fabs(inputSampleR); + if (fpFlip) + { + nibAR = nibAR + (absolute / nibDiv); + nibAR = nibAR / (1 + (1/nibDiv)); + nobAR = nobAR + (absolute / nobDiv); + nobAR = nobAR / (1 + (1/nobDiv)); + if (nobAR > 0) + { + nibnobFactor = nibAR / nobAR; + } + } + else + { + nibBR = nibBR + (absolute / nibDiv); + nibBR = nibBR / (1 + (1/nibDiv)); + nobBR = nobBR + (absolute / nobDiv); + nobBR = nobBR / (1 + (1/nobDiv)); + if (nobBR > 0) + { + nibnobFactor = nibBR / nobBR; + } + } + inputSampleR *= nibnobFactor; + + //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 Point::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 gaintrim = pow(10.0,((A*24.0)-12.0)/20); + double nibDiv = 1 / pow(C+0.2,7); + nibDiv /= overallscale; + double nobDiv; + if (((B*2.0)-1.0) > 0) nobDiv = nibDiv / (1.001-((B*2.0)-1.0)); + else nobDiv = nibDiv * (1.001-pow(((B*2.0)-1.0)*0.75,2)); + double nibnobFactor = 0.0; //start with the fallthrough value, why not + double absolute; + + 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; + + 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. + } + + inputSampleL *= gaintrim; + absolute = fabs(inputSampleL); + if (fpFlip) + { + nibAL = nibAL + (absolute / nibDiv); + nibAL = nibAL / (1 + (1/nibDiv)); + nobAL = nobAL + (absolute / nobDiv); + nobAL = nobAL / (1 + (1/nobDiv)); + if (nobAL > 0) + { + nibnobFactor = nibAL / nobAL; + } + } + else + { + nibBL = nibBL + (absolute / nibDiv); + nibBL = nibBL / (1 + (1/nibDiv)); + nobBL = nobBL + (absolute / nobDiv); + nobBL = nobBL / (1 + (1/nobDiv)); + if (nobBL > 0) + { + nibnobFactor = nibBL / nobBL; + } + } + inputSampleL *= nibnobFactor; + + + inputSampleR *= gaintrim; + absolute = fabs(inputSampleR); + if (fpFlip) + { + nibAR = nibAR + (absolute / nibDiv); + nibAR = nibAR / (1 + (1/nibDiv)); + nobAR = nobAR + (absolute / nobDiv); + nobAR = nobAR / (1 + (1/nobDiv)); + if (nobAR > 0) + { + nibnobFactor = nibAR / nobAR; + } + } + else + { + nibBR = nibBR + (absolute / nibDiv); + nibBR = nibBR / (1 + (1/nibDiv)); + nobBR = nobBR + (absolute / nobDiv); + nobBR = nobBR / (1 + (1/nobDiv)); + if (nobBR > 0) + { + nibnobFactor = nibBR / nobBR; + } + } + inputSampleR *= nibnobFactor; + + //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++; + } +}
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