diff options
Diffstat (limited to 'plugins/LinuxVST/src/Beam')
-rwxr-xr-x | plugins/LinuxVST/src/Beam/Beam.cpp | 142 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Beam/Beam.h | 67 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Beam/BeamProc.cpp | 266 |
3 files changed, 475 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/Beam/Beam.cpp b/plugins/LinuxVST/src/Beam/Beam.cpp new file mode 100755 index 0000000..183b5e5 --- /dev/null +++ b/plugins/LinuxVST/src/Beam/Beam.cpp @@ -0,0 +1,142 @@ +/* ======================================== + * Beam - Beam.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Beam_H +#include "Beam.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Beam(audioMaster);} + +Beam::Beam(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 1.0; + B = 0.5; + C = 0.0; + 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 +} + +Beam::~Beam() {} +VstInt32 Beam::getVendorVersion () {return 1000;} +void Beam::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Beam::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 Beam::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 Beam::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 Beam::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 Beam::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 Beam::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Quant", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Focus", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "DeRez", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void Beam::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 + case kParamB: float2string (B, 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 Beam::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 Beam::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Beam::getEffectName(char* name) { + vst_strncpy(name, "Beam", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Beam::getPlugCategory() {return kPlugCategEffect;} + +bool Beam::getProductString(char* text) { + vst_strncpy (text, "airwindows Beam", kVstMaxProductStrLen); return true; +} + +bool Beam::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/Beam/Beam.h b/plugins/LinuxVST/src/Beam/Beam.h new file mode 100755 index 0000000..30e1ccd --- /dev/null +++ b/plugins/LinuxVST/src/Beam/Beam.h @@ -0,0 +1,67 @@ +/* ======================================== + * Beam - Beam.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Beam_H +#define __Beam_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 = 'beam'; //Change this to what the AU identity is! + +class Beam : + public AudioEffectX +{ +public: + Beam(audioMasterCallback audioMaster); + ~Beam(); + 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; + float B; + float C; +}; + +#endif diff --git a/plugins/LinuxVST/src/Beam/BeamProc.cpp b/plugins/LinuxVST/src/Beam/BeamProc.cpp new file mode 100755 index 0000000..5871eb5 --- /dev/null +++ b/plugins/LinuxVST/src/Beam/BeamProc.cpp @@ -0,0 +1,266 @@ +/* ======================================== + * Beam - Beam.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Beam_H +#include "Beam.h" +#endif + +void Beam::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 ); + float sonority = B * 1.618033988749894848204586; + 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; + float scaleFactor; + if (highres) scaleFactor = 8388608.0; + else scaleFactor = 32768.0; + float derez = C; + if (derez > 0.0) scaleFactor *= pow(1.0-derez,6); + if (scaleFactor < 0.0001) scaleFactor = 0.0001; + float outScale = scaleFactor; + if (outScale < 8.0) outScale = 8.0; + + 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; + + inputSampleL *= scaleFactor; + inputSampleR *= scaleFactor; + //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 expectedSlewA = 0; + for(int x = 0; x < depth; x++) { + expectedSlewA += (lastSampleL[x+1] - lastSampleL[x]); + } + float expectedSlewB = expectedSlewA; + expectedSlewA += (lastSampleL[0] - quantA); + expectedSlewB += (lastSampleL[0] - quantB); + //now we have a collection of all slews, averaged and left at total scale + + float clamp = sonority; + if (fabs(inputSampleL) < sonority) clamp = fabs(inputSampleL); + + float testA = fabs(fabs(expectedSlewA)-clamp); + float testB = fabs(fabs(expectedSlewB)-clamp); + //doing this means the result will be lowest when it's reaching the target slope across + //the desired time range, either positively or negatively. Should produce the same target + //at whatever sample rate, as high rate stuff produces smaller increments. + + 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 + + expectedSlewA = 0; + for(int x = 0; x < depth; x++) { + expectedSlewA += (lastSampleR[x+1] - lastSampleR[x]); + } + expectedSlewB = expectedSlewA; + expectedSlewA += (lastSampleR[0] - quantA); + expectedSlewB += (lastSampleR[0] - quantB); + //now we have a collection of all slews, averaged and left at total scale + + clamp = sonority; + if (fabs(inputSampleR) < sonority) clamp = fabs(inputSampleR); + + testA = fabs(fabs(expectedSlewA)-clamp); + testB = fabs(fabs(expectedSlewB)-clamp); + //doing this means the result will be lowest when it's reaching the target slope across + //the desired time range, either positively or negatively. Should produce the same target + //at whatever sample rate, as high rate stuff produces smaller increments. + + 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 + + inputSampleL /= outScale; + inputSampleR /= outScale; + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Beam::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 ); + float sonority = B * 1.618033988749894848204586; + 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; + float scaleFactor; + if (highres) scaleFactor = 8388608.0; + else scaleFactor = 32768.0; + float derez = C; + if (derez > 0.0) scaleFactor *= pow(1.0-derez,6); + if (scaleFactor < 0.0001) scaleFactor = 0.0001; + float outScale = scaleFactor; + if (outScale < 8.0) outScale = 8.0; + + 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; + + inputSampleL *= scaleFactor; + inputSampleR *= scaleFactor; + //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 expectedSlewA = 0; + for(int x = 0; x < depth; x++) { + expectedSlewA += (lastSampleL[x+1] - lastSampleL[x]); + } + float expectedSlewB = expectedSlewA; + expectedSlewA += (lastSampleL[0] - quantA); + expectedSlewB += (lastSampleL[0] - quantB); + //now we have a collection of all slews, averaged and left at total scale + + float clamp = sonority; + if (fabs(inputSampleL) < sonority) clamp = fabs(inputSampleL); + + float testA = fabs(fabs(expectedSlewA)-clamp); + float testB = fabs(fabs(expectedSlewB)-clamp); + //doing this means the result will be lowest when it's reaching the target slope across + //the desired time range, either positively or negatively. Should produce the same target + //at whatever sample rate, as high rate stuff produces smaller increments. + + 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 + + expectedSlewA = 0; + for(int x = 0; x < depth; x++) { + expectedSlewA += (lastSampleR[x+1] - lastSampleR[x]); + } + expectedSlewB = expectedSlewA; + expectedSlewA += (lastSampleR[0] - quantA); + expectedSlewB += (lastSampleR[0] - quantB); + //now we have a collection of all slews, averaged and left at total scale + + clamp = sonority; + if (fabs(inputSampleR) < sonority) clamp = fabs(inputSampleR); + + testA = fabs(fabs(expectedSlewA)-clamp); + testB = fabs(fabs(expectedSlewB)-clamp); + //doing this means the result will be lowest when it's reaching the target slope across + //the desired time range, either positively or negatively. Should produce the same target + //at whatever sample rate, as high rate stuff produces smaller increments. + + 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 + + inputSampleL /= outScale; + inputSampleR /= outScale; + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} |