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author | Chris Johnson <jinx6568@sover.net> | 2018-07-15 21:49:47 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-07-15 21:49:47 -0400 |
commit | 4fead686b5cfb33bf2a2e41700134a4efd6f8fcf (patch) | |
tree | e0a3977e260d426e5dd23bd4862bb5f4d3d0c277 /plugins/LinuxVST/src/StereoFX | |
parent | 6dd0cc75eef5294133c324ca225275247923cccd (diff) | |
download | airwindows-lv2-port-4fead686b5cfb33bf2a2e41700134a4efd6f8fcf.tar.gz airwindows-lv2-port-4fead686b5cfb33bf2a2e41700134a4efd6f8fcf.tar.bz2 airwindows-lv2-port-4fead686b5cfb33bf2a2e41700134a4efd6f8fcf.zip |
StereoFX
Diffstat (limited to 'plugins/LinuxVST/src/StereoFX')
-rwxr-xr-x | plugins/LinuxVST/src/StereoFX/StereoFX.cpp | 140 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/StereoFX/StereoFX.h | 71 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp | 330 |
3 files changed, 541 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/StereoFX/StereoFX.cpp b/plugins/LinuxVST/src/StereoFX/StereoFX.cpp new file mode 100755 index 0000000..63c23db --- /dev/null +++ b/plugins/LinuxVST/src/StereoFX/StereoFX.cpp @@ -0,0 +1,140 @@ +/* ======================================== + * StereoFX - StereoFX.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __StereoFX_H +#include "StereoFX.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new StereoFX(audioMaster);} + +StereoFX::StereoFX(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 0.0; + C = 0.0; + iirSampleA = 0.0; + iirSampleB = 0.0; + fpNShapeLA = 0.0; + fpNShapeLB = 0.0; + fpNShapeRA = 0.0; + fpNShapeRB = 0.0; + flip = false; + //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 +} + +StereoFX::~StereoFX() {} +VstInt32 StereoFX::getVendorVersion () {return 1000;} +void StereoFX::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::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 StereoFX::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Wide", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "MonoBs", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "CSquish", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void StereoFX::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string (A, text, kVstMaxParamStrLen); break; + 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 StereoFX::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 StereoFX::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool StereoFX::getEffectName(char* name) { + vst_strncpy(name, "StereoFX", kVstMaxProductStrLen); return true; +} + +VstPlugCategory StereoFX::getPlugCategory() {return kPlugCategEffect;} + +bool StereoFX::getProductString(char* text) { + vst_strncpy (text, "airwindows StereoFX", kVstMaxProductStrLen); return true; +} + +bool StereoFX::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/StereoFX/StereoFX.h b/plugins/LinuxVST/src/StereoFX/StereoFX.h new file mode 100755 index 0000000..6748d1e --- /dev/null +++ b/plugins/LinuxVST/src/StereoFX/StereoFX.h @@ -0,0 +1,71 @@ +/* ======================================== + * StereoFX - StereoFX.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __StereoFX_H +#define __StereoFX_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 = 'stfx'; //Change this to what the AU identity is! + +class StereoFX : + public AudioEffectX +{ +public: + StereoFX(audioMasterCallback audioMaster); + ~StereoFX(); + 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 iirSampleA; + double iirSampleB; + long double fpNShapeLA; + long double fpNShapeLB; + long double fpNShapeRA; + long double fpNShapeRB; + bool flip; + //default stuff + + float A; + float B; + float C; +}; + +#endif diff --git a/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp b/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp new file mode 100755 index 0000000..49be44f --- /dev/null +++ b/plugins/LinuxVST/src/StereoFX/StereoFXProc.cpp @@ -0,0 +1,330 @@ +/* ======================================== + * StereoFX - StereoFX.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __StereoFX_H +#include "StereoFX.h" +#endif + +void StereoFX::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(); + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + long double inputSampleL; + long double inputSampleR; + long double mid; + long double side; + //High Impact section + double stereowide = A; + double centersquish = C; + double density = stereowide * 2.4; + double sustain = 1.0 - (1.0/(1.0 + (density/7.0))); + //this way, enhance increases up to 50% and then mid falls off beyond that + double bridgerectifier; + double count; + //Highpass section + double iirAmount = pow(B,3)/overallscale; + double tight = -0.33333333333333; + double offset; + //we are setting it up so that to either extreme we can get an audible sound, + //but sort of scaled so small adjustments don't shift the cutoff frequency yet. + + 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. + } + //assign working variables + mid = inputSampleL + inputSampleR; + side = inputSampleL - inputSampleR; + //assign mid and side. Now, High Impact code + count = density; + while (count > 1.0) + { + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (side > 0.0) side = bridgerectifier; + else side = -bridgerectifier; + count = count - 1.0; + } + //we have now accounted for any really high density settings. + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (side > 0) side = (side*(1-count))+(bridgerectifier*count); + else side = (side*(1-count))-(bridgerectifier*count); + //blend according to density control + //done first density. Next, sustain-reducer + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = (1-cos(bridgerectifier))*3.141592653589793; + if (side > 0) side = (side*(1-sustain))+(bridgerectifier*sustain); + else side = (side*(1-sustain))-(bridgerectifier*sustain); + //done with High Impact code + + //now, Highpass code + offset = 0.666666666666666 + ((1-fabs(side))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (flip) + { + iirSampleA = (iirSampleA * (1 - (offset * iirAmount))) + (side * (offset * iirAmount)); + side = side - iirSampleA; + } + else + { + iirSampleB = (iirSampleB * (1 - (offset * iirAmount))) + (side * (offset * iirAmount)); + side = side - iirSampleB; + } + //done with Highpass code + + bridgerectifier = fabs(mid)/1.273239544735162; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier)*1.273239544735162; + if (mid > 0) mid = (mid*(1-centersquish))+(bridgerectifier*centersquish); + else mid = (mid*(1-centersquish))-(bridgerectifier*centersquish); + //done with the mid saturating section. + + inputSampleL = (mid+side)/2.0; + inputSampleR = (mid-side)/2.0; + + //noise shaping to 32-bit floating point + if (flip) { + 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; + } + flip = !flip; + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void StereoFX::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 fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + long double inputSampleL; + long double inputSampleR; + long double mid; + long double side; + //High Impact section + double stereowide = A; + double centersquish = C; + double density = stereowide * 2.4; + double sustain = 1.0 - (1.0/(1.0 + (density/7.0))); + //this way, enhance increases up to 50% and then mid falls off beyond that + double bridgerectifier; + double count; + //Highpass section + double iirAmount = pow(B,3)/overallscale; + double tight = -0.33333333333333; + double offset; + //we are setting it up so that to either extreme we can get an audible sound, + //but sort of scaled so small adjustments don't shift the cutoff frequency yet. + + 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. + } + //assign working variables + mid = inputSampleL + inputSampleR; + side = inputSampleL - inputSampleR; + //assign mid and side. Now, High Impact code + count = density; + while (count > 1.0) + { + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (side > 0.0) side = bridgerectifier; + else side = -bridgerectifier; + count = count - 1.0; + } + //we have now accounted for any really high density settings. + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (side > 0) side = (side*(1-count))+(bridgerectifier*count); + else side = (side*(1-count))-(bridgerectifier*count); + //blend according to density control + //done first density. Next, sustain-reducer + bridgerectifier = fabs(side)*1.57079633; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = (1-cos(bridgerectifier))*3.141592653589793; + if (side > 0) side = (side*(1-sustain))+(bridgerectifier*sustain); + else side = (side*(1-sustain))-(bridgerectifier*sustain); + //done with High Impact code + + //now, Highpass code + offset = 0.666666666666666 + ((1-fabs(side))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (flip) + { + iirSampleA = (iirSampleA * (1 - (offset * iirAmount))) + (side * (offset * iirAmount)); + side = side - iirSampleA; + } + else + { + iirSampleB = (iirSampleB * (1 - (offset * iirAmount))) + (side * (offset * iirAmount)); + side = side - iirSampleB; + } + //done with Highpass code + + bridgerectifier = fabs(mid)/1.273239544735162; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier)*1.273239544735162; + if (mid > 0) mid = (mid*(1-centersquish))+(bridgerectifier*centersquish); + else mid = (mid*(1-centersquish))-(bridgerectifier*centersquish); + //done with the mid saturating section. + + inputSampleL = (mid+side)/2.0; + inputSampleR = (mid-side)/2.0; + + //noise shaping to 64-bit floating point + if (flip) { + 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; + } + flip = !flip; + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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