Merge pull request #55 from polimi-ispl/develop

Common libraries with Ebeam OSC controller
polimi-ispl · May 2, 2021 · e520a85 · e520a85
2 parents 9fbe68d + 7d37678
commit e520a85
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Showing 31 changed files with 502 additions and 2,399 deletions.
diff --git a/JuceLibraryCode/AppConfig.h b/JuceLibraryCode/AppConfig.h
@@ -48,6 +48,9 @@
 #define JUCE_PROJUCER_VERSION 0x60008
 
 //==============================================================================
+#define JUCE_MODULE_AVAILABLE_ebeamer_common                1
+#define JUCE_MODULE_AVAILABLE_ebeamer_gui                   1
+#define JUCE_MODULE_AVAILABLE_ebeamer_osc                   1
 #define JUCE_MODULE_AVAILABLE_juce_audio_basics             1
 #define JUCE_MODULE_AVAILABLE_juce_audio_devices            1
 #define JUCE_MODULE_AVAILABLE_juce_audio_formats            1

diff --git a/JuceLibraryCode/JuceHeader.h b/JuceLibraryCode/JuceHeader.h
@@ -14,6 +14,9 @@
 
 #include "AppConfig.h"
 
+#include <ebeamer_common/ebeamer_common.h>
+#include <ebeamer_gui/ebeamer_gui.h>
+#include <ebeamer_osc/ebeamer_osc.h>
 #include <juce_audio_basics/juce_audio_basics.h>
 #include <juce_audio_devices/juce_audio_devices.h>
 #include <juce_audio_formats/juce_audio_formats.h>
@@ -51,7 +54,7 @@ namespace ProjectInfo
 {
     const char* const  projectName    = "Ebeamer";
     const char* const  companyName    = "ISPL and Eventide";
-    const char* const  versionString  = "2.0.0";
-    const int          versionNumber  = 0x20000;
+    const char* const  versionString  = "2.1.0";
+    const int          versionNumber  = 0x20100;
 }
 #endif
diff --git a/JuceLibraryCode/JucePluginDefines.h b/JuceLibraryCode/JucePluginDefines.h
@@ -74,13 +74,13 @@
  #define JucePlugin_EditorRequiresKeyboardFocus  0
 #endif
 #ifndef  JucePlugin_Version
- #define JucePlugin_Version                2.0.0
+ #define JucePlugin_Version                2.1.0
 #endif
 #ifndef  JucePlugin_VersionCode
- #define JucePlugin_VersionCode            0x20000
+ #define JucePlugin_VersionCode            0x20100
 #endif
 #ifndef  JucePlugin_VersionString
- #define JucePlugin_VersionString          "2.0.0"
+ #define JucePlugin_VersionString          "2.1.0"
 #endif
 #ifndef  JucePlugin_VSTUniqueID
  #define JucePlugin_VSTUniqueID            JucePlugin_PluginCode

diff --git a/JuceLibraryCode/include_ebeamer_common.cpp b/JuceLibraryCode/include_ebeamer_common.cpp
@@ -0,0 +1,9 @@
+/*
+
+    IMPORTANT! This file is auto-generated each time you save your
+    project - if you alter its contents, your changes may be overwritten!
+
+*/
+
+#include "AppConfig.h"
+#include <ebeamer_common/ebeamer_common.cpp>
diff --git a/JuceLibraryCode/include_ebeamer_gui.cpp b/JuceLibraryCode/include_ebeamer_gui.cpp
@@ -0,0 +1,9 @@
+/*
+
+    IMPORTANT! This file is auto-generated each time you save your
+    project - if you alter its contents, your changes may be overwritten!
+
+*/
+
+#include "AppConfig.h"
+#include <ebeamer_gui/ebeamer_gui.cpp>
diff --git a/JuceLibraryCode/include_ebeamer_osc.cpp b/JuceLibraryCode/include_ebeamer_osc.cpp
@@ -0,0 +1,9 @@
+/*
+
+    IMPORTANT! This file is auto-generated each time you save your
+    project - if you alter its contents, your changes may be overwritten!
+
+*/
+
+#include "AppConfig.h"
+#include <ebeamer_osc/ebeamer_osc.cpp>
diff --git a/README.md b/README.md
@@ -10,6 +10,9 @@ A JUCE-based VST3 plug-in and standalone application to support the [Estick](htt
 - If you're developing on Windows, download and extract the [ASIO ASK](https://www.steinberg.net/en/company/developers.html) in the *ASIO/* folder.
 - Don't have the Esticks yet? Start developing with the [Estick simulator](https://github.com/luca-bondi/estick-simulator).
 
+## Dependencies
+- [Ebeamer JUCE modules](https://github.com/luca-bondi/ebeamer-modules)
+
 ## Contributing
 - Any contribution to the project is highly appreciated! Get in touch to know more.
 

diff --git a/Source/AudioBufferFFT.cpp b/Source/AudioBufferFFT.cpp
@@ -145,15 +145,49 @@ void AudioBufferFFT::prepareForConvolution() {
     }
 }
 
-void AudioBufferFFT::convolve(int outputChannel, const AudioBufferFFT &in_, int inChannel, AudioBufferFFT &filter_,
+void AudioBufferFFT::convolveAndAdd(int outputChannel, const AudioBufferFFT &in_, int inChannel, AudioBufferFFT &filter_,
                               int filterChannel) {
 
     jassert(in_.isReadyForConvolution());
     jassert(filter_.isReadyForConvolution());
 
-    clear(outputChannel, 0, getNumSamples());
     convolutionProcessingAndAccumulate(in_.getReadPointer(inChannel), filter_.getReadPointer(filterChannel),
                                        getWritePointer(outputChannel), fft->getSize());
 
     readyForConvolution = true;
 }
+
+void AudioBufferFFT::convolve(int outputChannel, const AudioBufferFFT &in_, int inChannel, AudioBufferFFT &filter_,
+                              int filterChannel) {
+
+    clear(outputChannel, 0, getNumSamples());
+    convolveAndAdd(outputChannel, in_, inChannel, filter_, filterChannel);
+}
+
+AudioBufferFFT& AudioBufferFFT::operator= (const AudioBufferFFT& other){
+
+    if (this != &other)
+    {
+        setSize (other.getNumChannels(), other.getNumSamples(), false, false, false);
+
+        if (other.hasBeenCleared())
+        {
+            clear();
+        }
+        else
+        {
+
+            for (int i = 0; i < getNumChannels(); ++i)
+                FloatVectorOperations::copy (getWritePointer(i), other.getReadPointer(i), getNumSamples());
+        }
+
+        readyForConvolution = other.readyForConvolution;
+        fft = other.fft;
+        convBuffer = other.convBuffer;
+
+    }
+
+    return *this;
+
+}
+
diff --git a/Source/AudioBufferFFT.h b/Source/AudioBufferFFT.h
@@ -32,24 +32,26 @@ class AudioBufferFFT : public AudioBuffer<float> {
 
     void
     convolve(int outputChannel, const AudioBufferFFT &in_, int inChannel, AudioBufferFFT &filter_, int filterChannel);
-
+
+    void
+    convolveAndAdd(int outputChannel, const AudioBufferFFT &in_, int inChannel, AudioBufferFFT &filter_, int filterChannel);
+
     void prepareForConvolution();
+
+    void updateSymmetricFrequency();
 
     bool isReadyForConvolution() const { return readyForConvolution; };
+
+    AudioBufferFFT& operator= (const AudioBufferFFT& other);
 
 private:
     AudioBuffer<float> convBuffer;
     std::shared_ptr<dsp::FFT> fft;
+    bool readyForConvolution = false;
 
     void prepareForConvolution(float *samples, int fftSize) const;
-
     void convolutionProcessingAndAccumulate(const float *input, const float *impulse, float *output, int fftSize) const;
-
     void updateSymmetricFrequencyDomainData(float *samples, int fftSize) const;
 
-    void updateSymmetricFrequency();
-
-    bool readyForConvolution = false;
-
 };
 
diff --git a/Source/Beamformer.cpp b/Source/Beamformer.cpp
@@ -7,32 +7,43 @@
 
 #include "Beamformer.h"
 
+#define NUM_DOAX 25
+#define NUM_DOAY 9
+
 BeamformerDoa::BeamformerDoa(Beamformer &b,
                              int numDoaHor_,
                              int numDoaVer_,
                              float sampleRate_,
                              int numActiveInputChannels,
                              int firLen,
+                             float expectedRate,
                              std::shared_ptr<dsp::FFT> fft_) : Thread("DOA"), beamformer(b) {
 
     numDoaHor = numDoaHor_;
     numDoaVer = numDoaVer_;
     fft = fft_;
     sampleRate = sampleRate_;
+    doaUpdateFrequency = expectedRate;
 
     /** Initialize levels and FIR */
     doaLevels.resize(numDoaVer,numDoaHor);
     doaLevels.setConstant(-100);
+    newDoaLevels.resize(numDoaVer,numDoaHor);
+    newDoaLevels.setConstant(-100);
     doaFirFFT.resize(numDoaHor*numDoaVer);
 
+    /** Time constants */
+    alpha = 1 - exp(-(1/doaUpdateFrequency) / timeConst);
+
     /** Allocate inputBuffer */
     inputBuffer = AudioBufferFFT(numActiveInputChannels, fft);
 
     /** Allocate convolution buffer */
     convolutionBuffer = AudioBufferFFT(1, fft);
 
-    /** Allocate DOA beam */
-    doaBeam.setSize(1, fft->getSize());
+    /* Determine frequency bins for energy average */
+    lowFreqIdx = lowFreq/sampleRate*fft->getSize();
+    numFreqBins = highFreq/sampleRate*fft->getSize() - lowFreqIdx;
 
     /** Compute FIR for DOA estimation */
     AudioBuffer<float> tmpFir(numActiveInputChannels, firLen);
@@ -56,6 +67,12 @@ void BeamformerDoa::run() {
 
     while (!threadShouldExit()){
 
+        /* Wait for previous doa to be consumed before computing a new one */
+        while (!threadShouldExit() && beamformer.isDoaOutputBufferNew())
+            sleep(5);
+        if (threadShouldExit())
+            return;
+
         const auto startTick = Time::getHighResolutionTicks();
 
         beamformer.getDoaInputBuffer(inputBuffer);
@@ -64,27 +81,35 @@ void BeamformerDoa::run() {
         for (auto vDirIdx = 0; vDirIdx < numDoaVer; vDirIdx++) {
             for (auto hDirIdx = 0; hDirIdx < numDoaHor; hDirIdx++) {
                 auto dirIdx = vDirIdx * numDoaHor + hDirIdx;
-                doaBeam.clear();
+
+                convolutionBuffer.clear();
                 for (auto inCh = 0; inCh < inputBuffer.getNumChannels(); inCh++) {
-                    /** Convolve inputs and DOA FIR */
-                    convolutionBuffer.convolve(0, inputBuffer, inCh, doaFirFFT[dirIdx], inCh);
-                    convolutionBuffer.addToTimeSeries(0, doaBeam, 0);
+                    /** Convolve inputs and DOA FIR and sum*/
+                    convolutionBuffer.convolveAndAdd(0, inputBuffer, inCh, doaFirFFT[dirIdx], inCh);
                 }
 
-                const Range<float> minMax = FloatVectorOperations::findMinAndMax(doaBeam.getReadPointer(0),
-                                                                                 doaBeam.getNumSamples());
-                const float dirEnergy = jmax(abs(minMax.getStart()), abs(minMax.getEnd()));
+                /** Back to regular FFT data */
+                convolutionBuffer.updateSymmetricFrequency();
+
+                std::complex<float>* cplxData = (std::complex<float>*)convolutionBuffer.getReadPointer(0);
+                Eigen::Map<CplxVec> doaBeamMap(cplxData,convolutionBuffer.getNumSamples()/2);
+
+                const float dirEnergy = doaBeamMap.segment(lowFreqIdx,numFreqBins).array().abs().sum()/float(numFreqBins);
                 const float dirEnergyDb = Decibels::gainToDecibels(dirEnergy);
-                doaLevels(vDirIdx,hDirIdx) = dirEnergyDb;
+                newDoaLevels(vDirIdx,hDirIdx) = dirEnergyDb;
             }
         }
+        doaLevels = (doaLevels * (1 - alpha)) + (newDoaLevels * alpha);
         beamformer.setDoaEnergy(doaLevels);
 
         const auto endTick = Time::getHighResolutionTicks();
         const float elapsedTime = Time::highResolutionTicksToSeconds(endTick-startTick);
-        const float expectedPeriod = 1.f/ENERGY_UPDATE_FREQ;
-        if (elapsedTime < expectedPeriod){
-            Time::waitForMillisecondCounter(Time::getMillisecondCounter() + (expectedPeriod-elapsedTime));
+        const float expectedPeriod = 1.f/doaUpdateFrequency;
+        const float sleepTime = expectedPeriod-elapsedTime;
+        if (sleepTime > 0){
+            sleep(expectedPeriod-elapsedTime);
+        }else{
+            //TODO: can't keep up, reduce complexity
         }
 
     }
@@ -95,7 +120,7 @@ BeamformerDoa::~BeamformerDoa(){
 }
 
 // ==============================================================================
-Beamformer::Beamformer(int numBeams_, MicConfig mic, double sampleRate_, int maximumExpectedSamplesPerBlock_) {
+Beamformer::Beamformer(int numBeams_, MicConfig mic, double sampleRate_, int maximumExpectedSamplesPerBlock_,float doaRefreshRate) {
 
     numBeams = numBeams_;
     numDoaVer = isLinearArray(mic) ? 1 : NUM_DOAY;
@@ -177,22 +202,12 @@ Beamformer::Beamformer(int numBeams_, MicConfig mic, double sampleRate_, int max
     beamBuffer.clear();
 
     /** Allocate DOA input buffer */
-    doaInputBuffer.setSize(numMic, maximumExpectedSamplesPerBlock);
-    doaInputBuffer.clear();
-
-    /** Set DOA input Filter  */
-    doaBPFilters.clear();
-    doaBPFilters.resize(numMic);
-    IIRCoefficients doaIIRCoeff = IIRCoefficients::makeBandPass(sampleRate, doaBPfreq, doaBPQ);
-    for (auto &f : doaBPFilters) {
-        f.setCoefficients(doaIIRCoeff);
-    }
+    doaInputBuffer = AudioBufferFFT(numMic, fft);
+    doaInputBuffer.prepareForConvolution();
 
     /** Prepare and start DOA thread */
-    doaThread = std::make_unique<BeamformerDoa>(*this, numDoaHor, numDoaVer, sampleRate, numMic, firLen, fft);
-#ifndef HEADLESS
+    doaThread = std::make_unique<BeamformerDoa>(*this, numDoaHor, numDoaVer, sampleRate, numMic, firLen, doaRefreshRate, fft);
     doaThread->startThread();
-#endif
 
 }
 
@@ -215,18 +230,16 @@ void Beamformer::setBeamParameters(int beamIdx, const BeamParameters &beamParams
 
 void Beamformer::processBlock(const AudioBuffer<float> &inBuffer) {
 
-    {
-        GenericScopedLock<SpinLock> lock(doaInputBufferLock);
-        for (auto chIdx = 0; chIdx < jmin(numMic, inBuffer.getNumChannels()); chIdx++) {
-            doaInputBuffer.copyFrom(chIdx, 0, inBuffer, chIdx, 0, inBuffer.getNumSamples());
-            doaBPFilters[chIdx].processSamples(doaInputBuffer.getWritePointer(chIdx), inBuffer.getNumSamples());
-        }
-    }
-
     /** Compute inputs FFT */
     inputBuffer.setTimeSeries(inBuffer);
     inputBuffer.prepareForConvolution();
 
+    if (!doaInputBufferNew){
+        GenericScopedLock<SpinLock> lock(doaInputBufferLock);
+        doaInputBuffer = inputBuffer;
+        doaInputBufferNew = true;
+    }
+
     for (auto beamIdx = 0; beamIdx < numBeams; beamIdx++) {
         for (auto inCh = 0; inCh < inputBuffer.getNumChannels(); inCh++) {
             /** Convolve inputs and FIR */
@@ -242,10 +255,10 @@ void Beamformer::getFir(AudioBuffer<float> &fir, const BeamParameters &params, f
     alg->getFir(fir, params, alpha);
 }
 
-void Beamformer::getDoaInputBuffer(AudioBufferFFT &dst) const {
+void Beamformer::getDoaInputBuffer(AudioBufferFFT &dst) {
     GenericScopedLock<SpinLock> lock(doaInputBufferLock);
-    dst.setTimeSeries(doaInputBuffer);
-    dst.prepareForConvolution();
+    dst = doaInputBuffer;
+    doaInputBufferNew = false;
 }
 
 void Beamformer::getBeams(AudioBuffer<float> &outBuffer) {
@@ -265,9 +278,27 @@ void Beamformer::getBeams(AudioBuffer<float> &outBuffer) {
 void Beamformer::setDoaEnergy(const Mtx &energy) {
     GenericScopedLock<SpinLock> lock(doaLock);
     doaLevels = energy;
+    doaOutputBufferNew = true;
 }
 
-void Beamformer::getDoaEnergy(Mtx &outDoaLevels) const {
+void Beamformer::getDoaEnergy(Mtx &outDoaLevels) {
     GenericScopedLock<SpinLock> lock(doaLock);
     outDoaLevels = doaLevels;
+    doaOutputBufferNew = false;
+}
+
+MemoryBlock Beamformer::getDoaEnergy(){
+    GenericScopedLock<SpinLock> lock(doaLock);
+
+    MemoryBlock mb(doaLevels.size()*sizeof(float)+2);
+    mb[0] = (uint8)doaLevels.rows();
+    mb[1] = (uint8)doaLevels.cols();
+    mb.copyFrom(doaLevels.data(), 2, doaLevels.size()*sizeof(float));
+
+    doaOutputBufferNew = false;
+    return mb;
+}
+
+bool Beamformer::isDoaOutputBufferNew() const{
+    return doaOutputBufferNew;
 }