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92
SoundSourceDraw.java
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92
SoundSourceDraw.java
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//package soundsourceloc;
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import java.awt.BasicStroke;
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import java.awt.BorderLayout;
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import java.awt.Dimension;
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import java.awt.Graphics;
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import java.awt.Graphics2D;
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import java.io.BufferedReader;
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import java.io.IOException;
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import java.io.InputStream;
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import java.io.InputStreamReader;
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import javax.swing.JFrame;
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import javax.swing.JPanel;
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public class SoundSourceDraw extends JFrame {
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private static final long serialVersionUID = 1L;
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private final SoundLocDraw _soundLocDraw;
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public SoundSourceDraw() throws Exception {
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super("Sound Source Localization");
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_soundLocDraw = new SoundLocDraw();
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getContentPane().add(_soundLocDraw, BorderLayout.CENTER);
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}
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public void run() throws IOException {
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ProcessBuilder pb = new ProcessBuilder("/home/quentin/Documents/Projet_localisation/code_v1.03/sound-source-loc");
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pb = pb.redirectErrorStream(true);
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Process p = pb.start();
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InputStream is = p.getInputStream();
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InputStreamReader isr = new InputStreamReader(is);
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BufferedReader br = new BufferedReader(isr);
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String line;
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while (( line = br.readLine()) != null) {
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int sep=line.indexOf(';');
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float angle=Float.parseFloat(line.substring(0,sep));
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float relativePower=Float.parseFloat(line.substring(sep+1));
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//System.out.println("received sound loc: "+line);
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_soundLocDraw.setSound(angle,relativePower);
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}
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}
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@SuppressWarnings("serial")
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static private class SoundLocDraw extends JPanel {
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// sound angle, between -PI/2...+PI/2
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private float _angle;
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// relative power with respect to mean power (1.0=mean power)
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private float _relativePower;
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public void setSound(float angle, float relativePower) {
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_angle = angle;
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_relativePower = relativePower;
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repaint();
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}
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@Override
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protected void paintComponent(Graphics g) {
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super.paintComponent(g);
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Graphics2D g2d = (Graphics2D) g;
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Dimension d = getSize();
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int radius = Math.min(d.height, d.width / 2);
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int cx = d.width / 2;
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int cy = 0;
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int tx = cx + (int) (Math.cos(_angle + Math.PI / 2) * radius);
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int ty = cy + (int) (Math.sin(_angle + Math.PI / 2) * radius);
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g2d.drawOval(cx - radius, cy - radius, radius * 2, radius * 2);
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// use larger strokes for louder sounds:
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g2d.setStroke(new BasicStroke(1 + (int) ((Math.max(_relativePower,
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1) - 1.0) * 10)));
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g2d.drawLine(cx, cy, tx, ty);
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}
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}
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/**
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* Entry point: create the frame, and start listening to sound until closed.
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*/
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public static void main(String[] args) throws Exception {
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SoundSourceDraw snd = new SoundSourceDraw();
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snd.setSize(800, 400);
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snd.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
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snd.setLocationRelativeTo(null);
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snd.setVisible(true);
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snd.run();
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}
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}
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225
sound-source-loc_v.1.0.3.cpp
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sound-source-loc_v.1.0.3.cpp
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#include <iostream>
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using namespace std;
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#include <stdio.h>
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#include <stdlib.h>
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#include <math.h>
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#include <limits.h>
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#include <alsa/asoundlib.h>
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#include "/usr/include/alsa/asoundlib.h"
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//#define SAMPLE_TYPE float
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//#define SAMPLE_TYPE_ALSA SND_PCM_FORMAT_FLOAT_LE
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#define SAMPLE_TYPE short
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#define SAMPLE_TYPE_ALSA SND_PCM_FORMAT_S16_LE
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class RunningAverage {
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int _nbValuesForAverage;
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int _nbValues;
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float _mean;
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public:
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RunningAverage(int nbValuesForAverage) {
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_nbValuesForAverage = nbValuesForAverage;
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_mean = 0;
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_nbValues = 0;
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}
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void newValue(SAMPLE_TYPE v) {
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if (_nbValues < _nbValuesForAverage)
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_nbValues++;
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_mean = ((_mean * (_nbValues - 1)) + v) / (float)_nbValues;
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}
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SAMPLE_TYPE getMean() {
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return (SAMPLE_TYPE) _mean;
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}
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};
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class SoundSourceLoc {
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static const int _nbSamplesMaxDiff = 13;
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static const int _bufferSize = 4096;
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static constexpr float _minLevelFactorForValidLoc = 1.05f;
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static constexpr float _soundSpeed = 344;
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/**
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* sound sampling rate in Hz
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*/
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unsigned int _soundSamplingRate;
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static constexpr float _distanceBetweenMicrophones = 0.1f;
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RunningAverage* _averageSoundLevel;
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snd_pcm_t* _capture_handle;
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SAMPLE_TYPE _rightBuffer[_bufferSize];
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SAMPLE_TYPE _leftBuffer[_bufferSize];
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public:
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SoundSourceLoc() {
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_averageSoundLevel = new RunningAverage(50);
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_soundSamplingRate = 44100;
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int err;
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snd_pcm_hw_params_t* hw_params;
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// ideally use "hw:0,0" for embedded, to limit processing. But check if card support our needs...
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const char* device = "plughw:1,0";
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if ((err = snd_pcm_open(&_capture_handle, device,
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SND_PCM_STREAM_CAPTURE, 0)) < 0) {
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fprintf(stderr, "cannot open audio device %s (%s)\n", device,
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snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
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fprintf(stderr,
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"cannot allocate hardware parameter structure (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_any(_capture_handle, hw_params)) < 0) {
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fprintf(stderr,
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"cannot initialize hardware parameter structure (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_set_access(_capture_handle, hw_params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED)) < 0) {
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fprintf(stderr, "cannot set access type (%s)\n", snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_set_format(_capture_handle, hw_params,
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SAMPLE_TYPE_ALSA)) < 0) {
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fprintf(stderr, "cannot set sample format (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_set_rate_near(_capture_handle, hw_params,
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&_soundSamplingRate, 0)) < 0) {
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fprintf(stderr, "cannot set sample rate (%s)\n", snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params_set_channels(_capture_handle, hw_params, 2))
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< 0) {
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fprintf(stderr, "cannot set channel count (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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if ((err = snd_pcm_hw_params(_capture_handle, hw_params)) < 0) {
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fprintf(stderr, "cannot set parameters (%s)\n", snd_strerror(err));
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exit(1);
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}
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snd_pcm_hw_params_free(hw_params);
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if ((err = snd_pcm_prepare(_capture_handle)) < 0) {
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fprintf(stderr, "cannot prepare audio interface for use (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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}
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/** Clean exit */
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~SoundSourceLoc() {
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snd_pcm_close(_capture_handle);
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delete _averageSoundLevel;
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}
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void run() {
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while (true) {
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processNextSoundBlock();
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}
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}
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private:
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void processNextSoundBlock() {
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SAMPLE_TYPE* bufs[2];
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bufs[0] = _rightBuffer;
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bufs[1] = _leftBuffer;
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int err;
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if ((err = snd_pcm_readn(_capture_handle, (void**) bufs, _bufferSize))
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!= _bufferSize) {
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fprintf(stderr, "read from audio interface failed (%s)\n",
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snd_strerror(err));
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exit(1);
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}
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SAMPLE_TYPE level = computeLevel(_rightBuffer, _leftBuffer);
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_averageSoundLevel->newValue(level);
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float relativeLevel = (float) level
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/ (float) _averageSoundLevel->getMean();
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int minDiff = INT_MAX;
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int minDiffTime = -1;
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for (int t = -_nbSamplesMaxDiff; t < _nbSamplesMaxDiff; t++) {
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int diff = 0;
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for (int i = _nbSamplesMaxDiff;
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i < _bufferSize - _nbSamplesMaxDiff - 1; i++) {
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diff += abs(_leftBuffer[i] - _rightBuffer[i + t]);
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}
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if (diff < minDiff) {
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minDiff = diff;
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minDiffTime = t;
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}
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}
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if ((relativeLevel > _minLevelFactorForValidLoc)
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&& (minDiffTime > -_nbSamplesMaxDiff)
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&& (minDiffTime < _nbSamplesMaxDiff)) {
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float angle =
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-(float) asin(
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(minDiffTime * _soundSpeed)
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/ (_soundSamplingRate
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* _distanceBetweenMicrophones));
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cout << angle << ";" << relativeLevel << endl;
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}
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}
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SAMPLE_TYPE computeLevel(SAMPLE_TYPE right[], SAMPLE_TYPE left[]) {
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float level = 0;
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for (int i = 0; i < _bufferSize; i++) {
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float s = (left[i] + right[i]) / 2;
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level += (s * s);
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}
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level /= _bufferSize;
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level = sqrt(level);
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return (SAMPLE_TYPE) level;
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}
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};
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int main(int argc, char *argv[]) {
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SoundSourceLoc soundLoc;
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soundLoc.run();
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exit(0);
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}
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