Transférer les fichiers vers ''

This commit is contained in:
QLeblanc 2020-05-25 21:51:11 +02:00
parent 7daafdbc0d
commit c645155ae2
2 changed files with 317 additions and 0 deletions

92
SoundSourceDraw.java Normal file
View file

@ -0,0 +1,92 @@
//package soundsourceloc;
import java.awt.BasicStroke;
import java.awt.BorderLayout;
import java.awt.Dimension;
import java.awt.Graphics;
import java.awt.Graphics2D;
import java.io.BufferedReader;
import java.io.IOException;
import java.io.InputStream;
import java.io.InputStreamReader;
import javax.swing.JFrame;
import javax.swing.JPanel;
public class SoundSourceDraw extends JFrame {
private static final long serialVersionUID = 1L;
private final SoundLocDraw _soundLocDraw;
public SoundSourceDraw() throws Exception {
super("Sound Source Localization");
_soundLocDraw = new SoundLocDraw();
getContentPane().add(_soundLocDraw, BorderLayout.CENTER);
}
public void run() throws IOException {
ProcessBuilder pb = new ProcessBuilder("/home/quentin/Documents/Projet_localisation/code_v1.03/sound-source-loc");
pb = pb.redirectErrorStream(true);
Process p = pb.start();
InputStream is = p.getInputStream();
InputStreamReader isr = new InputStreamReader(is);
BufferedReader br = new BufferedReader(isr);
String line;
while (( line = br.readLine()) != null) {
int sep=line.indexOf(';');
float angle=Float.parseFloat(line.substring(0,sep));
float relativePower=Float.parseFloat(line.substring(sep+1));
//System.out.println("received sound loc: "+line);
_soundLocDraw.setSound(angle,relativePower);
}
}
@SuppressWarnings("serial")
static private class SoundLocDraw extends JPanel {
// sound angle, between -PI/2...+PI/2
private float _angle;
// relative power with respect to mean power (1.0=mean power)
private float _relativePower;
public void setSound(float angle, float relativePower) {
_angle = angle;
_relativePower = relativePower;
repaint();
}
@Override
protected void paintComponent(Graphics g) {
super.paintComponent(g);
Graphics2D g2d = (Graphics2D) g;
Dimension d = getSize();
int radius = Math.min(d.height, d.width / 2);
int cx = d.width / 2;
int cy = 0;
int tx = cx + (int) (Math.cos(_angle + Math.PI / 2) * radius);
int ty = cy + (int) (Math.sin(_angle + Math.PI / 2) * radius);
g2d.drawOval(cx - radius, cy - radius, radius * 2, radius * 2);
// use larger strokes for louder sounds:
g2d.setStroke(new BasicStroke(1 + (int) ((Math.max(_relativePower,
1) - 1.0) * 10)));
g2d.drawLine(cx, cy, tx, ty);
}
}
/**
* Entry point: create the frame, and start listening to sound until closed.
*/
public static void main(String[] args) throws Exception {
SoundSourceDraw snd = new SoundSourceDraw();
snd.setSize(800, 400);
snd.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
snd.setLocationRelativeTo(null);
snd.setVisible(true);
snd.run();
}
}

View file

@ -0,0 +1,225 @@
#include <iostream>
using namespace std;
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <limits.h>
#include <alsa/asoundlib.h>
#include "/usr/include/alsa/asoundlib.h"
//#define SAMPLE_TYPE float
//#define SAMPLE_TYPE_ALSA SND_PCM_FORMAT_FLOAT_LE
#define SAMPLE_TYPE short
#define SAMPLE_TYPE_ALSA SND_PCM_FORMAT_S16_LE
class RunningAverage {
int _nbValuesForAverage;
int _nbValues;
float _mean;
public:
RunningAverage(int nbValuesForAverage) {
_nbValuesForAverage = nbValuesForAverage;
_mean = 0;
_nbValues = 0;
}
void newValue(SAMPLE_TYPE v) {
if (_nbValues < _nbValuesForAverage)
_nbValues++;
_mean = ((_mean * (_nbValues - 1)) + v) / (float)_nbValues;
}
SAMPLE_TYPE getMean() {
return (SAMPLE_TYPE) _mean;
}
};
class SoundSourceLoc {
static const int _nbSamplesMaxDiff = 13;
static const int _bufferSize = 4096;
static constexpr float _minLevelFactorForValidLoc = 1.05f;
static constexpr float _soundSpeed = 344;
/**
* sound sampling rate in Hz
*/
unsigned int _soundSamplingRate;
static constexpr float _distanceBetweenMicrophones = 0.1f;
RunningAverage* _averageSoundLevel;
snd_pcm_t* _capture_handle;
SAMPLE_TYPE _rightBuffer[_bufferSize];
SAMPLE_TYPE _leftBuffer[_bufferSize];
public:
SoundSourceLoc() {
_averageSoundLevel = new RunningAverage(50);
_soundSamplingRate = 44100;
int err;
snd_pcm_hw_params_t* hw_params;
// ideally use "hw:0,0" for embedded, to limit processing. But check if card support our needs...
const char* device = "plughw:1,0";
if ((err = snd_pcm_open(&_capture_handle, device,
SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf(stderr, "cannot open audio device %s (%s)\n", device,
snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
fprintf(stderr,
"cannot allocate hardware parameter structure (%s)\n",
snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_any(_capture_handle, hw_params)) < 0) {
fprintf(stderr,
"cannot initialize hardware parameter structure (%s)\n",
snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_set_access(_capture_handle, hw_params,
SND_PCM_ACCESS_RW_NONINTERLEAVED)) < 0) {
fprintf(stderr, "cannot set access type (%s)\n", snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_set_format(_capture_handle, hw_params,
SAMPLE_TYPE_ALSA)) < 0) {
fprintf(stderr, "cannot set sample format (%s)\n",
snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_set_rate_near(_capture_handle, hw_params,
&_soundSamplingRate, 0)) < 0) {
fprintf(stderr, "cannot set sample rate (%s)\n", snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params_set_channels(_capture_handle, hw_params, 2))
< 0) {
fprintf(stderr, "cannot set channel count (%s)\n",
snd_strerror(err));
exit(1);
}
if ((err = snd_pcm_hw_params(_capture_handle, hw_params)) < 0) {
fprintf(stderr, "cannot set parameters (%s)\n", snd_strerror(err));
exit(1);
}
snd_pcm_hw_params_free(hw_params);
if ((err = snd_pcm_prepare(_capture_handle)) < 0) {
fprintf(stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror(err));
exit(1);
}
}
/** Clean exit */
~SoundSourceLoc() {
snd_pcm_close(_capture_handle);
delete _averageSoundLevel;
}
void run() {
while (true) {
processNextSoundBlock();
}
}
private:
void processNextSoundBlock() {
SAMPLE_TYPE* bufs[2];
bufs[0] = _rightBuffer;
bufs[1] = _leftBuffer;
int err;
if ((err = snd_pcm_readn(_capture_handle, (void**) bufs, _bufferSize))
!= _bufferSize) {
fprintf(stderr, "read from audio interface failed (%s)\n",
snd_strerror(err));
exit(1);
}
SAMPLE_TYPE level = computeLevel(_rightBuffer, _leftBuffer);
_averageSoundLevel->newValue(level);
float relativeLevel = (float) level
/ (float) _averageSoundLevel->getMean();
int minDiff = INT_MAX;
int minDiffTime = -1;
for (int t = -_nbSamplesMaxDiff; t < _nbSamplesMaxDiff; t++) {
int diff = 0;
for (int i = _nbSamplesMaxDiff;
i < _bufferSize - _nbSamplesMaxDiff - 1; i++) {
diff += abs(_leftBuffer[i] - _rightBuffer[i + t]);
}
if (diff < minDiff) {
minDiff = diff;
minDiffTime = t;
}
}
if ((relativeLevel > _minLevelFactorForValidLoc)
&& (minDiffTime > -_nbSamplesMaxDiff)
&& (minDiffTime < _nbSamplesMaxDiff)) {
float angle =
-(float) asin(
(minDiffTime * _soundSpeed)
/ (_soundSamplingRate
* _distanceBetweenMicrophones));
cout << angle << ";" << relativeLevel << endl;
}
}
SAMPLE_TYPE computeLevel(SAMPLE_TYPE right[], SAMPLE_TYPE left[]) {
float level = 0;
for (int i = 0; i < _bufferSize; i++) {
float s = (left[i] + right[i]) / 2;
level += (s * s);
}
level /= _bufferSize;
level = sqrt(level);
return (SAMPLE_TYPE) level;
}
};
int main(int argc, char *argv[]) {
SoundSourceLoc soundLoc;
soundLoc.run();
exit(0);
}