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Capture 8kHz, 16-bit Linear Samples with 4 frames of 20ms audio in each RTP Packet

I need to write simple Java Client program to capture live audio streaming.
Requirement
RTP Audio Packets.
8kHz, 16-bit Linear Samples (Linear PCM).
4 frames of 20ms audio will be sent in each RTP Packet.
After some search I found sample code on internet to capture the audio but it play beep sound.
Code
import java.io.ByteArrayInputStream;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.SourceDataLine;
public class Server {
AudioInputStream audioInputStream;
static AudioInputStream ais;
static AudioFormat format;
static boolean status = true;
static int port = 31007;
static int sampleRate = 44100;
static DataLine.Info dataLineInfo;
static SourceDataLine sourceDataLine;
public static void main(String args[]) throws Exception
{
System.out.println("Server started at port:"+port);
#SuppressWarnings("resource")
DatagramSocket serverSocket = new DatagramSocket(port);
/**
* Formula for lag = (byte_size/sample_rate)*2
* Byte size 9728 will produce ~ 0.45 seconds of lag. Voice slightly broken.
* Byte size 1400 will produce ~ 0.06 seconds of lag. Voice extremely broken.
* Byte size 4000 will produce ~ 0.18 seconds of lag. Voice slightly more broken then 9728.
*/
byte[] receiveData = new byte[4096];
format = new AudioFormat(sampleRate, 16, 2, true, false);
dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceDataLine.open(format);
sourceDataLine.start();
//FloatControl volumeControl = (FloatControl) sourceDataLine.getControl(FloatControl.Type.MASTER_GAIN);
//volumeControl.setValue(1.00f);
DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);
ByteArrayInputStream baiss = new ByteArrayInputStream(receivePacket.getData());
while (status == true)
{
System.out.println("Reciving Packets");
serverSocket.receive(receivePacket);
ais = new AudioInputStream(baiss, format, receivePacket.getLength());
toSpeaker(receivePacket.getData());
}
sourceDataLine.drain();
sourceDataLine.close();
}
public static void toSpeaker(byte soundbytes[]) {
try
{
System.out.println("At the speaker");
sourceDataLine.write(soundbytes, 0, soundbytes.length);
} catch (Exception e) {
System.out.println("Not working in speakers...");
e.printStackTrace();
}
}
}
I think I can not able to find the proper format to capture packets send in given format ?
Can any one help me to find find the proper AudioFormat to capture this audio streaming or any link pointing to same will be helpful for me... Thanks... :)
Answer
float sampleRate = 8000;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
UDP + RTP Packet Format
While buffering minus 12 bytes from data as it contains RTP header information.
receivePacket = new DatagramPacket(receiveData, receiveData.length);
byte[] packet = new byte[receivePacket.getLength() - 12];
serverSocket.receive(receivePacket);
packet = Arrays.copyOfRange(receivePacket.getData(), 12, receivePacket.getLength());
hope this will help you in future or feel free to correct if its wrong Thanks..

You can try this implementation of Client and Server based on Datagram Sockets. It uses a mono 8000Hz 16bit signed big endian audio format. Server is running on port number 9786, while the client is using port number 8786. I guess the code is quite simple to understand.
Server:
import java.io.*;
import java.net.*;
import javax.sound.sampled.*;
public class Server {
ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
}
public static void main(String args[]) {
new Server().runVOIP();
}
public void runVOIP() {
try {
DatagramSocket serverSocket = new DatagramSocket(9786);
byte[] receiveData = new byte[4096];
while (true) {
DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);
serverSocket.receive(receivePacket);
System.out.println("RECEIVED: " + receivePacket.getAddress().getHostAddress() + " " + receivePacket.getPort());
try {
byte audioData[] = receivePacket.getData();
InputStream byteInputStream = new ByteArrayInputStream(audioData);
AudioFormat adFormat = getAudioFormat();
InputStream = new AudioInputStream(byteInputStream, adFormat, audioData.length / adFormat.getFrameSize());
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, adFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceLine.open(adFormat);
sourceLine.start();
Thread playThread = new Thread(new PlayThread());
playThread.start();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
} catch (Exception e) {
e.printStackTrace();
}
}
class PlayThread extends Thread {
byte tempBuffer[] = new byte[4096];
public void run() {
try {
int cnt;
while ((cnt = InputStream.read(tempBuffer, 0, tempBuffer.length)) != -1) {
if (cnt > 0) {
sourceLine.write(tempBuffer, 0, cnt);
}
}
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
}
}
Client:
import java.io.*;
import java.net.*;
import javax.sound.sampled.*;
public class Client {
boolean stopaudioCapture = false;
ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;
public static void main(String args[]) {
new Client();
}
public Client() {
captureAudio();
}
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
}
private void captureAudio() {
try {
adFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(TargetDataLine.class, adFormat);
targetDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
targetDataLine.open(adFormat);
targetDataLine.start();
Thread captureThread = new Thread(new CaptureThread());
captureThread.start();
} catch (Exception e) {
StackTraceElement stackEle[] = e.getStackTrace();
for (StackTraceElement val : stackEle) {
System.out.println(val);
}
System.exit(0);
}
}
class CaptureThread extends Thread {
byte tempBuffer[] = new byte[4096];
#Override
public void run() {
stopaudioCapture = false;
try {
DatagramSocket clientSocket = new DatagramSocket(8786);
InetAddress IPAddress = InetAddress.getByName("127.0.0.1");
int cnt;
while (!stopaudioCapture) {
cnt = targetDataLine.read(tempBuffer, 0, tempBuffer.length);
if (cnt > 0) {
DatagramPacket sendPacket = new DatagramPacket(tempBuffer, tempBuffer.length, IPAddress, 9786);
clientSocket.send(sendPacket);
}
}
} catch (Exception e) {
System.out.println("CaptureThread::run()" + e);
System.exit(0);
}
}
}
}

How can I print sound as byte array? or int array?

I'm making anti-phase sound with java.(anti-phase is reflected wave. x-coordinate is not changed, but y-coordinate is upside down.)
Before reflecting sound wave, I have to get byte array(or int array) from sound.
I get sound from microphone of my laptop.
Here is my CODE(I got original code, which record sound as file, in web. I modified it little)
public class NoiseController extends Thread{
private TargetDataLine line;
private AudioInputStream audioInputStream;
public NoiseController(TargetDataLine line) {
this.line = line;
this.audioInputStream = new AudioInputStream(line);
}
public void start() {
line.start();
super.start();
}
public void stopRecording() {
line.stop();
line.close();
}
public void run() {
try {
int packet;
while((packet = audioInputStream.read()) != -1)
System.out.println(packet);
}
catch(IOException ioe) {
ioe.getStackTrace();
}
}
public static void main(String[] args) {
AudioFormat audioFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100.0F, 16, 2, 4, 44100.0F, false);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, audioFormat);
TargetDataLine targetDataLine = null;
try {
targetDataLine = (TargetDataLine)AudioSystem.getLine(info);
targetDataLine.open(audioFormat);
}
catch(LineUnavailableException lue) {
out("unable to get a recording line");
lue.printStackTrace();
System.exit(-1);
}
AudioFileFormat.Type targetType = AudioFileFormat.Type.WAVE;
NoiseController recorder = new NoiseController(targetDataLine);
System.out.println(targetDataLine);
System.out.println(targetType);
out("Press ENTER to start the recording.");
try {
System.in.read();
}
catch(IOException ioe) {
ioe.printStackTrace();
}
recorder.start();
out("Recording...");
out("Press ENTER to stop the recording.");
try {
System.in.read();
System.in.read();
}
catch(IOException ioe) {
ioe.getStackTrace();
}
recorder.stopRecording();
out("Recording stopped.");
}
private static void out(String msg) {
System.out.println(msg);
}
}
However, Console doesn't print anything while recording...
It shows just
com.sun.media.sound.DirectAudioDevice$DirectTDL#25154f_
WAVE
Press ENTER to start the recording.
Recording...
Press ENTER to stop the recording.
Recording stopped.
If I edit run() like AudioSystem.write(stream, fileType, out);
instead of
int packet;
while((packet = audioInputStream.read()) != -1)
System.out.println(packet);
program saves wav file.
What is wrong in my program?

you are not printing the Exception as Uwe Allner said.
I´ve also try to correct it and I think the result must be like this:
public class NoiseController extends Thread {
private final TargetDataLine line;
private final AudioInputStream audioInputStream;
public NoiseController(final TargetDataLine line) {
this.line = line;
this.audioInputStream = new AudioInputStream(line);
}
#Override
public void start() {
line.start();
super.start();
}
public void stopRecording() {
line.stop();
line.close();
try {
audioInputStream.close();
} catch (final IOException e) {
e.printStackTrace();
}
}
#Override
public void run() {
try {
final int bufferSize = 1024;
int read = 0;
final byte[] frame = new byte[bufferSize];
while ((read = audioInputStream.read(frame)) != -1 && line.isOpen()) {
// only the first read bytes are valid
System.out.println(Arrays.toString(frame));
}
} catch (final IOException ioe) {
ioe.printStackTrace();
}
}
public static void main(final String[] args) {
final AudioFormat audioFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100.0F, 16, 2, 4, 44100.0F, false);
final DataLine.Info info = new DataLine.Info(TargetDataLine.class, audioFormat);
TargetDataLine targetDataLine = null;
try {
targetDataLine = (TargetDataLine) AudioSystem.getLine(info);
targetDataLine.open(audioFormat);
} catch (final LineUnavailableException lue) {
out("unable to get a recording line");
lue.printStackTrace();
System.exit(-1);
}
final AudioFileFormat.Type targetType = AudioFileFormat.Type.WAVE;
final NoiseController recorder = new NoiseController(targetDataLine);
System.out.println(targetDataLine);
System.out.println(targetType);
out("Press ENTER to start the recording.");
try {
System.in.read();
} catch (final IOException ioe) {
ioe.printStackTrace();
}
recorder.start();
out("Recording...");
out("Press ENTER to stop the recording.");
try {
System.in.read();
System.in.read();
} catch (final IOException ioe) {
ioe.printStackTrace();
}
recorder.stopRecording();
out("Recording stopped.");
}
private static void out(final String msg) {
System.out.println(msg);
}
}

The Most efficient way of playing wav sound Java

I am developing a drum metronome. So I have .wav sound files and I need to play them with using minimum computer memory and CPU because it's very important for the metronome to play the sample on the exact time. Currently I use the code from this link
How to play .wav files with java
class MakeSound {
private final int BUFFER_SIZE = 128000;
private File soundFile;
private AudioInputStream audioStream;
private AudioFormat audioFormat;
private SourceDataLine sourceLine;
/**
* #param filename the name of the file that is going to be played
*/
public void playSound(String filename){
String strFilename = filename;
try {
soundFile = new File(strFilename);
} catch (Exception e) {
e.printStackTrace();
System.exit(1);
}
try {
audioStream = AudioSystem.getAudioInputStream(soundFile);
} catch (Exception e){
e.printStackTrace();
System.exit(1);
}
audioFormat = audioStream.getFormat();
DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
try {
sourceLine = (SourceDataLine) AudioSystem.getLine(info);
sourceLine.open(audioFormat);
} catch (LineUnavailableException e) {
e.printStackTrace();
System.exit(1);
} catch (Exception e) {
e.printStackTrace();
System.exit(1);
}
sourceLine.start();
int nBytesRead = 0;
byte[] abData = new byte[BUFFER_SIZE];
while (nBytesRead != -1) {
try {
nBytesRead = audioStream.read(abData, 0, abData.length);
} catch (IOException e) {
e.printStackTrace();
}
if (nBytesRead >= 0) {
#SuppressWarnings("unused")
int nBytesWritten = sourceLine.write(abData, 0, nBytesRead);
}
}
sourceLine.drain();
sourceLine.close();
}
}
This code works but it's too slow for using it in a metronome app. So which is the fastest way of playing wav sound files. Please take into account that sometimes I need to play them simultaneously so the sound should play as a separate thread I think.
Thanks

i am assuming that you have a working method that looks like this:
example #1: (don't do this)
public void startMetronome(){
boolean abort = false;
String audoFileName = new String("myAudioFile);
do{
playSound(audoFileName );
while(abort != false);
}
or maybe you've done some better implementation like this:
exapmle#2: (don't do this either)
Runnable r = new Runnable(){
boolean abort = false;
String audoFileName = new String("myAudioFile);
do{
playSound(audoFileName );
try{
Thread.sleep(500);
}catch (Exception e);
while(abort != false);
}
new Thread(r).start();
in any case you're doing a big mistake: you initialize every time you play a sound the sound line and every time you load the file again and again.
but thats a toally wrong approach, you have to load the file once, open the line once and play the sound repetially on the line!
(cheap) solution #3:
adjust your playSoundmethode like this:
public void playSound(String filename){
//same as above
String strFilename = filename;
soundFile = new File(strFilename); //i've shorened the catch/throws, to make it more readable
audioStream = AudioSystem.getAudioInputStream(soundFile);
audioFormat = audioStream.getFormat();
DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(info);
sourceLine.open(audioFormat);
sourceLine.start();
//right now, the line is set up and all data is availible
boolean isRunning = true;
while(isRunning){ //this is an endless loop!
int nBytesRead = 0;
byte[] abData = new byte[BUFFER_SIZE];
while (nBytesRead != -1) {
nBytesRead = audioStream.read(abData, 0, abData.length);
//and your timing here
try{
Thread.sleep(500);
}catch (Exception e);
}
sourceLine.drain();
sourceLine.close();
}
and best practice would be to play the whole thing in an seperate thread... (as example #2)

Send audio over UDP

So, I was looking for a microphone data sending tut, but I haven't found any.
So I read Oracles tut about line opening and I am able to record the audio to a ByteArrayOutputStream, but now I have 2 problems!
First:
How to play the recorded audio.
Second: if I am recording it to a BAOS how would i dynamically send it.
I suppose I would send the data array, but would it be too processor hoggy to write to a BAOS every time I recieve it, or could I do it differently?
Current code:
import java.io.ByteArrayOutputStream;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
public class MicrophoneRecorder {
static boolean stopped = false;
public static void main(String[] args) {
AudioFormat format = new AudioFormat(8000.0f, 16, 1, true, true);
TargetDataLine line = null;
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
if (!AudioSystem.isLineSupported(info)) {
System.out.println("Not supported!");
}
try {
line = (TargetDataLine) AudioSystem.getLine(info);
line.open(format);
} catch (LineUnavailableException ex) {
ex.printStackTrace();
}
ByteArrayOutputStream out = new ByteArrayOutputStream();
int numBytesRead = 0;
byte[] data = new byte[line.getBufferSize() / 5];
line.start();
new Thread(new Runnable() {
#Override
public void run() {
try {
Thread.sleep(3000);
} catch (InterruptedException e) {
e.printStackTrace();
}
stopped = true;
}
}).start();
while (!stopped) {
numBytesRead = line.read(data, 0, data.length);
out.write(data, 0, numBytesRead);
}
}
}
Thanks for any help given.
Sincerely, Roberto Anić Banić
P.S.
Seen this, doesn't work http://javasolution.blogspot.com/2007/04/voice-chat-using-java.html
P.P.S.
Is UDP a good soulution or should I use RTSP

Here is a sample code that helped me in order to stream and consuming audio via UDP. You can changed the infinite loop in order to limit the duration of the audio stream. Below is the client and server code. The audio input is from microphone.
server:
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;
public class Sender {
public static void main(String[] args) throws IOException {
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, true);
TargetDataLine microphone;
SourceDataLine speakers;
try {
microphone = AudioSystem.getTargetDataLine(format);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
microphone = (TargetDataLine) AudioSystem.getLine(info);
microphone.open(format);
ByteArrayOutputStream out = new ByteArrayOutputStream();
int numBytesRead;
int CHUNK_SIZE = 1024;
byte[] data = new byte[microphone.getBufferSize() / 5];
microphone.start();
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
speakers = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
speakers.open(format);
speakers.start();
// Configure the ip and port
String hostname = "localhost";
int port = 5555;
InetAddress address = InetAddress.getByName(hostname);
DatagramSocket socket = new DatagramSocket();
byte[] buffer = new byte[1024];
for(;;) {
numBytesRead = microphone.read(data, 0, CHUNK_SIZE);
// bytesRead += numBytesRead;
// write the mic data to a stream for use later
out.write(data, 0, numBytesRead);
// write mic data to stream for immediate playback
speakers.write(data, 0, numBytesRead);
DatagramPacket request = new DatagramPacket(data,numBytesRead, address, port);
socket.send(request);
}
} catch (LineUnavailableException e) {
e.printStackTrace();
}
}}
client:
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.SocketTimeoutException;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;
public class UdpClient {
public static void main(String[] args) throws LineUnavailableException {
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, true);
TargetDataLine microphone;
SourceDataLine speakers;
microphone = AudioSystem.getTargetDataLine(format);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
microphone = (TargetDataLine) AudioSystem.getLine(info);
microphone.open(format);
ByteArrayOutputStream out = new ByteArrayOutputStream();
int numBytesRead;
int CHUNK_SIZE = 1024;
byte[] data = new byte[microphone.getBufferSize() / 5];
microphone.start();
int bytesRead = 0;
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
speakers = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
speakers.open(format);
speakers.start();
String hostname = "localhost";
int port = 5555;
try {
InetAddress address = InetAddress.getByName(hostname);
DatagramSocket socket = new DatagramSocket();
DatagramSocket serverSocket = new DatagramSocket(5555);
byte[] receiveData = new byte[1024];
byte[] sendData = new byte[1024];
while (true) {
byte[] buffer = new byte[1024];
DatagramPacket response = new DatagramPacket(buffer, buffer.length);
serverSocket.receive(response);
out.write(response.getData(), 0, response.getData().length);
speakers.write(response.getData(), 0, response.getData().length);
String quote = new String(buffer, 0, response.getLength());
System.out.println(quote);
System.out.println();
//Thread.sleep(10000);
}
} catch (SocketTimeoutException ex) {
System.out.println("Timeout error: " + ex.getMessage());
ex.printStackTrace();
} catch (IOException ex) {
System.out.println("Client error: " + ex.getMessage());
ex.printStackTrace();
}/* catch (InterruptedException ex) {
ex.printStackTrace();
}*/
}}

Here's an implementation of sending audio over UDP.
Below is the client and server code. Basically the client code sends captured audio to the server, which plays it on receiving. The client can also play the captured audio.
Client code: VUClient.java
import javax.swing.*;
import java.awt.*;
import java.awt.event.*;
import java.io.*;
import java.net.*;
import javax.sound.sampled.*;
public class VUClient extends JFrame {
boolean stopaudioCapture = false;
ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;
Graphics g;
public static void main(String args[]) {
new VUClient();
}
public VUClient() {
final JButton capture = new JButton("Capture");
final JButton stop = new JButton("Stop");
final JButton play = new JButton("Playback");
capture.setEnabled(true);
stop.setEnabled(false);
play.setEnabled(false);
capture.addActionListener(new ActionListener() {
public void actionPerformed(ActionEvent e) {
capture.setEnabled(false);
stop.setEnabled(true);
play.setEnabled(false);
captureAudio();
}
});
getContentPane().add(capture);
stop.addActionListener(new ActionListener() {
public void actionPerformed(ActionEvent e) {
capture.setEnabled(true);
stop.setEnabled(false);
play.setEnabled(true);
stopaudioCapture = true;
targetDataLine.close();
}
});
getContentPane().add(stop);
play.addActionListener(new ActionListener() {
#Override
public void actionPerformed(ActionEvent e) {
playAudio();
}
});
getContentPane().add(play);
getContentPane().setLayout(new FlowLayout());
setTitle("Capture/Playback Demo");
setDefaultCloseOperation(EXIT_ON_CLOSE);
setSize(400, 100);
getContentPane().setBackground(Color.white);
setVisible(true);
g = (Graphics) this.getGraphics();
}
private void captureAudio() {
try {
adFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(TargetDataLine.class, adFormat);
targetDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
targetDataLine.open(adFormat);
targetDataLine.start();
Thread captureThread = new Thread(new CaptureThread());
captureThread.start();
} catch (Exception e) {
StackTraceElement stackEle[] = e.getStackTrace();
for (StackTraceElement val : stackEle) {
System.out.println(val);
}
System.exit(0);
}
}
private void playAudio() {
try {
byte audioData[] = byteOutputStream.toByteArray();
InputStream byteInputStream = new ByteArrayInputStream(audioData);
AudioFormat adFormat = getAudioFormat();
InputStream = new AudioInputStream(byteInputStream, adFormat, audioData.length / adFormat.getFrameSize());
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, adFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceLine.open(adFormat);
sourceLine.start();
Thread playThread = new Thread(new PlayThread());
playThread.start();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
private AudioFormat getAudioFormat() {
float sampleRate = 16000.0F;
int sampleInbits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
return new AudioFormat(sampleRate, sampleInbits, channels, signed, bigEndian);
}
class CaptureThread extends Thread {
byte tempBuffer[] = new byte[10000];
public void run() {
byteOutputStream = new ByteArrayOutputStream();
stopaudioCapture = false;
try {
DatagramSocket clientSocket = new DatagramSocket(8786);
InetAddress IPAddress = InetAddress.getByName("127.0.0.1");
while (!stopaudioCapture) {
int cnt = targetDataLine.read(tempBuffer, 0, tempBuffer.length);
if (cnt > 0) {
DatagramPacket sendPacket = new DatagramPacket(tempBuffer, tempBuffer.length, IPAddress, 9786);
clientSocket.send(sendPacket);
byteOutputStream.write(tempBuffer, 0, cnt);
}
}
byteOutputStream.close();
} catch (Exception e) {
System.out.println("CaptureThread::run()" + e);
System.exit(0);
}
}
}
class PlayThread extends Thread {
byte tempBuffer[] = new byte[10000];
public void run() {
try {
int cnt;
while ((cnt = InputStream.read(tempBuffer, 0, tempBuffer.length)) != -1) {
if (cnt > 0) {
sourceLine.write(tempBuffer, 0, cnt);
}
}
// sourceLine.drain();
// sourceLine.close();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
}
}
Server code: VUServer.java
import java.io.*;
import java.net.*;
import javax.sound.sampled.*;
public class VUServer {
ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;
private AudioFormat getAudioFormat() {
float sampleRate = 16000.0F;
int sampleInbits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
return new AudioFormat(sampleRate, sampleInbits, channels, signed, bigEndian);
}
public static void main(String args[]) {
new VUServer().runVOIP();
}
public void runVOIP() {
try {
DatagramSocket serverSocket = new DatagramSocket(9786);
byte[] receiveData = new byte[10000];
while (true) {
DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);
serverSocket.receive(receivePacket);
System.out.println("RECEIVED: " + receivePacket.getAddress().getHostAddress() + " " + receivePacket.getPort());
try {
byte audioData[] = receivePacket.getData();
InputStream byteInputStream = new ByteArrayInputStream(audioData);
AudioFormat adFormat = getAudioFormat();
InputStream = new AudioInputStream(byteInputStream, adFormat, audioData.length / adFormat.getFrameSize());
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, adFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceLine.open(adFormat);
sourceLine.start();
Thread playThread = new Thread(new PlayThread());
playThread.start();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
} catch (Exception e) {
e.printStackTrace();
}
}
class PlayThread extends Thread {
byte tempBuffer[] = new byte[10000];
public void run() {
try {
int cnt;
while ((cnt = InputStream.read(tempBuffer, 0, tempBuffer.length)) != -1) {
if (cnt > 0) {
sourceLine.write(tempBuffer, 0, cnt);
}
}
// sourceLine.drain();
// sourceLine.close();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
}
}

Java Sound not very clear when streaming

Hi I've been writing a chat client and wanted to test the Java Sound API. I've managed to get sound working from the mic to the speakers on different computers via UDP. However the sound isn't very clear. To check whether this was because of lost packets etc in the UDP protocol I wrote a small test for the sound to go to the speakers on the same machine as the mic. The sound isn't any different which makes me think I have some settings wrong for reading or writing the sound. Can anybody have a look at my code and tell me how to make the sound clearer?
package test;
import java.awt.*;
import java.awt.event.*;
import java.io.*;
import javax.sound.sampled.*;
import javax.swing.*;
#SuppressWarnings("serial")
public class VoiceTest extends JFrame {
private JButton chat = new JButton("Voice");
private GUIListener gl = new GUIListener();
private IncomingSoundListener isl = new IncomingSoundListener();
private OutgoingSoundListener osl = new OutgoingSoundListener();
private boolean inVoice = true;
private boolean outVoice = false;
AudioFormat format = getAudioFormat();
ByteArrayOutputStream baos = new ByteArrayOutputStream();
public VoiceTest() throws IOException {
super ("Test");
//new Thread(tl).start();
new Thread(isl).start();
Container contentPane = this.getContentPane();
this.setSize(200,100);
this.setLocationRelativeTo(null);
this.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
chat.setBounds(10,10,80,30);
chat.addActionListener(gl);
contentPane.add(chat);
this.setVisible(true);
}
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
//AudioFormat.Encoding.ULAW
return new AudioFormat(sampleRate, sampleSizeBits, channels, signed, bigEndian);
}
class GUIListener implements ActionListener {
public void actionPerformed(ActionEvent actionevent) {
String action = actionevent.getActionCommand();
switch (action) {
case "Mute":
outVoice = false;
chat.setText("Voice");
break;
case "Voice":
new Thread(osl).start();
outVoice = true;
chat.setText("Mute");
break;
}
}
}
class IncomingSoundListener implements Runnable {
#Override
public void run() {
try {
System.out.println("Listening for incoming sound");
DataLine.Info speakerInfo = new DataLine.Info(SourceDataLine.class, format);
SourceDataLine speaker = (SourceDataLine) AudioSystem.getLine(speakerInfo);
speaker.open(format);
speaker.start();
while(inVoice) {
byte[] data = baos.toByteArray();
baos.reset();
ByteArrayInputStream bais = new ByteArrayInputStream(data);
AudioInputStream ais = new AudioInputStream(bais,format,data.length);
int numBytesRead = 0;
if ((numBytesRead = ais.read(data)) != -1) speaker.write(data, 0, numBytesRead);
ais.close();
bais.close();
}
speaker.drain();
speaker.close();
System.out.println("Stopped listening for incoming sound");
} catch (Exception e) {
e.printStackTrace();
}
}
}
class OutgoingSoundListener implements Runnable {
#Override
public void run() {
try {
System.out.println("Listening for outgoing sound");
DataLine.Info micInfo = new DataLine.Info(TargetDataLine.class, format);
TargetDataLine mic = (TargetDataLine) AudioSystem.getLine(micInfo);
mic.open(format);
byte tmpBuff[] = new byte[mic.getBufferSize()/5];
mic.start();
while(outVoice) {
int count = mic.read(tmpBuff,0,tmpBuff.length);
if (count > 0) baos.write(tmpBuff, 0, count);
}
mic.drain();
mic.close();
System.out.println("Stopped listening for outgoing sound");
} catch (Exception e) {
e.printStackTrace();
}
}
}
/**
* #param args
* #throws IOException
*/
public static void main(String[] args) throws IOException {
new VoiceTest();
}
}

You should try higher sampling rates and try to find acceptable quality/size ratio for your audio stream.
Checking the AudioFormat reference is also a good start for getting the idea.
Try changing local variables in your getAudioFormat() method to this:
private AudioFormat getAudioFormat() {
float sampleRate = 16000.0F;
int sampleSizeBits = 16;
int channels = 1;
...
}
This is equivalent to a 256 kbps Mono audio file.