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AudioStreaming mic to RTP server using java

I'd like to stream my microphone live to an rtp server which spread the stream to other clients.
It looks like someone tried this here 2 years ago. I tested the Client.java and Server.java, but the described problem in his Client.java still exists, no working microphone.
Client.java from another question:
Client:
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.SocketException;
import java.net.UnknownHostException;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
public class Mic
{
public byte[] buffer;
private int port;
static AudioInputStream ais;
public static void main(String[] args)
{
TargetDataLine line;
DatagramPacket dgp;
AudioFormat.Encoding encoding = AudioFormat.Encoding.PCM_SIGNED;
float rate = 44100.0f;
int channels = 2;
int sampleSize = 16;
boolean bigEndian = true;
InetAddress addr;
AudioFormat format = new AudioFormat(encoding, rate, sampleSize, channels, (sampleSize / 8) * channels, rate, bigEndian);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
if (!AudioSystem.isLineSupported(info)) {
System.out.println("Line matching " + info + " not supported.");
return;
}
try
{
line = (TargetDataLine) AudioSystem.getLine(info);
int buffsize = line.getBufferSize()/5;
buffsize += 512;
line.open(format);
line.start();
int numBytesRead;
byte[] data = new byte[buffsize];
addr = InetAddress.getByName("127.0.0.1");
DatagramSocket socket = new DatagramSocket();
while (true) {
// Read the next chunk of data from the TargetDataLine.
numBytesRead = line.read(data, 0, data.length);
// Save this chunk of data.
dgp = new DatagramPacket (data,data.length,addr,50005);
socket.send(dgp);
}
}catch (LineUnavailableException e) {
e.printStackTrace();
}catch (UnknownHostException e) {
// TODO: handle exception
} catch (SocketException e) {
// TODO: handle exception
} catch (IOException e2) {
// TODO: handle exception
}
}
}
I found that someone else got an working microphone client here, but it's just for local playback. What's the difference there so the microphone is working? I am new to java and cannot figure it out actually.
Client for playback from another question:
package audio;
import java.io.ByteArrayInputStream;
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.io.InputStream;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
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 AudioTest {
public static void main(String[] args) {
AudioFormat format = new AudioFormat(8000.0f, 16, 1, true, true);
TargetDataLine microphone;
AudioInputStream audioInputStream;
SourceDataLine sourceDataLine;
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();
int bytesRead = 0;
try {
while (bytesRead < 100000) { // Just so I can test if recording
// my mic works...
numBytesRead = microphone.read(data, 0, CHUNK_SIZE);
bytesRead = bytesRead + numBytesRead;
System.out.println(bytesRead);
out.write(data, 0, numBytesRead);
}
} catch (Exception e) {
e.printStackTrace();
}
byte audioData[] = out.toByteArray();
// Get an input stream on the byte array
// containing the data
InputStream byteArrayInputStream = new ByteArrayInputStream(
audioData);
audioInputStream = new AudioInputStream(byteArrayInputStream,format, audioData.length / format.getFrameSize());
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceDataLine.open(format);
sourceDataLine.start();
int cnt = 0;
byte tempBuffer[] = new byte[10000];
try {
while ((cnt = audioInputStream.read(tempBuffer, 0,tempBuffer.length)) != -1) {
if (cnt > 0) {
// Write data to the internal buffer of
// the data line where it will be
// delivered to the speaker.
sourceDataLine.write(tempBuffer, 0, cnt);
}// end if
}
} catch (IOException e) {
e.printStackTrace();
}
// Block and wait for internal buffer of the
// data line to empty.
sourceDataLine.drain();
sourceDataLine.close();
microphone.close();
} catch (LineUnavailableException e) {
e.printStackTrace();
}
}
}

Java broadcasting audio via UDP to Android

I'm doing an application to receive real time audio broadcasted by a Raspberry running a java server that I wrote and it have, necessarily, to be broadcasted to multiple clients. Now I'm using MulticastSocket to achieve it.
The problem so far is that the received audio is breaking and I dont know why. Another problem is with Marshmallow Android, it can't join a group of MulticastSocket, when other device joins the server, audio is automatically cut from Android.
Android side:
import android.content.Context;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
import android.util.Log;
import java.net.DatagramPacket;
import java.net.InetAddress;
import java.net.MulticastSocket;
public class Receiver implements Runnable {
private final String SERVER_ADDRESS = "224.0.0.3";
private final int SERVER_PORT = 48824;
private final int SAMPLE_RATE = 32000;
private final int PACKET_SIZE = 1200;
private final int CHANNEL_OUT_MONO = 4;
private final int CHANNEL_OUT_STEREO = 12;
private static boolean stopTrack = false;
public static AudioTrack track;
private static AudioManager am;
private static InetAddress inetAddress;
private static MulticastSocket multicastSocket;
private final int CHANNELS = CHANNEL_OUT_MONO;
#Override
public void run() {
//ESTABLISH CONNECTION
try{
inetAddress = InetAddress.getByName(SERVER_ADDRESS);
}catch(Exception ex){
Log.e("TCP","Error: server probably closed");
ex.printStackTrace();
}
//am = (AudioManager)Context.getSystemService(Context.AUDIO_SERVICE);
//int currentVolume = am.getStreamVolume(AudioManager.STREAM_MUSIC);
stopTrack = false;
int minSize = AudioTrack.getMinBufferSize(SAMPLE_RATE, CHANNELS, AudioFormat.ENCODING_PCM_16BIT);
Log.e("AUDIO","minSize = " + minSize);
track = new AudioTrack( AudioManager.STREAM_MUSIC, SAMPLE_RATE,
CHANNELS, AudioFormat.ENCODING_PCM_16BIT,
minSize, AudioTrack.MODE_STREAM);
track.play();
try{
multicastSocket = new MulticastSocket(SERVER_PORT);
multicastSocket.joinGroup(inetAddress);
//multicastSocket.setReuseAddress(true);
//multicastSocket.setBroadcast(true);
byte[] buffer = new byte[PACKET_SIZE];
while(!stopTrack){
DatagramPacket packet = new DatagramPacket(buffer, buffer.length);
multicastSocket.receive(packet);
track.write(packet.getData(), 0, buffer.length);
//Log.e("ERRO","SOCKET");
}
//multicastSocket.close();
} catch (Exception e) {
Log.e("UDP", "Receiver error", e);
}
}
public static void stopLiveAudioStream(){
track.stop();
multicastSocket.close();
stopTrack = true;
}
}
and the UDP Server:
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.Mixer;
import javax.sound.sampled.Port;
import java.util.*;
import java.net.*;
import java.io.*;
public class MulticastUDPServer {
private static final String INET_ADDR = "224.0.0.3";
private static final int SERVER_PORT = 48824;
private static final int BUFFER_SIZE = 4096;
private static final int PACKET_SIZE = 1200;
private static DatagramSocket socket;
private static InetAddress addr;
public static void main(String[] args){
Mixer.Info minfo[] = AudioSystem.getMixerInfo();
System.out.println("STARTING SERVER!");
try{
addr = InetAddress.getByName(INET_ADDR);
}catch(Exception ex){
System.out.println("ADDRESS ERROR: " + INET_ADDR);
}
try(DatagramSocket serverSocket = new DatagramSocket()){
AudioFormat format = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(TargetDataLine.class, format);
DataLine.Info sourceInfo = new DataLine.Info(SourceDataLine.class, format);
SourceDataLine sourceDataLine;
TargetDataLine targetDataLine = (TargetDataLine)AudioSystem.getLine(dataLineInfo);
targetDataLine.open(format, BUFFER_SIZE);
targetDataLine.start();
sourceDataLine = (SourceDataLine) AudioSystem.getLine(sourceInfo);
sourceDataLine.open(format, BUFFER_SIZE);
sourceDataLine.start();
byte[] targetData = new byte[PACKET_SIZE];
System.out.println("\n\nBROADCASTING!");
while(true){
targetDataLine.read(targetData,0,targetData.length);
serverSocket.send(new DatagramPacket(targetData, targetData.length, addr, SERVER_PORT));
}
}catch(Exception ex){
ex.printStackTrace();
}
}
private static AudioFormat getAudioFormat() {
float sampleRate = 32000.0F;
int sampleSizeBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
return new AudioFormat(sampleRate, sampleSizeBits, channels, signed, bigEndian);
}
}
Thanks in advance.

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);
}
}
}
}

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.