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How to capture pc sound and send it via sockets using java

I am trying to capture audio from pc(from speaker/headphones) and send it via socket(UDP, if possible) to another computer, which must play it back. I have found some code to do this:
Server:
import javax.sound.sampled.*;
import java.io.BufferedInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.net.ServerSocket;
import java.net.Socket;
public class Server {
ServerSocket MyService;
Socket clientSocket = null;
InputStream input;
AudioFormat audioFormat;
SourceDataLine sourceDataLine;
byte tempBuffer[] = new byte[10000];
static Mixer.Info[] mixerInfo = AudioSystem.getMixerInfo();
Server() throws LineUnavailableException {
try {
Mixer mixer_ = AudioSystem.getMixer(mixerInfo[0]);
audioFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, audioFormat);
sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceDataLine.open(audioFormat);
sourceDataLine.start();
MyService = new ServerSocket(500);
clientSocket = MyService.accept();
input = new BufferedInputStream(clientSocket.getInputStream());
while (input.read(tempBuffer) != -1) {
sourceDataLine.write(tempBuffer, 0, 10000);
}
} catch (IOException e) {
e.printStackTrace();
}
}
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 8;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
return new AudioFormat(
sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}
public static void main(String s[]) throws LineUnavailableException {
Server s2 = new Server();
}}
Client:
import javax.sound.sampled.*;
import java.io.BufferedInputStream;
import java.io.BufferedOutputStream;
import java.net.Socket;
public class Client {
boolean stopCapture = false;
AudioFormat audioFormat;
TargetDataLine targetDataLine;
BufferedOutputStream out = null;
BufferedInputStream in = null;
Socket sock = null;
public static void main(String[] args) {
Client tx = new Client();
tx.captureAudio();
}
private void captureAudio() {
try {
sock = new Socket("192.168.1.38", 500);
out = new BufferedOutputStream(sock.getOutputStream());
in = new BufferedInputStream(sock.getInputStream());
Mixer.Info[] mixerInfo = AudioSystem.getMixerInfo();
audioFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(
TargetDataLine.class, audioFormat);
Mixer mixer = AudioSystem.getMixer(mixerInfo[2]);
targetDataLine = (TargetDataLine) mixer.getLine(dataLineInfo);
targetDataLine.open(audioFormat);
targetDataLine.start();
Thread captureThread = new CaptureThread();
captureThread.start();
} catch (Exception e) {
e.printStackTrace();
System.exit(0);
}
}
class CaptureThread extends Thread {
byte tempBuffer[] = new byte[10000];
#Override
public void run() {
stopCapture = false;
try {
while (!stopCapture) {
int cnt = targetDataLine.read(tempBuffer, 0,
tempBuffer.length);
out.write(tempBuffer);
}
} catch (Exception e) {
e.printStackTrace();
System.exit(0);
}
}
}
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 8;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed,
bigEndian);
}}
But client throws
java.lang.IllegalArgumentException: Line unsupported: interface TargetDataLine supporting format PCM_SIGNED 8000.0 Hz, 8 bit, mono, 1 bytes/frame,
at java.desktop/com.sun.media.sound.DirectAudioDevice.getLine(DirectAudioDevice.java:175)
at Client.captureAudio(Client.java:28)
at Client.main(Client.java:15)
and i do not know what to do(I know it is not UDP socket, but i firstly want to have some code which work). Advance thanks.

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 to play a mono sound with Java in a 4.0 speakers configuration

I need to play a .wav mono file in Java (possibly preloading it in RAM) to all the speakers of a 4.0 speakers configuration. I also need to adjust the volume of each of the channels independently while the audio is playing, leaving the system volume unchanged.
This is my current code. It's nowhere near what I need.
How can I do that? Thanks.
import java.io.File;
import java.io.IOException;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.Mixer;
import javax.sound.sampled.SourceDataLine;
public class AudioPlayer extends Thread {
private final int BUFFER_SIZE = 128000;
private File soundFile;
private AudioInputStream audioStream;
private AudioFormat audioFormat;
private SourceDataLine sourceLine;
String filename;
Mixer mixer;
public void load(String filename) {
this.filename = filename;
}
public AudioPlayer() {
}
public void run() {
try {
soundFile = new File(filename);
audioStream = AudioSystem.getAudioInputStream(soundFile);
audioFormat = audioStream.getFormat();
DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(info);
sourceLine.open(audioFormat);
} 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();
}
}

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);
}
}
}
}

getting mp3 audio signal as array in java

I have been trying to get audio stream of mp3 as array of floating points. I have got the array with the below sample code. I am not sure whether I can use this array for applying FFT. Because this array is not matching[or similar] to the one which I got from C++'s code which uses LAME.
import java.io.File;
import java.io.IOException;
import java.io.PrintStream;
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.UnsupportedAudioFileException;
import javazoom.spi.mpeg.sampled.file.MpegAudioFileReader;
import org.tritonus.share.sampled.FloatSampleTools;
public class onjava {
public static void main(String[] args) {
try {
File file = new File("mymp3file.mp3");
MpegAudioFileReader mpegAudioFileReader = new MpegAudioFileReader();
int fl = mpegAudioFileReader.getAudioFileFormat(file).getFrameLength();
System.out.println(fl);
AudioInputStream in= AudioSystem.getAudioInputStream(file);
AudioInputStream din = null;
AudioFormat baseFormat = in.getFormat();
onjava oj = new onjava();
AudioFormat decodedFormat =
new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
baseFormat.getSampleRate(),
16,
baseFormat.getChannels(),
baseFormat.getChannels() * 2,
baseFormat.getSampleRate(),
false);
int len = (int)file.length();
din = AudioSystem.getAudioInputStream(decodedFormat, in);
oj.rawplay(decodedFormat, din, len, fl);
in.close();
} catch (IOException e) {
e.printStackTrace();
} catch (LineUnavailableException e) {
e.printStackTrace();
} catch (UnsupportedAudioFileException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
private void rawplay(AudioFormat targetFormat, AudioInputStream din , int len, int frameLength)
throws IOException, LineUnavailableException {
byte[] data = new byte[len-1];
float[] floatArray = new float[len-1];
SourceDataLine line = getLine(targetFormat);
File textFile = new File("outputfile.txt");
PrintStream printStream = new PrintStream(textFile);
System.setOut(printStream);
if (line != null) {
// Start
line.start();
int nBytesRead = 0, nBytesWritten = 0;
while (nBytesRead != -1) {
nBytesRead = din.read(data, 0, 8000);
if (nBytesRead != -1) {
//Please tell me if something is wrong with the arguments passed below
FloatSampleTools.byte2floatGeneric(data, 0, targetFormat.getFrameSize(), floatArray, 0, 1000, targetFormat);
for (int i = 0; i < nBytesRead; i++) {
if(floatArray[i] != 0.0)
System.out.println(floatArray[i]);
}
}
}
// Stop
line.drain();
line.stop();
line.close();
din.close();
}
}
private SourceDataLine getLine(AudioFormat audioFormat)
throws LineUnavailableException {
SourceDataLine res = null;
DataLine.Info info = new DataLine.Info(SourceDataLine.class,
audioFormat);
res = (SourceDataLine) AudioSystem.getLine(info);
res.open(audioFormat);
return res;
}
}
Please suggest if anything is wrong in the above code. Also if any other pure java API is available for processing mp3 file. I need array of floats from mp3 audio stream.
Also let me know about pure java implementation of LAME if available.
Thanks!!

javazoom.spi.mpeg.sampled.file.MpegAudioFileReader
&
org.tritonus.share.sampled.FloatSampleTools
are not found. Specify the library you have used for those functions.