I need to write simple Java Client program to capture live audio streaming.
Requirement
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);
}
}
}
}