此系列文章拖了N久，有好多人发邮件来询问我第五次的文章为什么没有写，其实非常抱歉，本人学生一个，暑假一直
去公司实习，最近又忙着各种招聘找工作，没有时间好好写，现在抽空把最后一篇补上，水平有限，如过有不对的，请
各位指正～
前四篇文章分别介绍了 “代码结构”，“程序流程”，以及”发送方的处理”，现在就把接收方的处理流程做个介绍；
如上图所示，接收方的操作有三个类：AudioDecoder(负责解码），AudioPlayer(负责播放解码后的音频），
AudioReceiver(负责从服务器接收音频数据包），这三个类的流程在第三篇中有详细的介绍。
1.AudioReceiver代码：
AudioReceiver使用UDP方式从服务端接收音频数据，其过程比较简单，直接上代码：
package xmu.swordbearer.audio.receiver;import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.SocketException;import xmu.swordbearer.audio.MyConfig;
import android.util.Log;public class AudioReceiver implements Runnable {String LOG = "NET Reciever ";int port = MyConfig.CLIENT_PORT;// 接收的端口DatagramSocket socket;DatagramPacket packet;boolean isRunning = false;private byte[] packetBuf = new byte[1024];private int packetSize = 1024;/** 开始接收数据*/public void startRecieving() {if (socket == null) {try {socket = new DatagramSocket(port);packet = new DatagramPacket(packetBuf, packetSize);} catch (SocketException e) {}}new Thread(this).start();}/** 停止接收数据*/public void stopRecieving() {isRunning = false;}/** 释放资源*/private void release() {if (packet != null) {packet = null;}if (socket != null) {socket.close();socket = null;}}public void run() {// 在接收前，要先启动解码器AudioDecoder decoder = AudioDecoder.getInstance();decoder.startDecoding();isRunning = true;try {while (isRunning) {socket.receive(packet);// 每接收一个UDP包，就交给解码器，等待解码decoder.addData(packet.getData(), packet.getLength());}} catch (IOException e) {Log.e(LOG, LOG + "RECIEVE ERROR!");}// 接收完成，停止解码器，释放资源decoder.stopDecoding();release();Log.e(LOG, LOG + "stop recieving");}}
2.AudioDecoder代码：
解码的过程也很简单，由于接收端接收到了音频数据，然后就把数据交给解码器，所以解码的主要工作就是把接收端的数
据取出来进行解码，如果解码正确，就将解码后的数据再转交给AudioPlayer去播放，这三个类之间是依次传递的 :
AudioReceiver---->AudioDecoder--->AudioPlayer
下面代码中有个List变量 private List<AudioData> dataList = null;这个就是用来存放数据的，每次解码时，dataList.remove(0)，
从最前端取出数据进行解码：
package xmu.swordbearer.audio.receiver;import java.util.Collections;
import java.util.LinkedList;
import java.util.List;import xmu.swordbearer.audio.AudioCodec;
import xmu.swordbearer.audio.data.AudioData;
import android.util.Log;public class AudioDecoder implements Runnable {String LOG = "CODEC Decoder ";private static AudioDecoder decoder;private static final int MAX_BUFFER_SIZE = 2048;private byte[] decodedData = new byte[1024];// data of decodedprivate boolean isDecoding = false;private List<AudioData> dataList = null;public static AudioDecoder getInstance() {if (decoder == null) {decoder = new AudioDecoder();}return decoder;}private AudioDecoder() {this.dataList = Collections.synchronizedList(new LinkedList<AudioData>());}/** add Data to be decoded* * @ data:the data recieved from server* * @ size:data size*/public void addData(byte[] data, int size) {AudioData adata = new AudioData();adata.setSize(size);byte[] tempData = new byte[size];System.arraycopy(data, 0, tempData, 0, size);adata.setRealData(tempData);dataList.add(adata);System.out.println(LOG + "add data once");}/** start decode AMR data*/public void startDecoding() {System.out.println(LOG + "start decoder");if (isDecoding) {return;}new Thread(this).start();}public void run() {// start player firstAudioPlayer player = AudioPlayer.getInstance();player.startPlaying();//this.isDecoding = true;// init ILBC parameter:30 ,20, 15AudioCodec.audio_codec_init(30);Log.d(LOG, LOG + "initialized decoder");int decodeSize = 0;while (isDecoding) {while (dataList.size() > 0) {AudioData encodedData = dataList.remove(0);decodedData = new byte[MAX_BUFFER_SIZE];byte[] data = encodedData.getRealData();//decodeSize = AudioCodec.audio_decode(data, 0,encodedData.getSize(), decodedData, 0);if (decodeSize > 0) {// add decoded audio to playerplayer.addData(decodedData, decodeSize);// clear datadecodedData = new byte[decodedData.length];}}}System.out.println(LOG + "stop decoder");// stop playback audioplayer.stopPlaying();}public void stopDecoding() {this.isDecoding = false;}
}3.AudioPlayer代码：
播放器的工作流程其实和解码器一模一样，都是启动一个线程，然后不断从自己的 dataList中提取数据。
不过要注意，播放器的一些参数配置非常的关键；
播放声音时，使用了Android自带的 AudioTrack 这个类，它有这个方法：
public int write(byte[] audioData,int offsetInBytes, int sizeInBytes)可以直接播放；
所有播放器的代码如下：
package xmu.swordbearer.audio.receiver;import java.util.Collections;
import java.util.LinkedList;
import java.util.List;import xmu.swordbearer.audio.data.AudioData;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioRecord;
import android.media.AudioTrack;
import android.util.Log;public class AudioPlayer implements Runnable {String LOG = "AudioPlayer ";private static AudioPlayer player;private List<AudioData> dataList = null;private AudioData playData;private boolean isPlaying = false;private AudioTrack audioTrack;private static final int sampleRate = 8000;// 注意：参数配置private static final int channelConfig = AudioFormat.CHANNEL_IN_MONO;private static final int audioFormat = AudioFormat.ENCODING_PCM_16BIT;private AudioPlayer() {dataList = Collections.synchronizedList(new LinkedList<AudioData>());}public static AudioPlayer getInstance() {if (player == null) {player = new AudioPlayer();}return player;}public void addData(byte[] rawData, int size) {AudioData decodedData = new AudioData();decodedData.setSize(size);byte[] tempData = new byte[size];System.arraycopy(rawData, 0, tempData, 0, size);decodedData.setRealData(tempData);dataList.add(decodedData);}/** init Player parameters*/private boolean initAudioTrack() {int bufferSize = AudioRecord.getMinBufferSize(sampleRate,channelConfig, audioFormat);if (bufferSize < 0) {Log.e(LOG, LOG + "initialize error!");return false;}audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,channelConfig, audioFormat, bufferSize, AudioTrack.MODE_STREAM);// set volume:设置播放音量audioTrack.setStereoVolume(1.0f, 1.0f);audioTrack.play();return true;}private void playFromList() {while (dataList.size() > 0 && isPlaying) {playData = dataList.remove(0);audioTrack.write(playData.getRealData(), 0, playData.getSize());}}public void startPlaying() {if (isPlaying) {return;}new Thread(this).start();}public void run() {this.isPlaying = true;if (!initAudioTrack()) {Log.e(LOG, LOG + "initialized player error!");return;}while (isPlaying) {if (dataList.size() > 0) {playFromList();} else {try {Thread.sleep(20);} catch (InterruptedException e) {}}}if (this.audioTrack != null) {if (this.audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {this.audioTrack.stop();this.audioTrack.release();}}Log.d(LOG, LOG + "end playing");}public void stopPlaying() {this.isPlaying = false;}
}
为了方便测试，我自己用Java 写了一个UDP的服务器，其功能非常的弱，就是接收，然后转发给另一方：
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.SocketException;
import java.net.UnknownHostException;public class AudioServer implements Runnable {DatagramSocket socket;DatagramPacket packet;// 从客户端接收到的UDP包DatagramPacket sendPkt;// 转发给另一个客户端的UDP包byte[] pktBuffer = new byte[1024];int bufferSize = 1024;boolean isRunning = false;int myport = 5656;// ///String clientIpStr = "192.168.1.104";InetAddress clientIp;int clientPort = 5757;public AudioServer() {try {clientIp = InetAddress.getByName(clientIpStr);} catch (UnknownHostException e1) {e1.printStackTrace();}try {socket = new DatagramSocket(myport);packet = new DatagramPacket(pktBuffer, bufferSize);} catch (SocketException e) {e.printStackTrace();}System.out.println("服务器初始化完成");}public void startServer() {this.isRunning = true;new Thread(this).start();}public void run() {try {while (isRunning) {socket.receive(packet);sendPkt = new DatagramPacket(packet.getData(),packet.getLength(), packet.getAddress(), clientPort);socket.send(sendPkt);try {Thread.sleep(20);} catch (InterruptedException e) {e.printStackTrace();}}} catch (IOException e) {}}// mainpublic static void main(String[] args) {new AudioServer().startServer();}
}
5.结语：
Android使用 ILBC 进行语音通话的大致过程就讲述完了，此系列只是做一个ILBC 使用原理的介绍，距离真正的语音
通话还有很多工作要做，缺点还是很多的：
1. 文章中介绍的只是单方通话，如果要做成双方互相通话或者一对多的通话，就需要增加更多的流程处理，其服务端
也要做很多工作；
2. 实时性：本程序在局域网中使用时，实时性还是较高的，但是再广域网中，效果可能会有所下降，除此之外，本
程序还缺少时间戳的处理，如果网络状况不理想，或者数据延迟，就会导致语音播放前后混乱；
3. 服务器很弱：真正的流媒体服务器，需要很强的功能，来对数据进行处理，我是为了方便，就写了一个简单的，
最近打算移植live555，用来做专门的流媒体服务器，用RTP协议对数据进行封装，这样效果应该会好很多。
现在，整个工程的代码都完成了，全部源代码可以在这里下载： http://download.csdn.net/detail/ranxiedao/4923759
BY:http://blog.csdn.net/ranxiedao