Android ilbc 语音对话示范（四）发送方代码

来源：互联网发布：apache tiles 编辑：程序博客网时间：2024/04/30 11:51

上一文章中提到：
发送端有三个主要的类：AudioRecorder(负责音频采集），AudioEncoder(负责音频编码），AudioSender(负责将编码后的数据发送出去）；这三个类中各有一个线程，录制开始后，这三个线程一起运行，分别执行各自的任务， AudioRecorder采集音频后，添加到AudioEncoder 的音频数据的List中，而AudioEncoder 的编码线程不断从List头部取出数据，调用 ilbc 的底层\函数进行编码，编码后的数据则又添加到下一级的AudioSender的 List中，AudioSender又不断从头部取出数据，然后发送出去；

1. 先建立一个 AudioData的类，代表一段音频数据：

public class AudioData {
int size;
byte[] realData;
//long timestamp;
public int getSize() {
return size;
}
public void setSize(int size) {
this.size = size;
}
public byte[] getRealData() {
return realData;
}
public void setRealData(byte[] realData) {
this.realData = realData;
}
//public long getTimestamp() {
// return timestamp;
// }
//
// public void setTimestamp(long timestamp) {
// this.timestamp = timestamp;
// }
}

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2.AudioRecorder 类，使用Android系统自带的AudioRecord来采集音频，每采集一次，就交给编码器编码。

public class AudioRecorder implements Runnable {
String LOG = "Recorder ";
private boolean isRecording = false;
private AudioRecord audioRecord;
private static final int audioSource = MediaRecorder.AudioSource.MIC;
private static final int sampleRate = 8000;
private static final int channelConfig = AudioFormat.CHANNEL_IN_MONO;
private static final int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
private static final int BUFFER_FRAME_SIZE =960;
private int audioBufSize = 0;
//
private byte[] samples;// 缓冲区
private int bufferRead = 0;// 从recorder中读取的samples的大小
private int bufferSize = 0;// samples的大小
// 开始录制
public void startRecording() {
bufferSize = BUFFER_FRAME_SIZE;
audioBufSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig,
audioFormat);
if (audioBufSize == AudioRecord.ERROR_BAD_VALUE) {
Log.e(LOG, "audioBufSize error");
return;
}
samples = new byte[audioBufSize];
// 初始化recorder
if (null == audioRecord) {
audioRecord = new AudioRecord(audioSource, sampleRate,
channelConfig, audioFormat, audioBufSize);
}
new Thread(this).start();
}
// 停止录制
public void stopRecording() {
this.isRecording = false;
}
public boolean isRecording() {
return isRecording;
}
// run
public void run() {
// 录制前，先启动解码器
AudioEncoder encoder = AudioEncoder.getInstance();
encoder.startEncoding();
System.out.println(LOG + "audioRecord startRecording()");
audioRecord.startRecording();
this.isRecording = true;
while (isRecording) {
bufferRead = audioRecord.read(samples, 0, bufferSize);
if (bufferRead > 0) {
// 将数据添加给解码器
encoder.addData(samples, bufferRead);
}
try {
Thread.sleep(20);
} catch (InterruptedException e) {
e.printStackTrace();
}
}
System.out.println(LOG + "录制结束");
audioRecord.stop();
encoder.stopEncoding();
}
}

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3. AudioEncoder，负责调用NDK 方法实现音频的编码，每编码一次，就交给AudioSender 去发送：

public class AudioEncoder implements Runnable {
String LOG = "AudioEncoder";
private static AudioEncoder encoder;
private boolean isEncoding = false;
private List<AudioData> dataList = null;// 存放数据
public static AudioEncoder getInstance() {
if (encoder == null) {
encoder = new AudioEncoder();
}
return encoder;
}
private AudioEncoder() {
dataList = Collections.synchronizedList(new LinkedList<AudioData>());
}
public void addData(byte[] data, int size) {
AudioData rawData = new AudioData();
rawData.setSize(size);
byte[] tempData = new byte[size];
System.arraycopy(data, 0, tempData, 0, size);
rawData.setRealData(tempData);
dataList.add(rawData);
}
// 开始编码
public void startEncoding() {
System.out.println(LOG + "解码线程启动");
if (isEncoding) {
Log.e(LOG, "编码器已经启动，不能再次启动");
return;
}
new Thread(this).start();
}
// 结束
public void stopEncoding() {
this.isEncoding = false;
}
public void run() {
// 先启动发送端
AudioSender sender = new AudioSender();
sender.startSending();
int encodeSize = 0;
byte[] encodedData = new byte[256];
// 初始化编码器
AudioCodec.audio_codec_init(30);
isEncoding = true;
while (isEncoding) {
if (dataList.size() == 0) {
try {
Thread.sleep(20);
} catch (InterruptedException e) {
e.printStackTrace();
}
continue;
}
if (isEncoding) {
AudioData rawData = dataList.remove(0);
encodedData = new byte[rawData.getSize()];
//
encodeSize = AudioCodec.audio_encode(rawData.getRealData(), 0,
rawData.getSize(), encodedData, 0);
System.out.println();
if (encodeSize > 0) {
sender.addData(encodedData, encodeSize);
// 清空数据
encodedData = new byte[encodedData.length];
}
}
}
System.out.println(LOG + "编码结束");
sender.stopSending();
}
}

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4. AudioSender类，负责音频数据的发送，使用UDP协议将编码后的AMR音频数据发送到服务器端，这个类功能简单：

public class AudioSender implements Runnable {
String LOG = "AudioSender ";
private boolean isSendering = false;
private List<AudioData> dataList;
DatagramSocket socket;
DatagramPacket dataPacket;
private InetAddress ip;
private int port;
public AudioSender() {
dataList = Collections.synchronizedList(new LinkedList<AudioData>());
try {
try {
ip = InetAddress.getByName(MyConfig.SERVER_HOST);
this.port = MyConfig.SERVER_PORT;
socket = new DatagramSocket();
} catch (UnknownHostException e) {
e.printStackTrace();
}
} catch (SocketException e) {
e.printStackTrace();
}
}
// 添加数据
public void addData(byte[] data, int size) {
AudioData encodedData = new AudioData();
encodedData.setSize(size);
byte[] tempData = new byte[size];
System.arraycopy(data, 0, tempData, 0, size);
encodedData.setRealData(tempData);
dataList.add(encodedData);
}
// 发送数据
private void sendData(byte[] data, int size) {
try {
dataPacket = new DatagramPacket(data, size, ip, port);
dataPacket.setData(data);
socket.send(dataPacket);
} catch (IOException e) {
e.printStackTrace();
}
}
// 开始发送
public void startSending() {
System.out.println(LOG + "发送线程启动");
new Thread(this).start();
}
// 停止发送
public void stopSending() {
this.isSendering = false;
}
// run
public void run() {
this.isSendering = true;
System.out.println(LOG + "开始发送数据");
while (isSendering) {
if (dataList.size() > 0) {
AudioData encodedData = dataList.remove(0);
sendData(encodedData.getRealData(), encodedData.getSize());
}
}
System.out.println(LOG + "发送结束");
}
}

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5. 另外，上述类中有一个 MyConfig 类，主要放一些配置参数：

public class MyConfig {
public static String SERVER_HOST = "192.168.1.130";// 服务器的IP
public static final int SERVER_PORT = 5656;// 服务器的监听端口
public static final int CLIENT_PORT = 5757;//
public static final int AUDIO_STATUS_RECORDING = 0;//手机端的状态：录音 or 播放
public static final int AUDIO_STATUS_LISTENING = 1;
public static void setServerHost(String ip) {
System.out.println("修改后的服务器网址为 " + ip);
SERVER_HOST = ip;
}
}

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上述代码实现了发送端的功能，现在代码结构如下：

本实例中对音频没有添加时间戳处理，实际测试中没有太大的影响，可以听的清楚双方的语音对话，如果想要添加时间戳的话，就在音频录制 AudioRecord的 read方法出得到时间戳，然后附加给UDP包。取自：http://www.cnblogs.com/swordbearer/archive/2012/08/10/2633083.html