android中音频的采集与传输的初步实现

最近一直在尝试android下的音视频通话，觉得有几个任务模块得完成：

1、数据的采集与播放

2、数据的编解码

3、音视频流的实时传输

以下是音频流采集与播放的简单思路，使用AudioRecord采集音频流，使用jlibrtp进行传输，使用AudioTrack播放接收到的音频流。

这里只是初步的实现，还是存在着很大的时延，应该还得实现流的编解码，毕竟直接录制的原始流不适宜进行传输。

欢迎大家指出我这里实现存在的问题

主界面Activity

public class MainActivity extends Activity {
	private String TAG = "MainActivity";

	private boolean isRecording = false;
	private Button btn_record;
	private Button btn_record_finish;
	private Context context;
//	private AudioTrack audioTrack;
	private SoundReceiver receiver;

	/**
	 * 音频采样率
	 */
	private int frequency = 11025;

	@Override
	protected void onCreate(Bundle savedInstanceState) {
		super.onCreate(savedInstanceState);
		setContentView(R.layout.activity_main);
		context = this;
		btn_record = (Button) findViewById(R.id.btn_record);
		btn_record_finish = (Button) findViewById(R.id.btn_record_finish);

		btn_record.setOnClickListener(clickListener);
		btn_record_finish.setOnClickListener(clickListener);

		new Thread() {

			@Override
			public void run() {
				receiver = new SoundReceiver(10003, 10004);
			}

		}.start();
	}

	@Override
	protected void onDestroy() {
		super.onDestroy();
		receiver.destroySession();
	}

	private OnClickListener clickListener = new OnClickListener() {

		@Override
		public void onClick(View v) {
			switch (v.getId()) {
			case R.id.btn_record:
				record();
				break;
			case R.id.btn_record_finish:
				isRecording = false;
				break;
			}
		}

	};

	public void record() {
		new Thread() {
			@Override
			public void run() {
				int channelConfiguration = AudioFormat.CHANNEL_IN_STEREO;
				int audioEncoding = AudioFormat.ENCODING_PCM_16BIT;
				Participant participant = new Participant("127.0.0.1", 10003, 10004);
				SoundSender sender = new SoundSender(10001, 10002, participant);
				int bufferSize = AudioRecord.getMinBufferSize(frequency, channelConfiguration, audioEncoding);
				AudioRecord audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, frequency, channelConfiguration, audioEncoding, bufferSize);

				byte[] buffer = new byte[bufferSize];
				try {
					audioRecord.startRecording();
//					audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, frequency, AudioFormat.CHANNEL_CONFIGURATION_MONO,
//							AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
//					audioTrack.play();
					isRecording = true;
					while (isRecording) {
						int bufferReadResult = audioRecord.read(buffer, 0, bufferSize);
						Log.d(TAG, "send buffer length = " + bufferSize);
						sender.startSend(buffer);
					}
				} catch (Throwable t) {
					Log.e("AudioRecord", "Recording Failed");
				} finally {
//					audioTrack.stop();
//					audioTrack.release();
					sender.destroySession();
					audioRecord.stop();
					audioRecord.release();
					receiver.destroySession();
				}
			}
		}.start();
	}

}

SoundSender类：使用jlibrtp发送音频流数据

public class SoundSender implements RTPAppIntf {
	private String TAG = "SoundSender";
	private RTPSession rtpSession = null;

	public SoundSender(int rtpPort, int rtcpPort, Participant... participants) {
		DatagramSocket rtpSocket = null;
		DatagramSocket rtcpSocket = null;
		try {
			rtpSocket = new DatagramSocket(rtpPort);
			rtcpSocket = new DatagramSocket(rtcpPort);
		} catch (Exception e) {
			e.printStackTrace();
			System.out.println("RTPSession failed to obtain sender port");
		}

		rtpSession = new RTPSession(rtpSocket, rtcpSocket);
		rtpSession.RTPSessionRegister(this, null, null);
		System.out.println("CNAME: " + rtpSession.CNAME());

		// 添加要传输的目标地址
		for (int i = 0; i < participants.length; i++) {
			rtpSession.addParticipant(participants[i]);
		}
	}

	public void startSend(byte[] data) {
		rtpSession.sendData(data);
	}
	
	public void destroySession(){
		rtpSession.endSession();
	}

	@Override
	public void receiveData(DataFrame frame, Participant participant) {

	}

	@Override
	public void userEvent(int type, Participant[] participant) {
		switch (type) {
		case 1:
			Log.d(TAG, TAG + " Bye.");
			break;
		case 2:
			break;
		case 3:
			break;
		case 4:
			Log.d(TAG, "SDES packet received");
			break;
		case 5:
			Log.d(TAG, "Matched SSRC to ip-address provided by application");
			break;
		default:
			break;
		}
	}

	@Override
	public int frameSize(int payloadType) {
		return 1;
	}

}

SoundReceiver：接收音频流数据，并使用AudioTrack进行播放

public class SoundReceiver implements RTPAppIntf {
	private String TAG = "SoundReceiver";

	private RTPSession rtpSession = null;
	private AudioTrack audioTrack;
	private int bufferSize;
	/**
	 * 音频采样率
	 */
	private int frequency = 11025;

	public SoundReceiver(int rtpPort, int rtcpPort) {
		DatagramSocket rtpSocket = null;
		DatagramSocket rtcpSocket = null;

		try {
			rtpSocket = new DatagramSocket(rtpPort);
			rtcpSocket = new DatagramSocket(rtcpPort);
		} catch (Exception e) {
			e.printStackTrace();
			System.out.println("RTPSession failed to obtain receiver port");
		}

		rtpSession = new RTPSession(rtpSocket, rtcpSocket);
		rtpSession.naivePktReception(true);
		rtpSession.RTPSessionRegister(this, null, null);

		bufferSize = AudioTrack.getMinBufferSize(frequency, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT);

		audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, frequency, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
				bufferSize, AudioTrack.MODE_STREAM);
		audioTrack.play();
	}

	public void destroySession() {
		rtpSession.endSession();
	}

	@Override
	public void receiveData(DataFrame frame, Participant participant) {
		if (audioTrack != null) {
			byte[] data = frame.getConcatenatedData();
			audioTrack.write(data, 0, data.length);
		}
	}

	@Override
	public void userEvent(int type, Participant[] participant) {
		switch (type) {
		case 1:
			Log.d(TAG, TAG + " Bye.");
			audioTrack.stop();
			audioTrack.release();
			break;
		case 2:
			break;
		case 3:
			break;
		case 4:
			Log.d(TAG, "SDES packet received");
			break;
		case 5:
			Log.d(TAG, "Matched SSRC to ip-address provided by application");
			break;
		default:
			break;
		}
	}

	@Override
	public int frameSize(int payloadType) {
		return 1;
	}

}