我创建了一个应用程序，它应该从用户的麦克风流式传输音频。应用程序中有两个部分：
- 安卓应用
- 用于监听应用程序发送的音频的 Java 类
我面临的问题是，流正在发生，但流的质量很差，而且流中也有明显的中断。
android应用程序中的活动：
public class SendActivity extends Activity {
private Button startButton, stopButton;
public byte[] buffer;
public static DatagramSocket socket;
private int port = 50005;
AudioRecord recorder;
private int sampleRate = 8000;
@SuppressWarnings("deprecation")
private int channelConfig = AudioFormat.CHANNEL_IN_MONO;
private int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
int minBufSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig,
audioFormat);
private boolean status = true;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_send);
startButton = (Button) findViewById(R.id.start_button);
stopButton = (Button) findViewById(R.id.stop_button);
startButton.setOnClickListener(new View.OnClickListener() {
@Override
public void onClick(View v) {
status = true;
startStreaming();
}
});
stopButton.setOnClickListener(new View.OnClickListener() {
@Override
public void onClick(View v) {
status = false;
recorder.release();
Log.d("VS", "Recorder released");
}
});
minBufSize += 2048;
System.out.println("minBufSize: " + minBufSize);
}
public void startStreaming() {
Thread streamThread = new Thread(new Runnable() {
@Override
public void run() {
try {
DatagramSocket socket = new DatagramSocket();
Log.d("VS", "Socket Created");
byte[] buffer = new byte[minBufSize];
Log.d("VS", "Buffer created of size " + minBufSize);
DatagramPacket packet;
final InetAddress destination = InetAddress
.getByName("192.168.1.20");
Log.d("VS", "Address retrieved");
recorder = new AudioRecord(MediaRecorder.AudioSource.MIC,
sampleRate, channelConfig, audioFormat,
minBufSize * 10);
Log.d("VS", "Recorder initialized");
recorder.startRecording();
while (status == true) {
// reading data from MIC into buffer
minBufSize = recorder.read(buffer, 0, buffer.length);
// putting buffer in the packet
packet = new DatagramPacket(buffer, buffer.length,
destination, port);
socket.send(packet);
System.out.println("MinBufferSize: " + minBufSize);
}
} catch (UnknownHostException e) {
Log.e("VS", "UnknownHostException");
} catch (IOException e) {
e.printStackTrace();
Log.e("VS", "IOException");
}
}
});
streamThread.start();
}
}
布局文件：
<RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android"
xmlns:tools="http://schemas.android.com/tools"
android:layout_width="match_parent"
android:layout_height="match_parent"
android:paddingBottom="@dimen/activity_vertical_margin"
android:paddingLeft="@dimen/activity_horizontal_margin"
android:paddingRight="@dimen/activity_horizontal_margin"
android:paddingTop="@dimen/activity_vertical_margin"
tools:context=".SendActivity" >
<Button
android:id="@+id/stop_button"
android:layout_width="wrap_content"
android:layout_height="wrap_content"
android:layout_alignBaseline="@+id/start_button"
android:layout_alignBottom="@+id/start_button"
android:layout_toRightOf="@+id/start_button"
android:text="Stop" />
<Button
android:id="@+id/start_button"
android:layout_width="wrap_content"
android:layout_height="wrap_content"
android:layout_alignParentLeft="true"
android:layout_alignParentTop="true"
android:layout_marginLeft="79dp"
android:layout_marginTop="163dp"
android:text="Start" />
</RelativeLayout>
用于收听音频的java类：
class Server {
AudioInputStream audioInputStream;
static AudioInputStream ais;
static AudioFormat format;
static boolean status = true;
static int port = 50005;
static int sampleRate = 8000;
public static void main(String args[]) throws Exception {
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[9000];
format = new AudioFormat(sampleRate, 16, 1, true, false);
while (status == true) {
DatagramPacket receivePacket = new DatagramPacket(receiveData,
receiveData.length);
serverSocket.receive(receivePacket);
ByteArrayInputStream baiss = new ByteArrayInputStream(
receivePacket.getData());
ais = new AudioInputStream(baiss, format, receivePacket.getLength());
//System.out.println("Reading...");
toSpeaker(receivePacket.getData());
}
}
public static void toSpeaker(byte soundbytes[]) {
try {
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
SourceDataLine sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceDataLine.open(format);
FloatControl volumeControl = (FloatControl) sourceDataLine.getControl(FloatControl.Type.MASTER_GAIN);
volumeControl.setValue(6.0206f);
sourceDataLine.start();
sourceDataLine.open(format);
sourceDataLine.start();
System.out.println("format? :" + sourceDataLine.getFormat());
sourceDataLine.write(soundbytes, 0, soundbytes.length);
System.out.println(soundbytes.toString());
sourceDataLine.drain();
sourceDataLine.close();
} catch (Exception e) {
System.out.println("Not working in speakers...");
e.printStackTrace();
}
}
}
我应该怎么做才能获得从应用程序到课堂的清晰连续的音频流？