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我正嘗試通過WiFi將音頻從麥克風從1 Android傳輸到另一個Android。 在看了一些例子之後，我在每個應用中做了一個單獨的活動，1個捕獲併發送音頻，另一個接收。通過WiFi在Android手機之間傳輸語音
我已經使用Audiorecord和Audiotrack類來捕捉和播放。然而，我只是聽到一些crack啪聲（現在停止後，我做了一些改變，雖然我恢復了回來）
活動發送語音。
public class VoiceSenderActivity extends Activity {
private EditText target;
private TextView streamingLabel;
private Button startButton,stopButton;
public byte[] buffer;
public static DatagramSocket socket;
private int port=50005; //which port??
AudioRecord recorder;
//Audio Configuration.
private int sampleRate = 8000; //How much will be ideal?
private int channelConfig = AudioFormat.CHANNEL_CONFIGURATION_MONO;
private int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
private boolean status = true;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
target = (EditText) findViewById (R.id.target_IP);
streamingLabel = (TextView) findViewById(R.id.streaming_label);
startButton = (Button) findViewById (R.id.start_button);
stopButton = (Button) findViewById (R.id.stop_button);
streamingLabel.setText("Press Start! to begin");
startButton.setOnClickListener (startListener);
stopButton.setOnClickListener (stopListener);
}
private final OnClickListener stopListener = new OnClickListener() {
@Override
public void onClick(View arg0) {
status = false;
recorder.release();
Log.d("VS","Recorder released");
}
};
private final OnClickListener startListener = new OnClickListener() {
@Override
public void onClick(View arg0) {
status = true;
startStreaming();
}
};
public void startStreaming() {
Thread streamThread = new Thread(new Runnable() {
@Override
public void run() {
try {
int minBufSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat);
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(target.getText().toString());
Log.d("VS", "Address retrieved");
recorder = new AudioRecord(MediaRecorder.AudioSource.MIC,sampleRate,channelConfig,audioFormat,minBufSize);
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);
}
} catch(UnknownHostException e) {
Log.e("VS", "UnknownHostException");
} catch (IOException e) {
Log.e("VS", "IOException");
}
}
});
streamThread.start();
}
}
活動接收語音
public class VoiceReceiverActivity extends Activity {
private Button receiveButton,stopButton;
public static DatagramSocket socket;
private AudioTrack speaker;
//Audio Configuration.
private int sampleRate = 8000; //How much will be ideal?
private int channelConfig = AudioFormat.CHANNEL_CONFIGURATION_MONO;
private int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
private boolean status = true;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
receiveButton = (Button) findViewById (R.id.receive_button);
stopButton = (Button) findViewById (R.id.stop_button);
findViewById(R.id.receive_label);
receiveButton.setOnClickListener(receiveListener);
stopButton.setOnClickListener(stopListener);
}
private final OnClickListener stopListener = new OnClickListener() {
@Override
public void onClick(View v) {
status = false;
speaker.release();
Log.d("VR","Speaker released");
}
};
private final OnClickListener receiveListener = new OnClickListener() {
@Override
public void onClick(View arg0) {
status = true;
startReceiving();
}
};
public void startReceiving() {
Thread receiveThread = new Thread (new Runnable() {
@Override
public void run() {
try {
DatagramSocket socket = new DatagramSocket(50005);
Log.d("VR", "Socket Created");
byte[] buffer = new byte[256];
//minimum buffer size. need to be careful. might cause problems. try setting manually if any problems faced
int minBufSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat);
speaker = new AudioTrack(AudioManager.STREAM_MUSIC,sampleRate,channelConfig,audioFormat,minBufSize,AudioTrack.MODE_STREAM);
speaker.play();
while(status == true) {
try {
DatagramPacket packet = new DatagramPacket(buffer,buffer.length);
socket.receive(packet);
Log.d("VR", "Packet Received");
//reading content from packet
buffer=packet.getData();
Log.d("VR", "Packet data read into buffer");
//sending data to the Audiotrack obj i.e. speaker
speaker.write(buffer, 0, minBufSize);
Log.d("VR", "Writing buffer content to speaker");
} catch(IOException e) {
Log.e("VR","IOException");
}
}
} catch (SocketException e) {
Log.e("VR", "SocketException");
}
}
});
receiveThread.start();
}
}
我使用Wireshark來檢查是否正在發送的數據包，我可以看到的數據包。但是，源是發送設備的MAC地址，目標也是物理地址。不知道這是否相關。
那麼問題是什麼？
至少有三個問題需要處理：數據延遲（或丟失），總體數據吞吐量以及稍微不匹配採樣頻率的可能性。實際的IP電話必須具備處理這三者的手段。不匹配的時鐘是非常棘手的 - 最初你可以引入一個延遲來給出一些緩衝允許，但是如果發送者速度較慢，你將用盡緩衝區，並且接收器將會缺乏數據;而如果發送者速度更快，則緩衝區最終會隨未播放數據溢出。 – 2012-12-01 17:32:50
我確實設法實現了這一點。沒有真正存在不匹配頻率的問題。數據延遲，是的。有一種我自己的協議來匹配接收器/發送器時鐘。最後，它確實工作，但只有一些滯後（隨着距離無線路由器的增加而增加） – Alabhya 2012-12-02 18:37:41
嘿，那裏，我爲上面的代碼實現了一個測試應用程序，在下面提出了所有必要的修改，但我仍然有問題。兩個手機之間的通訊沒有問題，但我認爲麥克風不能正常錄音，因爲我聽不到另一端的聲音。您是否可以鏈接到我可以看一下的示例解決方案？ – chuckliddell0 2013-03-11 14:48:16