一、簡介
WebRTCAndroid是一個開源的Android應用程序開發庫,支持視頻和音頻實時通話。它由Google開發並維護,以簡化實時通信應用程序的開發。
在WebRTCAndroid中,我們可以使用許多官方提供的API來實現實時通信,並且可以很容易地與其他平台(如Web、iOS等)進行交互。這些API包括MediaStream、RTCPeerConnection和RTCDataChannel等,所有這些API都是Java介面,易於開發和集成。
二、使用WebRTCAndroid實現實時視頻通話
以下是使用WebRTCAndroid實現實時視頻通話的示例代碼:
private PeerConnectionFactory peerConnectionFactory;
private PeerConnection peerConnection;
private localMediaStream localStream;
private VideoCapturer videoCapturer;
private VideoSource videoSource;
private VideoTrack localVideoTrack;
private SurfaceTextureHelper surfaceTextureHelper;
public void start() {
// 初始化PeerConnectionFactory
PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());
PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
options.disableEncryption = true;
options.disableNetworkMonitor = true;
peerConnectionFactory = new PeerConnectionFactory(options);
// 創建本地視頻流
surfaceTextureHelper = SurfaceTextureHelper.create("CaptureThread", null);
videoCapturer = createCameraCapturer(new Camera1Enumerator(false));
videoSource = peerConnectionFactory.createVideoSource(false);
videoCapturer.initialize(surfaceTextureHelper, context, videoSource.getCapturerObserver());
videoCapturer.startCapture(640, 480, 30);
localVideoTrack = peerConnectionFactory.createVideoTrack("100", videoSource);
localVideoTrack.addSink(localView); // localView是本地的SurfaceView
// 創建PeerConnection
peerConnection = peerConnectionFactory.createPeerConnection(null, new CustomPeerConnectionObserver(TAG), new CustomSdpObserver(TAG));
MediaStreamManager localMediaStreamManager = new MediaStreamManager(peerConnection);
localMediaStreamManager.addTrack(localVideoTrack); // 添加本地視頻軌道到媒體流中
peerConnection.createOffer(new CustomSdpObserver(TAG) {
@Override
public void onCreateSuccess(SessionDescription sessionDescription) {
super.onCreateSuccess(sessionDescription);
peerConnection.setLocalDescription(new CustomSdpObserver(TAG), sessionDescription);
remoteDescription(sessionDescription); // 發送SDP offer到遠程連接
}
}, new MediaConstraints());
}
三、使用WebRTCAndroid實現實時音頻通話
以下是使用WebRTCAndroid實現實時音頻通話的示例代碼:
private PeerConnectionFactory peerConnectionFactory;
private PeerConnection peerConnection;
private localMediaStream localStream;
private AudioTrack localAudioTrack;
private AudioSource audioSource;
public void start() {
// 初始化PeerConnectionFactory
PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());
PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
options.disableEncryption = true;
options.disableNetworkMonitor = true;
peerConnectionFactory = new PeerConnectionFactory(options);
// 創建音頻軌道
audioSource = peerConnectionFactory.createAudioSource(new MediaConstraints());
localAudioTrack = peerConnectionFactory.createAudioTrack("101", audioSource);
// 創建PeerConnection
peerConnection = peerConnectionFactory.createPeerConnection(null, new CustomPeerConnectionObserver(TAG), new CustomSdpObserver(TAG));
MediaStreamManager localMediaStreamManager = new MediaStreamManager(peerConnection);
localMediaStreamManager.addTrack(localAudioTrack); // 添加本地音頻軌道到媒體流中
peerConnection.createOffer(new CustomSdpObserver(TAG) {
@Override
public void onCreateSuccess(SessionDescription sessionDescription) {
super.onCreateSuccess(sessionDescription);
peerConnection.setLocalDescription(new CustomSdpObserver(TAG), sessionDescription);
remoteDescription(sessionDescription); // 發送SDP offer到遠程連接
}
}, new MediaConstraints());
}
四、WebRTCAndroid的使用示例
在WebRTCAndroid中,我們可以使用許多官方提供的API來實現實時通信。下面是一個使用WebRTCAndroid實現實時視頻通話的示例:
public class CallActivity extends Activity {
private View hangupButton;
private VideoRenderer remoteVideoView;
private VideoRenderer localVideoView;
private VideoCapturerAndroid videoCapturer;
private PeerConnectionFactory factory;
private PeerConnection peerConnection;
private MediaStream localStream;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_call);
hangupButton = findViewById(R.id.hangup_button);
remoteVideoView = (VideoRendererGui.ScalingType scalingType, boolean mirror) ->
VideoRendererGui.create(remoteVideoView, 0, 0, 100, 100, scalingType, mirror);
localVideoView = (VideoRendererGui.ScalingType scalingType, boolean mirror) ->
VideoRendererGui.create(localVideoView, 0, 0, 100, 100, scalingType, mirror);
PeerConnectionFactory.initializeAndroidGlobals(this);
factory = new PeerConnectionFactory();
// 創建本地視頻流
videoCapturer = new VideoCapturerAndroid(CameraEnumerationAndroid.getNameOfFrontFacingDevice(), null);
VideoSource localVideoSource = factory.createVideoSource(videoCapturer, new MediaConstraints());
VideoTrack localVideoTrack = factory.createVideoTrack("video", localVideoSource);
localVideoTrack.addRenderer(new VideoRenderer(localVideoView));
localStream = factory.createLocalMediaStream("stream");
localStream.addTrack(localVideoTrack);
// 設置遠程和本地視頻視圖
VideoRendererGui.setView(remoteVideoView, null);
VideoRendererGui.setView(localVideoView, null);
// 創建peerConnection,添加本地媒體流
peerConnection = factory.createPeerConnection(getIceServerList(), getMediaConstraints(), new CustomPeerConnectionObserver(TAG));
if (peerConnection != null) {
peerConnection.addStream(localStream, new MediaConstraints());
}
// 建立SDP offer
peerConnection.createOffer(new CustomSdpObserver(TAG) {
@Override
public void onCreateSuccess(SessionDescription sessionDescription) {
super.onCreateSuccess(sessionDescription);
peerConnection.setLocalDescription(new CustomSdpObserver(TAG), sessionDescription);
sendOfferToRemote(peerConnection, sessionDescription.description);
}
}, new MediaConstraints());
hangupButton.setOnClickListener(v -> {
disconnectPeerConnection(peerConnection);
});
}
private List getIceServerList() {
List iceServers = new ArrayList();
iceServers.add(new PeerConnection.IceServer("stun:stun.l.google.com:19302"));
iceServers.add(new PeerConnection.IceServer("turn:192.158.29.39:3478?transport=udp",
"username",
"password"));
return iceServers;
}
private MediaConstraints getMediaConstraints() {
MediaConstraints mediaConstraints = new MediaConstraints();
mediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"));
mediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"));
mediaConstraints.optional.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
return mediaConstraints;
}
private void sendOfferToRemote(PeerConnection peerConnection, String sdp){
// 發送SDP offer到遠程連接
}
private void disconnectPeerConnection(PeerConnection peerConnection) {
if (peerConnection != null) {
peerConnection.dispose();
peerConnection = null;
}
if (videoCapturer != null) {
try {
videoCapturer.stopCapture();
} catch (InterruptedException e) {
e.printStackTrace();
}
videoCapturer.dispose();
videoCapturer = null;
}
if (localStream != null) {
localStream.dispose();
localStream = null;
}
VideoRendererGui.remove(remoteVideoView);
VideoRendererGui.remove(localVideoView);
}
}
五、總結
WebRTCAndroid是一個非常有用的開源庫,提供了許多API和示例代碼,可以讓開發人員輕鬆地實現實時通信應用程序。在使用WebRTCAndroid時,需要注意實時通信的性能和穩定性,以便提供更好的用戶體驗。
原創文章,作者:小藍,如若轉載,請註明出處:https://www.506064.com/zh-tw/n/156707.html
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