一、简介
WebRTCAndroid是一个开源的Android应用程序开发库,支持视频和音频实时通话。它由Google开发并维护,以简化实时通信应用程序的开发。
在WebRTCAndroid中,我们可以使用许多官方提供的API来实现实时通信,并且可以很容易地与其他平台(如Web、iOS等)进行交互。这些API包括MediaStream、RTCPeerConnection和RTCDataChannel等,所有这些API都是Java接口,易于开发和集成。
二、使用WebRTCAndroid实现实时视频通话
以下是使用WebRTCAndroid实现实时视频通话的示例代码:
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); // 创建本地视频流 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 // 创建PeerConnection peerConnection = peerConnectionFactory.createPeerConnection(null, new CustomPeerConnectionObserver(TAG), new CustomSdpObserver(TAG)); MediaStreamManager localMediaStreamManager = new MediaStreamManager(peerConnection); localMediaStreamManager.addTrack(localVideoTrack); // 添加本地视频轨道到媒体流中 peerConnection.createOffer(new CustomSdpObserver(TAG) { @Override public void onCreateSuccess(SessionDescription sessionDescription) { super.onCreateSuccess(sessionDescription); peerConnection.setLocalDescription(new CustomSdpObserver(TAG), sessionDescription); remoteDescription(sessionDescription); // 发送SDP offer到远程连接 } }, new MediaConstraints()); }
三、使用WebRTCAndroid实现实时音频通话
以下是使用WebRTCAndroid实现实时音频通话的示例代码:
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); // 创建音频轨道 audioSource = peerConnectionFactory.createAudioSource(new MediaConstraints()); localAudioTrack = peerConnectionFactory.createAudioTrack("101", audioSource); // 创建PeerConnection peerConnection = peerConnectionFactory.createPeerConnection(null, new CustomPeerConnectionObserver(TAG), new CustomSdpObserver(TAG)); MediaStreamManager localMediaStreamManager = new MediaStreamManager(peerConnection); localMediaStreamManager.addTrack(localAudioTrack); // 添加本地音频轨道到媒体流中 peerConnection.createOffer(new CustomSdpObserver(TAG) { @Override public void onCreateSuccess(SessionDescription sessionDescription) { super.onCreateSuccess(sessionDescription); peerConnection.setLocalDescription(new CustomSdpObserver(TAG), sessionDescription); remoteDescription(sessionDescription); // 发送SDP offer到远程连接 } }, new MediaConstraints()); }
四、WebRTCAndroid的使用示例
在WebRTCAndroid中,我们可以使用许多官方提供的API来实现实时通信。下面是一个使用WebRTCAndroid实现实时视频通话的示例:
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(); // 创建本地视频流 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); // 设置远程和本地视频视图 VideoRendererGui.setView(remoteVideoView, null); VideoRendererGui.setView(localVideoView, null); // 创建peerConnection,添加本地媒体流 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 ListgetIceServerList() { 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){ // 发送SDP offer到远程连接 } 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、总结
WebRTCAndroid是一个非常有用的开源库,提供了许多API和示例代码,可以让开发人员轻松地实现实时通信应用程序。在使用WebRTCAndroid时,需要注意实时通信的性能和稳定性,以便提供更好的用户体验。