Android WebRTC
原文链接：WebRTC之Android客户端
六、回调函数
1.WebSocket回调接口与主要消息处理
Candidate、answer、offer、bye四大类消息
private class WebSocketObserverimplements WebSocketConnectionObserver {
@Override
public void onOpen() {
Log.d(TAG, "WebSocket connection opened to: " + wsServerUrl);
executor.execute(new Runnable() {
@Override
public void run() {
state = WebSocketConnectionState.CONNECTED;
// Check if we have pending register request.
if (roomID !=null && clientID != null) {
register(roomID, clientID);
}
}
});
}
@Override
public void onClose(WebSocketCloseNotification code, String reason) {
Log.d(TAG, "WebSocket connection closed. Code: " + code
+ ". Reason: " + reason + ". State: " + state);
synchronized (closeEventLock) {
closeEvent = true;
closeEventLock.notify();
}
executor.execute(new Runnable() {
@Override
public void run() {
if (state != WebSocketConnectionState.CLOSED) {
state = WebSocketConnectionState.CLOSED;
events.onWebSocketClose();
}
}
});
}
@Override
public void onTextMessage(String payload) {
Log.d(TAG, "WSS->C: " + payload);
final String message = payload;
executor.execute(new Runnable() {
@Override
public void run() {
if (state == WebSocketConnectionState.CONNECTED
|| state == WebSocketConnectionState.REGISTERED) {
events.onWebSocketMessage(message);
}
}
});
}
@Override
public void onRawTextMessage(byte[] payload) {
}
@Override
public void onBinaryMessage(byte[] payload) {
}
}
@Override
public void onWebSocketMessage(final String msg) {
if (wsClient.getState() != WebSocketConnectionState.REGISTERED) {
Log.e(TAG, "Got WebSocket message in non registered state.");
return;
}
try {
JSONObject json = new JSONObject(msg);
String msgText = json.getString("msg");
String errorText = json.optString("error");
if (msgText.length() > 0) {
json = new JSONObject(msgText);
String type = json.optString("type");
if (type.equals("candidate")) {
IceCandidate candidate = new IceCandidate(
json.getString("id"),
json.getInt("label"),
json.getString("candidate"));
events.onRemoteIceCandidate(candidate);
} else if (type.equals("answer")) {
if (initiator) {
SessionDescription sdp = new SessionDescription(
SessionDescription.Type.fromCanonicalForm(type),
json.getString("sdp"));
events.onRemoteDescription(sdp);
} else {
reportError("Received answer for call initiator: " + msg);
}
} else if (type.equals("offer")) {
if (!initiator) {
SessionDescription sdp = new SessionDescription(
SessionDescription.Type.fromCanonicalForm(type),
json.getString("sdp"));
events.onRemoteDescription(sdp);
} else {
reportError("Received offer for call receiver: " + msg);
}
} else if (type.equals("bye")) {
events.onChannelClose();
} else {
reportError("Unexpected WebSocket message: " + msg);
}
} else {
if (errorText !=null && errorText.length() > 0) {
reportError("WebSocket error message: " + errorText);
} else {
reportError("Unexpected WebSocket message: " + msg);
}
}
} catch (JSONException e) {
reportError("WebSocket message JSON parsing error: " + e.toString());
}
}
2.Observer接口
主要是连接建立完成后Ice的改变和流信息的改变引起的回调
public static interface Observer {
/** Triggered when the SignalingState changes. */
public void onSignalingChange(SignalingState newState);
/** Triggered when the IceConnectionState changes. */
public void onIceConnectionChange(IceConnectionState newState);
/** Triggered when the ICE connection receiving status changes. */
public void onIceConnectionReceivingChange(boolean receiving);
/** Triggered when the IceGatheringState changes. */
public void onIceGatheringChange(IceGatheringState newState);
/** Triggered when a new ICE candidate has been found. */
public void onIceCandidate(IceCandidate candidate);
/** Triggered when media is received on a new stream from remote peer. */
public void onAddStream(MediaStream stream);
/** Triggered when a remote peer close a stream. */
public void onRemoveStream(MediaStream stream);
/** Triggered when a remote peer opens a DataChannel. */
public void onDataChannel(DataChannel dataChannel);
/** Triggered when renegotiation is necessary. */
public void onRenegotiationNeeded();
}
3.SDP接口
主要是连接建立的过程中引起的回调
/** Interface for observing SDP-related events. */
public interface SdpObserver {
/** Called on success of Create{Offer,Answer}(). */
public void onCreateSuccess(SessionDescription sdp);
/** Called on success of Set{Local,Remote}Description(). */
public void onSetSuccess();
/** Called on error of Create{Offer,Answer}(). */
public void onCreateFailure(String error);
/** Called on error of Set{Local,Remote}Description(). */
public void onSetFailure(String error);
}
4.PeerConnectionClient
生成PeerConnection，实现相关的回调，完成整个业务逻辑
private final PCObserverpcObserver = new PCObserver();（Observer ）
private final SDPObserversdpObserver = new SDPObserver();（SdpObserver）
private PeerConnectionFactoryfactory;
private PeerConnectionpeerConnection;
5.CallActivity
private PeerConnectionClientpeerConnectionClient = null;
private AppRTCClientappRtcClient;
七、Native函数之信令协商
6.1 加载so文件
static {
System.loadLibrary("jingle_peerconnection_so");
}
6.2 PeerConnectionFactory相关Native函数
6.2.1网络接口相关参数
public static class Options {
// Keep in sync with webrtc/base/network.h!
static final int ADAPTER_TYPE_UNKNOWN = 0;
static final int ADAPTER_TYPE_ETHERNET = 1 << 0;
static final int ADAPTER_TYPE_WIFI = 1 << 1;
static final int ADAPTER_TYPE_CELLULAR = 1 << 2;
static final int ADAPTER_TYPE_VPN = 1 << 3;
static final int ADAPTER_TYPE_LOOPBACK = 1 << 4;
public int networkIgnoreMask;
public boolean disableEncryption;
}
6.2.2初始化PeerConnectionFactory
// |context| is an android.content.Context object, but we keep it untyped here
// to allow building on non-Android platforms.
// Callers may specify either |initializeAudio| or |initializeVideo| as false
// to skip initializing the respective engine (and avoid the need for the
// respective permissions).
// |renderEGLContext| can be provided to suport HW video decoding to
// texture and will be used to create a shared EGL context on video
// decoding thread.
public static native boolean initializeAndroidGlobals(Object context, boolean initializeAudio, boolean initializeVideo,boolean videoHwAcceleration);
Context：简单的ApplicationContext，或者其他Context相关的上下文。
initializeAudio：初始化音频部分。（boolean）
videoHwAcceleration：是否启用硬件加速。（boolean）
6.2.3初始化音视频轨
private static final String FIELD_TRIAL_VP9 = "WebRTC-SupportVP9/Enabled/";
// Field trial initialization. Must be called before PeerConnectionFactory
// is created.
public static native void initializeFieldTrials(String fieldTrialsInitString);
6.2.4 PeerConnectionFactory其他函数
//创建Factory
private static native long nativeCreatePeerConnectionFactory();
//创建指令回调接口（与ICE服务器进行交互的指令）
private static native long nativeCreateObserver(PeerConnection.Observer observer);
//创建PeerConnection
private static native long nativeCreatePeerConnection(long nativeFactory, PeerConnection.RTCConfiguration rtcConfig, ediaConstraints constraints, long nativeObserver);
//创建本地音视频流
private static native long nativeCreateLocalMediaStream(long nativeFactory, String label);
//创建本地视频源
private static native long nativeCreateVideoSource(long nativeFactory, long nativeVideoCapturer, MediaConstraints constraints);
//创建视频轨
private static native long nativeCreateVideoTrack(long nativeFactory, String id, long nativeVideoSource);
//创建本地音频流
private static native long nativeCreateAudioSource(long nativeFactory, MediaConstraints constraints);
//创建音频轨
private static native long nativeCreateAudioTrack(long nativeFactory, String id, long nativeSource);
//设置相关网络参数
public native void nativeSetOptions(long nativeFactory, Options options);
//设置视频硬件加速参数
private static native void nativeSetVideoHwAccelerationOptions(long nativeFactory, Object renderEGLContext);
//回收PeerConnectionFactory
private static native void freeFactory(long nativeFactory);
6.3 PeerConnection相关Native函数
6.3.1相关信令状态
//检测本地candidate的状态：刚刚创建、正在收集、完成收集
** Tracks PeerConnectionInterface::IceGatheringState */
public enum IceGatheringState { NEW, GATHERING, COMPLETE };
//检测远端candidate的状态
/** Tracks PeerConnectionInterface::IceConnectionState */
public enum IceConnectionState {
NEW, CHECKING, CONNECTED, COMPLETED, FAILED, DISCONNECTED, CLOSED
};
//检测与Sigal信令服务器连接的状态
/** Tracks PeerConnectionInterface::SignalingState */
public enum SignalingState {
STABLE, HAVE_LOCAL_OFFER, HAVE_LOCAL_PRANSWER, HAVE_REMOTE_OFFER,
HAVE_REMOTE_PRANSWER, CLOSED
};
6.3.2 Native函数介绍
//得到本地sdp描述
public native SessionDescription getLocalDescription();
//得到远端sdp描述
public native SessionDescription getRemoteDescription();
//创建数据通道
public native DataChannel createDataChannel(String label, DataChannel.Init init);
//创建offer消息
public native void createOffer(SdpObserver observer, MediaConstraints constraints);
//创建answer消息
public native void createAnswer(SdpObserver observer, MediaConstraints constraints);
//设置本地sdp
public native void setLocalDescription(SdpObserver observer, SessionDescription sdp);
//设置远端sdp
public native void setRemoteDescription(SdpObserver observer, SessionDescription sdp);
//更新IceServer
public native boolean updateIce(List<IceServer> iceServers, MediaConstraints constraints);
//得到信令状态
public native SignalingState signalingState();
//获得远端连接状态
public native IceConnectionState iceConnectionState();
//获得本地连接状态
public native IceGatheringState iceGatheringState();
//关闭与Ice服务器的连接
public native void close();
//释放PeerConnection
private static native void freePeerConnection(long nativePeerConnection);
//释放Observer
private static native void freeObserver(long nativeObserver);
//添加新的Candidate
private native boolean nativeAddIceCandidate(String sdpMid, int sdpMLineIndex, String iceCandidateSdp);
//添加本地流
private native boolean nativeAddLocalStream(long nativeStream);
//移除本地流
private native void nativeRemoveLocalStream(long nativeStream);
//得到StatsObserver的状态
private native boolean nativeGetStats(StatsObserver observer, long nativeTrack);
八、Native函数之音视频
一旦有了peerConnectionFactory实例，就应该从你的设备上获取音频和视频了，最终渲染到屏幕上。VideoCapturerAndroid，VideoSource，VideoTrack和VideoRenderer，都是以VideoCapturerAndroid开始。
8.1 VideoCapturerAndroid
VideoCapturerAndroid类是一个相机的包装类，提供访问设备相机数据流的江边方法。允许你获取设备数量，获取前置后置摄像头
// Returns the number of camera devices
VideoCapturerAndroid.getDeviceCount();
// Returns the front face device name
VideoCapturerAndroid.getNameOfFrontFacingDevice();
// Returns the back facing device name
VideoCapturerAndroid.getNameOfBackFacingDevice();
// Creates a VideoCapturerAndroid instance for the device name
VideoCapturerAndroid.create(name);
使用VideoCapturerAndroid类的实例，可以创建包含相机视频流的MediaStream，你可以给对方发送数据。
8.2 VideoSource/VideoTrack
VideoSource可以开始或停止你的设备。在无用停止抓取信息有助于电池使用寿命的延长。
VideoTrack是一个添加VideoSource到MediaStream对象的一个包装。
8.3 AudioSource/AudioTrack
除了不需要AudioCapturer获取麦克风数据，AudioSource/AudioTrack和VideoSource/VideoTrack很类似。audioConstraints是MediaContraints的实例。
8.4 VideoRenderer
VideoRendererGui是一个GLSurfaceView，在这之上，可以显示视频流,增加我们的renderer到VideoTrack上。
// To create our VideoRenderer, we can use the// included VideoRendererGui for simplicity// First we need to set the GLSurfaceView that it should render to
GLSurfaceView videoView = (GLSurfaceView) findViewById(R.id.glview_call);
// Then we set that view, and pass a Runnable// to run once the surface is ready
VideoRendererGui.setView(videoView, runnable);
// Now that VideoRendererGui is ready, we can get our VideoRenderer
VideoRenderer renderer = VideoRendererGui.createGui(x, y, width, height);
// And finally, with our VideoRenderer ready, we// can add our renderer to the VideoTrack.
localVideoTrack.addRenderer(renderer);
8.5 MediaConstraints
这个MediaConstraints是WebRTC支持将视频和音频放入MediaStream的方式。看这个支持的规范，大多数方法都需要MediaContraints的实例。
8.6 MediaStream
getUserMedia直接返回一个MediaStream，可以直接将其添加到RTCPeerConnection中发送给对端。
// We start out with an empty MediaStream object,
// created with help from our PeerConnectionFactory
// Note that LOCAL_MEDIA_STREAM_ID can be any string
MediaStream mediaStream = peerConnectionFactory.createLocalMediaStream(LOCAL_MEDIA_STREAM_ID);
// Now we can add our tracks.
mediaStream.addTrack(localVideoTrack);
mediaStream.addTrack(localAudioTrack);