标签：error: report share glob ilbc sed log res city
一.WebRTC的Android客户端搭建
1.libjingle_peerconnection_so.so
2.libjingle_peerconnection.jar
3.客户端源码一份（可以在github上下载）
二、相关概念介绍
1.P2P：点对点通讯；
2.STUN：提供反射地址使双方可以进行P2P通讯；
3.TURN：在反射地址方式失败情况下的补充方案，即使用中继器，使双方百分之百能够通讯；
4.ICE：综合STUN与TURN两种方案，找出一种最合理最廉价的可行路径；
5.SIP/SDP：SIP一种音视频通讯的协议，SDP为SIP协议中对音视频的描述；
6.PeerConnectionFactory/PeerConnection：整个WebRTC中最核心的类，有了这个类才能获得音视频相关的其他操作，
7.MediaStream：流媒体，包括音频和视频；
8.Track：流媒体中的轨，封装了流媒体的数据，包括音频数据中的声道，视频数据中的YUV场；
9.Session Management：抽象的会话层；
10.RTP：流媒体协议；
11.iSAC：音频语音编码器；
12.iLBC：音频语音解码器；
13.VP8：视频图像编解码器；
14.Room：在web客户端会生成一个随机的房间号，在android客户端需要输入该房间号进行相互连接进行通讯；
15.Observe：观察连接方的数据，就是一个连接后数据的回调接口；
三、相关类的介绍
1.AppRTCAudioManager：音频管理类，全部调用Android SDK已有的方法；
2.AppRTCClient：自定义的与服务端进行通讯和消息回调的接口；
3.AppRTCProximitySensor：Android手机上的距离传感器，使手机靠近耳朵时让按键失效；
4.CallActivity：建立通讯连接，并且显示音视频的界面；
5.CallFragment：通讯界面，CallActivity的子界面；
6.CaptureQualityController：控制界面的画质分辨率大小，用seekbar显示；
7.ConnectActivity：初始化相机和通信的一些参数，点击呼叫按钮时跳转到CallActivity界面进行呼叫；
8.CpuMonitor：显示CPU的当前使用性能数据；
9.HudFragment：参数的界面显示，包括上行宽带和丢包率；
10.PeerConnectionClient：PeerConnection的实现，有了这个类才能进行音视频相关数据通讯；
11.PercentFrameLayout：封装了FrameLayout；
12.RoomParametersFetcher：解析服务器的json数据，并转发房间的参数和地址；
13.SettingsActivity：设置Activity；
14.SettingsFragment：设置Fragment，一些音视频通讯的参数设置；
15.UnhandledExceptionHandler：crash时抓异常，不会直接抛出异常；
16.WebSocketChannelClient：websockot封装；
17.WebSocketRTCClient：对WebSocketChannelClient使用，进行websockt方式发送数据；
工具类：
1.AppRTCUtils：
2.AsyncHttpURLConnection：
3.LooperExecutor：
四、WebSocket
如果下载的代码中含有autobanh.jar这个jar包，那么可以断定该Android客户端与WebRTC的通讯是采用WebSocket的方式进行通讯的，其实现主要在WebSocketChannelClient和WebSocketRTCClient这两个类中，其回调接口则在WebSocketConnectionObserver的三个回调函数里面，这个类被封装在了WebSocketChannelClient里面。（WebSocket的详细知识不做介绍）
五、WebRTC呼叫流程介绍（转载）
上述序列中，标注的场景是ClientA向ClientB发起对聊请求，调用描述如下：
·ClientA首先创建PeerConnection对象，然后打开本地音视频设备，将音视频数据封装成MediaStream添加到PeerConnection中。
·ClientA调用PeerConnection的CreateOffer方法创建一个用于offer的SDP对象，SDP对象中保存当前音视频的相关参数。ClientA通过PeerConnection的SetLocalDescription方法将该SDP对象保存起来，并通过Signal服务器发送给ClientB。
·ClientB接收到ClientA发送过的offer SDP对象，通过PeerConnection的SetRemoteDescription方法将其保存起来，并调用PeerConnection的CreateAnswer方法创建一个应答的SDP对象，通过PeerConnection的SetLocalDescription的方法保存该应答SDP对象并将它通过Signal服务器发送给ClientA。
·ClientA接收到ClientB发送过来的应答SDP对象，将其通过PeerConnection的SetRemoteDescription方法保存起来。
·在SDP信息的offer/answer流程中，ClientA和ClientB已经根据SDP信息创建好相应的音频Channel和视频Channel并开启Candidate数据的收集，Candidate数据可以简单地理解成Client端的IP地址信息（本地IP地址、公网IP地址、Relay服务端分配的地址）。
·当ClientA收集到Candidate信息后，PeerConnection会通过OnIceCandidate接口给ClientA发送通知，ClientA将收到的Candidate信息通过Signal服务器发送给ClientB，ClientB通过PeerConnection的AddIceCandidate方法保存起来。同样的操作ClientB对ClientA再来一次。
·这样ClientA和ClientB就已经建立了音视频传输的P2P通道，ClientB接收到ClientA传送过来的音视频流，会通过PeerConnection的OnAddStream回调接口返回一个标识ClientA端音视频流的MediaStream对象，在ClientB端渲染出来即可。同样操作也适应ClientB到ClientA的音视频流的传输。
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;
static {
System.loadLibrary("jingle_peerconnection_so");
}
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;
}
// |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）
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);
//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);
//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
};
//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);
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。
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。
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。
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。
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);
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。
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);
标签：error: report share glob ilbc sed log res city
原文地址：http://www.cnblogs.com/idignew/p/7248257.html
