WebRTC之Android客户端

.WebRTCAndroid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)

 WebRTC之Android客户端_第1张图片

 

二、相关概念介绍

WebRTC之Android客户端_第2张图片 

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STUNTURN两种方案,找出一种最合理最廉价的可行路径;

5.SIP/SDPSIP一种音视频通讯的协议,SDPSIP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.AppRTCProximitySensorAndroid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.PeerConnectionClientPeerConnection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.UnhandledExceptionHandlercrash时抓异常,不会直接抛出异常;

16.WebSocketChannelClientwebsockot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WebSocketChannelClientWebSocketRTCClient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)

  WebRTC之Android客户端_第3张图片

上述序列中,标注的场景是ClientAClientB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PeerConnectionCreateOffer方法创建一个用于offerSDP对象,SDP对象中保存当前音视频的相关参数。ClientA通过PeerConnectionSetLocalDescription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PeerConnectionSetRemoteDescription方法将其保存起来,并调用PeerConnectionCreateAnswer方法创建一个应答的SDP对象,通过PeerConnectionSetLocalDescription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PeerConnectionSetRemoteDescription方法保存起来。

·在SDP信息的offer/answer流程中,ClientAClientB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ClientBClientB通过PeerConnectionAddIceCandidate方法保存起来。同样的操作ClientBClientA再来一次。

·这样ClientAClientB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PeerConnectionOnAddStream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ClientBClientA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

Candidateanswerofferbye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 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);


你可能感兴趣的:(音视频)