Add support for dynamic channels in WebrtcTransport.

remoting/protocol/webrtc_transport.h

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
 #define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
 
 #include <memory>
 #include <string>
 
@@ skipping 12 matching lines @@
 
 namespace webrtc {
 class FakeAudioDeviceModule;
 }  // namespace webrtc
 
 namespace remoting {
 namespace protocol {
 
 class TransportContext;
 class MessageChannelFactory;
+class MessagePipe;
 
 class WebrtcTransport : public Transport,
                         public webrtc::PeerConnectionObserver {
  public:
   class EventHandler {
    public:
     // Called after |peer_connection| has been created but before handshake. The
     // handler should create data channels and media streams. Renegotiation will
     // be required in two cases after this method returns:
     //   1. When the first data channel is created, if it wasn't created by this
     //      event handler.
     //   2. Whenever a media stream is added or removed.
     virtual void OnWebrtcTransportConnecting() = 0;
 
     // Called when the transport is connected.
     virtual void OnWebrtcTransportConnected() = 0;
 
     // Called when there is an error connecting the session.
     virtual void OnWebrtcTransportError(ErrorCode error) = 0;
 
+    // Called when a new data channel is created by the peer.
+    virtual void OnWebrtcTransportIncomingDataChannel(
+        const std::string& name,
+        std::unique_ptr<MessagePipe> pipe) = 0;
+
     // Called when an incoming media stream is added or removed.
     virtual void OnWebrtcTransportMediaStreamAdded(
         scoped_refptr<webrtc::MediaStreamInterface> stream) = 0;
     virtual void OnWebrtcTransportMediaStreamRemoved(
         scoped_refptr<webrtc::MediaStreamInterface> stream) = 0;
+
+   protected:
+    virtual ~EventHandler() {}
   };
 
   WebrtcTransport(rtc::Thread* worker_thread,
                   scoped_refptr<TransportContext> transport_context,
                   EventHandler* event_handler);
   ~WebrtcTransport() override;
 
   webrtc::PeerConnectionInterface* peer_connection() {
     return peer_connection_;
   }
   webrtc::PeerConnectionFactoryInterface* peer_connection_factory() {
     return peer_connection_factory_;
   }
   remoting::WebrtcVideoEncoderFactory* video_encoder_factory() {
     return video_encoder_factory_;
   }
 
-  // Factories for outgoing and incoming data channels. Must be used only after
-  // the transport is connected.
+  // Factory for outgoing data channels. Must be used only after the transport
+  // is connected.
   MessageChannelFactory* outgoing_channel_factory() {
-    return &outgoing_data_stream_adapter_;
-  }
-  MessageChannelFactory* incoming_channel_factory() {
-    return &incoming_data_stream_adapter_;
+    return data_stream_adapter_.get();
   }
 
   // Transport interface.
   void Start(Authenticator* authenticator,
              SendTransportInfoCallback send_transport_info_callback) override;
   bool ProcessTransportInfo(buzz::XmlElement* transport_info) override;
   void Close(ErrorCode error);
 
  private:
   void OnLocalSessionDescriptionCreated(
@@ skipping 45 matching lines @@
 
   bool negotiation_pending_ = false;
 
   bool connected_ = false;
 
   std::unique_ptr<buzz::XmlElement> pending_transport_info_message_;
   base::OneShotTimer transport_info_timer_;
 
   ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_;
 
-  WebrtcDataStreamAdapter outgoing_data_stream_adapter_;
-  WebrtcDataStreamAdapter incoming_data_stream_adapter_;
+  std::unique_ptr<WebrtcDataStreamAdapter> data_stream_adapter_;
 
   base::WeakPtrFactory<WebrtcTransport> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(WebrtcTransport);
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_