Ensure that connection can be closed in response to a data message.

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>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_vector.h"
 #include "base/memory/weak_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "base/timer/timer.h"
 #include "crypto/hmac.h"
 #include "remoting/protocol/transport.h"
 #include "remoting/protocol/webrtc_data_stream_adapter.h"
 #include "remoting/protocol/webrtc_video_encoder_factory.h"
 #include "remoting/signaling/signal_strategy.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 
-namespace webrtc {
-class FakeAudioDeviceModule;
-}  // namespace webrtc
-
 namespace remoting {
 namespace protocol {
 
 class TransportContext;
 class MessagePipe;
 
-class WebrtcTransport : public Transport,
-                        public webrtc::PeerConnectionObserver {
+class WebrtcTransport : public Transport {
  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;
@@ skipping 17 matching lines @@
 
    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_;
-  }
+  webrtc::PeerConnectionInterface* peer_connection();
+  webrtc::PeerConnectionFactoryInterface* peer_connection_factory();
   remoting::WebrtcVideoEncoderFactory* video_encoder_factory() {
     return video_encoder_factory_;
   }
 
   // Creates outgoing data channel. The channel is created in CONNECTING state.
   // The caller must wait for OnMessagePipeOpen() notification before sending
   // any messages.
   std::unique_ptr<MessagePipe> CreateOutgoingChannel(const std::string& name);
 
   // Transport interface.
   void Start(Authenticator* authenticator,
              SendTransportInfoCallback send_transport_info_callback) override;
   bool ProcessTransportInfo(buzz::XmlElement* transport_info) override;
   void Close(ErrorCode error);
 
  private:
+  // PeerConnectionWrapper is responsible for PeerConnection creation,
+  // ownership. It passes all events to the corresponding methods below. This is
+  // necessary to make it possible to close and destroy PeerConnection
+  // asynchronously, as it may be on stack when the transport is destroyed.
+  class PeerConnectionWrapper;
+  friend class PeerConnectionWrapper;
+
   void OnLocalSessionDescriptionCreated(
       std::unique_ptr<webrtc::SessionDescriptionInterface> description,
       const std::string& error);
   void OnLocalDescriptionSet(bool success, const std::string& error);
   void OnRemoteDescriptionSet(bool send_answer,
                               bool success,
                               const std::string& error);
 
-  // webrtc::PeerConnectionObserver interface.
+  // PeerConnection event handlers, called by PeerConnectionWrapper.
   void OnSignalingChange(
-      webrtc::PeerConnectionInterface::SignalingState new_state) override;
+      webrtc::PeerConnectionInterface::SignalingState new_state);
   void OnAddStream(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream);
   void OnRemoveStream(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream);
   void OnDataChannel(
-      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
-  void OnRenegotiationNeeded() override;
+      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel);
+  void OnRenegotiationNeeded();
   void OnIceConnectionChange(
-      webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
+      webrtc::PeerConnectionInterface::IceConnectionState new_state);
   void OnIceGatheringChange(
-      webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
-  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+      webrtc::PeerConnectionInterface::IceGatheringState new_state);
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
 
   void RequestNegotiation();
   void SendOffer();
   void EnsurePendingTransportInfoMessage();
   void SendTransportInfo();
   void AddPendingCandidatesIfPossible();
 
   base::ThreadChecker thread_checker_;
 
   rtc::Thread* worker_thread_;
   scoped_refptr<TransportContext> transport_context_;
   EventHandler* event_handler_ = nullptr;
   SendTransportInfoCallback send_transport_info_callback_;
 
   crypto::HMAC handshake_hmac_;
 
-  std::unique_ptr<webrtc::FakeAudioDeviceModule> fake_audio_device_module_;
-
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
-      peer_connection_factory_;
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  std::unique_ptr<PeerConnectionWrapper> peer_connection_wrapper_;
 
   remoting::WebrtcVideoEncoderFactory* video_encoder_factory_;
 
   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_;
 
   base::WeakPtrFactory<WebrtcTransport> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(WebrtcTransport);
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_