Revert of Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/webrtcsession.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 96 matching lines @@
 
  protected:
   ~IceObserver() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
 };
 
 class WebRtcSession : public cricket::BaseSession,
                       public AudioProviderInterface,
+                      public DataChannelFactory,
                       public VideoProviderInterface,
                       public DtmfProviderInterface,
                       public DataChannelProviderInterface {
  public:
   WebRtcSession(cricket::ChannelManager* channel_manager,
                 rtc::Thread* signaling_thread,
                 rtc::Thread* worker_thread,
-                cricket::PortAllocator* port_allocator);
+                cricket::PortAllocator* port_allocator,
+                MediaStreamSignaling* mediastream_signaling);
   virtual ~WebRtcSession();
 
   bool Initialize(
       const PeerConnectionFactoryInterface::Options& options,
       const MediaConstraintsInterface* constraints,
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
   // Deletes the voice, video and data channel and changes the session state
   // to STATE_RECEIVEDTERMINATE.
   void Terminate();
 
   void RegisterIceObserver(IceObserver* observer) {
     ice_observer_ = observer;
   }
 
   virtual cricket::VoiceChannel* voice_channel() {
     return voice_channel_.get();
   }
   virtual cricket::VideoChannel* video_channel() {
     return video_channel_.get();
   }
   virtual cricket::DataChannel* data_channel() {
     return data_channel_.get();
   }
 
+  virtual const MediaStreamSignaling* mediastream_signaling() const {
+    return mediastream_signaling_;
+  }
+
   void SetSdesPolicy(cricket::SecurePolicy secure_policy);
   cricket::SecurePolicy SdesPolicy() const;
 
   // Get current ssl role from transport.
   bool GetSslRole(rtc::SSLRole* role);
 
   // Generic error message callback from WebRtcSession.
   // TODO - It may be necessary to supply error code as well.
   sigslot::signal0<> SignalError;
 
   void CreateOffer(
       CreateSessionDescriptionObserver* observer,
-      const PeerConnectionInterface::RTCOfferAnswerOptions& options,
-      const cricket::MediaSessionOptions& session_options);
+      const PeerConnectionInterface::RTCOfferAnswerOptions& options);
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
-                    const MediaConstraintsInterface* constraints,
-                    const cricket::MediaSessionOptions& session_options);
+                    const MediaConstraintsInterface* constraints);
   // The ownership of |desc| will be transferred after this call.
   bool SetLocalDescription(SessionDescriptionInterface* desc,
                            std::string* err_desc);
   // The ownership of |desc| will be transferred after this call.
   bool SetRemoteDescription(SessionDescriptionInterface* desc,
                             std::string* err_desc);
   bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
 
   bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
 
@@ skipping 63 matching lines @@
 
   // virtual so it can be mocked in unit tests
   virtual bool GetLocalCertificate(
       const std::string& transport_name,
       rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
 
   // Caller owns returned certificate
   virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
                                        rtc::SSLCertificate** cert);
 
+  // Implements DataChannelFactory.
+  rtc::scoped_refptr<DataChannel> CreateDataChannel(
+      const std::string& label,
+      const InternalDataChannelInit* config) override;
+
   cricket::DataChannelType data_channel_type() const;
 
   bool IceRestartPending() const;
 
   void ResetIceRestartLatch();
 
   // Called when an RTCCertificate is generated or retrieved by
   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
   void OnCertificateReady(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
 
-  // Called when voice_channel_, video_channel_ and data_channel_ are created
-  // and destroyed. As a result of, for example, setting a new description.
-  sigslot::signal0<> SignalVoiceChannelCreated;
-  sigslot::signal0<> SignalVoiceChannelDestroyed;
-  sigslot::signal0<> SignalVideoChannelCreated;
-  sigslot::signal0<> SignalVideoChannelDestroyed;
-  sigslot::signal0<> SignalDataChannelCreated;
-  sigslot::signal0<> SignalDataChannelDestroyed;
-
-  // Called when a valid data channel OPEN message is received.
-  // std::string represents the data channel label.
-  sigslot::signal2<const std::string&, const InternalDataChannelInit&>
-      SignalDataChannelOpenMessage;
-
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
     kOffer,
     kPrAnswer,
     kAnswer,
   };
 
   bool UpdateSessionState(Action action, cricket::ContentSource source,
@@ skipping 89 matching lines @@
   // Gather the usage of IPv4/IPv6 as best connection.
   void ReportBestConnectionState(const cricket::TransportStats& stats);
 
   void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
   rtc::scoped_ptr<MediaControllerInterface> media_controller_;
   rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
   rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
   rtc::scoped_ptr<cricket::DataChannel> data_channel_;
   cricket::ChannelManager* channel_manager_;
+  MediaStreamSignaling* mediastream_signaling_;
   IceObserver* ice_observer_;
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
   bool ice_connection_receiving_;
   rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
   rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
   // Candidates that arrived before the remote description was set.
   std::vector<IceCandidateInterface*> saved_candidates_;
   // If the remote peer is using a older version of implementation.
   bool older_version_remote_peer_;
   bool dtls_enabled_;
   // Specifies which kind of data channel is allowed. This is controlled
   // by the chrome command-line flag and constraints:
   // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
   // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
   // not set or false, SCTP is allowed (DCT_SCTP);
   // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
   // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
   cricket::DataChannelType data_channel_type_;
   rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
 
   rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
       webrtc_session_desc_factory_;
 
+  sigslot::signal0<> SignalVoiceChannelDestroyed;
+  sigslot::signal0<> SignalVideoChannelDestroyed;
+  sigslot::signal0<> SignalDataChannelDestroyed;
+
   // Member variables for caching global options.
   cricket::AudioOptions audio_options_;
   cricket::VideoOptions video_options_;
   MetricsObserverInterface* metrics_observer_;
 
   // Declares the bundle policy for the WebRTCSession.
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_