MediaController/Call instantiation.

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 14 matching lines @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_
 #define TALK_APP_WEBRTC_WEBRTCSESSION_H_
 
 #include <string>
 
 #include "talk/app/webrtc/datachannel.h"
 #include "talk/app/webrtc/dtmfsender.h"
+#include "talk/app/webrtc/mediacontroller.h"
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/statstypes.h"
 #include "talk/media/base/mediachannel.h"
 #include "webrtc/p2p/base/session.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 
@@ skipping 143 matching lines @@
     return BaseSession::local_description();
   }
   const cricket::SessionDescription* base_remote_description() const {
     return BaseSession::remote_description();
   }
 
   // Get the id used as a media stream track's "id" field from ssrc.
   virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
   virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
 
-
   // AudioMediaProviderInterface implementation.
   void SetAudioPlayout(uint32 ssrc,
                        bool enable,
                        cricket::AudioRenderer* renderer) override;
   void SetAudioSend(uint32 ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer) override;
   void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
 
@@ skipping 154 matching lines @@
                                  bool* valid);
 
   std::string GetSessionErrorMsg();
 
   // Invoked when OnTransportCompleted is signaled to 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_;
@@ skipping 28 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_