Delete MediaController class, move Call ownership to PeerConnection.

webrtc/pc/peerconnection.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 372 matching lines @@
       int candidate_pool_size,
       bool prune_turn_ports);
 
   // Starts recording an Rtc EventLog using the supplied platform file.
   // This function should only be called from the worker thread.
   bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes);
   // Starts recording an Rtc EventLog using the supplied platform file.
   // This function should only be called from the worker thread.
   void StopRtcEventLog_w();
 
+  // Creates the |*call_| object. Must only be called from the worker thread.
+  void CreateCall_w();
+
   // Storing the factory as a scoped reference pointer ensures that the memory
   // in the PeerConnectionFactoryImpl remains available as long as the
   // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
   // However, since the reference counting is done in the
   // PeerConnectionFactoryInterface all instances created using the raw pointer
   // will refer to the same reference count.
   rtc::scoped_refptr<PeerConnectionFactory> factory_;
   PeerConnectionObserver* observer_;
   UMAObserver* uma_observer_;
   SignalingState signaling_state_;
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
   PeerConnectionInterface::RTCConfiguration configuration_;
 
   std::unique_ptr<cricket::PortAllocator> port_allocator_;
-  // The EventLog needs to outlive the media controller.
+  // The EventLog needs to outlive |call_|.
   std::unique_ptr<RtcEventLog> event_log_;
-  std::unique_ptr<MediaControllerInterface> media_controller_;
 
   // One PeerConnection has only one RTCP CNAME.
   // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
   std::string rtcp_cname_;
 
   // Streams added via AddStream.
   rtc::scoped_refptr<StreamCollection> local_streams_;
   // Streams created as a result of SetRemoteDescription.
   rtc::scoped_refptr<StreamCollection> remote_streams_;
 
@@ skipping 12 matching lines @@
   std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
 
   bool remote_peer_supports_msid_ = false;
 
   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
       senders_;
   std::vector<
       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
       receivers_;
   std::unique_ptr<WebRtcSession> session_;
+  std::unique_ptr<Call> call_;
   std::unique_ptr<StatsCollector> stats_;
   rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_PEERCONNECTION_H_