Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/api/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 11 matching lines @@
 #include "webrtc/api/rtpreceiver.h"
 #include "webrtc/api/rtpsender.h"
 #include "webrtc/api/statscollector.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/webrtcsession.h"
 
 namespace webrtc {
 
 class MediaStreamObserver;
 class VideoRtpReceiver;
+class RtcEventLog;
 
 // Populates |session_options| from |rtc_options|, and returns true if options
 // are valid.
 // |session_options|->transport_options map entries must exist in order for
 // them to be populated from |rtc_options|.
 bool ExtractMediaSessionOptions(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     bool is_offer,
     cricket::MediaSessionOptions* session_options);
 
@@ skipping 343 matching lines @@
   rtc::scoped_refptr<PeerConnectionFactory> factory_;
   PeerConnectionObserver* observer_;
   UMAObserver* uma_observer_;
   SignalingState signaling_state_;
   // TODO(bemasc): Remove ice_state_.
   IceState ice_state_;
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
 
   std::unique_ptr<cricket::PortAllocator> port_allocator_;
+  // The EventLog needs to outlive the media controller.
+  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 14 matching lines @@
 
   bool remote_peer_supports_msid_ = false;
 
   bool enable_ice_renomination_ = 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<StatsCollector> stats_;
   rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTION_H_