Don't delete UMA observer until after PeerConnection.

content/renderer/media/rtc_peer_connection_handler.h

 // Copyright (c) 2012 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 CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
 
 #include <stddef.h>
 
 #include <map>
@@ skipping 244 matching lines @@
 
   MediaStreamTrackMetrics track_metrics_;
 
   // Counter for a UMA stat reported at destruction time.
   int num_data_channels_created_ = 0;
 
   // Counter for number of IPv4 and IPv6 local candidates.
   int num_local_candidates_ipv4_ = 0;
   int num_local_candidates_ipv6_ = 0;
 
-  // To make sure the observer is released after the native_peer_connection_,
-  // it has to come first.
+  // To make sure the observers are released after native_peer_connection_,
+  // they have to come first.
   scoped_refptr<Observer> peer_connection_observer_;
+  scoped_refptr<webrtc::UMAObserver> uma_observer_;
 
   // |native_peer_connection_| is the libjingle native PeerConnection object.
   scoped_refptr<webrtc::PeerConnectionInterface> native_peer_connection_;
 
   // The last applied configuration. Used so that the constraints
   // used when constructing the PeerConnection carry over when
   // SetConfiguration is called.
   webrtc::PeerConnectionInterface::RTCConfiguration configuration_;
 
   // Record info about the first SessionDescription from the local and
   // remote side to record UMA stats once both are set.  We only check
   // for the first offer or answer.  "pranswer"s and "unknown"s (from
   // unit tests) are ignored.
   std::unique_ptr<FirstSessionDescription> first_local_description_;
   std::unique_ptr<FirstSessionDescription> first_remote_description_;
 
   std::map<webrtc::MediaStreamInterface*,
            std::unique_ptr<content::RemoteMediaStreamImpl>>
       remote_streams_;
-  scoped_refptr<webrtc::UMAObserver> uma_observer_;
   base::TimeTicks ice_connection_checking_start_;
 
   // Track which ICE Connection state that this PeerConnection has gone through.
   bool ice_state_seen_[webrtc::PeerConnectionInterface::kIceConnectionMax] = {};
 
   base::WeakPtrFactory<RTCPeerConnectionHandler> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_