Initialize PeerConnection members in declaration order and destroy them in reverse order.

webrtc/pc/peerconnection.cc

 /*
  *  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 377 matching lines @@
   if (!constraints) {
     return true;
   }
   return mandatory_constraints_satisfied == constraints->GetMandatory().size();
 }
 
 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
+      event_log_(RtcEventLog::Create()),
       signaling_state_(kStable),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
-      event_log_(RtcEventLog::Create()),
       rtcp_cname_(GenerateRtcpCname()),
       local_streams_(StreamCollection::Create()),
       remote_streams_(StreamCollection::Create()) {}
 
 PeerConnection::~PeerConnection() {
   TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // Need to detach RTP senders/receivers from WebRtcSession,
   // since it's about to be destroyed.
   for (const auto& sender : senders_) {
@@ skipping 870 matching lines @@
   return session_->pending_remote_description();
 }
 
 void PeerConnection::Close() {
   TRACE_EVENT0("webrtc", "PeerConnection::Close");
   // Update stats here so that we have the most recent stats for tracks and
   // streams before the channels are closed.
   stats_->UpdateStats(kStatsOutputLevelStandard);
 
   session_->Close();
-  event_log_.reset();
   network_thread()->Invoke<void>(
       RTC_FROM_HERE,
       rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
                 port_allocator_.get()));
 
   factory_->worker_thread()->Invoke<void>(RTC_FROM_HERE,
                                           [this] { call_.reset(); });
+
+  // The event log must outlive call (and any other object that uses it).
+  event_log_.reset();
 }
 
 void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
                                           WebRtcSession::State state) {
   switch (state) {
     case WebRtcSession::STATE_INIT:
       ChangeSignalingState(PeerConnectionInterface::kStable);
       break;
     case WebRtcSession::STATE_SENTOFFER:
       ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
@@ skipping 1026 matching lines @@
   call_config.audio_state =
       factory_->channel_manager() ->media_engine()->GetAudioState();
   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
 
   call_.reset(webrtc::Call::Create(call_config));
 }
 
 }  // namespace webrtc