Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/api/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 19 matching lines @@
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/videocapturertracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/sctp/sctpdataengine.h"
 #include "webrtc/pc/channelmanager.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::RtpSenderInternal;
 using webrtc::RtpSenderInterface;
@@ skipping 512 matching lines @@
 }
 
 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
+      event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
       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 28 matching lines @@
   port_allocator_ = std::move(allocator);
 
   // The port allocator lives on the network thread and should be initialized
   // there.
   if (!network_thread()->Invoke<bool>(
           RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
                                    this, configuration))) {
     return false;
   }
 
-  media_controller_.reset(
-      factory_->CreateMediaController(configuration.media_config));
+  media_controller_.reset(factory_->CreateMediaController(
+      configuration.media_config, event_log_.get()));
 
   session_.reset(new WebRtcSession(
       media_controller_.get(), factory_->network_thread(),
       factory_->worker_thread(), factory_->signaling_thread(),
       port_allocator_.get(),
       std::unique_ptr<cricket::TransportController>(
           factory_->CreateTransportController(
               port_allocator_.get(),
               configuration.redetermine_role_on_ice_restart))));
 
@@ skipping 1702 matching lines @@
   // Call this last since it may create pooled allocator sessions using the
   // candidate filter set above.
   port_allocator_->SetConfiguration(stun_servers, turn_servers,
                                     configuration.ice_candidate_pool_size,
                                     configuration.prune_turn_ports);
   return true;
 }
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
-  return media_controller_->call_w()->StartEventLog(file, max_size_bytes);
+  return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
-  media_controller_->call_w()->StopEventLog();
+  event_log_->StopLogging();
 }
 }  // namespace webrtc