Avoid construction of unused RtcEventLogNullImpl object

webrtc/pc/peerconnectionfactory.cc

 /*
  *  Copyright 2004 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.
  */
 
 #include "webrtc/pc/peerconnectionfactory.h"
 
 #include <utility>
 
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/mediastreamproxy.h"
 #include "webrtc/api/mediastreamtrackproxy.h"
 #include "webrtc/api/peerconnectionfactoryproxy.h"
 #include "webrtc/api/peerconnectionproxy.h"
 #include "webrtc/api/videosourceproxy.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/rtc_base/bind.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/ptr_util.h"
 // Adding 'nogncheck' to disable the gn include headers check to support modular
 // WebRTC build targets.
 // TODO(zhihuang): This wouldn't be necessary if the interface and
 // implementation of the media engine were in separate build targets.
 #include "webrtc/media/engine/webrtcmediaengine.h"             // nogncheck
 #include "webrtc/media/engine/webrtcvideodecoderfactory.h"     // nogncheck
 #include "webrtc/media/engine/webrtcvideoencoderfactory.h"     // nogncheck
 #include "webrtc/modules/audio_device/include/audio_device.h"  // nogncheck
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/client/basicportallocator.h"
@@ skipping 219 matching lines @@
   }
 
   if (!allocator) {
     allocator.reset(new cricket::BasicPortAllocator(
         default_network_manager_.get(), default_socket_factory_.get()));
   }
   network_thread_->Invoke<void>(
       RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask,
                                allocator.get(), options_.network_ignore_mask));
 
-  std::unique_ptr<RtcEventLog> event_log(new RtcEventLogNullImpl());
-  if (event_log_factory_) {
-    event_log = event_log_factory_->CreateRtcEventLog();
-  }
+  std::unique_ptr<RtcEventLog> event_log =
+      event_log_factory_ ? event_log_factory_->CreateRtcEventLog()
+                         : rtc::MakeUnique<RtcEventLogNullImpl>();
 
   std::unique_ptr<Call> call = worker_thread_->Invoke<std::unique_ptr<Call>>(
       RTC_FROM_HERE,
       rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get()));
 
   rtc::scoped_refptr<PeerConnection> pc(
       new rtc::RefCountedObject<PeerConnection>(this, std::move(event_log),
                                                 std::move(call)));
 
   if (!pc->Initialize(configuration, std::move(allocator),
@@ skipping 66 matching lines @@
   }
   call_config.audio_state = 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;
 
   return std::unique_ptr<Call>(call_factory_->CreateCall(call_config));
 }
 
 }  // namespace webrtc