Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop f…

webrtc/api/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.
  */
 
@@ skipping 202 matching lines @@
                                          int64_t max_size_bytes) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   return channel_manager_->StartAecDump(file, max_size_bytes);
 }
 
 void PeerConnectionFactory::StopAecDump() {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   channel_manager_->StopAecDump();
 }
 
-bool PeerConnectionFactory::StartRtcEventLog(rtc::PlatformFile file,
-                                             int64_t max_size_bytes) {
-  RTC_DCHECK(signaling_thread_->IsCurrent());
-  return channel_manager_->StartRtcEventLog(file, max_size_bytes);
-}
-
-void PeerConnectionFactory::StopRtcEventLog() {
-  RTC_DCHECK(signaling_thread_->IsCurrent());
-  channel_manager_->StopRtcEventLog();
-}
-
 rtc::scoped_refptr<PeerConnectionInterface>
 PeerConnectionFactory::CreatePeerConnection(
     const PeerConnectionInterface::RTCConfiguration& configuration_in,
     const MediaConstraintsInterface* constraints,
     std::unique_ptr<cricket::PortAllocator> allocator,
     std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
     PeerConnectionObserver* observer) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
 
   // We merge constraints and configuration into a single configuration.
@@ skipping 85 matching lines @@
 cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   return cricket::WebRtcMediaEngineFactory::Create(
       default_adm_.get(),
       audio_decoder_factory_,
       video_encoder_factory_.get(),
       video_decoder_factory_.get());
 }
 
 }  // namespace webrtc