Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 334 matching lines @@
   }
   WEBRTC_STUB(StopSend, (int channel));
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
   WEBRTC_FUNC(AssociateSendChannel, (int channel,
                                      int accociate_send_channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->associate_send_channel = accociate_send_channel;
     return 0;
   }
-  webrtc::RtcEventLog* GetEventLog() { return nullptr; }
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     // To match the behavior of the real implementation.
     if (_stricmp(codec.plname, "telephone-event") == 0 ||
         _stricmp(codec.plname, "audio/telephone-event") == 0 ||
         _stricmp(codec.plname, "CN") == 0 ||
@@ skipping 407 matching lines @@
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_