Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/audio/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_state.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
 class CongestionController;
 class RemoteBitrateEstimator;
+class RtcEventLog;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   AudioReceiveStream(CongestionController* congestion_controller,
                      const webrtc::AudioReceiveStream::Config& config,
-                     const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
+                     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                     webrtc::RtcEventLog* event_log);
   ~AudioReceiveStream() override;
 
   // webrtc::AudioReceiveStream implementation.
   void Start() override;
   void Stop() override;
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
   void SignalNetworkState(NetworkState state);
@@ skipping 12 matching lines @@
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_