Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/channel_proxy.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_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 namespace webrtc {
 
 class AudioSinkInterface;
 class PacketRouter;
+class RtcEventLog;
 class RtpPacketSender;
 class TransportFeedbackObserver;
 
 namespace voe {
 
 class Channel;
 
 // This class provides the "view" of a voe::Channel that we need to implement
 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
 // purposes:
@@ skipping 29 matching lines @@
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int32_t GetSpeechOutputLevelFullRange() const;
   virtual uint32_t GetDelayEstimate() const;
 
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
   virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
 
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
+  virtual void SetRtcEventLog(RtcEventLog* event_log);
+
  private:
   Channel* channel() const;
 
   rtc::ThreadChecker thread_checker_;
   ChannelOwner channel_owner_;
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_