Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

 /*
  *  Copyright (c) 2012 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 18 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class ModuleRtpRtcpImpl;
 class RTCPReceiver;
-class RtcEventLog;
+class RtcEventLogProxy;
 
 class NACKStringBuilder {
  public:
   NACKStringBuilder();
   ~NACKStringBuilder();
 
   void PushNACK(uint16_t nack);
   std::string GetResult();
 
  private:
@@ skipping 22 matching lines @@
     RtcpReceiveTimeInfo last_xr_rr;
 
     // Used when generating TMMBR.
     ModuleRtpRtcpImpl* module;
   };
 
   RTCPSender(bool audio,
              Clock* clock,
              ReceiveStatistics* receive_statistics,
              RtcpPacketTypeCounterObserver* packet_type_counter_observer,
-             RtcEventLog* event_log,
+             RtcEventLogProxy* event_log,
              Transport* outgoing_transport);
   virtual ~RTCPSender();
 
   RtcpMode Status() const;
   void SetRTCPStatus(RtcpMode method);
 
   bool Sending() const;
   int32_t SetSendingStatus(const FeedbackState& feedback_state,
                            bool enabled);  // combine the functions
 
@@ skipping 102 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   rtc::scoped_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
  private:
   const bool audio_;
   Clock* const clock_;
   Random random_ GUARDED_BY(critical_section_rtcp_sender_);
   RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
 
-  RtcEventLog* const event_log_;
+  RtcEventLogProxy* const event_log_;
   Transport* const transport_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
   bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
   bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
   bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
 
   int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
 
   uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
@@ skipping 68 matching lines @@
 
   typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
       const RtcpContext&);
   std::map<RTCPPacketType, BuilderFunc> builders_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_