Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtp_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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 
 #include <list>
 #include <map>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/call/rtc_event_log_proxy.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/transport.h"
 
@@ skipping 58 matching lines @@
   RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
             RtpPacketSender* paced_sender,
             TransportSequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log);
+            RtcEventLogProxy* event_log);
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
@@ skipping 347 matching lines @@
 
   // Statistics
   rtc::scoped_ptr<CriticalSectionWrapper> statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
-  RtcEventLog* const event_log_;
+  RtcEventLogProxy* const event_log_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
   uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
   SSRCDatabase* const ssrc_db_;
   uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
   bool sequence_number_forced_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
   bool ssrc_forced_ GUARDED_BY(send_critsect_);
@@ skipping 15 matching lines @@
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_