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.
  */
 
@@ skipping 78 matching lines @@
  public:
   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);
+            SendSideDelayObserver* send_side_delay_observer);
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
@@ skipping 194 matching lines @@
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
   uint32_t BitrateSent() const;
 
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
   CVOMode ActivateCVORtpHeaderExtension() override;
 
+  void SetRtcEventLog(webrtc::RtcEventLog* event_log);
+
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
   size_t CreateRtpHeader(uint8_t* header,
@@ skipping 133 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_;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> event_log_crit_;
+  RtcEventLog* event_log_ GUARDED_BY(event_log_crit_);

[the sun, 2016/03/03 09:25:13]

Can you use send_critsect_ instead?

[ivoc, 2016/03/10 13:15:36]

The critical section is no longer needed with the proxy object.

 
   // 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_