Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 0e1242837c863ce9a024a229b0d45ba4a943fbbc..5793d2faf4a30988e2499dc98407d18de58f3bd0 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -96,8 +96,7 @@ class RTPSender : public RTPSenderInterface {
             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();
@@ -312,6 +311,8 @@ class RTPSender : public RTPSenderInterface {
   RtpState GetRtxRtpState() const;
   CVOMode ActivateCVORtpHeaderExtension() override;
 
+  void SetRtcEventLog(webrtc::RtcEventLog* event_log);
+
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
@@ -465,7 +466,9 @@ class RTPSender : public RTPSenderInterface {
   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_);