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..b19ae0c9eb09ce69ae380238a43ec5a05d0f3818 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -19,6 +19,7 @@
 #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"
@@ -97,7 +98,7 @@ class RTPSender : public RTPSenderInterface {
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log);
+            RtcEventLogProxy* event_log);
   virtual ~RTPSender();
 
   void ProcessBitrate();
@@ -465,7 +466,7 @@ 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_;
+  RtcEventLogProxy* const event_log_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);