Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 97469ca35ecf27defdfc971376e4d6ff6399b23d..3f7dd76e868408ddb6abb12f5cb8c79323109014 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -114,8 +114,7 @@ RTPSender::RTPSender(
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
-    SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log)
+    SendSideDelayObserver* send_side_delay_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
@@ -153,7 +152,8 @@ RTPSender::RTPSender(
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
-      event_log_(event_log),
+      event_log_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      event_log_(nullptr),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -757,6 +757,7 @@ bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
     bytes_sent = transport_->SendRtp(packet, size, options)
                      ? static_cast<int>(size)
                      : -1;
+    CriticalSectionScoped lock(event_log_crit_.get());
     if (event_log_ && bytes_sent > 0) {
       event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet, size);
     }
@@ -1917,4 +1918,8 @@ RtpState RTPSender::GetRtxRtpState() const {
   return state;
 }
 
+void RTPSender::SetRtcEventLog(webrtc::RtcEventLog* event_log) {
+  CriticalSectionScoped lock(event_log_crit_.get());
+  event_log_ = event_log;
+}
 }  // namespace webrtc