Fixes a bug which incorrectly logs incoming RTCP as outgoing.

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 6ad666b01aaba1a77b08d0a1b951ff124d44e025..9aa1003408fdedfebaa868ac8064ba9ec2a104dc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -17,6 +17,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
@@ -122,7 +124,8 @@ RTPSender::RTPSender(
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
-    SendSideDelayObserver* send_side_delay_observer)
+    SendSideDelayObserver* send_side_delay_observer,
+    RtcEventLog* event_log)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
@@ -161,6 +164,7 @@ RTPSender::RTPSender(
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
+      event_log_(event_log),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -1059,6 +1063,10 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     paced_sender_->InsertPacket(priority, rtp_header.ssrc,
                                 rtp_header.sequenceNumber, corrected_time_ms,
                                 payload_length, false);
+    if (event_log_) {
+      event_log_->LogRtpHeader(true, MediaType::ANY, buffer,

[the sun, 2016/01/11 12:36:16]

Add a test case for this which uses a MockRtcEventLog

[terelius, 2016/01/13 17:08:18]

Done.

+                               payload_length + rtp_header_length);
+    }
     if (last_capture_time_ms_sent_ == 0 ||
         corrected_time_ms > last_capture_time_ms_sent_) {
       last_capture_time_ms_sent_ = corrected_time_ms;
@@ -1074,6 +1082,9 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
 
   size_t length = payload_length + rtp_header_length;
   bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
+  if (event_log_) {
+    event_log_->LogRtpHeader(true, MediaType::ANY, buffer, length);
+  }
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be