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 9eb3cbf038c2b279cf3b6a755c441f684b1d112d..f58b8dc82f907d838aa41dc516d2a0159955a934 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -13,6 +13,7 @@ |
#include <algorithm> |
#include <utility> |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
@@ -26,6 +27,7 @@ |
#include "webrtc/rtc_base/arraysize.h" |
#include "webrtc/rtc_base/checks.h" |
#include "webrtc/rtc_base/logging.h" |
+#include "webrtc/rtc_base/ptr_util.h" |
#include "webrtc/rtc_base/rate_limiter.h" |
#include "webrtc/rtc_base/safe_minmax.h" |
#include "webrtc/rtc_base/timeutils.h" |
@@ -642,7 +644,8 @@ bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet, |
? static_cast<int>(packet.size()) |
: -1; |
if (event_log_ && bytes_sent > 0) { |
- event_log_->LogOutgoingRtpHeader(packet, pacing_info.probe_cluster_id); |
+ event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>( |
+ packet, pacing_info.probe_cluster_id)); |
} |
} |
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |