Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1165)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Issue 3016473002: Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead (Closed)
Patch Set: Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender.cc ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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"),
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender.cc ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698