Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(45)

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

Issue 1571283002: Fixes a bug which incorrectly logs incoming RTCP as outgoing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 11 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/rtp_rtcp_impl.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender.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.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 3c623365071760fa9aff08ecee59656391839c23..a672a06398a5772b9ef64914914a5946e8410536 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -34,6 +34,7 @@ class BitrateAggregator;
class CriticalSectionWrapper;
class RTPSenderAudio;
class RTPSenderVideo;
+class RtcEventLog;
class RTPSenderInterface {
public:
@@ -96,7 +97,8 @@ class RTPSender : public RTPSenderInterface {
TransportFeedbackObserver* transport_feedback_callback,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer,
- SendSideDelayObserver* send_side_delay_observer);
+ SendSideDelayObserver* send_side_delay_observer,
+ RtcEventLog* event_log);
virtual ~RTPSender();
void ProcessBitrate();
@@ -433,6 +435,7 @@ 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_;
// RTP variables
bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
@@ -464,6 +467,8 @@ class RTPSender : public RTPSenderInterface {
// that the target bitrate is still valid.
rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698