webrtc/modules/rtp_rtcp/source/rtp_sender_video.h - Issue 1765443002: Added log messages for important call setup events: first packet sent/received

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

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

Issue 1765443002: Added log messages for important call setup events: first packet sent/received (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Added the non-thread-safe version of OneTimeEvent. Created 4 years, 9 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

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

index dc1088a3f7cf150d731377094c1ce8f2bfde02dc..3ea9168a45d880fc513fffd8da16ebea6156ab46 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -13,6 +13,7 @@

#include <list>

+#include "webrtc/base/onetimeevent.h"

#include "webrtc/base/scoped_ptr.h"

#include "webrtc/base/thread_annotations.h"

#include "webrtc/common_types.h"

@@ -116,6 +117,7 @@ class RTPSenderVideo {

Bitrate _fecOverheadRate;

// Bitrate used for video payload and RTP headers

Bitrate _videoBitrate;

+ OneTimeEvent first_frame_sent_;

};

} // namespace webrtc

« webrtc/base/onetimeevent.h ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »