Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(471)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/video_send_stream.h

Issue 2960363002: Implement RTP keepalive in native stack. (Closed)
Patch Set: Cleanup Created 3 years, 5 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/video/video_send_stream_tests.cc ('k') | no next file » | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video_send_stream.h
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 266112e5871550d9406a945981647724170d1bfa..53be83f2aa876a0c2dca5fcdb9602f6b928c1e4e 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -168,6 +168,8 @@ class VideoSendStream {
int payload_type = -1;
} rtx;
+ RtpKeepAliveConfig keep_alive;
+
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
« no previous file with comments | « webrtc/video/video_send_stream_tests.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698