| commit | a68379bfe27e461f4ca544fbbdb7f72e0c97ac52 | [log] [tgz] |
|---|---|---|
| author | jiayl@webrtc.org <jiayl@webrtc.org> | Mon Feb 10 19:12:14 2014 +0000 |
| committer | jiayl@webrtc.org <jiayl@webrtc.org> | Mon Feb 10 19:12:14 2014 +0000 |
| tree | df5050c2b8811b432d16c34b3278047d76448ba5 | |
| parent | bac08b333bdd6e719e9c3c0369f6b13259c500f1 [diff] [blame] |
Add stats of incoming frame delays for debugging bandwidth estimation. BUG=crbug/338380 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/vie_rtp_rtcp_impl.h b/video_engine/vie_rtp_rtcp_impl.h index caabd2c..74e69ce 100644 --- a/video_engine/vie_rtp_rtcp_impl.h +++ b/video_engine/vie_rtp_rtcp_impl.h
@@ -110,6 +110,8 @@ virtual int GetEstimatedReceiveBandwidth( const int video_channel, unsigned int* estimated_bandwidth) const; + virtual int GetReceiveBandwidthEstimatorStats( + const int video_channel, ReceiveBandwidthEstimatorStats* output) const; virtual int StartRTPDump(const int video_channel, const char file_nameUTF8[1024], RTPDirections direction);