Commit Graph

4 Commits

Author SHA1 Message Date
0c43f779f8 Update video histograms that do not have a minimum lifetime limit before being recorded.
Updated histograms:
"WebRTC.Video.ReceivedPacketsLostInPercent" (two RTCP RR previously needed)
"WebRTC.Video.ReceivedFecPacketsInPercent" (one received packet previously needed)
"WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" (one received FEC packet previously needed)

Prevents logging stats if call was shortly in use.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2536653002
Cr-Commit-Position: refs/heads/master@{#15315}
2016-11-30 09:42:32 +00:00
46c4e3caa3 Add unit tests to ReceiveStatisticsProxy class.
BUG=none

Review-Url: https://codereview.webrtc.org/2450363003
Cr-Commit-Position: refs/heads/master@{#14911}
2016-11-03 13:48:36 +00:00
de9e5fffa2 Add stats for frequency offset when converting RTP timestamp to NTP time.
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"

  The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.

Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.

- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).

BUG=webrtc:6579

Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
2016-11-02 14:14:10 +00:00
e5ba44eab1 Implement framesDecoded stat in video receive ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2423823003
Cr-Commit-Position: refs/heads/master@{#14789}
2016-10-26 14:09:29 +00:00