| commit | de9e5fffa21401df94fd89b15be5e549cbe3e142 | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Wed Nov 02 14:14:03 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Nov 02 14:14:10 2016 |
| tree | 163166867e25e25f8d4a39e51377bf99a1665a3c | |
| parent | 0dbcfa51a27dc3efa936c2facc49e9fe9c066531 [diff] |
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}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.