commit | 18bc3e19c42915ebdbbd5cc3dffc749f55c07178 | [log] [tgz] |
---|---|---|
author | Sergey Silkin <ssilkin@webrtc.org> | Wed Jan 17 13:15:57 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 17 13:16:07 2018 |
tree | 3599d4d382ee7ec23614c2aeffffc057fd989741 | |
parent | 53d877c0f85fc690d8ea64d12bf939fdcc076e75 [diff] |
Revert "Updated analysis in videoprocessor." This reverts commit 1880c7162bd3637c433f9421c798808cd6eacaf7. Reason for revert: breaks internal tests Original change's description: > Updated analysis in videoprocessor. > > - Run analysis after all frames are processed. Before part of it was > done at bitrate change points; > - Analysis is done for whole stream as well as for each rate update > interval; > - Changed units from number of frames to time units for some metrics > and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to > 'time to reach target bitrate, sec'; > - Changed data type of FrameStatistic::max_nalu_length (renamed to > max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to > use such advanced data type in such low level data structure. > > Bug: webrtc:8524 > Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f > Reviewed-on: https://webrtc-review.googlesource.com/31901 > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21653} TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8524 Reviewed-on: https://webrtc-review.googlesource.com/40220 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21656}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.