commit | 1880c7162bd3637c433f9421c798808cd6eacaf7 | [log] [tgz] |
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author | Sergey Silkin <ssilkin@webrtc.org> | Wed Jan 17 10:36:27 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 17 12:44:06 2018 |
tree | d93256d8a0ab6c83a6f6f21653251acb42dfdcad | |
parent | 8d0f1db3191ea97382d3b0a2cbe5d34e252646a1 [diff] |
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}
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.