commit | a6e7b88198441d4209388c0559c6e765b4387ecc | [log] [tgz] |
---|---|---|
author | Åsa Persson <asapersson@webrtc.org> | Fri Jan 19 13:57:10 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Jan 22 09:02:56 2018 |
tree | 7da45a24bc3644bb1ddcbdae6df4c668bd1a96a3 | |
parent | 7cfff23fe1a8f5f8d9a7e1bf92855c009b49c5ee [diff] |
Move rtp_timestamp_to_frame_num_ map from VideoProcessor to Stats class. Let Stats class handle rtp timestamp to frame number mapping. Bug: none Change-Id: I2a29c89a25c75c4bbd6c6368a5d10514f90b3c42 Reviewed-on: https://webrtc-review.googlesource.com/41220 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21709}
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.