| commit | 1723cf9fa2e5b215e9a45923aaa264c1b9dd7dcb | [log] [tgz] |
|---|---|---|
| author | Sergey Silkin <ssilkin@webrtc.org> | Mon Jan 22 14:49:55 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jan 22 15:45:58 2018 |
| tree | 00fe33fc219e004818799c8b53bb75fc391a50c5 | |
| parent | e93de5ff70eaffcb413c620ab71664673b8e70bd [diff] |
Get rid of packet loss related stuff from videoprocessor. This feature is not needed in video codec testing framework. In WebRTC video codecs never deal with packet loss. Packet loss is handled by jitter buffer which prevents passing of incomplete frames to decoder. Bug: webrtc:8768 Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6 Reviewed-on: https://webrtc-review.googlesource.com/40740 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21722}
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.