| commit | d2a6686a1066f52e1edf861465afff18223388aa | [log] [tgz] |
|---|---|---|
| author | Chen Xing <chxg@google.com> | Mon Jun 03 12:53:42 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jun 03 14:37:01 2019 |
| tree | 8fd9d92bddf1525d044388d710310f9a52aed91d | |
| parent | 1df841d446ebc5ef8f2e27e03025d5670d67c7de [diff] |
Add RtpPacketInfo to hold information about a received RtpPacket. This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet. Bug: webrtc:10668 Change-Id: I962df493a76692f668314f78d6792d7636c5a31b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28138}
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.