| commit | cd8a6e2f38cb3caf66b57bd0336211cef2b8e7f0 | [log] [tgz] |
|---|---|---|
| author | Chen Xing <chxg@google.com> | Mon Jul 01 08:56:51 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jul 03 14:07:36 2019 |
| tree | e68c09aa443aa86e0c4ee4f5880225591d66932b | |
| parent | 53d45baa50361b65691582579e39e51e835641a5 [diff] |
Add writing and parsing of the `abs-capture-time` RTP header extension. This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at: http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time We are still missing the code to: - Negotiate the header extension. - Collect capture time for audio and video and have the info sent with the header extension. - Receive the header extension and use its info. Bug: webrtc:10739 Change-Id: I75af492e994367f45a5bdc110af199900327b126 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#28468}
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.