| commit | 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a | [log] [tgz] |
|---|---|---|
| author | Minyue Li <minyue@webrtc.org> | Thu Jan 23 12:45:50 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jan 23 16:06:12 2020 |
| tree | bf8c4008fa96ec309dbe23055b8fd89d1672f4e1 | |
| parent | cdd73e095cd4279c80697692a8652cf6129171db [diff] |
Send absolute capture time through audio coding module. Bug: webrtc:10739 Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30363}
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.