commit | d74c56fcd025ba104489995cfe5ffe36922062f7 | [log] [tgz] |
---|---|---|
author | Ruslan Burakov <kuddai@webrtc.org> | Tue Jan 07 16:40:17 2020 +0300 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Jan 21 13:06:18 2020 +0000 |
tree | ca2793f21607a7cc2426cefa71741571169432e5 | |
parent | ccbe95fd8a9e12dc519904b9d16c41590c2a16b6 [diff] |
Add absolute capture time to audio sender path. WebRTC prototype: https://webrtc-review.googlesource.com/c/src/+/158520 Bug: webrtc:10739 Change-Id: I07b7a60602b41dc04292a91923e878a8d753486f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161732 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Ruslan Burakov <kuddai@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30335}
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.