commit | b8c775aeaf41d796a85402116e7353efb3fb2202 | [log] [tgz] |
---|---|---|
author | Tim Na <natim@webrtc.org> | Fri Jan 10 18:33:05 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Jan 13 18:31:30 2020 |
tree | c86c3ad8552fbf743bda597b2e72fe66d40bcda3 | |
parent | 8234b92ba3aa454c7ed63a7d168f4cab8d3f439e [diff] |
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238}
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.