| commit | 470a5eae9306f8b2b95613e5df9f64aa0438ebd1 | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Wed Jan 23 12:37:49 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 23 12:13:29 2019 +0000 |
| tree | beec8569413befa79eaa633d0358957803488596 | |
| parent | 33b716f7ddb98c2c14be750b4243e49ec0fd31d5 [diff] |
Introduces common AudioAllocationSettings class. This class collects the field trial based configuration of audio allocation and bandwidth in one place. This makes it easier overview and prepares for future cleanup of the trials. Bug: webrtc:9718 Change-Id: I34a441c0165b423f1e2ee63894337484684146ac Reviewed-on: https://webrtc-review.googlesource.com/c/118282 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26370}
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.