commit | 67b011d22c1b343aedad667b387a6baa1cfcb796 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Mon Oct 22 11:00:40 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Oct 22 12:58:33 2018 |
tree | 6675ce118a6eeb6fb4a4e70a374d0f77ccf46805 | |
parent | ff292f30d9a4b7a56aea872fe488d342f47202a3 [diff] |
Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream Followup to cl https://webrtc-review.googlesource.com/70880, which introduced the interface. Intended to enable tests using MockBitrateAllocator. Bug: None Change-Id: I0a784106acf37ff9aca118297233ebd2f2259ae4 Reviewed-on: https://webrtc-review.googlesource.com/c/107342 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25290}
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.