commit | cc8e8bb73f8384045d808eace924a955422bc111 | [log] [tgz] |
---|---|---|
author | Piotr (Peter) Slatala <psla@webrtc.org> | Thu Nov 15 08:26:19 2018 -0800 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Nov 15 17:36:48 2018 +0000 |
tree | 30314e52dfe60868f1ae2e3a2c04236eeed875de | |
parent | 86336a50bd43c3eff64ef97dfda419c5ad5a1ed1 [diff] |
Pass the media transport from JsepTransportController to Call. Add TargetRateObservers for media transport in the call object. Bug: webrtc:9719 Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7 Reviewed-on: https://webrtc-review.googlesource.com/c/110622 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25662}
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.