commit | 741164813a08195394ed26c8b0bb85051015886a | [log] [tgz] |
---|---|---|
author | Steve Anton <steveanton@webrtc.org> | Mon Dec 18 19:00:14 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Dec 18 23:37:47 2017 |
tree | 21cec238236aeb87eee36bcda14990e7db84c414 | |
parent | d5247510dc9cf1b1f8eb1f79310bef702b7804c1 [diff] |
Remove SessionStats.proxy_to_transport The stats collectors would only ever call this on the signaling thread, so they might as well just ask the voice/video channel their transport_name directly. Bug: None Change-Id: I8dd36210ff22d222b45b5f5e22c253f601cdc79e Reviewed-on: https://webrtc-review.googlesource.com/34581 Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21339}
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.