commit | 98d9c306efbeb487a36c59116c1c5f5b959c3cc4 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Wed Dec 16 10:53:09 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Dec 16 14:01:02 2020 |
tree | cec6124caeedc4d1ca966e7356fd5f6e2c29800f | |
parent | 7bb3afac79f8da7ae30658b0decae2401e14bd42 [diff] |
Don't instantiate any CallStats object in RtpVideoSender tests Tests appear to not depend on any CallStats behaviour, and the usage is not compatible with the threading requirements of the new internal::CallStats class. Bug: webrtc:11581 Change-Id: I8802a46842930eb58bd7609bcd8384ae97e3cf59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197814 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32847}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.