| commit | d21f7ab1746c456a84be60d2d692456d8a5edb66 | [log] [tgz] |
|---|---|---|
| author | Tomas Gunnarsson <tommi@webrtc.org> | Sat Jun 27 16:02:39 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jun 29 09:52:44 2020 |
| tree | 64393042048625e8b3bec00cf8091c62e43b571e | |
| parent | f4b956c02671188ba7f4f06a3a2b6ed271565115 [diff] |
Remove media_has_been_sent from RtpState. The field is unused and the way it's currently laid out in the code, it maps to a state in the RtpSenderEgress class - which in turn puts unnecessary threading restrictions on that class. Bug: webrtc:11581 Change-Id: I41a4740c3277317f33f8e815d8c12c70b355c1db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177426 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31577}
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.