commit | bf46cfef2201acdc0c98c00c6e0e484ff4bf42da | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Mon May 11 16:22:02 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Mon May 11 17:14:33 2020 |
tree | 1e5560df6ca27aa3a54b58e9ccf565a162bb2921 | |
parent | 8e321cd690ad2789c43bf1e2e2ce4267cd130eb8 [diff] |
Refactors send rate statistics in RtpSenderEgress When FEC generation is moved to egress, we'll need to poll bitrates from there instead of the RtpVideoSender. In preparation, refactoring some getter methods. For context, see https://webrtc-review.googlesource.com/c/src/+/173708 Bug: webrtc:11340 Change-Id: Ibc27362361ee9640d9fce676fc8e1093a579344f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174202 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31214}
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.