Erik Språng bf46cfef22 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}
2020-05-11 17:14:33 +00:00
2020-04-21 09:06:37 +00:00
2018-10-05 14:40:21 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
2020-03-18 18:04:41 +00:00
2020-05-07 17:04:15 +00:00
2020-05-08 19:06:36 +00:00
2020-03-30 12:15:56 +00:00
2018-07-23 15:28:48 +00:00
2020-04-16 11:08:43 +00:00
2020-05-11 05:38:59 +00:00

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.

Development

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.

More info

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%