Markus Handell 8935a7e8eb VideoStreamEncoder: expose screenshare UMA stats.
This change adds a few UMAs to inform on the usage of frame
rate constraints related to screenshare sessions.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: Icdd011a8e7df837416d603beeb0866d9eb1918e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235368
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35238}
2021-10-19 12:29:43 +00:00
2021-08-23 19:52:17 +00:00
2021-10-14 12:28:24 +00:00
2021-10-11 09:07:14 +00:00
2021-01-20 15:01:07 +00:00
2021-07-22 16:41:26 +00:00
2021-09-15 16:56:30 +00:00
2021-09-15 16:56:30 +00:00
2021-08-12 18:37:10 +00:00
2020-07-13 11:42:07 +00:00
2021-08-23 13:37:55 +00:00
2021-10-12 15:10:50 +00:00
2021-09-24 20:09:34 +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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%