Sergey Silkin 26d3e569be Add AV1 screencast perf test
Bug: b/348784414
Change-Id: If1b3bf2439280eba65cf66cc3699e11a0ef412f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355300
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42524}
2024-06-24 11:55:59 +00:00
2024-06-14 08:12:55 +00:00
2024-05-31 13:20:31 +00:00
2023-10-30 14:56:36 +00:00
2024-06-24 11:55:59 +00:00
2022-12-02 09:21:47 +00:00
2024-05-22 13:32:41 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +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%