Sergey Silkin 1fdafaeb21 Calculate bitrate and frame rate mismatches in video codec tests
Bug: webrtc:10812
Change-Id: I3408c0d7adefc37d088a5c6e10fce4f95aa1b668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228943
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34782}
2021-08-17 10:33:08 +00:00
2021-08-05 07:39:25 +00:00
2021-08-17 10:03:08 +00:00
2021-01-20 15:01:07 +00:00
2021-07-22 16:41:26 +00:00
2021-08-12 18:37:10 +00:00
2021-08-16 09:54:27 +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
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