Artem Titov 484acf2723 Add ability to configure sampling rate for input/output video dumps in PC level framework
Bug: b/179986638
Change-Id: I9ab960840e4b8f912abe4fb79cfd9278f4d4562a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33350}
2021-02-26 11:24:52 +00:00
2021-02-18 08:28:24 +00:00
2021-02-24 06:47:39 +00:00
2021-01-20 15:01:07 +00:00
2020-07-13 11:42:07 +00:00
2021-02-14 19:14:44 +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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