Andrey Logvin 76ad2e0c05 Add jitter buffer delay metric to pc level tests
Bug: webrtc:11701
Change-Id: I45db3d179150dbad87e1b85a91d9d11feed1cb89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179065
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31699}
2020-07-10 11:30:39 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
.gn
2020-03-18 18:04:41 +00:00
2020-07-06 10:08:09 +00:00
2020-03-30 12:15:56 +00:00
2020-04-16 11:08:43 +00:00
2020-06-15 08:40:44 +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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