Danil Chapovalov 2454d85bb6 Cleanup rtp_rtcp mocks
Modernise function to unified MOCK_METHOD macro, delete few deprecated functions on the way.
Remove default constructors to stress they do nothing special

Bug: None
Change-Id: Ie126f38f0589acb65886f25f754ca575c17af29b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174583
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31191}
2020-05-08 13:43:15 +00:00
2020-04-21 09:06:37 +00:00
2020-05-08 13:43:15 +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
2020-05-07 17:04:15 +00:00
2020-03-30 12:15:56 +00:00
2020-04-16 11:08:43 +00:00
2020-04-21 13:15:09 +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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