Niels Möller e7b9e6b17d Move RtpSenderVideo tests to separate file.
Also refactor most of the RtpSender tests to not use the frame-level
method RTPSender::SendOutgoingData.

Bug: webrtc:7135
Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7
Reviewed-on: https://webrtc-review.googlesource.com/c/121461
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26577}
2019-02-06 18:00:39 +00:00
2018-10-05 14:40:21 +00:00
2019-02-06 10:24:07 +00:00
2019-02-04 19:46:30 +00:00
2019-01-29 12:16:19 +00:00
.gn
2018-08-13 13:54:05 +00:00
2019-01-30 15:45:10 +00:00
2018-12-18 12:30:58 +00:00
2018-11-09 14:23:59 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2019-01-31 22:09:16 +00:00
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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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