Danil Chapovalov 731d1cabeb Reduce flakiness of asynchronous RtcpTransceiver tests
Restructure tests to never wait for no packets,
Greatly increase wait timeout.
(Reduce expectation of synchronous primitives precision)

Bug: webrtc:8494
Change-Id: I9a80fda3a2bf527d8b7337ecabaf625e543b8c62
Reviewed-on: https://webrtc-review.googlesource.com/20502
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20584}
2017-11-07 13:08:05 +00:00
2017-11-07 08:57:21 +00:00
2017-10-31 21:08:43 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-10-31 17:46:42 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

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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