philipel 20d05a9f71 Now expect the correct number of streams in EndToEndTest.GetStats.
The rtx streams were not included in the number of expected streams
but the test passed most of the time anyway due to how the checking was done.
Flake was caused when the number of registered streams jumped passed the
number of expected send streams excluding the number of rtx streams.

BUG=webrtc:6879

Review-Url: https://codereview.webrtc.org/2580343002
Cr-Commit-Position: refs/heads/master@{#15671}
2016-12-19 12:17:27 +00:00
2016-12-18 21:14:50 +00:00
2016-06-14 09:39:40 +00:00
2016-12-18 21:14:50 +00:00
2015-09-11 09:04:09 +00:00
2016-11-23 16:42:57 +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. This page is maintained by the Google Chrome team.

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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