ehmaldonado ebb0b8ec9a Increase the threshold for RunPlayoutAndRecordingInFullDuplex.
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.

This test ensures that audio starts pumping. The exact performance is
not that important.

R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True

Review-Url: https://codereview.webrtc.org/2391563002
Cr-Commit-Position: refs/heads/master@{#14492}
2016-10-04 08:59:05 +00:00
2016-08-11 14:01:03 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +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. 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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