commit | 73276ad7ed1e70ab764cd02d7189ed5839fadc20 | [log] [tgz] |
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author | Fredrik Solenberg <solenberg@webrtc.org> | Thu Sep 14 12:46:47 2017 |
committer | Fredrik Solenberg <solenberg@webrtc.org> | Thu Sep 14 12:46:50 2017 |
tree | 2a2e9edab7a943bea5d6189befacf1d66d8aadff | |
parent | 7d1f493a8b042b9ccf6993bb078c83bb0d992094 [diff] |
- Removes voe_conference_test. - Adds a new AudioStatsTest, with better coverage of the same features, based on call_test. - Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses. BUG=webrtc:4690 R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3008273002 . Cr-Commit-Position: refs/heads/master@{#19833}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.