f948eb66aa
The DefaultAudioQualityAnalyzer will read stats reports (temporarily using the old PeerConnectionInterface::GetStats) and for each audio stream it will collect some NetEq related stats. When DefaultAudioQualityAnalyzer::Stop is invoked by the framework, it will report the following metrics: - expand_rate - accelerate_rate - preemptive_rate - speech_expand_rate - preferred_buffer_size_ms Bug: webrtc:10138 Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27474}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct