commit | f948eb66aa2cf7ab125c9068d9f2bf8b78df9aca | [log] [tgz] |
---|---|---|
author | Mirko Bonadei <mbonadei@webrtc.org> | Fri Apr 05 13:13:23 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Sun Apr 07 14:32:33 2019 |
tree | e51c029fa48e27f2042319818d3cf125b7957187 | |
parent | 0b2bf9590fc15605a9aea599e7d88180af0976b4 [diff] |
Implement DefaultAudioQualityAnalyzer. 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.
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.