| commit | 9a394f0649d39737a791d17e73ad2603e3d162fb | [log] [tgz] |
|---|---|---|
| author | hbos <hbos@webrtc.org> | Wed Dec 14 07:58:22 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Dec 14 15:58:30 2016 +0000 |
| tree | 4eb58d8f51a3895940115a7cf2af0a0905632cbc | |
| parent | c3c2f318521270ac42e3721d707259a29288e98c [diff] |
Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. Due to the Chromium implementation[1] of GetAudioProcesssingStats, echoReturnLoss and echoReturnLossEnhancement could default to -100 when no value was available. This should be improved by using rtc::Optional or AudioProcessorInterface::GetStats being able to return false, but this requires a bunch of refactoring. In the meantime we "blacklist" the value -100 which is a nonsense value anyway. In that case echoReturnLoss[Enhancement] is correctly left undefined. [1] https://cs.chromium.org/chromium/src/content/renderer/media/media_stream_audio_processor_options.cc?sq=package:chromium&dr=C&rcl=1481530670&l=461 BUG=chromium:669877 Review-Url: https://codereview.webrtc.org/2573443002 Cr-Commit-Position: refs/heads/master@{#15611}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.