commit | a27cfbffdfa0bf359628d2164db5b9d6321f9c9c | [log] [tgz] |
---|---|---|
author | Taylor Brandstetter <deadbeef@webrtc.org> | Sun Jun 20 21:59:48 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jun 21 21:18:02 2021 |
tree | 818564525064e919f6701cca44501d42f6f19b90 | |
parent | 2e3edc1da90a6c050af0bd29ef631073b128da07 [diff] |
Fix echo return loss stats and add to RTCAudioSourceStats. This solves two problems: * Echo return loss stats weren't being gathered in Chrome, because they need to be taken from the audio processor attached to the track rather than the audio send stream. * The standardized location is in RTCAudioSourceStats, not RTCMediaStreamTrackStats. For now, will populate the stats in both locations. Bug: webrtc:12770 Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182 Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34344}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.