| commit | 3ef3bfc2aafa707985c9e9dcd4cfb6ccbc525628 | [log] [tgz] |
|---|---|---|
| author | Henrik Lundin <henrik.lundin@webrtc.org> | Tue Apr 10 13:10:26 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Apr 10 21:32:55 2018 |
| tree | f0ae34d12d636be7cf1515576d4331761e246839 | |
| parent | f0482ea9dd5772e98c18550b443f516d92c372ff [diff] |
Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent These two new histograms relate to the packet-loss concealment that happens when audio packets are lost or late for decoding, and the NetEq must resort to extrapolating audio from the previously decoded data. Bug: webrtc:9126 Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b Reviewed-on: https://webrtc-review.googlesource.com/67861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22812}
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.