| commit | fe047757d65cbb2eff3f337552692e148a0e46d9 | [log] [tgz] |
|---|---|---|
| author | Henrik Lundin <henrik.lundin@webrtc.org> | Tue Nov 19 12:58:11 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Nov 19 12:58:50 2019 +0000 |
| tree | fda647db16b8660761a1f936e0050cb0ddd9aba5 | |
| parent | 29e07e508044236a20c28c3aa18a69d34d66e9b9 [diff] |
Fix a bug in interruption metrics The reported audio interruption metrics are too high. If GetAudio calls start before the first packets are arriving, and the sample rate of the encoded audio is different from the one used to initialize NetEq (default 16 kHz), the initial silent period of GetAudio calls will be reported as an interruption. Modifying a unit test to trigger the bug, and make sure it won't come back. Bug: webrtc:11094, b/144567257 Change-Id: Id540422cb7f35d3bef68b9e7c03c6e7c8bdb8d97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159980 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29831}
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.