commit | 00a6ab568bf021a2861ef15e4a679812c58250cb | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Wed Jan 09 15:35:07 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 09 16:21:11 2019 |
tree | 39db6d60ce31022e6045c9ed89f22fe30d52d149 | |
parent | 278f82516f6a1b7656053c46f44401090b4415ca [diff] |
Check timestamp difference when choosing to extract multiple packets from the jitter buffer. This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls. Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1 Bug: webrtc:10167 Reviewed-on: https://webrtc-review.googlesource.com/c/115419 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26179}
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.