| commit | 26c59ff6caa4555289c70b2e8ad226fa473f77c2 | [log] [tgz] |
|---|---|---|
| author | Jakob Ivarsson <jakobi@webrtc.org> | Thu Feb 28 08:55:49 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 28 15:51:31 2019 |
| tree | 3911990a436a96dcbf1110fc9a20d0869e816c79 | |
| parent | c58c01d6d4c34e373002f69619553b58c94b57f9 [diff] |
Fix jitter buffer delay reporting. Previously, if more than one packet is extracted in a GetAudio call then an incorrect number of samples will be reported. Bug: webrtc:10363 Change-Id: Ia1bcc87a0e0082060e4f746d37a4008735eec6b3 Reviewed-on: https://webrtc-review.googlesource.com/c/124829 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26903}
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.