commit | 053c371552cf9f15bab6ff5264d5ac03621aa71a | [log] [tgz] |
---|---|---|
author | Karl Wiberg <kwiberg@webrtc.org> | Thu May 16 13:24:17 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue May 21 03:10:49 2019 |
tree | e4d75b5896d9cfdbe1905ffdd941e1c02e2ff6b6 | |
parent | d9f02f64e80611d8b387ce37f96b7d0480296677 [diff] |
Audio coding: Don't choke when RTP timestamp rate > sample rate Bug: webrtc:10631 Change-Id: If0422786172502f039acc2cac5e8c13b637af54c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137048 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27998}
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.