commit | 85ba9972c42544c4771e394c9aa1d20bf5d09a91 | [log] [tgz] |
---|---|---|
author | Johannes Kron <kron@webrtc.org> | Wed Aug 28 12:18:08 2019 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 28 11:00:02 2019 +0000 |
tree | cf9f7f094b7a37f55968a735059e294f6f6c567a | |
parent | 5e8ddc360ba44ccb6965a6ae691b08178a65f544 [diff] |
Preserve min and max playout delay from RTP header extension Audio and video synchronization can sometimes override the minimum and maximum playout delay that is set through the RTP header extension. This CL makes sure that the playout delay always is within the limits set by the RTP header extension. Bug: webrtc:10886 Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28980}
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.