| commit | 32d989b3f2d168327ed43d0e4c493550ccee4179 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Tue Dec 15 14:55:10 2015 |
| committer | Stefan Holmer <stefan@webrtc.org> | Tue Dec 15 14:55:20 2015 |
| tree | 20f91a2ffaea44a6d7fa707229e303e994cded08 | |
| parent | 10aea22c8476bc4666ac9b3a90f48b4741091c0f [diff] |
Disable transport sequence numbers for audio. Since this isn't fully wired up yet it shouldn't be part of the SendSideBwe experiment yet. BUG=webrtc:5263 R=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1523283002 . Cr-Commit-Position: refs/heads/master@{#11029}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.