| commit | 1e80ce438eefab6394fb176fdcb8938a13dda16c | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Fri Feb 19 15:02:15 2016 |
| committer | Danil Chapovalov <danilchap@webrtc.org> | Fri Feb 19 15:02:24 2016 |
| tree | 7b834ab761cb9631ecb605d0e77e49b04d3cc94c | |
| parent | c51d6947e4f2e9faabc5518f7f33aa60f4e1ae0b [diff] |
webrtc::RtpPacket name freed for better RtpPacket There were two different structures named RtpPacket in webrtc namespace: RtpPacket defined in fec_test_helper renamed to test::RawRtpPacket RtpPacket defined in rtp_sender_video and producer_fec removed as unused BUG=webrtc:5261 R=sprang@google.com, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1710103004 . Cr-Commit-Position: refs/heads/master@{#11682}
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.