| commit | 70ffead25675c4761467755f9be844335dd59dba | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Wed Jul 20 13:26:59 2016 |
| committer | Danil Chapovalov <danilchap@webrtc.org> | Wed Jul 20 13:27:09 2016 |
| tree | babc739674a6b2b6a47271251dc87f7f1bac9ebb | |
| parent | f39f7d931c11045fe0ba842e4eba9b816f0288ca [diff] |
Reimplemented fix for bogus RTP timestamp in RTCP packet created before RTP packet. Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too. It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet. BUG=webrtc:1600 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1639253007 . Cr-Commit-Position: refs/heads/master@{#13503}
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.