| commit | dbdb3a00797d28c2e21191efd859d01b2533c003 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Tue Jul 17 14:03:46 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Jul 17 14:46:15 2018 |
| tree | 65dea86b28ed87f0c415e6b1240bc7a8e755ce87 | |
| parent | e1d7b23915c87cf52d70ef0f53680bc613ddfa90 [diff] |
Refactoring PayloadRouter. - Move PayloadRouter to RtpTransportControllerInterface. - Move RetransmissionLimiter inside RtpTransportControllerSend from VideoSendStreamImpl. - Move video RTP specifics into PayloadRouter, in particular ownership of the RTP modules. - PayloadRouter now contains all video specific RTP code, and will be renamed in a follow-up to VideoRtpSender. - Introduce VideoRtpSenderInterface. Bug: webrtc:9517 Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38 Reviewed-on: https://webrtc-review.googlesource.com/88240 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24009}
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.