| commit | 5fe9510efb5a737c1b089d5c36173ed9ea923185 | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Mon Mar 04 15:49:25 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 04 16:57:49 2019 |
| tree | bfb20156e96f569eb6dd842a553ffef41cf09a87 | |
| parent | ac6cf7f089a5c437d1c90e24d1135d6b6be3b25d [diff] |
Move ownership of RTPSenderVideo one more level up, to RtpVideoSender The idea is to let the RtpRtcp and RTPSender classes be responsible for media-agnostic RTP transport, and move out the media-specific processing, such as packetization and media-specific headers. Bug: webrtc:7135 Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26954}
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.