commit | 59ab1cf081967353c1ae31f9d8111ec9b17284b2 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Wed Feb 06 21:48:11 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 07 13:31:48 2019 |
tree | 5558c307f46a71b2c160a59643dfa433ddc412c9 | |
parent | 938dd9f1e80c11264eac511ec4ab3bf42a160fcf [diff] |
Move ownership of RTPSenderVideo and RTPSenderAudio one level up From RTPSender to RtpRtcpImpl. Makes RTPSender operate on packets only, not frames. Bug: webrtc:7135 Change-Id: Ia9a11456404c3b322d873d4f8fb828742296b26d Reviewed-on: https://webrtc-review.googlesource.com/c/120044 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26586}
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.