commit | f7f8a1f1764755f6079c6a1c0c5ab49f961e2ba8 | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Tue Aug 28 17:45:31 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 29 08:44:08 2018 |
tree | ef555b2e477a2030c7d85bdab9332725e6f46f00 | |
parent | 2a4906532f9de4b247349af6f8f75d8931fb9e38 [diff] |
Cleanup RtpPacketizer interface merge construction and call to SetPayloadData Add NumPackets instead of SetPayloadData Remove virtual ToString() as unused move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo Bug: webrtc:9680 Change-Id: I074644e048c797eb836f79979df363fe1ea0075e Reviewed-on: https://webrtc-review.googlesource.com/96543 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24474}
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.