| commit | cebdbf650dbd3a718b4b77389a183ebf8928ed99 | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Fri Dec 13 16:08:18 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Dec 16 12:08:11 2019 +0000 |
| tree | c2c824fdd372cc86cd7e28b9a4107ed7aa38d2fc | |
| parent | 0f6bf75ab4a7fec6a943ebb976d926eebff8ec22 [diff] |
switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface instead of creating each time an object with RtpDepacketizer interface this moves packet payload memcpy from RtpVideoStreamReceiver into the depacketizers with possibility to remove it from there in follow ups. Bug: webrtc:11152 Change-Id: If474207eb84d7e9d0207075bd395e60895f0d842 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162185 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30095}
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.