commit | 51bf200294622a45444b68ad1498a41f8a860df3 | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Fri Oct 11 08:53:27 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Oct 11 10:59:21 2019 |
tree | 9f68e49ffd8dff87fde21cf3bd5b3cfc49f575e6 | |
parent | 4b64411406af7b35e330708b2eddf82d468cf32b [diff] |
Reduce number of RTPVideoSender::SendVideo parameters use frame_type from the RTPVideoHeader instead of as an extra parameter merge payload data and payload size into single argument pass RTPVideoHeader by value (relying on copy elision) Bug: None Change-Id: Ie7970af3b198b83b723d84c7a8b047219c4b38c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156400 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29445}
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.