| commit | 12d6a49e97701eebc753e13822355b823190ef4c | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Thu Mar 22 11:41:48 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 22 12:34:34 2018 |
| tree | 40ea6bd2e4097f92e3266f82051031fa083c2ce5 | |
| parent | 9cfb18c5b3812a1bb609ddfac618c9cabef4cf0d [diff] |
Add payload_name and payload_type to VideoSendStream::Config::Rtp. Another step of the transition needed to reland cl https://webrtc-review.googlesource.com/62062, and move payload_name and payload_type out of VideoSendStream::Config::EncoderSettings. If the new fields are set, values of the old fields are ignored. Bug: webrtc:8830 Change-Id: I1f0cd56fd6b13b05608b284afc92523707887e25 Reviewed-on: https://webrtc-review.googlesource.com/64101 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22562}
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.