commit | bc900cb1d1810fcf678fe41cf1e3966daa39c88c | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Wed Mar 21 11:50:22 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Mar 21 12:55:08 2018 |
tree | fb29d8f8e5a0a7fbfa976cbca0cba14509510d46 | |
parent | c3d1e09a25107618b743b6b36e4212321048a917 [diff] |
Move rtp-specific config out of EncoderSettings. In VideoSendStream::Config, move payload_name and payload_type from EncoderSettings to Rtp. EncoderSettings now contains configuration for VideoStreamEncoder only, and should perhaps be renamed in a follow up cl. It's no longer passed as an argument to VideoCodecInitializer::SetupCodec. The latter then needs a different way to know the codec type, which is provided by a new codec_type member in VideoEncoderConfig. Bug: webrtc:8830 Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 Reviewed-on: https://webrtc-review.googlesource.com/62062 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22532}
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.