| commit | 9c166e064fb2da8273e2d997ce182de49091dbd5 | [log] [tgz] |
|---|---|---|
| author | Per K <perkj@webrtc.org> | Fri Jan 26 07:34:34 2024 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Jan 26 09:19:50 2024 |
| tree | 47eceb3804185d601ce878ce5df4d70474a11290 | |
| parent | 9a953b28f96a03cc1b95ddff7e167700f4279d34 [diff] |
Remove VideoSendStream::StartPerRtpStream Instead, always use VideoSendStream::Start. VideoSendStream::StartPerRtpStream was used for controlling if individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers. With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not. The cleanup is done to simplify code and in the future allow sending probe packet on a RtpTransceiver that allows sending, regardless of the RtpEncodingParameter.active flag. Bug: webrtc:14928 Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41619}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.