| commit | 25d31ec440d1aeb6ea06bd42f3b2b6dbf484b23c | [log] [tgz] |
|---|---|---|
| author | philipel <philipel@webrtc.org> | Wed Aug 08 16:33:01 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 08 15:28:20 2018 +0000 |
| tree | 135fd5df1c7737bd12d1fe8a01b5d002b78645ff | |
| parent | 0b9e01d605a174a52635626c885738a222abff46 [diff] |
Add shared frame id state to RtpVideoSender. When using the generic descriptor we want all simulcast streams to share one frame id space (so that the SFU can switch stream without having to rewrite the frame id). The state also needs to be restored when the RtpVideoSender is recreated. Note that |shared_simulcast_frame_id_| is only added, but not used in this CL. Actually using it will be part of the next CL. Bug: webrtc:9361 Change-Id: I7192a06d6ae4cab118ca5996ed99a56888ad1d97 Reviewed-on: https://webrtc-review.googlesource.com/92803 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24233}
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.