cc7125f240
When a simulcast stream is enabled or disabled, we want this state change to be reflected properly in the RtpRtcp modules. Each video send stream can contain multiple rtp_rtcp_modules pertaining to different simulcast streams. These modules are currently all turned on/off when the send stream is started and stopped. This change allows for individual modules to be turned on/off. This means if a module stops sending it will send a bye message, so the receiving side will not expect more frames to be sent when the stream is inactive and the encoder is no longer encoding/sending images. Bug: webrtc:8653 Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4 Reviewed-on: https://webrtc-review.googlesource.com/42841 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21880}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct