62ec0f6130
This CL adds a cooldown of 0.5 seconds where if the WebRtcVideoChannel created an unsignalled receive stream within that amount of time, if we receive even more unknown ssrcs we simply drop those RTP packets. This prevents getting into a state of spawning new decoders on every single packet which could happen e.g. if PT based demuxing is enabled and MIDs are missing from the packets. Bug: webrtc:12815 Change-Id: Id7675fb0cbfbc72281dcfe030d1a35629df3eb9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221520 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34263}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs