eea605deeb
Previously this functionality only worked correctly with a single Transport instance, meaning a single video track. This CL moves the transport pointer from being a member in FakeNetworkPipe to being set on each packet, so that e.g. audio packets point to the audio transport and video packet to the video transport. This means we need a separate adapter per stream in DegradedCall. Additionally, since Transport instances can potentially be destroyed before it's time to forward the message to it, we need to keep track of which instance that are live and ignore packets we can't forward. Bug: None Change-Id: I314d431c04ff81c3859cf661e2722c99342f785e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148586 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28831}
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