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Functionality wise, there should be no change with this CL, aside from updating tests to anticipate OnPacketReceived to handle the packet asynchronously (as already was the case via BaseChannel). This only removes the network->worker hop out of the BaseChannel class into the WebRTC MediaChannel implementations. However, it updates the interface contract between BaseChannel and MediaChannel to align with how we want things to work down the line, i.e. avoid hopping to the worker thread for every rtp packet. The following steps will be to update the video and voice channel classes to call Call::DeliverPacket on the network thread and only handle unsignalled SSRCs on the worker (exception case). Bug: webrtc:11993 Change-Id: If0540874444565dc93773aee89d862f3bfc9c502 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33040}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs