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RtpTransportInternal does not need to expose these. They are only used by tests and for setting options. Instead, it can expose a SetRtpOption and SetRtcpOption to set options relevant to each of its transports. Also updates tests to work around no longer having access to internals. This will simplify the composite needed during negotiation of different RTP transport types, as we no longer need to have composites of both RtpTransport and PacketTransport. Bug: webrtc:9719 Change-Id: I91bfa6e95b7aa384d10497f47e7d2483c2e0bef2 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138282 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28066}
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