f39c815a1d
This was just checked in all places were it was used, moving the check into RtpRtcp reduces the boiler plate required at the call sites. Also changing to always register and unregister extensions by URI to synchronize the code in AudioSendStream with the code in RtpVideoSender. This prepares for reducing the scope of ChannelSend. Bug: webrtc:9883 Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29490}
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