fd77b78821
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and RtpPayloadRegistry. Only intended change in behavior is that packets with unknown payload types are not discarded at this level of the stack. They are discarded higher up, in Channel::ReceivePacket (audio) and RtpVideoStreamReceiver::ReceivePacket (video). Bug: webrtc:8995 Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf Reviewed-on: https://webrtc-review.googlesource.com/92087 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24196}
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