3ea55d56eb
This is a reland of 171df9326200d1e01bce530e2ff01ac5890e6cb7 Original change's description: > Delete RtpUtility::Payload, and refactor RTPSender to not use it > > Replaced by a payload type --> video codec map in RTPSenderVideo, > where it is used to select the right packetizer. > > Bug: webrtc:6883 > Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f > Reviewed-on: https://webrtc-review.googlesource.com/c/119263 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26380} Tbr: danilchap@webrtc.org Bug: webrtc:6883 Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4 Reviewed-on: https://webrtc-review.googlesource.com/c/119661 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26394}
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