RtpPacket test for the abs-capture-time extension
Clarify when the RTP header extension can be set depending on the value of the `extmap-allow-mixed` option and on whether the header extension ID is for one-byte or two-bytes extensions. Bug: b/270541827 Change-Id: I4b939f6862d1f19cbfea11518a1cc1507beb2362 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294920 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39399}
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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.
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- 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
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Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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