In current state, the SDP parser in webrtc is not backward compatible with clients that might still be using RTP data channels. Obviously, this isn't there is no such usecase in webrtc since the code is deleted, but in Meta we still use it and would like to be able to negotiate between clients that offer RTP data channels. Instead of erroring the parsing procedure, we can parse it as unsupported media in the client that no longer supports RTP data channels. Replaced the existing test that expects parsing failures with a test that validates that the content was parsed as unsupported media. Bug: webrtc:14872 Change-Id: I4c105cf55e33b8c19b2849e16148b8175053c40c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291190 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39252}
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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.
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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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