This is a reland of b141c162ee2ef88a7498ba8cb8bc852287f93ad2 Original change's description: > Take out listen support from AsyncPacketSocket > > Moved to new interface class AsyncListenSocket. > > Bug: webrtc:13065 > Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35234} Bug: webrtc:13065 Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35260}
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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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