This is a reland of 52fd96fb731d9a911bd064a9718d96fef0bd5b24 Original change's description: > Migrate test/time_controller to webrtc::Mutex. > > Bug: webrtc:11567 > Change-Id: I26fb07bf84ed197ce667290aa0bf4816bc9c5c06 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178818 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Markus Handell <handellm@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31660} Bug: webrtc:11567 Change-Id: I4979b6be8ac5ec661e0f18cca4d0c185916233bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178876 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31687}
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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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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