2d6c4d07129f88d599fc41965c22c7faaf09e793

Avoids using webrtc::GlobalMutex. Since state is allocated on first use and never destroyed, we avoid an exit-time destructor when building with absl::Mutex. Bug: webrtc:11567 Change-Id: Ib9c6480ab0474e37a853460115b35d961b93009c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258080 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36455}
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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
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