072c0086a9004f8f2945bcd6c112abd4d7912999

This is a reland of f8da43d179043f1df2e1c3e2c49494bc23f4ec28 Original change's description: > Replace RecursiveCriticalSection with Mutex in RTCAudioSession. > > Bug: webrtc:11567 > Change-Id: I2a2ddbce57d070d6cbad5a64defb4c27be77a665 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206472 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33259} Bug: webrtc:11567 Change-Id: I4f7235dd164d8f698fe0bedea8c5dca50849f6d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207432 Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33302}
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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
- Reporting bugs
Description
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