Niels Möller 2d6c4d0712 Move global libsrtp usage count into a singleton class
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}
2022-04-06 07:41:52 +00:00
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2020-07-13 11:42:07 +00:00
2021-08-23 13:37:55 +00:00
2021-12-16 17:45:31 +00:00

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

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