Niels Möller f8da43d179 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}
2021-02-15 14:35:38 +00:00
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2020-07-13 11:42:07 +00:00
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2021-02-14 19:14:44 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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