commit | f8da43d179043f1df2e1c3e2c49494bc23f4ec28 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Mon Feb 15 11:44:27 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 15 14:35:38 2021 |
tree | f315b05cdad2b811a84740a56eefed2f1ab5c8c4 | |
parent | a24f3d035f7d9acee551d5547f4fb16838cf98d6 [diff] |
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}
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.
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.