Use RunLoop in audio/ tests Bug: webrtc:469327588 Change-Id: I6f7a6f3d88184aa38f5a8597fc29acc56a6a6964 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461346 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47316}
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.