Adding guards around audio_unit_. Changes: 1. Moving audio_is_initialized_ = true/false to InitPlayOrRecord and ShutdownPlayOrRecord respectively, for clarity. 2. Checking audio_is_initialized_ when accessed, not just DCHECKing. Bug: webrtc:408364830 Change-Id: Ibd383a2baaa074e5515cc48f50a52738200ebf83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/384440 Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Reviewed-by: Anders Lilienthal <andersc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44310}
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.