e5b4e941a0
With this change, we catch audio unit start errors and pipe them to the audio session. The audio session notifies its delegate, which can then take appropriate action based on the error code. The signal follows the same path as the playout glitch detection. Bug: webrtc:13119 Change-Id: I8c9f9d2a1e3457447d0ce61ad197f7e1c6392837 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230240 Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Xavier Lepaul <xalep@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34862}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation