14137a1064
Goal is to be able to retrieve more details about possible microphone conflicts in cases where Init/Start of audio recording fails. Only supported on Android N and higher. Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession. Its value is stored after the recording session has been stopped. Does not affect the media flow or functionality of the ADM. Time to start audio should not be affected either since the new check and logging takes place on a separate ExecutorService thread. See go/webrtc-adm-android for more details and examples. Bug: webrtc:10971 Change-Id: Ia80c1534e326907a1582824225d5f58caa016922 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Alex Glaznev <glaznev@webrtc.org> Reviewed-by: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29236}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct