bdb5830d69
The Android native audio code asks the OS to provide an appropriate buffer size for real-time audio playout. We should add logging for this value so we can see what values are used in practice. Bug: b/157429867 Change-Id: I111a74faefc0e77b5c98921804d6625cba1b84af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176126 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreasson <henrika@chromium.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31368}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct