69662a99d30da64cbe064f5cc646bb75d77c0b71

AudioAttributes::getAllowedCapturePolicy was added in API Level 29. Update WebRtcAudioTrack to add API Level check before using the API. Bug: webrtc:12250 Change-Id: Ica6604eb1d7fa736a0e64729a022eefcfb7b3020 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195941 Commit-Queue: Gaurav Vaish <gvaish@chromium.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32735}
Document purpose and wire format of extension http://www.webrtc.org/experiments/rtp-hdrext/video-layers-allocation00
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
- Reporting bugs
Description
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