As the exposure of power efficient stats to JavaScript are limited as to reduce the fingerprinting surface to getStats, a new RTCStatsMember derivation, RTCLimitedStatsMember, was added in this change. This sets the exposure criteria of the stat on the type, which keeps the size of the RTCStatsMember class the same and allows for extension in the future for new types of stat restrictions. Bug: webrtc:14483 Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021 Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38576}
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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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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