a006ba152fb20cbf04da78bd4dbac7ffd3a16b65

The new frame buffer is already the default. Sync decoding can now be inferred by the presence of a metronome rather than using the field trial. Tests have been updated to use the DecodeSynchronizer rather than the field trial. Bug: webrtc:14003 Change-Id: I33b457feaf4eac1500f3bf828680e445ae4d93cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274163 Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38011}
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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
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