This reverts commit d2d165d47cc7a2aaa53596ad8055ddc30b76101b. Reason for revert: All the regressions! Original change's description: > Always use AV1 specific bitrate limits when spatial layers are used. > > Bug: b/295129711 > Change-Id: I93569027bea34c43e2a3c4de0875e8bbddd5b64e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319283 > Reviewed-by: Michael Horowitz <mhoro@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40719} Bug: b/295129711 Change-Id: I5776edbaba33e86eb10414062ef2b29510f40b8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319880 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40730}
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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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