This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3. Reason for revert: Broke internal tests. Original change's description: > Use PayloadTypePicker for video PT assignment > > This includes changes that change the order of codecs. > It is preparatory to doing late assignment of video PTs. > > Bug: webrtc:360058654 > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43489} Bug: webrtc:360058654 Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43490}
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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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