requested_resolution.
According to spec, if you ask for three encodings you get three encodings (duh). But according to legacy code, if you ask for three encodings AND codec is VP9, then surely you meant a single encoding that is kSVC where the other encodings influence the scalability mode of the first encoding. Standard simulcast support in VP9 was shipped as an opt-in feature where you have to specify `scalability_mode` and `scale_resolution_down_by` in order to let WebRTC know that you want to disable the legacy path. But `scale_resolution_down_by` is not the only way to configure resolution, there is also the `requested_resolution` code path. This CL adds standard simulcast support for this code path as well. Prior to this change, our parameterized test would have passed in VP8 but failed in VP9. With this change the test passes for all codecs. Bug: webrtc:361124448 Change-Id: Ic5a7136de8abf430813fd01342862775fca145fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360100 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42822}
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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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