This is a reland of f394f65b7195a5f9f71c9f0dec3bb68300590b5b Original change's description: > Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler. > > In addition restore call to ConfigureQualityScaler in SetSource, which > is needed if degradation preferences change mid-stream. > > Fixes a regressions from https://webrtc-review.googlesource.com/70740, > The encoder's GetScalingSettings may depend on arguments to > InitEncode, so configuring the quality scaler only at encoder creation > time isn't enough. > > Bug: webrtc:8830 > Change-Id: I48f66cde219c56272f44441fdb26ec64c6002068 > Reviewed-on: https://webrtc-review.googlesource.com/72000 > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22993} Bug: webrtc:8830 Change-Id: I96f0b8c27e3b58b9e9268117267abcb63ee7fe73 Reviewed-on: https://webrtc-review.googlesource.com/72320 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23071}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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