0c1a934e260ee257ed9c8f2c312873f54195ba0e

This removes the unused field trials `WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` and `WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` as well as the `video_hysteresis` and `screenshare_hysteresis` parameters in `WebRTC-VideoRateControl`. The hysteresis parameters in `WebRTC-StableTargetRate` are currently left, their future is unclear... Bug: webrtc:9734 Change-Id: I9e6bbe4b630a0501d365bf69e87e65164c500122 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269207 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Auto-Submit: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37635}
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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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