1. Add loss threshold for high bandwidth preference. If the average loss ratio is less than the threshold, then the model prefers higher bandwidth candidates. Otherwise, it prefers lower bandwidth candidates. Before, it always prefers higher bandwidth candidate. The default value is 0.99, means it always prefers high bandwidth candidates. 2. Only increase the estimate if the inherent loss (random loss) is equal to/greater than the average loss. If the inherent loss is less than the average loss, then it is oversending, thus should not increase the estimate. Bug: webrtc:12707 Change-Id: I37eb536679ca29e017a4a47703b417efd4d103ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269101 Commit-Queue: Diep Bui <diepbp@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37608}
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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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