This change adds a median filter that can replace the IIR filter that is currently used to estimate the avg frame size (in bytes). It is enabled through a boolean, and reuses the window length from the max percentile filter. The median filter is only used by the delay calculation in `CalculateEstimate()`. It does not replaced the use of the IIR estimate in the size outlier rejection heuristic. Bug: webrtc:14151 Change-Id: I519b6b57a8bee3c41a300ed2e92a1981c61cca15 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275121 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38077}
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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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