This CL: - makes it easier to understand the (nontrivial) metric interpretation - corrects the computation of BufferDelay to use 0 for absent delay - deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701 - updates the unit test to directly test metric reporting Corresponding update to histograms.xml: https://crrev.com/c/3944909 Bug: webrtc:8671, chromium:1349051 Change-Id: If73b6fca4de7343bff2c53f72cedda458d36c599 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278782 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38362}
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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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