Added script 'apm_quality_assessment_optimize' for finding parameters that minimize a custom function of the scores generated by APM-QA. The script reuses the existing functionality for filtering the data on configs/scores/outputs. To archieve that, some modularization has been done: the part from apm_quality_assessment_export that reads in data into a pandas.DataFrame has been moved into quality_assessment.collect_data. TESTED = though extensive manual tests. Unit tests for the user scripts and 'collect_data' are missing, because we don't have a test framework for loading/exporting fake data. BUG=webrtc:7218 Change-Id: I5521b952970243da05fc4db1b9feef87a2e5ccad Reviewed-on: https://chromium-review.googlesource.com/643292 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19780}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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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