Prior to this change, the receiver_time metric had huge outliers whenever FlexFEC was enabled. This was due to a measurement problem, where the time of the incoming packet was incorrectly set to zero. This happened for packets that were lost in transit, but recovered through FEC. This CL fixes this problem by simply not recording samples where the incoming packet time is undefined. The CL also removes the possibility of timestamp collisions in the data structures. TESTED=Ran './webrtc_perf_tests --gtest_filter="*ForemanCifPlr5H264Flexfec*" | grep receiver_time' locally 10 times, without experiencing any outliers. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2596793002 Cr-Commit-Position: refs/heads/master@{#15735}
Revert of Refactor webrtc/modules/video_processing for GN check (patchset #3 id:40001 of https://codereview.webrtc.org/2595543002/ )
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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. This page is maintained by the Google Chrome team.
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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