92594a30ce02aed75f8a2a9f21e5b8c5c4e5f199

Currently, FFT is performance when AEC buffers farend signal. This has some drawbacks 1. memory inefficiency: two ring buffers are needed; 2. computation inefficiency: if ringbuffer gets wrapped around, some FFT computation will be wasted; 3. accessibility: the main AEC function looses accessibility to the time-domain signal. Therefore, this CL tries to buffer time domain data, which is buffered any way if a debugging macro is defined, and calculate the FFTs where they are actually used. BUG= Review URL: https://codereview.webrtc.org/1512573003 Cr-Commit-Position: refs/heads/master@{#11091}
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
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
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