The test code created an AudioBuffer object inside the work loop. This turned out to be expensive, since the AudioBuffer ctor implicitly called memset on all of the audio data array. The obvious remedy is to create the buffer outside of the loop. This does not have any impact apart from the performance boost, since the output data from NetEq is not even considered in the test. BUG=chromium:592907,webrtc:5647 TBR=ivoc@webrtc.org NOTRY=true Review URL: https://codereview.webrtc.org/1782803002 Cr-Commit-Position: refs/heads/master@{#11940}
Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ )
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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