0cf99ce8656b78863b0f180baeba18cb3cd6fe82
This allows the user to run audioproc_f with various field trials set. The approach is copied from test/test_main_lib.cc. Tested: 1. Verified bitexactness vs ToT audioproc_f on a large dataset of aecdumps 2. Ran it with flags --aec=1 --force_fieldtrials="WebRTC-Aec3ClampInstQualityToZeroKillSwitch/Enabled/WebRTC-Aec3ClampInstQualityToOneKillSwitch/Enabled/" and verified in GDB that the AEC3 config was changed accordingly. No-Try: True Bug: webrtc:5298 Change-Id: I70eec7777f70893b36af33794a5842f67d56af31 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172623 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30976}
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.
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
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