This CL corrects the behavior in AEC3 during buffer overruns and underruns in three ways. 1) When there is no render signal available (due to a buffering issues, a zero block is inserted instead of the previous render block. This avoids the same block being repeatedly inserted when there are many back-to-back calls. 2) The internal counters in the main adaptive filter gain are also reset when the filter is reset. 3) The internal counters in the shadow adaptive filter gain are reset when the filter is reset. BUG=chromium:717920,webrtc:7559 Review-Url: https://codereview.webrtc.org/2862533002 Cr-Commit-Position: refs/heads/master@{#17991}
Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ )
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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