Per Åhgren 12eb85881c Separating the AEC3 suppressor gain rampup behavior for call startup and in-call resets
This CL introduces a different rampup behavir for the call startup and after resets
that may occur due to delay changes, clock-drift and audio path glitches.

Bug: chromium:819111, webrtc:8979
Change-Id: Ied1d7896be7f0c69aa6deb61475117021ca6ab09
Reviewed-on: https://webrtc-review.googlesource.com/60002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22312}
2018-03-06 15:48:41 +00:00
2018-03-02 12:16:29 +00:00
.gn
2018-02-19 15:07:45 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-29 11:18:18 +00:00
2017-09-15 04:25:06 +00:00
2018-02-19 14:35:45 +00:00
2018-02-23 10:34:16 +00:00

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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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