Alessio Bazzica eeb223557f Retune AGC2 input volume controller speech ratio threshold
Based on offline testing; needed to allow input volume adaptations
more frequently. Note that if the estimated speech level falls in
the target range, the recommended input volume won't change and
hence the new lower threshold won't necessarily increase the
number of adjustments.

Bug: webrtc:7494
Change-Id: Iabb501c188da238ea7b7137175bcfe09239c90a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291110
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39161}
2023-01-20 14:03:58 +00:00
2022-12-16 10:19:13 +00:00
2023-01-18 11:14:50 +00:00
2021-01-20 15:01:07 +00:00
.gn
2022-09-14 08:49:56 +00:00
2022-02-20 14:22:13 +00:00
2021-12-08 08:53:00 +00:00
2022-12-02 09:21:47 +00:00
2022-12-02 09:21:47 +00:00
2022-05-13 09:01:34 +00:00
2020-07-13 11:42:07 +00:00
2022-11-17 21:29:53 +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 here 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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