Alessio Bazzica 10f6eadd48 AGC2 Saturation Protector: switch to ring buffer
Even if small, the peak delay buffer copies N-1 elements for each frame
whereas a ring buffer is copy-free and scales better if the buffer size
increases.

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32165}
2020-09-22 19:34:04 +00:00
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2020-09-06 10:13:23 +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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