ivoc d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 11:06:41 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

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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