Christoffer Jansson 2ae4ed223a Fix the last checksum
This should be the last checksum CL for audio tests.

Bug: webrtc:12882
Change-Id: Ie7033434e920a2f923c521cca00d1c270c406370
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224086
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/master@{#34391}
2021-06-30 07:32:00 +00:00
2021-06-30 07:32:00 +00:00
2021-06-29 10:19:11 +00:00
2021-06-21 22:21:04 +00:00
2021-01-20 15:01:07 +00:00
2021-04-26 16:39:07 +00:00
2020-07-13 11:42:07 +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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