Danil Chapovalov 28da36a6ea Add unittest for av1 wrappers to test Encode and Decode functions
while helpful by itself, it is also a preparation
for adding unittests for (to be added) svc features of the encoder.

Bug: webrtc:11404
Change-Id: I62b0645f44579f21f228d406a206b4c01d80dd02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174580
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31189}
2020-05-08 11:57:27 +00:00
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2018-10-05 14:40:21 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
.gn
2020-03-18 18:04:41 +00:00
2020-05-07 17:04:15 +00:00
2020-03-30 12:15:56 +00:00
2018-07-23 15:28:48 +00:00
2020-04-16 11:08:43 +00:00
2020-04-21 13:15:09 +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.

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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