kjellander 676e08f3b6 Refactor webrtc/{api,audio} and modules/audio_coding for GN check
This moves some GN check configurations out of .gn to individual targets.
The now checked targets are:
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/modules/audio_coding/*",

Many targets were fixed by adding dependencies, but the ones that
requires more refactorings are left with the check_includes attribute
set to false instead.

Make //webrtc/test:test_support a public dep of //webrtc/test:test_main
to avoid having to add that to all users of it.

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2556943003
Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 16:23:35 +00:00
2016-12-07 11:00:25 +00:00
2016-12-02 11:30:51 +00:00
2016-11-29 10:23:05 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +00:00
2016-12-02 11:30:51 +00:00
2016-11-23 16:42:57 +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
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