kjellander@webrtc.org e281f7fba3 GN: Update webrtc/base to recent GYP changes.
Update the webrtc/base/BUILD.gn file to reflect
webrtc/base/base.gyp changes between r6438 and r7011.

BUG=3441
TESTED= Trybots + compilation with a standalone WebRTC checkout:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default

Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 11:22:06 +00:00
..
2014-07-07 04:45:15 +00:00

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.