Henrik Kjellander 8108764552 Analyze support in gyp_webrtc
BUG=chromium:482463
TESTED=Manually tested using the JSON files attached to https://code.google.com/p/chromium/issues/detail?id=482463#c2 and:
webrtc/build/gyp_webrtc --analyzer nothing-files.json nothing-files-RESULT.json
webrtc/build/gyp_webrtc --analyzer everything-files.json everything-files-RESULT.json
webrtc/build/gyp_webrtc --analyzer test_support_unittests-files.json test_support_unittests-files-RESULT.json
Then I verified the result-json contained the expected output.

R=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1369683004 .

Cr-Commit-Position: refs/heads/master@{#10097}
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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.