Remove <(webrtc_root) from source file entries.
This required to move the AGC tools source files into webrtc/tools and create a new agc_test_utils target. Since audio_codec_speed_tests.gypi referenced sources above, the best approach I could come up with was to add an audio_coding.gypi file at a higher level and move the targets in there (+ the includes from modules.gyp which is an improvement IMO). I also added a PRESUBMIT.py check to prevent new source entries being added with <(webrtc_root) in the path. BUG=4185 R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37859004 Cr-Commit-Position: refs/heads/master@{#8197} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -61,6 +61,7 @@
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'dependencies': [
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(webrtc_root)/test/test.gyp:rtp_test_utils',
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'bwe_tools_util',
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'rtp_rtcp',
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],
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@ -71,8 +72,6 @@
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},
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'sources': [
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'tools/rtp_to_text.cc',
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'<(webrtc_root)/test/rtp_file_reader.cc',
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'<(webrtc_root)/test/rtp_file_reader.h',
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], # source
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},
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{
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@ -84,6 +83,7 @@
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'dependencies': [
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(webrtc_root)/test/test.gyp:rtp_test_utils',
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'bwe_tools_util',
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'rtp_rtcp',
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],
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@ -94,8 +94,6 @@
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},
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'sources': [
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'tools/bwe_rtp_play.cc',
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'<(webrtc_root)/test/rtp_file_reader.cc',
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'<(webrtc_root)/test/rtp_file_reader.h',
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], # source
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},
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], # targets
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