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1 include_rules = [ | 1 include_rules = [ |
2 "+third_party/libjpeg", | 2 "+third_party/libjpeg", |
3 "+third_party/libjpeg_turbo", | 3 "+third_party/libjpeg_turbo", |
4 "+webrtc/base", | |
5 "+webrtc/call", | 4 "+webrtc/call", |
6 "+webrtc/common_audio", | 5 "+webrtc/common_audio", |
7 "+webrtc/common_video", | 6 "+webrtc/common_video", |
8 "+webrtc/logging/rtc_event_log", | 7 "+webrtc/logging/rtc_event_log", |
9 "+webrtc/media/base", | 8 "+webrtc/media/base", |
10 "+webrtc/modules/audio_coding", | 9 "+webrtc/modules/audio_coding", |
11 "+webrtc/modules/audio_device", | 10 "+webrtc/modules/audio_device", |
12 "+webrtc/modules/audio_mixer", | 11 "+webrtc/modules/audio_mixer", |
13 "+webrtc/modules/audio_processing", | 12 "+webrtc/modules/audio_processing", |
14 "+webrtc/modules/media_file", | 13 "+webrtc/modules/media_file", |
15 "+webrtc/modules/rtp_rtcp", | 14 "+webrtc/modules/rtp_rtcp", |
16 "+webrtc/modules/video_capture", | 15 "+webrtc/modules/video_capture", |
17 "+webrtc/modules/video_coding", | 16 "+webrtc/modules/video_coding", |
18 "+webrtc/sdk", | 17 "+webrtc/sdk", |
19 "+webrtc/system_wrappers", | 18 "+webrtc/system_wrappers", |
20 "+webrtc/voice_engine", | 19 "+webrtc/voice_engine", |
21 ] | 20 ] |
22 | 21 |
23 specific_include_rules = { | 22 specific_include_rules = { |
24 "gmock\.h": [ | 23 "gmock\.h": [ |
25 "+testing/gmock/include/gmock", | 24 "+testing/gmock/include/gmock", |
26 ], | 25 ], |
27 "gtest\.h": [ | 26 "gtest\.h": [ |
28 "+testing/gtest/include/gtest", | 27 "+testing/gtest/include/gtest", |
29 ], | 28 ], |
30 } | 29 } |
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