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