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