| OLD | NEW |
| 1 include_rules = [ | 1 include_rules = [ |
| 2 "+webrtc/base", | 2 "+webrtc/base", |
| 3 "+webrtc/voice_engine", | 3 "+webrtc/voice_engine", |
| 4 "+webrtc/modules/audio_coding/codecs/mock", | 4 "+webrtc/modules/audio_coding/codecs/mock", |
| 5 "+webrtc/call", | 5 "+webrtc/call", |
| 6 "+webrtc/logging/rtc_event_log", |
| 6 "+webrtc/modules/bitrate_controller", | 7 "+webrtc/modules/bitrate_controller", |
| 7 "+webrtc/modules/congestion_controller", | 8 "+webrtc/modules/congestion_controller", |
| 8 "+webrtc/modules/pacing", | 9 "+webrtc/modules/pacing", |
| 9 "+webrtc/modules/remote_bitrate_estimator", | 10 "+webrtc/modules/remote_bitrate_estimator", |
| 10 "+webrtc/modules/rtp_rtcp", | 11 "+webrtc/modules/rtp_rtcp", |
| 11 "+webrtc/system_wrappers", | 12 "+webrtc/system_wrappers", |
| 12 "+webrtc/voice_engine", | 13 "+webrtc/voice_engine", |
| 13 ] | 14 ] |
| 14 | |
| 15 specific_include_rules = { | |
| 16 "audio_receive_stream_unittest\.cc": [ | |
| 17 "+webrtc/call/mock", | |
| 18 ], | |
| 19 "audio_send_stream_unittest\.cc": [ | |
| 20 "+webrtc/call/mock", | |
| 21 ], | |
| 22 } | |
| OLD | NEW |