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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/modules/audio_device", |
| 7 "+webrtc/modules/audio_mixer", |
| 8 "+webrtc/modules/audio_processing", |
6 "+webrtc/modules/bitrate_controller", | 9 "+webrtc/modules/bitrate_controller", |
7 "+webrtc/modules/congestion_controller", | 10 "+webrtc/modules/congestion_controller", |
8 "+webrtc/modules/pacing", | 11 "+webrtc/modules/pacing", |
9 "+webrtc/modules/remote_bitrate_estimator", | 12 "+webrtc/modules/remote_bitrate_estimator", |
10 "+webrtc/modules/rtp_rtcp", | 13 "+webrtc/modules/rtp_rtcp", |
11 "+webrtc/system_wrappers", | 14 "+webrtc/system_wrappers", |
12 "+webrtc/voice_engine", | 15 "+webrtc/voice_engine", |
13 ] | 16 ] |
14 | 17 |
15 specific_include_rules = { | 18 specific_include_rules = { |
16 "audio_receive_stream_unittest\.cc": [ | 19 "audio_receive_stream_unittest\.cc": [ |
17 "+webrtc/call/mock", | 20 "+webrtc/call/mock", |
18 ], | 21 ], |
19 "audio_send_stream_unittest\.cc": [ | 22 "audio_send_stream_unittest\.cc": [ |
20 "+webrtc/call/mock", | 23 "+webrtc/call/mock", |
21 ], | 24 ], |
22 } | 25 } |
OLD | NEW |