| include_rules = [ | |
| "+webrtc/base", | |
| "+webrtc/voice_engine", | |
| "+webrtc/modules/audio_coding/codecs/mock", | |
| "+webrtc/call", | |
| "+webrtc/modules/bitrate_controller", | |
| "+webrtc/modules/congestion_controller", | |
| "+webrtc/modules/pacing", | |
| "+webrtc/modules/remote_bitrate_estimator", | |
| "+webrtc/modules/rtp_rtcp", | |
| "+webrtc/system_wrappers", | |
| "+webrtc/voice_engine", | |
| ] | |
| specific_include_rules = { | |
| "audio_receive_stream_unittest\.cc": [ | |
| "+webrtc/call/mock", | |
| ], | |
| "audio_send_stream_unittest\.cc": [ | |
| "+webrtc/call/mock", | |
| ], | |
| } |