| include_rules = [ |
| "+third_party/libsrtp" |
| "+webrtc/api", |
| "+webrtc/base", |
| "+webrtc/call", |
| "+webrtc/common_video/h264", |
| "+webrtc/logging/rtc_event_log", |
| "+webrtc/logging/rtc_event_log", |
| "+webrtc/media", |
| "+webrtc/modules/audio_device", |
| "+webrtc/modules/rtp_rtcp", |
| "+webrtc/modules/video_coding", |
| "+webrtc/modules/video_render", |
| "+webrtc/p2p", |
| "+webrtc/system_wrappers", |
| ] |
| |
| specific_include_rules = { |
| "androidtestinitializer\.cc": [ |
| "+base/android", # Allowed only for Android tests. |
| "+webrtc/voice_engine", |
| ], |
| "srtpfilter_unittest\.cc": [ |
| "+crypto", |
| ], |
| "peerconnectionfactory\.cc": [ |
| "+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h", |
| ], |
| } |
