| include_rules = [ |
| "+third_party/libjpeg", |
| "+third_party/libjpeg_turbo", |
| "+webrtc/call", |
| "+webrtc/common_audio", |
| "+webrtc/common_video", |
| "+webrtc/logging/rtc_event_log", |
| "+webrtc/media/base", |
| "+webrtc/modules/audio_coding", |
| "+webrtc/modules/audio_device", |
| "+webrtc/modules/audio_mixer", |
| "+webrtc/modules/audio_processing", |
| "+webrtc/modules/media_file", |
| "+webrtc/modules/rtp_rtcp", |
| "+webrtc/modules/video_capture", |
| "+webrtc/modules/video_coding", |
| "+webrtc/sdk", |
| "+webrtc/system_wrappers", |
| "+webrtc/voice_engine", |
| ] |
| |
| specific_include_rules = { |
| "gmock\.h": [ |
| "+testing/gmock/include/gmock", |
| ], |
| "gtest\.h": [ |
| "+testing/gtest/include/gtest", |
| ], |
| ".*congestion_controller_feedback_fuzzer\.cc": [ |
| "+webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h", |
| "+webrtc/modules/pacing/packet_router.h", |
| "+webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h", |
| ], |
| ".*pseudotcp_parser_fuzzer\.cc": [ |
| "+webrtc/p2p/base/pseudotcp.h", |
| ], |
| ".*stun_parser_fuzzer\.cc": [ |
| "+webrtc/p2p/base/stun.h", |
| ], |
| ".*stun_validator_fuzzer\.cc": [ |
| "+webrtc/p2p/base/stun.h", |
| ], |
| } |