| # Define rules for which include paths are allowed in our source. |
| include_rules = [ |
| # Base is only used to build Android APK tests and may not be referenced by |
| # WebRTC production code. |
| "-base", |
| "-chromium", |
| "+external/webrtc/webrtc", # Android platform build. |
| "+gflags", |
| "+libyuv", |
| "+testing", |
| "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. |
| # Individual headers that will be moved out of here, see webrtc: |
| "+webrtc/audio_receive_stream.h", |
| "+webrtc/audio_send_stream.h", |
| "+webrtc/audio_sink.h", |
| "+webrtc/audio_state.h", |
| "+webrtc/call.h", |
| "+webrtc/common.h", |
| "+webrtc/common_types.h", |
| "+webrtc/config.h", |
| "+webrtc/engine_configurations.h", |
| "+webrtc/transport.h", |
| "+webrtc/typedefs.h", |
| "+webrtc/video_decoder.h", |
| "+webrtc/video_encoder.h", |
| "+webrtc/video_frame.h", |
| "+webrtc/video_receive_stream.h", |
| "+webrtc/video_renderer.h", |
| "+webrtc/video_send_stream.h", |
| |
| "+WebRTC", |
| "+webrtc/base", |
| "+webrtc/modules/include", |
| "+webrtc/test", |
| "+webrtc/tools", |
| ] |
| |
| # The below rules will be removed when webrtc: is fixed. |
| specific_include_rules = { |
| "audio_send_stream\.h": [ |
| "+webrtc/modules/audio_coding", |
| ], |
| "audio_receive_stream\.h": [ |
| "+webrtc/modules/audio_coding/codecs/audio_decoder_factory.h", |
| ], |
| "video_frame\.h": [ |
| "+webrtc/common_video", |
| ], |
| "video_receive_stream\.h": [ |
| "+webrtc/common_video/include", |
| "+webrtc/media/base", |
| ], |
| "video_send_stream\.h": [ |
| "+webrtc/common_video/include", |
| "+webrtc/media/base", |
| ], |
| } |