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1 # Define rules for which include paths are allowed in our source. | 1 # Define rules for which include paths are allowed in our source. |
2 include_rules = [ | 2 include_rules = [ |
3 # Base is only used to build Android APK tests and may not be referenced by | 3 # Base is only used to build Android APK tests and may not be referenced by |
4 # WebRTC production code. | 4 # WebRTC production code. |
5 "-base", | 5 "-base", |
6 "-chromium", | 6 "-chromium", |
7 "+external/webrtc/webrtc", # Android platform build. | 7 "+external/webrtc/webrtc", # Android platform build. |
8 "+gflags", | 8 "+gflags", |
9 "+libyuv", | 9 "+libyuv", |
10 "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. | 10 "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. |
11 # Individual headers that will be moved out of here, see webrtc:4243. | 11 # Individual headers that will be moved out of here, see webrtc:4243. |
| 12 "+webrtc/call/rtp_config.h", |
12 "+webrtc/common_types.h", | 13 "+webrtc/common_types.h", |
13 "+webrtc/config.h", | |
14 "+webrtc/transport.h", | 14 "+webrtc/transport.h", |
15 "+webrtc/typedefs.h", | 15 "+webrtc/typedefs.h", |
16 "+webrtc/voice_engine_configurations.h", | 16 "+webrtc/voice_engine_configurations.h", |
17 | 17 |
18 "+WebRTC", | 18 "+WebRTC", |
19 "+webrtc/api", | 19 "+webrtc/api", |
20 "+webrtc/modules/include", | 20 "+webrtc/modules/include", |
21 "+webrtc/rtc_base", | 21 "+webrtc/rtc_base", |
22 "+webrtc/test", | 22 "+webrtc/test", |
23 "+webrtc/rtc_tools", | 23 "+webrtc/rtc_tools", |
24 ] | 24 ] |
25 | 25 |
26 # The below rules will be removed when webrtc:4243 is fixed. | 26 # The below rules will be removed when webrtc:4243 is fixed. |
27 specific_include_rules = { | 27 specific_include_rules = { |
28 "video_receive_stream\.h": [ | 28 "video_receive_stream\.h": [ |
29 "+webrtc/call/video_receive_stream.h", | 29 "+webrtc/call/video_receive_stream.h", |
30 ], | 30 ], |
31 "video_send_stream\.h": [ | 31 "video_send_stream\.h": [ |
32 "+webrtc/call/video_send_stream.h", | 32 "+webrtc/call/video_send_stream.h", |
33 ], | 33 ], |
34 } | 34 } |
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