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1 # Define rules for which include paths are allowed in our source. | |
2 include_rules = [ | |
3 # Base is only used to build Android APK tests and may not be referenced by | |
4 # WebRTC production code. | |
5 "-base", | |
6 "-chromium", | |
7 "+external/webrtc/webrtc", # Android platform build. | |
8 "+gflags", | |
9 "+libyuv", | |
10 "+testing", | |
11 "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. | |
12 # Individual headers that will be moved out of here, see webrtc: | |
13 "+webrtc/audio_receive_stream.h", | |
14 "+webrtc/audio_send_stream.h", | |
15 "+webrtc/audio_sink.h", | |
16 "+webrtc/audio_state.h", | |
17 "+webrtc/call.h", | |
18 "+webrtc/common.h", | |
19 "+webrtc/common_types.h", | |
20 "+webrtc/config.h", | |
21 "+webrtc/engine_configurations.h", | |
22 "+webrtc/frame_callback.h", | |
23 "+webrtc/stream.h", | |
24 "+webrtc/transport.h", | |
25 "+webrtc/typedefs.h", | |
26 "+webrtc/video_decoder.h", | |
27 "+webrtc/video_encoder.h", | |
28 "+webrtc/video_frame.h", | |
29 "+webrtc/video_receive_stream.h", | |
30 "+webrtc/video_renderer.h", | |
31 "+webrtc/video_send_stream.h", | |
32 | |
33 "+webrtc/base", | |
34 "+webrtc/modules/include", | |
35 "+webrtc/test", | |
36 "+webrtc/tools", | |
37 ] | |
38 | |
39 # The below rules will be removed when webrtc: is fixed. | |
40 specific_include_rules = { | |
41 "audio_send_stream\.h": [ | |
42 "+webrtc/modules/audio_coding", | |
43 ], | |
44 "video_frame\.h": [ | |
45 "+webrtc/common_video", | |
46 ], | |
47 } | |
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