Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(18)

Side by Side Diff: webrtc/DEPS

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Rebase Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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/common_types.h", 12 "+webrtc/common_types.h",
13 "+webrtc/config.h", 13 "+webrtc/call/rtp_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 }
OLDNEW
« no previous file with comments | « webrtc/BUILD.gn ('k') | webrtc/api/BUILD.gn » ('j') | webrtc/modules/audio_coding/BUILD.gn » ('J')

Powered by Google App Engine
This is Rietveld 408576698