Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

36 lines
839 B
Python

include_rules = [
"+third_party/libsrtp",
"+api",
"+call",
"+common_video/h264",
"+logging/rtc_event_log",
"+logging/rtc_event_log",
"+media",
"+modules/audio_device",
"+modules/audio_processing",
"+modules/rtp_rtcp",
"+modules/video_coding",
"+modules/video_render",
"+p2p",
"+system_wrappers",
]
specific_include_rules = {
"androidtestinitializer\.cc": [
"+base/android", # Allowed only for Android tests.
"+voice_engine",
],
"srtpfilter_unittest\.cc": [
"+crypto",
],
# TODO(ossu): Remove these exceptions when audio_encoder_factory.h
# has moved to api/.
"peerconnectionfactory\.cc": [
"+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
],
"peerconnectioninterface_unittest\.cc": [
"+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
],
}