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}
diff --git a/ortc/DEPS b/ortc/DEPS
index b72aa6b..bebf030 100644
--- a/ortc/DEPS
+++ b/ortc/DEPS
@@ -1,17 +1,17 @@
include_rules = [
- "+webrtc/api",
- "+webrtc/call",
- "+webrtc/logging/rtc_event_log",
- "+webrtc/media",
- "+webrtc/modules/audio_coding",
- "+webrtc/modules/audio_processing",
- "+webrtc/p2p",
- "+webrtc/pc",
+ "+api",
+ "+call",
+ "+logging/rtc_event_log",
+ "+media",
+ "+modules/audio_coding",
+ "+modules/audio_processing",
+ "+p2p",
+ "+pc",
- "+webrtc/modules/rtp_rtcp",
- "+webrtc/system_wrappers",
+ "+modules/rtp_rtcp",
+ "+system_wrappers",
- "+webrtc/modules/audio_device",
- "+webrtc/modules/video_coding",
- "+webrtc/modules/video_render",
+ "+modules/audio_device",
+ "+modules/video_coding",
+ "+modules/video_render",
]