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",
 ]