commit 92ea95e34af5966555903026f45164afbd7e2088
author Mirko Bonadei <mbonadei@webrtc.org> Fri Sep 15 04:47:31 2017
committer Commit Bot <commit-bot@chromium.org> Fri Sep 15 05:02:56 2017
tree 3befb3fcfefff3b35da8a834052543338e15bcc0
parent bb547203bfebcc478b263c4e9ca173c6fd5a0c5d

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}

call/DEPS

diff --git a/call/DEPS b/call/DEPS
index 54dcebb..7622e24 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -1,27 +1,27 @@
 include_rules = [
-  "+webrtc/audio",
-  "+webrtc/logging/rtc_event_log",
-  "+webrtc/modules/audio_coding",
-  "+webrtc/modules/audio_device",
-  "+webrtc/modules/audio_mixer",
-  "+webrtc/modules/audio_processing",
-  "+webrtc/modules/bitrate_controller",
-  "+webrtc/modules/congestion_controller",
-  "+webrtc/modules/pacing",
-  "+webrtc/modules/rtp_rtcp",
-  "+webrtc/modules/utility",
-  "+webrtc/system_wrappers",
-  "+webrtc/voice_engine",
-  "+webrtc/video",
+  "+audio",
+  "+logging/rtc_event_log",
+  "+modules/audio_coding",
+  "+modules/audio_device",
+  "+modules/audio_mixer",
+  "+modules/audio_processing",
+  "+modules/bitrate_controller",
+  "+modules/congestion_controller",
+  "+modules/pacing",
+  "+modules/rtp_rtcp",
+  "+modules/utility",
+  "+system_wrappers",
+  "+voice_engine",
+  "+video",
 ]
 
 specific_include_rules = {
   "video_receive_stream\.h": [
-    "+webrtc/common_video/include",
-    "+webrtc/media/base",
+    "+common_video/include",
+    "+media/base",
   ],
   "video_send_stream\.h": [
-    "+webrtc/common_video/include",
-    "+webrtc/media/base",
+    "+common_video/include",
+    "+media/base",
   ],
 }