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/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index 4bf4c98..d896e76 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -20,10 +20,10 @@
#endif
#include "audio_coding_module.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "system_wrappers/include/rw_lock_wrapper.h"
// TODO(tlegrand): Consider removing usage of gtest.
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {