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/rtc_base/asyncsocket.h b/rtc_base/asyncsocket.h
index 7284506..c018c23 100644
--- a/rtc_base/asyncsocket.h
+++ b/rtc_base/asyncsocket.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_RTC_BASE_ASYNCSOCKET_H_
-#define WEBRTC_RTC_BASE_ASYNCSOCKET_H_
+#ifndef RTC_BASE_ASYNCSOCKET_H_
+#define RTC_BASE_ASYNCSOCKET_H_
-#include "webrtc/rtc_base/sigslot.h"
-#include "webrtc/rtc_base/socket.h"
+#include "rtc_base/sigslot.h"
+#include "rtc_base/socket.h"
namespace rtc {
@@ -80,4 +80,4 @@
} // namespace rtc
-#endif // WEBRTC_RTC_BASE_ASYNCSOCKET_H_
+#endif // RTC_BASE_ASYNCSOCKET_H_