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/pc/rtpsender.h b/pc/rtpsender.h
index de679cf..672637f 100644
--- a/pc/rtpsender.h
+++ b/pc/rtpsender.h
@@ -12,22 +12,22 @@
// An RtpSender associates a MediaStreamTrackInterface with an underlying
// transport (provided by AudioProviderInterface/VideoProviderInterface)
-#ifndef WEBRTC_PC_RTPSENDER_H_
-#define WEBRTC_PC_RTPSENDER_H_
+#ifndef PC_RTPSENDER_H_
+#define PC_RTPSENDER_H_
#include <memory>
#include <string>
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/criticalsection.h"
+#include "api/mediastreaminterface.h"
+#include "api/rtpsenderinterface.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/criticalsection.h"
// Adding 'nogncheck' to disable the gn include headers check to support modular
// WebRTC build targets.
-#include "webrtc/media/base/audiosource.h" // nogncheck
-#include "webrtc/pc/channel.h"
-#include "webrtc/pc/dtmfsender.h"
-#include "webrtc/pc/statscollector.h"
+#include "media/base/audiosource.h" // nogncheck
+#include "pc/channel.h"
+#include "pc/dtmfsender.h"
+#include "pc/statscollector.h"
namespace webrtc {
@@ -258,4 +258,4 @@
} // namespace webrtc
-#endif // WEBRTC_PC_RTPSENDER_H_
+#endif // PC_RTPSENDER_H_