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_