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/call/rtx_receive_stream.h b/call/rtx_receive_stream.h
index c288a27..8ffa440 100644
--- a/call/rtx_receive_stream.h
+++ b/call/rtx_receive_stream.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#ifndef CALL_RTX_RECEIVE_STREAM_H_
+#define CALL_RTX_RECEIVE_STREAM_H_
 
 #include <map>
 
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/rtp_packet_sink_interface.h"
 
 namespace webrtc {
 
@@ -47,4 +47,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#endif  // CALL_RTX_RECEIVE_STREAM_H_