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_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 03bd8c8..d530376 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
@@ -20,12 +20,12 @@
 #include <string.h>
 #include <vector>
 
-#include "webrtc/modules/audio_processing/beamformer/array_util.h"
-#include "webrtc/modules/audio_processing/include/config.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/platform_file.h"
-#include "webrtc/rtc_base/refcount.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_processing/beamformer/array_util.h"
+#include "modules/audio_processing/include/config.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/platform_file.h"
+#include "rtc_base/refcount.h"
+#include "typedefs.h"
 
 namespace webrtc {
 
@@ -1138,4 +1138,4 @@
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_