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/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index de070f3..b6e56c5 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -15,37 +15,37 @@
 #include <memory>
 #include <queue>
 
-#include "webrtc/common_audio/include/audio_util.h"
-#include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
-#include "webrtc/modules/audio_processing/audio_processing_impl.h"
-#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
-#include "webrtc/modules/audio_processing/common.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
-#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
-#include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/gtest_prod_util.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
-#include "webrtc/rtc_base/safe_minmax.h"
-#include "webrtc/rtc_base/task_queue.h"
-#include "webrtc/rtc_base/thread.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
+#include "modules/audio_processing/common.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/level_controller/level_controller_constants.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
+#include "rtc_base/safe_minmax.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/trace.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
 
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
 #else
-#include "webrtc/modules/audio_processing/test/unittest.pb.h"
+#include "modules/audio_processing/test/unittest.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()