commit 92ea95e34af5966555903026f45164afbd7e2088
author Mirko Bonadei <mbonadei@webrtc.org> Fri Sep 15 06:47:31 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Fri Sep 15 05:02:56 2017 +0000
tree 3befb3fcfefff3b35da8a834052543338e15bcc0
parent bb547203bfebcc478b263c4e9ca173c6fd5a0c5d

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}

modules/audio_processing/audio_processing_impl.h

diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 56c1afc..4ad9f7f 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -8,26 +8,26 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <memory>
 #include <vector>
 
-#include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/include/aec_dump.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
-#include "webrtc/modules/audio_processing/rms_level.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/function_view.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/swap_queue.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/render_queue_item_verifier.h"
+#include "modules/audio_processing/rms_level.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/function_view.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
+#include "rtc_base/swap_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {
 
@@ -415,4 +415,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#endif  // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_