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_coding/test/EncodeDecodeTest.h

diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index f9a9a5b..941a92f 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
 
 #include <stdio.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/RTPFile.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/RTPFile.h"
+#include "typedefs.h"
 
 namespace webrtc {
 
@@ -120,4 +120,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#endif  // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_