audio_coding: remove "main" directory

webrtc/modules/audio_coding/test/RTPFile.h

Index: webrtc/modules/audio_coding/test/RTPFile.h
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/test/RTPFile.h
similarity index 92%
rename from webrtc/modules/audio_coding/main/test/RTPFile.h
rename to webrtc/modules/audio_coding/test/RTPFile.h
index 6bad755af9d390ca8330ea117898683cc14f6859..696d41ebd230e72e70ff0e5ecfcc2ee55ca5850d 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/test/RTPFile.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
 
 #include <stdio.h>
 #include <queue>
 
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -123,4 +123,4 @@ class RTPFile : public RTPStream {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_