audio_coding: remove "main" directory

webrtc/modules/audio_coding/test/EncodeDecodeTest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 #include <sstream>
 #include <stdio.h>
 #include <stdlib.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
     : _rtpStream(rtpStream),
       _frequency(frequency),
       _seqNo(0) {
 }
@@ skipping 309 matching lines @@
   if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
 
   return fileName;
 }
 
 }  // namespace webrtc