audio_coding: remove "main" directory

webrtc/modules/audio_coding/test/PacketLossTest.h

 /*
  *  Copyright (c) 2014 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.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
 #include <string>
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
 
 class ReceiverWithPacketLoss : public Receiver {
  public:
   ReceiverWithPacketLoss();
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
              std::string out_file_name, int channels, int loss_rate,
              int burst_length);
   bool IncomingPacket() override;
@@ skipping 30 matching lines @@
   int sample_rate_hz_;
   rtc::scoped_ptr<SenderWithFEC> sender_;
   rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
   int expected_loss_rate_;
   int actual_loss_rate_;
   int burst_length_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_