audio_coding: remove "main" directory

webrtc/modules/audio_coding/acm2/acm_resampler.h

 /*
  *  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.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace acm2 {
 
 class ACMResampler {
  public:
   ACMResampler();
   ~ACMResampler();
 
   int Resample10Msec(const int16_t* in_audio,
                      int in_freq_hz,
                      int out_freq_hz,
                      int num_audio_channels,
                      size_t out_capacity_samples,
                      int16_t* out_audio);
 
  private:
   PushResampler<int16_t> resampler_;
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_