audio_coding: remove "main" directory

webrtc/modules/audio_coding/acm2/call_statistics.cc

 /*
  *  Copyright (c) 2013 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/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 #include <assert.h>
 
 namespace webrtc {
 
 namespace acm2 {
 
 void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
   ++decoding_stat_.calls_to_neteq;
   switch (speech_type) {
@@ skipping 24 matching lines @@
   ++decoding_stat_.calls_to_silence_generator;
 }
 
 const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
   return decoding_stat_;
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc