modules: more interface -> include renames

webrtc/modules/audio_coding/main/test/RTPFile.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_TEST_RTPFILE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
 
 #include <stdio.h>
 #include <queue>
 
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class RTPStream {
  public:
   virtual ~RTPStream() {
   }
 
@@ skipping 88 matching lines @@
   bool EndOfFile() const override { return _rtpEOF; }
 
  private:
   FILE* _rtpFile;
   bool _rtpEOF;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_