audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/isac/main/include/isac.h

 /*
  *  Copyright (c) 2011 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_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
 
 #include <stddef.h>
 
 #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/typedefs.h"
 
 typedef struct WebRtcISACStruct    ISACStruct;
 
 #if defined(__cplusplus)
 extern "C" {
@@ skipping 691 matching lines @@
   /* If |inst| is a decoder but not an encoder: tell it what sample rate the
      encoder is using, for bandwidth estimation purposes. */
   void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */