audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/isac/main/source/structs.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.
  */
 
 /*
  * structs.h
  *
  * This header file contains all the structs used in the ISAC codec
  *
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 
 #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h"
 #include "webrtc/typedefs.h"
 
 typedef struct Bitstreamstruct {
 
   uint8_t   stream[STREAM_SIZE_MAX];
   uint32_t  W_upper;
   uint32_t  streamval;
   uint32_t  stream_index;
 
@@ skipping 453 matching lines @@
   int16_t               maxPayloadSizeBytes;
   /* The expected sampling rate of the input signal. Valid values are 16000
    * and 32000. This is not the operation sampling rate of the codec. */
   uint16_t in_sample_rate_hz;
 
   // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
   TransformTables transform_tables;
 } ISACMainStruct;
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */