Delete full-band mode from the iSAC codec

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.
  */
 
@@ skipping 466 matching lines @@
   // encoder and decoder.
   int16_t               initFlag;
 
   // Flag to to indicate signal bandwidth switch
   int16_t               resetFlag_8kHz;
 
   // Maximum allowed rate, measured in Bytes per 30 ms.
   int16_t               maxRateBytesPer30Ms;
   // Maximum allowed payload-size, measured in Bytes.
   int16_t               maxPayloadSizeBytes;
-  /* The expected sampling rate of the input signal. Valid values are 16000,
-   * 32000 and 48000. This is not the operation sampling rate of the codec.
-   * Input signals at 48 kHz are resampled to 32 kHz, then encoded. */
+  /* 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;
-  /* State for the input-resampler. It is only used for 48 kHz input signals. */
-  int16_t state_in_resampler[SIZE_RESAMPLER_STATE];
 
   // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
   TransformTables transform_tables;
 } ISACMainStruct;
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */