Move AudioDecoderIsac* to its own files

webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h

 /*
  *  Copyright (c) 2014 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_AUDIO_ENCODER_ISAC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
 
 namespace webrtc {
 
-struct IsacFloat {
-  typedef ISACStruct instance_type;
-  static const bool has_swb = true;
-  static inline int16_t Control(instance_type* inst,
-                                int32_t rate,
-                                int framesize) {
-    return WebRtcIsac_Control(inst, rate, framesize);
-  }
-  static inline int16_t ControlBwe(instance_type* inst,
-                                   int32_t rate_bps,
-                                   int frame_size_ms,
-                                   int16_t enforce_frame_size) {
-    return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
-                                 enforce_frame_size);
-  }
-  static inline int16_t Create(instance_type** inst) {
-    return WebRtcIsac_Create(inst);
-  }
-  static inline int DecodeInternal(instance_type* inst,
-                                   const uint8_t* encoded,
-                                   size_t len,
-                                   int16_t* decoded,
-                                   int16_t* speech_type) {
-    return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
-  }
-  static inline size_t DecodePlc(instance_type* inst,
-                                 int16_t* decoded,
-                                 size_t num_lost_frames) {
-    return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
-  }
-
-  static inline void DecoderInit(instance_type* inst) {
-    WebRtcIsac_DecoderInit(inst);
-  }
-  static inline int Encode(instance_type* inst,
-                           const int16_t* speech_in,
-                           uint8_t* encoded) {
-    return WebRtcIsac_Encode(inst, speech_in, encoded);
-  }
-  static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
-    return WebRtcIsac_EncoderInit(inst, coding_mode);
-  }
-  static inline uint16_t EncSampRate(instance_type* inst) {
-    return WebRtcIsac_EncSampRate(inst);
-  }
-
-  static inline int16_t Free(instance_type* inst) {
-    return WebRtcIsac_Free(inst);
-  }
-  static inline void GetBandwidthInfo(instance_type* inst,
-                                      IsacBandwidthInfo* bwinfo) {
-    WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
-  }
-  static inline int16_t GetErrorCode(instance_type* inst) {
-    return WebRtcIsac_GetErrorCode(inst);
-  }
-
-  static inline int16_t GetNewFrameLen(instance_type* inst) {
-    return WebRtcIsac_GetNewFrameLen(inst);
-  }
-  static inline void SetBandwidthInfo(instance_type* inst,
-                                      const IsacBandwidthInfo* bwinfo) {
-    WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
-  }
-  static inline int16_t SetDecSampRate(instance_type* inst,
-                                       uint16_t sample_rate_hz) {
-    return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
-  }
-  static inline int16_t SetEncSampRate(instance_type* inst,
-                                       uint16_t sample_rate_hz) {
-    return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
-  }
-  static inline void SetEncSampRateInDecoder(instance_type* inst,
-                                                uint16_t sample_rate_hz) {
-    WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
-  }
-  static inline void SetInitialBweBottleneck(
-      instance_type* inst,
-      int bottleneck_bits_per_second) {
-    WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
-  }
-  static inline int16_t UpdateBwEstimate(instance_type* inst,
-                                         const uint8_t* encoded,
-                                         size_t packet_size,
-                                         uint16_t rtp_seq_number,
-                                         uint32_t send_ts,
-                                         uint32_t arr_ts) {
-    return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
-                                       rtp_seq_number, send_ts, arr_ts);
-  }
-  static inline int16_t SetMaxPayloadSize(instance_type* inst,
-                                          int16_t max_payload_size_bytes) {
-    return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
-  }
-  static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
-    return WebRtcIsac_SetMaxRate(inst, max_bit_rate);
-  }
-};
-
 using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
-using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_