Move AudioDecoderIsac* to its own files

webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h

Index: webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
similarity index 84%
copy from webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
copy to webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
index da363d8c91a8e61a628520f8e81d0ace3a4a190b..0335548be7f948db53166003a3a3e3341620f076 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 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
@@ -8,18 +8,15 @@
  *  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_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_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"
 
 namespace webrtc {
 
 struct IsacFloat {
-  typedef ISACStruct instance_type;
+  using instance_type = ISACStruct;
   static const bool has_swb = true;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
@@ -91,12 +88,11 @@ struct IsacFloat {
     return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
   }
   static inline void SetEncSampRateInDecoder(instance_type* inst,
-                                                uint16_t sample_rate_hz) {
+                                             uint16_t sample_rate_hz) {
     WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
   }
-  static inline void SetInitialBweBottleneck(
-      instance_type* inst,
-      int bottleneck_bits_per_second) {
+  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,
@@ -117,8 +113,5 @@ struct IsacFloat {
   }
 };
 
-using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
-using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
-
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_