diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 5d85db3e1a..5725ec03df 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -412,12 +412,14 @@ config("isac_config") {
 
 source_set("isac") {
   sources = [
+    "codecs/isac/main/interface/audio_decoder_isac.h",
     "codecs/isac/main/interface/audio_encoder_isac.h",
     "codecs/isac/main/interface/isac.h",
     "codecs/isac/main/source/arith_routines.c",
     "codecs/isac/main/source/arith_routines.h",
     "codecs/isac/main/source/arith_routines_hist.c",
     "codecs/isac/main/source/arith_routines_logist.c",
+    "codecs/isac/main/source/audio_decoder_isac.cc",
     "codecs/isac/main/source/audio_encoder_isac.cc",
     "codecs/isac/main/source/bandwidth_estimator.c",
     "codecs/isac/main/source/bandwidth_estimator.h",
@@ -439,6 +441,7 @@ source_set("isac") {
     "codecs/isac/main/source/filterbanks.c",
     "codecs/isac/main/source/intialize.c",
     "codecs/isac/main/source/isac.c",
+    "codecs/isac/main/source/isac_float_type.h",
     "codecs/isac/main/source/lattice.c",
     "codecs/isac/main/source/lpc_analysis.c",
     "codecs/isac/main/source/lpc_analysis.h",
@@ -493,12 +496,14 @@ config("isac_fix_config") {
 
 source_set("isac_fix") {
   sources = [
+    "codecs/isac/fix/interface/audio_decoder_isacfix.h",
     "codecs/isac/fix/interface/audio_encoder_isacfix.h",
     "codecs/isac/fix/interface/isacfix.h",
     "codecs/isac/fix/source/arith_routines.c",
     "codecs/isac/fix/source/arith_routines_hist.c",
     "codecs/isac/fix/source/arith_routines_logist.c",
     "codecs/isac/fix/source/arith_routins.h",
+    "codecs/isac/fix/source/audio_decoder_isacfix.cc",
     "codecs/isac/fix/source/audio_encoder_isacfix.cc",
     "codecs/isac/fix/source/bandwidth_estimator.c",
     "codecs/isac/fix/source/bandwidth_estimator.h",
@@ -516,6 +521,7 @@ source_set("isac_fix") {
     "codecs/isac/fix/source/filterbanks.c",
     "codecs/isac/fix/source/filters.c",
     "codecs/isac/fix/source/initialize.c",
+    "codecs/isac/fix/source/isac_fix_type.h",
     "codecs/isac/fix/source/isacfix.c",
     "codecs/isac/fix/source/lattice.c",
     "codecs/isac/fix/source/lattice_c.c",
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
new file mode 100644
index 0000000000..714fe2d22a
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -0,0 +1,53 @@
+/*
+ *  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
+ *  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_AUDIO_DECODER_ISAC_T_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+
+namespace webrtc {
+
+template <typename T>
+class AudioDecoderIsacT final : public AudioDecoder {
+ public:
+  AudioDecoderIsacT();
+  explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
+  ~AudioDecoderIsacT() override;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  int IncomingPacket(const uint8_t* payload,
+                     size_t payload_len,
+                     uint16_t rtp_sequence_number,
+                     uint32_t rtp_timestamp,
+                     uint32_t arrival_timestamp) override;
+  int ErrorCode() override;
+  size_t Channels() const override;
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  typename T::instance_type* isac_state_;
+  LockedIsacBandwidthInfo* bwinfo_;
+  int decoder_sample_rate_hz_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
new file mode 100644
index 0000000000..126c129814
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -0,0 +1,108 @@
+/*
+ *  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
+ *  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_AUDIO_DECODER_ISAC_T_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT()
+    : AudioDecoderIsacT(nullptr) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
+    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+  RTC_CHECK_EQ(0, T::Create(&isac_state_));
+  T::DecoderInit(isac_state_);
+  if (bwinfo_) {
+    IsacBandwidthInfo bi;
+    T::GetBandwidthInfo(isac_state_, &bi);
+    bwinfo_->Set(bi);
+  }
+}
+
+template <typename T>
+AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
+  RTC_CHECK_EQ(0, T::Free(isac_state_));
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
+  // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
+  if (sample_rate_hz == 48000)
+    sample_rate_hz = 32000;
+  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  if (sample_rate_hz != decoder_sample_rate_hz_) {
+    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
+    decoder_sample_rate_hz_ = sample_rate_hz;
+  }
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+template <typename T>
+bool AudioDecoderIsacT<T>::HasDecodePlc() const {
+  return false;
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return T::DecodePlc(isac_state_, decoded, num_frames);
+}
+
+template <typename T>
+void AudioDecoderIsacT<T>::Reset() {
+  T::DecoderInit(isac_state_);
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
+                                         size_t payload_len,
+                                         uint16_t rtp_sequence_number,
+                                         uint32_t rtp_timestamp,
+                                         uint32_t arrival_timestamp) {
+  int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
+                                rtp_sequence_number, rtp_timestamp,
+                                arrival_timestamp);
+  if (bwinfo_) {
+    IsacBandwidthInfo bwinfo;
+    T::GetBandwidthInfo(isac_state_, &bwinfo);
+    bwinfo_->Set(bwinfo);
+  }
+  return ret;
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::ErrorCode() {
+  return T::GetErrorCode(isac_state_);
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 50d369ba35..6b361b4146 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -13,7 +13,6 @@
 
 #include <vector>
 
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
@@ -94,36 +93,5 @@ class AudioEncoderIsacT final : public AudioEncoder {
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
-template <typename T>
-class AudioDecoderIsacT final : public AudioDecoder {
- public:
-  AudioDecoderIsacT();
-  explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
-  ~AudioDecoderIsacT() override;
-
-  bool HasDecodePlc() const override;
-  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
-  void Reset() override;
-  int IncomingPacket(const uint8_t* payload,
-                     size_t payload_len,
-                     uint16_t rtp_sequence_number,
-                     uint32_t rtp_timestamp,
-                     uint32_t arrival_timestamp) override;
-  int ErrorCode() override;
-  size_t Channels() const override;
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  typename T::instance_type* isac_state_;
-  LockedIsacBandwidthInfo* bwinfo_;
-  int decoder_sample_rate_hz_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
-};
-
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 4122ee0bc5..fbc1ba9139 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -13,10 +13,7 @@
 
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
 
-#include <algorithm>
-
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
 
 namespace webrtc {
 
@@ -193,92 +190,6 @@ void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
   config_ = config;
 }
 
-template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT()
-    : AudioDecoderIsacT(nullptr) {}
-
-template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
-    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
-  RTC_CHECK_EQ(0, T::Create(&isac_state_));
-  T::DecoderInit(isac_state_);
-  if (bwinfo_) {
-    IsacBandwidthInfo bi;
-    T::GetBandwidthInfo(isac_state_, &bi);
-    bwinfo_->Set(bi);
-  }
-}
-
-template <typename T>
-AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
-  RTC_CHECK_EQ(0, T::Free(isac_state_));
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
-                                         size_t encoded_len,
-                                         int sample_rate_hz,
-                                         int16_t* decoded,
-                                         SpeechType* speech_type) {
-  // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
-  // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
-  if (sample_rate_hz == 48000)
-    sample_rate_hz = 32000;
-  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
-      << "Unsupported sample rate " << sample_rate_hz;
-  if (sample_rate_hz != decoder_sample_rate_hz_) {
-    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
-    decoder_sample_rate_hz_ = sample_rate_hz;
-  }
-  int16_t temp_type = 1;  // Default is speech.
-  int ret =
-      T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-template <typename T>
-bool AudioDecoderIsacT<T>::HasDecodePlc() const {
-  return false;
-}
-
-template <typename T>
-size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
-  return T::DecodePlc(isac_state_, decoded, num_frames);
-}
-
-template <typename T>
-void AudioDecoderIsacT<T>::Reset() {
-  T::DecoderInit(isac_state_);
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
-                                         size_t payload_len,
-                                         uint16_t rtp_sequence_number,
-                                         uint32_t rtp_timestamp,
-                                         uint32_t arrival_timestamp) {
-  int ret = T::UpdateBwEstimate(
-      isac_state_, payload, payload_len,
-      rtp_sequence_number, rtp_timestamp, arrival_timestamp);
-  if (bwinfo_) {
-    IsacBandwidthInfo bwinfo;
-    T::GetBandwidthInfo(isac_state_, &bwinfo);
-    bwinfo_->Set(bwinfo);
-  }
-  return ret;
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::ErrorCode() {
-  return T::GetErrorCode(isac_state_);
-}
-
-template <typename T>
-size_t AudioDecoderIsacT<T>::Channels() const {
-  return 1;
-}
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h
new file mode 100644
index 0000000000..ed162a10ae
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h
@@ -0,0 +1,22 @@
+/*
+ *  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
+ *  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_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
index 5bca23ec4e..00c0987749 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
@@ -11,116 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_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/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
 
 namespace webrtc {
 
-struct IsacFix {
-  typedef ISACFIX_MainStruct instance_type;
-  static const bool has_swb = false;
-  static const uint16_t kFixSampleRate = 16000;
-  static inline int16_t Control(instance_type* inst,
-                                int32_t rate,
-                                int framesize) {
-    return WebRtcIsacfix_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 WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
-                                    enforce_frame_size);
-  }
-  static inline int16_t Create(instance_type** inst) {
-    return WebRtcIsacfix_Create(inst);
-  }
-  static inline int DecodeInternal(instance_type* inst,
-                                   const uint8_t* encoded,
-                                   size_t len,
-                                   int16_t* decoded,
-                                   int16_t* speech_type) {
-    return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
-  }
-  static inline size_t DecodePlc(instance_type* inst,
-                                 int16_t* decoded,
-                                 size_t num_lost_frames) {
-    return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
-  }
-  static inline void DecoderInit(instance_type* inst) {
-    WebRtcIsacfix_DecoderInit(inst);
-  }
-  static inline int Encode(instance_type* inst,
-                           const int16_t* speech_in,
-                           uint8_t* encoded) {
-    return WebRtcIsacfix_Encode(inst, speech_in, encoded);
-  }
-  static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
-    return WebRtcIsacfix_EncoderInit(inst, coding_mode);
-  }
-  static inline uint16_t EncSampRate(instance_type* inst) {
-    return kFixSampleRate;
-  }
-
-  static inline int16_t Free(instance_type* inst) {
-    return WebRtcIsacfix_Free(inst);
-  }
-  static inline void GetBandwidthInfo(instance_type* inst,
-                                      IsacBandwidthInfo* bwinfo) {
-    WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo);
-  }
-  static inline int16_t GetErrorCode(instance_type* inst) {
-    return WebRtcIsacfix_GetErrorCode(inst);
-  }
-
-  static inline int16_t GetNewFrameLen(instance_type* inst) {
-    return WebRtcIsacfix_GetNewFrameLen(inst);
-  }
-  static inline void SetBandwidthInfo(instance_type* inst,
-                                      const IsacBandwidthInfo* bwinfo) {
-    WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo);
-  }
-  static inline int16_t SetDecSampRate(instance_type* inst,
-                                       uint16_t sample_rate_hz) {
-    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
-    return 0;
-  }
-  static inline int16_t SetEncSampRate(instance_type* inst,
-                                       uint16_t sample_rate_hz) {
-    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
-    return 0;
-  }
-  static inline void SetEncSampRateInDecoder(instance_type* inst,
-                                             uint16_t sample_rate_hz) {
-    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
-  }
-  static inline void SetInitialBweBottleneck(
-      instance_type* inst,
-      int bottleneck_bits_per_second) {
-    WebRtcIsacfix_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 WebRtcIsacfix_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 WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes);
-  }
-  static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
-    return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
-  }
-};
-
 using AudioEncoderIsacFix = AudioEncoderIsacT<IsacFix>;
-using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
new file mode 100644
index 0000000000..8b4741b2cb
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
@@ -0,0 +1,20 @@
+/*
+ *  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
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+
+namespace webrtc {
+
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFix>;
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
index 9b2f3174b8..ad75ea867e 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
@@ -14,10 +14,7 @@
 
 namespace webrtc {
 
-const uint16_t IsacFix::kFixSampleRate;
-
 // Explicit instantiation:
 template class AudioEncoderIsacT<IsacFix>;
-template class AudioDecoderIsacT<IsacFix>;
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
new file mode 100644
index 0000000000..69c73d6904
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
@@ -0,0 +1,123 @@
+/*
+ *  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
+ *  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_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+
+namespace webrtc {
+
+class IsacFix {
+ public:
+  using instance_type = ISACFIX_MainStruct;
+  static const bool has_swb = false;
+  static inline int16_t Control(instance_type* inst,
+                                int32_t rate,
+                                int framesize) {
+    return WebRtcIsacfix_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 WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
+                                    enforce_frame_size);
+  }
+  static inline int16_t Create(instance_type** inst) {
+    return WebRtcIsacfix_Create(inst);
+  }
+  static inline int DecodeInternal(instance_type* inst,
+                                   const uint8_t* encoded,
+                                   size_t len,
+                                   int16_t* decoded,
+                                   int16_t* speech_type) {
+    return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
+  }
+  static inline size_t DecodePlc(instance_type* inst,
+                                 int16_t* decoded,
+                                 size_t num_lost_frames) {
+    return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
+  }
+  static inline void DecoderInit(instance_type* inst) {
+    WebRtcIsacfix_DecoderInit(inst);
+  }
+  static inline int Encode(instance_type* inst,
+                           const int16_t* speech_in,
+                           uint8_t* encoded) {
+    return WebRtcIsacfix_Encode(inst, speech_in, encoded);
+  }
+  static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
+    return WebRtcIsacfix_EncoderInit(inst, coding_mode);
+  }
+  static inline uint16_t EncSampRate(instance_type* inst) {
+    return kFixSampleRate;
+  }
+
+  static inline int16_t Free(instance_type* inst) {
+    return WebRtcIsacfix_Free(inst);
+  }
+  static inline void GetBandwidthInfo(instance_type* inst,
+                                      IsacBandwidthInfo* bwinfo) {
+    WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo);
+  }
+  static inline int16_t GetErrorCode(instance_type* inst) {
+    return WebRtcIsacfix_GetErrorCode(inst);
+  }
+
+  static inline int16_t GetNewFrameLen(instance_type* inst) {
+    return WebRtcIsacfix_GetNewFrameLen(inst);
+  }
+  static inline void SetBandwidthInfo(instance_type* inst,
+                                      const IsacBandwidthInfo* bwinfo) {
+    WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo);
+  }
+  static inline int16_t SetDecSampRate(instance_type* inst,
+                                       uint16_t sample_rate_hz) {
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+    return 0;
+  }
+  static inline int16_t SetEncSampRate(instance_type* inst,
+                                       uint16_t sample_rate_hz) {
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+    return 0;
+  }
+  static inline void SetEncSampRateInDecoder(instance_type* inst,
+                                             uint16_t sample_rate_hz) {
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+  }
+  static inline void SetInitialBweBottleneck(instance_type* inst,
+                                             int bottleneck_bits_per_second) {
+    WebRtcIsacfix_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 WebRtcIsacfix_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 WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes);
+  }
+  static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
+    return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
+  }
+
+ private:
+  enum { kFixSampleRate = 16000 };
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/isac.gypi b/webrtc/modules/audio_coding/codecs/isac/isac.gypi
index 8ecc2dc414..354719f330 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isac.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isac.gypi
@@ -28,11 +28,13 @@
         ],
       },
       'sources': [
+        'main/interface/audio_decoder_isac.h',
         'main/interface/audio_encoder_isac.h',
         'main/interface/isac.h',
         'main/source/arith_routines.c',
         'main/source/arith_routines_hist.c',
         'main/source/arith_routines_logist.c',
+        'main/source/audio_decoder_isac.cc',
         'main/source/audio_encoder_isac.cc',
         'main/source/bandwidth_estimator.c',
         'main/source/crc.c',
@@ -46,6 +48,7 @@
         'main/source/filterbank_tables.c',
         'main/source/intialize.c',
         'main/source/isac.c',
+        'main/source/isac_float_type.h',
         'main/source/filterbanks.c',
         'main/source/pitch_lag_tables.c',
         'main/source/lattice.c',
diff --git a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
index 81b4375c21..4355fa92cc 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
@@ -27,11 +27,13 @@
         ],
       },
       'sources': [
+        'fix/interface/audio_decoder_isacfix.h',
         'fix/interface/audio_encoder_isacfix.h',
         'fix/interface/isacfix.h',
         'fix/source/arith_routines.c',
         'fix/source/arith_routines_hist.c',
         'fix/source/arith_routines_logist.c',
+        'fix/source/audio_decoder_isacfix.cc',
         'fix/source/audio_encoder_isacfix.cc',
         'fix/source/bandwidth_estimator.c',
         'fix/source/decode.c',
@@ -44,6 +46,7 @@
         'fix/source/filterbanks.c',
         'fix/source/filters.c',
         'fix/source/initialize.c',
+        'fix/source/isac_fix_type.h',
         'fix/source/isacfix.c',
         'fix/source/lattice.c',
         'fix/source/lattice_c.c',
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h
new file mode 100644
index 0000000000..b224296ad4
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h
@@ -0,0 +1,22 @@
+/*
+ *  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
+ *  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_DECODER_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_DECODER_ISAC_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
index da363d8c91..f6ef7c42fd 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
@@ -11,114 +11,12 @@
 #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_
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
new file mode 100644
index 0000000000..89879adafb
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
@@ -0,0 +1,20 @@
+/*
+ *  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
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+
+namespace webrtc {
+
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFloat>;
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
index eac7cc79e7..c72af307b7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
@@ -16,6 +16,5 @@ namespace webrtc {
 
 // Explicit instantiation:
 template class AudioEncoderIsacT<IsacFloat>;
-template class AudioDecoderIsacT<IsacFloat>;
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
new file mode 100644
index 0000000000..0335548be7
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
@@ -0,0 +1,117 @@
+/*
+ *  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
+ *  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_SOURCE_ISAC_FLOAT_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+
+namespace webrtc {
+
+struct IsacFloat {
+  using instance_type = ISACStruct;
+  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);
+  }
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
index c07ecec850..669eadb68e 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
@@ -16,7 +16,9 @@
 #include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 1428eaceda..52f327bc61 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -22,9 +22,11 @@
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
 #endif
 #ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
 #endif
 #ifdef WEBRTC_CODEC_OPUS
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index d7446a826e..537f16cca4 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -24,7 +24,9 @@
 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"