Revert 7623 "Remove the state_ member from AudioDecoder"

Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...

> Remove the state_ member from AudioDecoder
> 
> The subclasses that need a state pointer should declare them---with
> the right type, not void*, to get rid of all those casts.
> 
> Two small but not quite trivial cleanups are included because they
> blocked the state_ removal:
> 
>   - AudioDecoderG722Stereo now inherits directly from AudioDecoder
>     instead of being a subclass of AudioDecoderG722.
> 
>   - AudioDecoder now has a CngDecoderInstance member function, which
>     is implemented only by AudioDecoderCng. This replaces the previous
>     practice of calling AudioDecoder::state() and casting the result
>     to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
>     plainly visible in the AudioDecoder class declaration.
> 
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/24169005

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7629 4adac7df-926f-26a2-2b94-8c16560cd09d

6 files changed, 68 insertions, 84 deletions

diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc
index 8fa96e50..bc20c961 100644
--- a/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/modules/audio_coding/main/acm2/acm_isac.cc
@@ -277,6 +277,7 @@ ACMISAC::ACMISAC(int16_t codec_id)
     return;
   }
   codec_inst_ptr_->inst = NULL;
+  state_ = codec_inst_ptr_;
 }
 
 ACMISAC::~ACMISAC() {
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc
index 4fcae054..07b1b4be 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -103,17 +103,17 @@ AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) {
 // iLBC
 #ifdef WEBRTC_CODEC_ILBC
 AudioDecoderIlbc::AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_));
 }
 
 AudioDecoderIlbc::~AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderFree(dec_state_);
+  WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_));
 }
 
 int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIlbcfix_Decode(dec_state_,
+  int16_t ret = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_),
                                      reinterpret_cast<const int16_t*>(encoded),
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -122,11 +122,12 @@ int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
-  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+  return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_),
+                                decoded, num_frames);
 }
 
 int AudioDecoderIlbc::Init() {
-  return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+  return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_));
 }
 #endif
 
@@ -134,18 +135,19 @@ int AudioDecoderIlbc::Init() {
 #ifdef WEBRTC_CODEC_ISAC
 AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) {
   DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000);
-  WebRtcIsac_Create(&isac_state_);
-  WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz);
+  WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_));
+  WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_),
+                            decode_sample_rate_hz);
 }
 
 AudioDecoderIsac::~AudioDecoderIsac() {
-  WebRtcIsac_Free(isac_state_);
+  WebRtcIsac_Free(static_cast<ISACStruct*>(state_));
 }
 
 int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsac_Decode(isac_state_,
+  int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
                                   encoded,
                                   static_cast<int16_t>(encoded_len), decoded,
                                   &temp_type);
@@ -157,7 +159,7 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
                                       size_t encoded_len, int16_t* decoded,
                                       SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsac_DecodeRcu(isac_state_,
+  int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
                                      encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -166,11 +168,12 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
 }
 
 int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) {
-  return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames);
+  return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_),
+                                 decoded, num_frames);
 }
 
 int AudioDecoderIsac::Init() {
-  return WebRtcIsac_DecoderInit(isac_state_);
+  return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_));
 }
 
 int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
@@ -178,7 +181,7 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
                                      uint16_t rtp_sequence_number,
                                      uint32_t rtp_timestamp,
                                      uint32_t arrival_timestamp) {
-  return WebRtcIsac_UpdateBwEstimate(isac_state_,
+  return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_),
                                      payload,
                                      static_cast<int32_t>(payload_len),
                                      rtp_sequence_number,
@@ -187,24 +190,24 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
 }
 
 int AudioDecoderIsac::ErrorCode() {
-  return WebRtcIsac_GetErrorCode(isac_state_);
+  return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_));
 }
 #endif
 
 // iSAC fix
 #ifdef WEBRTC_CODEC_ISACFX
 AudioDecoderIsacFix::AudioDecoderIsacFix() {
-  WebRtcIsacfix_Create(&isac_state_);
+  WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_));
 }
 
 AudioDecoderIsacFix::~AudioDecoderIsacFix() {
-  WebRtcIsacfix_Free(isac_state_);
+  WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_));
 }
 
 int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
                                 int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsacfix_Decode(isac_state_,
+  int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
                                      encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -213,7 +216,7 @@ int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderIsacFix::Init() {
-  return WebRtcIsacfix_DecoderInit(isac_state_);
+  return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_));
 }
 
 int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
@@ -222,32 +225,32 @@ int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
                                         uint32_t rtp_timestamp,
                                         uint32_t arrival_timestamp) {
   return WebRtcIsacfix_UpdateBwEstimate(
-      isac_state_,
+      static_cast<ISACFIX_MainStruct*>(state_),
       payload,
       static_cast<int32_t>(payload_len),
       rtp_sequence_number, rtp_timestamp, arrival_timestamp);
 }
 
 int AudioDecoderIsacFix::ErrorCode() {
-  return WebRtcIsacfix_GetErrorCode(isac_state_);
+  return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_));
 }
 #endif
 
 // G.722
 #ifdef WEBRTC_CODEC_G722
 AudioDecoderG722::AudioDecoderG722() {
-  WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_));
 }
 
 AudioDecoderG722::~AudioDecoderG722() {
-  WebRtcG722_FreeDecoder(dec_state_);
+  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_));
 }
 
 int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
   int16_t ret = WebRtcG722_Decode(
-      dec_state_,
+      static_cast<G722DecInst*>(state_),
       const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)),
       static_cast<int16_t>(encoded_len), decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -255,7 +258,7 @@ int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderG722::Init() {
-  return WebRtcG722_DecoderInit(dec_state_);
+  return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_));
 }
 
 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@@ -264,15 +267,18 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(2 * encoded_len / channels_);
 }
 
-AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
+AudioDecoderG722Stereo::AudioDecoderG722Stereo()
+    : AudioDecoderG722(),
+      state_left_(state_),  // Base member |state_| is used for left channel.
+      state_right_(NULL) {
   channels_ = 2;
-  WebRtcG722_CreateDecoder(&dec_state_left_);
-  WebRtcG722_CreateDecoder(&dec_state_right_);
+  // |state_left_| already created by the base class AudioDecoderG722.
+  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_));
 }
 
 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
-  WebRtcG722_FreeDecoder(dec_state_left_);
-  WebRtcG722_FreeDecoder(dec_state_right_);
+  // |state_left_| will be freed by the base class AudioDecoderG722.
+  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_));
 }
 
 int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
@@ -283,13 +289,13 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
   SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
   // Decode left and right.
   int16_t ret = WebRtcG722_Decode(
-      dec_state_left_,
+      static_cast<G722DecInst*>(state_left_),
       reinterpret_cast<int16_t*>(encoded_deinterleaved),
       static_cast<int16_t>(encoded_len / 2), decoded, &temp_type);
   if (ret >= 0) {
     int decoded_len = ret;
     ret = WebRtcG722_Decode(
-      dec_state_right_,
+      static_cast<G722DecInst*>(state_right_),
       reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]),
       static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type);
     if (ret == decoded_len) {
@@ -311,10 +317,11 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderG722Stereo::Init() {
-  int r = WebRtcG722_DecoderInit(dec_state_left_);
-  if (r != 0)
-    return r;
-  return WebRtcG722_DecoderInit(dec_state_right_);
+  int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_));
+  if (ret != 0) {
+    return ret;
+  }
+  return AudioDecoderG722::Init();
 }
 
 // Split the stereo packet and place left and right channel after each other
@@ -394,17 +401,18 @@ int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) {
 AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
   DCHECK(num_channels == 1 || num_channels == 2);
   channels_ = num_channels;
-  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+  WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_),
+                           static_cast<int>(channels_));
 }
 
 AudioDecoderOpus::~AudioDecoderOpus() {
-  WebRtcOpus_DecoderFree(dec_state_);
+  WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_));
 }
 
 int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeNew(dec_state_, encoded,
+  int16_t ret = WebRtcOpus_DecodeNew(static_cast<OpusDecInst*>(state_), encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   if (ret > 0)
@@ -417,7 +425,7 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
                                       size_t encoded_len, int16_t* decoded,
                                       SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
+  int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   if (ret > 0)
@@ -427,12 +435,12 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
 }
 
 int AudioDecoderOpus::Init() {
-  return WebRtcOpus_DecoderInitNew(dec_state_);
+  return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_));
 }
 
 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) {
-  return WebRtcOpus_DurationEst(dec_state_,
+  return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_),
                                 encoded, static_cast<int>(encoded_len));
 }
 
@@ -450,15 +458,19 @@ bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
 #endif
 
 AudioDecoderCng::AudioDecoderCng() {
-  DCHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
+  WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_));
+  assert(state_);
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
-  WebRtcCng_FreeDec(dec_state_);
+  if (state_) {
+    WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_));
+  }
 }
 
 int AudioDecoderCng::Init() {
-  return WebRtcCng_InitDec(dec_state_);
+  assert(state_);
+  return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_));
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h
index b30331f3..214392e7 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -19,22 +19,6 @@
 #include "webrtc/engine_configurations.h"
 #endif
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
-#endif
-#ifdef WEBRTC_CODEC_ISACFX
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#endif
 #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
 #include "webrtc/typedefs.h"
 
@@ -125,7 +109,6 @@ class AudioDecoderIlbc : public AudioDecoder {
   virtual int Init();
 
  private:
-  iLBC_decinst_t* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
 };
 #endif
@@ -150,7 +133,6 @@ class AudioDecoderIsac : public AudioDecoder {
   virtual int ErrorCode();
 
  private:
-  ISACStruct* isac_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsac);
 };
 #endif
@@ -171,7 +153,6 @@ class AudioDecoderIsacFix : public AudioDecoder {
   virtual int ErrorCode();
 
  private:
-  ISACFIX_MainStruct* isac_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix);
 };
 #endif
@@ -188,11 +169,10 @@ class AudioDecoderG722 : public AudioDecoder {
   virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len);
 
  private:
-  G722DecInst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
 };
 
-class AudioDecoderG722Stereo : public AudioDecoder {
+class AudioDecoderG722Stereo : public AudioDecoderG722 {
  public:
   AudioDecoderG722Stereo();
   virtual ~AudioDecoderG722Stereo();
@@ -209,8 +189,8 @@ class AudioDecoderG722Stereo : public AudioDecoder {
   void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len,
                          uint8_t* encoded_deinterleaved);
 
-  G722DecInst* dec_state_left_;
-  G722DecInst* dec_state_right_;
+  void* const state_left_;
+  void* state_right_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
 };
@@ -249,7 +229,6 @@ class AudioDecoderOpus : public AudioDecoder {
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
  private:
-  OpusDecInst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
 };
 #endif
@@ -273,10 +252,7 @@ class AudioDecoderCng : public AudioDecoder {
                              uint32_t rtp_timestamp,
                              uint32_t arrival_timestamp) { return -1; }
 
-  virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; }
-
  private:
-  CNG_dec_inst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
 };
 
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index e2be066e..31bb40c9 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -36,7 +36,7 @@ int ComfortNoise::UpdateParameters(Packet* packet) {
     return kUnknownPayloadType;
   }
   decoder_database_->SetActiveCngDecoder(packet->header.payloadType);
-  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
+  CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
   int16_t ret = WebRtcCng_UpdateSid(cng_inst,
                                     packet->payload,
                                     packet->payload_length);
@@ -72,7 +72,7 @@ int ComfortNoise::Generate(size_t requested_length,
   if (!cng_decoder) {
     return kUnknownPayloadType;
   }
-  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
+  CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
   // The expression &(*output)[0][0] is a pointer to the first element in
   // the first channel.
   if (WebRtcCng_Generate(cng_inst, &(*output)[0][0],
diff --git a/modules/audio_coding/neteq/interface/audio_decoder.h b/modules/audio_coding/neteq/interface/audio_decoder.h
index b36d215e..16d78c9e 100644
--- a/modules/audio_coding/neteq/interface/audio_decoder.h
+++ b/modules/audio_coding/neteq/interface/audio_decoder.h
@@ -13,9 +13,7 @@
 
 #include <stdlib.h>  // NULL
 
-#include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -65,7 +63,7 @@ class AudioDecoder {
   // Used by PacketDuration below. Save the value -1 for errors.
   enum { kNotImplemented = -2 };
 
-  AudioDecoder() : channels_(1) {}
+  AudioDecoder() : channels_(1), state_(NULL) {}
   virtual ~AudioDecoder() {}
 
   // Decodes |encode_len| bytes from |encoded| and writes the result in
@@ -116,12 +114,8 @@ class AudioDecoder {
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
-  // If this is a CNG decoder, return the underlying CNG_dec_inst*. If this
-  // isn't a CNG decoder, don't call this method.
-  virtual CNG_dec_inst* CngDecoderInstance() {
-    FATAL() << "Not a CNG decoder";
-    return NULL;
-  }
+  // Returns the underlying decoder state.
+  void* state() { return state_; }
 
   // Returns true if |codec_type| is supported.
   static bool CodecSupported(NetEqDecoder codec_type);
@@ -140,6 +134,7 @@ class AudioDecoder {
   static SpeechType ConvertSpeechType(int16_t type);
 
   size_t channels_;
+  void* state_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index ca2c1ee5..46d03fb8 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -147,9 +147,9 @@ int Normal::Process(const int16_t* input,
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
 
     if (cng_decoder) {
+      CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
       // Generate long enough for 32kHz.
-      if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
-                             kCngLength, 0) < 0) {
+      if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }