diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 480b1aa26d..5e9e33d47b 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -64,8 +64,8 @@ class AudioDecoder {
   // one or several lost packets.
   virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
 
-  // Initializes the decoder.
-  virtual int Init() = 0;
+  // Resets the decoder state (empty buffers etc.).
+  virtual void Reset() = 0;
 
   // Notifies the decoder of an incoming packet to NetEQ.
   virtual int IncomingPacket(const uint8_t* payload,
diff --git a/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
index 24095402e5..1061dca69a 100644
--- a/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -194,7 +194,7 @@ TEST_F(CngTest, CngUpdateSid) {
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Run normal Encode and UpdateSid.
   EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@@ -205,7 +205,7 @@ TEST_F(CngTest, CngUpdateSid) {
   // Reinit with new length.
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsHigh));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Expect 0 because of unstable parameters after switching length.
   EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
@@ -242,7 +242,7 @@ TEST_F(CngTest, CngUpdateSidErroneous) {
   EXPECT_EQ(6220, WebRtcCng_GetErrorCodeDec(cng_dec_inst_));
 
   // Initialize decoder.
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // First run with valid parameters, then with too many CNG parameters.
   // The function will operate correctly by only reading the maximum number of
@@ -268,7 +268,7 @@ TEST_F(CngTest, CngGenerate) {
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Normal Encode.
   EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@@ -301,7 +301,7 @@ TEST_F(CngTest, CngAutoSid) {
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Normal Encode, 100 msec, where no SID data should be generated.
   for (int i = 0; i < 10; i++) {
@@ -328,7 +328,7 @@ TEST_F(CngTest, CngAutoSidShort) {
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidShortIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // First call will never generate SID, unless forced to.
   EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
index 6c7e50bf92..fe87fc90cc 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
@@ -70,7 +70,7 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
 
 int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
                       int16_t quality);
-int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
+void WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
 
 /****************************************************************************
  * WebRtcCng_FreeEnc/Dec(...)
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index a0c166a6c3..8dddc5c717 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -169,7 +169,7 @@ int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
   return 0;
 }
 
-int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
+void WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
   int i;
 
   WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
@@ -188,8 +188,6 @@ int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
   inst->dec_used_reflCoefs[0] = 0;
   inst->dec_used_energy = 0;
   inst->initflag = 1;
-
-  return 0;
 }
 
 /****************************************************************************
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
index 8fdeec162b..952a7d037f 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
@@ -157,11 +157,7 @@ static void block4(G722DecoderState *s, int band, int d)
 G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
                                           int rate,
                                           int options) {
-    if (s == NULL)
-    {
-        if ((s = (G722DecoderState *) malloc(sizeof(*s))) == NULL)
-            return NULL;
-    }
+    s = s ? s : malloc(sizeof(*s));
     memset(s, 0, sizeof(*s));
     if (rate == 48000)
         s->bits_per_sample = 6;
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index f6b9842eb4..4244d5c809 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -66,17 +66,10 @@ int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
     }
 }
 
-int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
-{
-    // Create and/or reset the G.722 decoder
-    // Bitrate 64 kbps and wideband mode (2)
-    G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
-        (G722DecoderState*) G722dec_inst, 64000, 2);
-    if (G722dec_inst == NULL) {
-        return -1;
-    } else {
-        return 0;
-    }
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+  // Create and/or reset the G.722 decoder
+  // Bitrate 64 kbps and wideband mode (2)
+  WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
 }
 
 int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
index fa4a48c297..e3133d6cf7 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -113,22 +113,16 @@ size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
  */
 int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
 
-
 /****************************************************************************
  * WebRtcG722_DecoderInit(...)
  *
- * This function initializes a G729 instance
+ * This function initializes a G722 instance
  *
  * Input:
- *     - G729_decinst_t    : G729 instance, i.e. the user that should receive
- *                           be initialized
- *
- * Return value            :  0 - Ok
- *                           -1 - Error
+ *     - inst      : G722 instance
  */
 
-int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
-
+void WebRtcG722_DecoderInit(G722DecInst* inst);
 
 /****************************************************************************
  * WebRtcG722_FreeDecoder(...)
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index c565a246db..6cd9a723fd 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -131,13 +131,11 @@ int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
     return(-1);
   }
 }
-int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst) {
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
-  return(0);
 }
-int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
-  return(0);
 }
 
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
index be0b121abb..ba31f18ba5 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
@@ -159,8 +159,8 @@ extern "C" {
 
   int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance *iLBCdec_inst,
                                     int16_t frameLen);
-  int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst);
-  int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst);
+  void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
+  void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
 
   /****************************************************************************
    * WebRtcIlbcfix_Decode(...)
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 a2c43a6c25..a8498fa224 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
@@ -95,7 +95,7 @@ class AudioDecoderIsacT final : public AudioDecoder {
 
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
-  int Init() override;
+  void Reset() override;
   int IncomingPacket(const uint8_t* payload,
                      size_t payload_len,
                      uint16_t rtp_sequence_number,
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 93fbde9c48..98f3ed9652 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
@@ -185,7 +185,7 @@ template <typename T>
 AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
     : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
   CHECK_EQ(0, T::Create(&isac_state_));
-  CHECK_EQ(0, T::DecoderInit(isac_state_));
+  T::DecoderInit(isac_state_);
   if (bwinfo_) {
     IsacBandwidthInfo bwinfo;
     T::GetBandwidthInfo(isac_state_, &bwinfo);
@@ -232,8 +232,8 @@ size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
 }
 
 template <typename T>
-int AudioDecoderIsacT<T>::Init() {
-  return T::DecoderInit(isac_state_);
+void AudioDecoderIsacT<T>::Reset() {
+  T::DecoderInit(isac_state_);
 }
 
 template <typename T>
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 6c619153b1..0fd05da682 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
@@ -50,8 +50,8 @@ struct IsacFix {
                                  size_t num_lost_frames) {
     return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
   }
-  static inline int16_t DecoderInit(instance_type* inst) {
-    return WebRtcIsacfix_DecoderInit(inst);
+  static inline void DecoderInit(instance_type* inst) {
+    WebRtcIsacfix_DecoderInit(inst);
   }
   static inline int Encode(instance_type* inst,
                            const int16_t* speech_in,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
index eec4a39553..013ab7f13d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -174,14 +174,9 @@ extern "C" {
    *
    * Input:
    *  - ISAC_main_inst : ISAC instance.
-   *
-   * Return value
-   *       :  0 - Ok
-   *         -1 - Error
    */
 
-  int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
-
+  void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst);
 
   /****************************************************************************
    * WebRtcIsacfix_UpdateBwEstimate1(...)
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 4a663d12cf..21911dd058 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -568,13 +568,9 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
  *
  * Input:
  *      - ISAC_main_inst    : ISAC instance.
- *
- * Return value
- *                          :  0 - Ok
- *                            -1 - Error
  */
 
-int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
+void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -597,8 +593,6 @@ int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
   WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACdec_obj.decimatorstr_obj);
 #endif
-
-  return 0;
 }
 
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index fc7588dac8..adee3376b5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -48,7 +48,7 @@ void IsacSpeedTest::SetUp() {
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
   EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
-  EXPECT_EQ(0, WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_));
+  WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_);
   // Set bitrate and block length.
   EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
                                      block_duration_ms_));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index 6a947c8f01..d0f508f759 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -539,12 +539,7 @@ int main(int argc, char* argv[])
         printf("\n\n Error in encoderinit: %d.\n\n", errtype);
       }
 
-      err = WebRtcIsacfix_DecoderInit(ISAC_main_inst);
-      /* Error check */
-      if (err < 0) {
-        errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
-        printf("\n\n Error in decoderinit: %d.\n\n", errtype);
-      }
+      WebRtcIsacfix_DecoderInit(ISAC_main_inst);
     }
 
 
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 1bfd149b27..58abbdf3c3 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
@@ -50,8 +50,8 @@ struct IsacFloat {
     return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
   }
 
-  static inline int16_t DecoderInit(instance_type* inst) {
-    return WebRtcIsac_DecoderInit(inst);
+  static inline void DecoderInit(instance_type* inst) {
+    WebRtcIsac_DecoderInit(inst);
   }
   static inline int Encode(instance_type* inst,
                            const int16_t* speech_in,
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
index 0597de8ae8..1f5aeb36f4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -157,15 +157,9 @@ extern "C" {
    *
    * Input:
    *        - ISAC_main_inst    : ISAC instance.
-   *
-   * Return value
-   *                            : 0 - Ok
-   *                             -1 - Error
    */
 
-  int16_t WebRtcIsac_DecoderInit(
-      ISACStruct* ISAC_main_inst);
-
+  void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
 
   /******************************************************************************
    * WebRtcIsac_UpdateBwEstimate(...)
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index 190277eb66..0a5f75a901 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -924,12 +924,8 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
  *
  * Input:
  *        - ISAC_main_inst    : ISAC instance.
- *
- * Return value
- *                            :  0 - Ok
- *                              -1 - Error
  */
-static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
+static void DecoderInitLb(ISACLBStruct* instISAC) {
   int i;
   /* Initialize stream vector to zero. */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -940,10 +936,9 @@ static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
   WebRtcIsac_InitPostFilterbank(
     &instISAC->ISACdecLB_obj.postfiltbankstr_obj);
   WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
-  return 0;
 }
 
-static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
+static void DecoderInitUb(ISACUBStruct* instISAC) {
   int i;
   /* Init stream vector to zero */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -953,24 +948,18 @@ static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
   WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
   WebRtcIsac_InitPostFilterbank(
     &instISAC->ISACdecUB_obj.postfiltbankstr_obj);
-  return (0);
 }
 
-int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
+void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
-  if (DecoderInitLb(&instISAC->instLB) < 0) {
-    return -1;
-  }
+  DecoderInitLb(&instISAC->instLB);
   if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
     memset(instISAC->synthesisFBState1, 0,
            FB_STATE_SIZE_WORD32 * sizeof(int32_t));
     memset(instISAC->synthesisFBState2, 0,
            FB_STATE_SIZE_WORD32 * sizeof(int32_t));
-
-    if (DecoderInitUb(&(instISAC->instUB)) < 0) {
-      return -1;
-    }
+    DecoderInitUb(&(instISAC->instUB));
   }
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
     WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
@@ -979,7 +968,6 @@ int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
   }
   instISAC->initFlag |= BIT_MASK_DEC_INIT;
   instISAC->resetFlag_8kHz = 0;
-  return 0;
 }
 
 
@@ -2353,9 +2341,7 @@ int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
       memset(instISAC->synthesisFBState2, 0,
              FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
-      if (DecoderInitUb(&(instISAC->instUB)) < 0) {
-        return -1;
-      }
+      DecoderInitUb(&instISAC->instUB);
   }
   instISAC->decoderSamplingRateKHz = decoder_operational_rate;
   return 0;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index d385ff40c3..2e5badd82c 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -499,13 +499,8 @@ int main(int argc, char* argv[]) {
       return 0;
     }
   }
-  if (testNum != 2) {
-    if (WebRtcIsac_DecoderInit(ISAC_main_inst) < 0) {
-      printf("Error could not initialize the decoder \n");
-      cout << flush;
-      return 0;
-    }
-  }
+  if (testNum != 2)
+    WebRtcIsac_DecoderInit(ISAC_main_inst);
   if (CodingMode == 1) {
     err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
     if (err < 0) {
@@ -570,13 +565,7 @@ int main(int argc, char* argv[]) {
         cout << flush;
       }
 
-      err = WebRtcIsac_DecoderInit(ISAC_main_inst);
-      /* Error check */
-      if (err < 0) {
-        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
-        printf("\n\n Error in decoderinit: %d.\n\n", errtype);
-        cout << flush;
-      }
+      WebRtcIsac_DecoderInit(ISAC_main_inst);
     }
 
     cur_framesmpls = 0;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 08061ac532..a53e7bd0b5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -166,13 +166,7 @@ int main(int argc, char* argv[])
       return -1;
     }
 
-    // Initialize Decoder
-    if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0)
-    {
-      printf("Could not initialize decoder of client %d\n",
-             clientCntr + 1);
-      return -1;
-    }
+    WebRtcIsac_DecoderInit(codecInstance[clientCntr]);
 
     // setup Rate if in Instantaneous mode
     if(codingMode != 0)
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 2f44ca88ae..e8116ffdf8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -253,10 +253,7 @@ int main(int argc, char* argv[]) {
     printf("cannot initialize encoder\n");
     return -1;
   }
-  if (WebRtcIsac_DecoderInit(ISAC_main_inst) < 0) {
-    printf("cannot initialize decoder\n");
-    return -1;
-  }
+  WebRtcIsac_DecoderInit(ISAC_main_inst);
 
   // {
   //   int32_t b1, b2;
diff --git a/webrtc/modules/audio_coding/codecs/isac/unittest.cc b/webrtc/modules/audio_coding/codecs/isac/unittest.cc
index d05ffa6e48..673d2906ae 100644
--- a/webrtc/modules/audio_coding/codecs/isac/unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/unittest.cc
@@ -111,7 +111,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
   typename T::instance_type* encdec;
   ASSERT_EQ(0, T::Create(&encdec));
   ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
-  ASSERT_EQ(0, T::DecoderInit(encdec));
+  T::DecoderInit(encdec);
   ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
   if (adaptive)
     ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
@@ -129,7 +129,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
     ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
   typename T::instance_type* dec;
   ASSERT_EQ(0, T::Create(&dec));
-  ASSERT_EQ(0, T::DecoderInit(dec));
+  T::DecoderInit(dec);
   T::SetInitialBweBottleneck(dec, bit_rate);
   T::SetEncSampRateInDecoder(dec, sample_rate_hz);
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
index 007f5c53f4..ded8b6f8eb 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
@@ -212,11 +212,8 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
  *
  * Input:
  *      - inst               : Decoder context
- *
- * Return value              :  0 - Success
- *                             -1 - Error
  */
-int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
+void WebRtcOpus_DecoderInit(OpusDecInst* inst);
 
 /****************************************************************************
  * WebRtcOpus_Decode(...)
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index e2a8383c4b..4f6e22f77d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -250,13 +250,9 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
   return inst->channels;
 }
 
-int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
-  int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
-  if (error == OPUS_OK) {
-    inst->in_dtx_mode = 0;
-    return 0;
-  }
-  return -1;
+void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  inst->in_dtx_mode = 0;
 }
 
 /* For decoder to determine if it is to output speech or comfort noise. */
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index 2208f741f2..d2fd009055 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -376,7 +376,7 @@ TEST_P(OpusTest, OpusDecodeInit) {
                 kOpus20msFrameSamples, opus_decoder_, output_data_decode,
                 &audio_type)));
 
-  EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_decoder_));
+  WebRtcOpus_DecoderInit(opus_decoder_);
 
   EXPECT_EQ(kOpus20msFrameSamples,
             static_cast<size_t>(WebRtcOpus_Decode(
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 74f65a99f4..4b53a524e9 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -988,9 +988,6 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutputExternalDecoder) {
   MockAudioDecoder mock_decoder;
   // Set expectations on the mock decoder and also delegate the calls to the
   // real decoder.
-  EXPECT_CALL(mock_decoder, Init())
-      .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Init));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, _, _, _, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::IncomingPacket));
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index 79124aa7f3..d6482dd447 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -84,8 +84,8 @@ void OpusTest::Perform() {
   // Create Opus decoders for mono and stereo for stand-alone testing of Opus.
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
-  ASSERT_GT(WebRtcOpus_DecoderInit(opus_mono_decoder_), -1);
-  ASSERT_GT(WebRtcOpus_DecoderInit(opus_stereo_decoder_), -1);
+  WebRtcOpus_DecoderInit(opus_mono_decoder_);
+  WebRtcOpus_DecoderInit(opus_stereo_decoder_);
 
   ASSERT_TRUE(acm_receiver_.get() != NULL);
   EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 769f0b0fa8..592f17b097 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -39,8 +39,7 @@ namespace webrtc {
 
 // PCMu
 
-int AudioDecoderPcmU::Init() {
-  return 0;
+void AudioDecoderPcmU::Reset() {
 }
 size_t AudioDecoderPcmU::Channels() const {
   return 1;
@@ -70,8 +69,7 @@ size_t AudioDecoderPcmUMultiCh::Channels() const {
 
 // PCMa
 
-int AudioDecoderPcmA::Init() {
-  return 0;
+void AudioDecoderPcmA::Reset() {
 }
 size_t AudioDecoderPcmA::Channels() const {
   return 1;
@@ -103,8 +101,7 @@ size_t AudioDecoderPcmAMultiCh::Channels() const {
 #ifdef WEBRTC_CODEC_PCM16
 AudioDecoderPcm16B::AudioDecoderPcm16B() {}
 
-int AudioDecoderPcm16B::Init() {
-  return 0;
+void AudioDecoderPcm16B::Reset() {
 }
 size_t AudioDecoderPcm16B::Channels() const {
   return 1;
@@ -143,6 +140,7 @@ size_t AudioDecoderPcm16BMultiCh::Channels() const {
 #ifdef WEBRTC_CODEC_ILBC
 AudioDecoderIlbc::AudioDecoderIlbc() {
   WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
 AudioDecoderIlbc::~AudioDecoderIlbc() {
@@ -170,8 +168,8 @@ size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
   return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
-int AudioDecoderIlbc::Init() {
-  return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+void AudioDecoderIlbc::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
 size_t AudioDecoderIlbc::Channels() const {
@@ -183,6 +181,7 @@ size_t AudioDecoderIlbc::Channels() const {
 #ifdef WEBRTC_CODEC_G722
 AudioDecoderG722::AudioDecoderG722() {
   WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_DecoderInit(dec_state_);
 }
 
 AudioDecoderG722::~AudioDecoderG722() {
@@ -206,8 +205,8 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
-int AudioDecoderG722::Init() {
-  return WebRtcG722_DecoderInit(dec_state_);
+void AudioDecoderG722::Reset() {
+  WebRtcG722_DecoderInit(dec_state_);
 }
 
 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@@ -223,6 +222,8 @@ size_t AudioDecoderG722::Channels() const {
 AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
   WebRtcG722_CreateDecoder(&dec_state_left_);
   WebRtcG722_CreateDecoder(&dec_state_right_);
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
 }
 
 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
@@ -265,11 +266,9 @@ size_t AudioDecoderG722Stereo::Channels() const {
   return 2;
 }
 
-int AudioDecoderG722Stereo::Init() {
-  int r = WebRtcG722_DecoderInit(dec_state_left_);
-  if (r != 0)
-    return r;
-  return WebRtcG722_DecoderInit(dec_state_right_);
+void AudioDecoderG722Stereo::Reset() {
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
 }
 
 // Split the stereo packet and place left and right channel after each other
@@ -306,6 +305,7 @@ AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
     : channels_(num_channels) {
   DCHECK(num_channels == 1 || num_channels == 2);
   WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+  WebRtcOpus_DecoderInit(dec_state_);
 }
 
 AudioDecoderOpus::~AudioDecoderOpus() {
@@ -348,8 +348,8 @@ int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
   return ret;
 }
 
-int AudioDecoderOpus::Init() {
-  return WebRtcOpus_DecoderInit(dec_state_);
+void AudioDecoderOpus::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
 }
 
 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
@@ -381,14 +381,15 @@ size_t AudioDecoderOpus::Channels() const {
 
 AudioDecoderCng::AudioDecoderCng() {
   CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
+  WebRtcCng_InitDec(dec_state_);
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
   WebRtcCng_FreeDec(dec_state_);
 }
 
-int AudioDecoderCng::Init() {
-  return WebRtcCng_InitDec(dec_state_);
+void AudioDecoderCng::Reset() {
+  WebRtcCng_InitDec(dec_state_);
 }
 
 int AudioDecoderCng::IncomingPacket(const uint8_t* payload,
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 427a0a6006..f2ca711383 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -37,7 +37,7 @@ namespace webrtc {
 class AudioDecoderPcmU : public AudioDecoder {
  public:
   AudioDecoderPcmU() {}
-  int Init() override;
+  void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 
@@ -55,7 +55,7 @@ class AudioDecoderPcmU : public AudioDecoder {
 class AudioDecoderPcmA : public AudioDecoder {
  public:
   AudioDecoderPcmA() {}
-  int Init() override;
+  void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 
@@ -102,7 +102,7 @@ class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
 class AudioDecoderPcm16B : public AudioDecoder {
  public:
   AudioDecoderPcm16B();
-  int Init() override;
+  void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 
@@ -138,7 +138,7 @@ class AudioDecoderIlbc : public AudioDecoder {
   ~AudioDecoderIlbc() override;
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
-  int Init() override;
+  void Reset() override;
   size_t Channels() const override;
 
  protected:
@@ -160,7 +160,7 @@ class AudioDecoderG722 : public AudioDecoder {
   AudioDecoderG722();
   ~AudioDecoderG722() override;
   bool HasDecodePlc() const override;
-  int Init() override;
+  void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 
@@ -180,7 +180,7 @@ class AudioDecoderG722Stereo : public AudioDecoder {
  public:
   AudioDecoderG722Stereo();
   ~AudioDecoderG722Stereo() override;
-  int Init() override;
+  void Reset() override;
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
@@ -212,7 +212,7 @@ class AudioDecoderOpus : public AudioDecoder {
   explicit AudioDecoderOpus(size_t num_channels);
   ~AudioDecoderOpus() override;
 
-  int Init() override;
+  void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   int PacketDurationRedundant(const uint8_t* encoded,
                               size_t encoded_len) const override;
@@ -248,7 +248,7 @@ class AudioDecoderCng : public AudioDecoder {
  public:
   explicit AudioDecoderCng();
   ~AudioDecoderCng() override;
-  int Init() override;
+  void Reset() override;
   int IncomingPacket(const uint8_t* payload,
                      size_t payload_len,
                      uint16_t rtp_sequence_number,
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index a2ef9d1cb1..392e3dccd3 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -171,7 +171,6 @@ class AudioDecoderTest : public ::testing::Test {
     size_t processed_samples = 0u;
     encoded_bytes_ = 0u;
     InitEncoder();
-    EXPECT_EQ(0, decoder_->Init());
     std::vector<int16_t> input;
     std::vector<int16_t> decoded;
     while (processed_samples + frame_size_ <= data_length_) {
@@ -220,7 +219,7 @@ class AudioDecoderTest : public ::testing::Test {
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     size_t dec_len;
     AudioDecoder::SpeechType speech_type1, speech_type2;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
     dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                frame_size_ * channels_ * sizeof(int16_t),
@@ -228,7 +227,7 @@ class AudioDecoderTest : public ::testing::Test {
     ASSERT_LE(dec_len, frame_size_ * channels_);
     EXPECT_EQ(frame_size_ * channels_, dec_len);
     // Re-init decoder and decode again.
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
     dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                frame_size_ * channels_ * sizeof(int16_t),
@@ -249,7 +248,7 @@ class AudioDecoderTest : public ::testing::Test {
         input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     AudioDecoder::SpeechType speech_type;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
     size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                       frame_size_ * channels_ * sizeof(int16_t),
@@ -341,7 +340,7 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
         input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     AudioDecoder::SpeechType speech_type;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
     size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                       frame_size_ * channels_ * sizeof(int16_t),
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.cc b/webrtc/modules/audio_coding/neteq/decoder_database.cc
index 18eee067e3..97dc00d7a6 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -72,7 +72,6 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
   if (!decoder) {
     return kInvalidPointer;
   }
-  decoder->Init();
   std::pair<DecoderMap::iterator, bool> ret;
   DecoderInfo info(codec_type, fs_hz, decoder, true);
   ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
@@ -136,7 +135,6 @@ AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) {
     AudioDecoder* decoder = CreateAudioDecoder(info->codec_type);
     assert(decoder);  // Should not be able to have an unsupported codec here.
     info->decoder = decoder;
-    info->decoder->Init();
   }
   return info->decoder;
 }
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
index d26e2a19aa..90f132b0f7 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -27,7 +27,7 @@ class MockAudioDecoder : public AudioDecoder {
       int(const uint8_t*, size_t, int, size_t, int16_t*, SpeechType*));
   MOCK_CONST_METHOD0(HasDecodePlc, bool());
   MOCK_METHOD2(DecodePlc, size_t(size_t, int16_t*));
-  MOCK_METHOD0(Init, int());
+  MOCK_METHOD0(Reset, void());
   MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
                                    uint32_t));
   MOCK_METHOD0(ErrorCode, int());
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
index f239b4aa3a..fca1c2d6ee 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
@@ -28,7 +28,7 @@ using ::testing::Invoke;
 class ExternalPcm16B : public AudioDecoder {
  public:
   ExternalPcm16B() {}
-  virtual int Init() { return 0; }
+  void Reset() override {}
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
@@ -58,8 +58,8 @@ class MockExternalPcm16B : public ExternalPcm16B {
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
     ON_CALL(*this, DecodePlc(_, _))
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodePlc));
-    ON_CALL(*this, Init())
-        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Init));
+    ON_CALL(*this, Reset())
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Reset));
     ON_CALL(*this, IncomingPacket(_, _, _, _, _))
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket));
     ON_CALL(*this, ErrorCode())
@@ -79,8 +79,7 @@ class MockExternalPcm16B : public ExternalPcm16B {
       bool());
   MOCK_METHOD2(DecodePlc,
       size_t(size_t num_frames, int16_t* decoded));
-  MOCK_METHOD0(Init,
-      int());
+  MOCK_METHOD0(Reset, void());
   MOCK_METHOD5(IncomingPacket,
       int(const uint8_t* payload, size_t payload_len,
           uint16_t rtp_sequence_number, uint32_t rtp_timestamp,
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 3c945f92dd..2a116163bf 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -40,8 +40,6 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
         payload_size_bytes_(0),
         last_send_time_(0),
         last_arrival_time_(0) {
-    // Init() will trigger external_decoder_->Init().
-    EXPECT_CALL(*external_decoder_, Init());
     // NetEq is not allowed to delete the external decoder (hence Times(0)).
     EXPECT_CALL(*external_decoder_, Die()).Times(0);
     Init();
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 22e71f7f1d..cf7afbc8d9 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -1178,15 +1178,14 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
 
   if (reset_decoder_) {
     // TODO(hlundin): Write test for this.
-    // Reset decoder.
-    if (decoder) {
-      decoder->Init();
-    }
+    if (decoder)
+      decoder->Reset();
+
     // Reset comfort noise decoder.
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
-    if (cng_decoder) {
-      cng_decoder->Init();
-    }
+    if (cng_decoder)
+      cng_decoder->Reset();
+
     reset_decoder_ = false;
   }
 
@@ -1896,11 +1895,9 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
     mute_factor_array_[i] = 16384;  // 1.0 in Q14.
   }
 
-  // Reset comfort noise decoder, if there is one active.
   AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
-  if (cng_decoder) {
-    cng_decoder->Init();
-  }
+  if (cng_decoder)
+    cng_decoder->Reset();
 
   // Reinit post-decode VAD with new sample rate.
   assert(vad_.get());  // Cannot be NULL here.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 006a5ad542..ee04a6f255 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -444,10 +444,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
       return encoded_len;
     }
 
-    virtual int Init() {
-      next_value_ = 1;
-      return 0;
-    }
+    void Reset() override { next_value_ = 1; }
 
     size_t Channels() const override { return 1; }
 
@@ -524,7 +521,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
@@ -690,7 +687,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
@@ -829,9 +826,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
 
   class MockAudioDecoder : public AudioDecoder {
    public:
-    int Init() override {
-      return 0;
-    }
+    void Reset() override {}
     MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
     MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
                                      SpeechType*));
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e1a0f69dfa..139106bd3c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -33,7 +33,7 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
   virtual ~MockAudioDecoderOpus() { Die(); }
   MOCK_METHOD0(Die, void());
 
-  MOCK_METHOD0(Init, int());
+  MOCK_METHOD0(Reset, void());
 
   int PacketDuration(const uint8_t* encoded,
                      size_t encoded_len) const override {
@@ -271,7 +271,6 @@ struct NetEqNetworkStatsCheck {
 
 TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
   MockAudioDecoderOpus decoder(1);
-  EXPECT_CALL(decoder, Init());
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
@@ -279,7 +278,6 @@ TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
 
 TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
   MockAudioDecoderOpus decoder(2);
-  EXPECT_CALL(decoder, Init());
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
@@ -287,7 +285,6 @@ TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
   MockAudioDecoderOpus decoder(1);
-  EXPECT_CALL(decoder, Init());
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);