diff --git a/src/modules/audio_coding/neteq/codec_db.c b/src/modules/audio_coding/neteq/codec_db.c
index e91e37297e..5369cfd271 100644
--- a/src/modules/audio_coding/neteq/codec_db.c
+++ b/src/modules/audio_coding/neteq/codec_db.c
@@ -726,12 +726,18 @@ WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType)
         {
             switch (i)
             {
+#ifdef NETEQ_WIDEBAND
                 case 1:
                     return 16000;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
                 case 2:
                     return 32000;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
                 case 3:
                     return 48000;
+#endif
                 default:
                     return 8000;
             }
diff --git a/src/modules/audio_coding/neteq/dtmf_tonegen.c b/src/modules/audio_coding/neteq/dtmf_tonegen.c
index a52f9bc635..c9ad232173 100644
--- a/src/modules/audio_coding/neteq/dtmf_tonegen.c
+++ b/src/modules/audio_coding/neteq/dtmf_tonegen.c
@@ -173,8 +173,8 @@ WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, WebRtc_Wor
     WebRtc_Word16 a2 = 0; /* a-coefficient for second tone (high tone) */
     int i;
     int frameLen; /* number of samples to generate */
-    int lowIndex;
-    int highIndex;
+    int lowIndex = 0;  /* Default to avoid compiler warnings. */
+    int highIndex = 4;  /* Default to avoid compiler warnings. */
     WebRtc_Word32 tempVal;
     WebRtc_Word16 tempValLow;
     WebRtc_Word16 tempValHigh;
@@ -309,10 +309,6 @@ WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, WebRtc_Wor
             highIndex = 7;/* high frequency: 1633 Hz */
             break;
         }
-        default:
-        {
-            return DTMF_DEC_PARAMETER_ERROR;
-        }
     } /* end switch */
 
     /* select coefficients based on results from switches above */
diff --git a/src/modules/audio_coding/neteq/rtp.c b/src/modules/audio_coding/neteq/rtp.c
index bd4f9a2260..63cdf652fc 100644
--- a/src/modules/audio_coding/neteq/rtp.c
+++ b/src/modules/audio_coding/neteq/rtp.c
@@ -138,7 +138,7 @@ int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
     else
     {
         /* Discard all but the two last payloads. */
-        while (((pw16_data[2] & 0x8000) == 1)&&
+        while (((pw16_data[2] & 0x8000) != 0) &&
             (pw16_data<((RTPheader[0]->payload)+((RTPheader[0]->payloadLen+1)>>1))))
         {
             i_discardedBlockLength += (4+(((WebRtc_UWord16)pw16_data[1]) & 0x3FF));
@@ -167,7 +167,7 @@ int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
     else
     {
         /* Discard all but the two last payloads. */
-        while (((pw16_data[2] & 0x80) == 1) && (pw16_data < ((RTPheader[0]->payload)
+        while (((pw16_data[2] & 0x80) != 0) && (pw16_data < ((RTPheader[0]->payload)
             + ((RTPheader[0]->payloadLen + 1) >> 1))))
         {
             i_discardedBlockLength += (4 + ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
diff --git a/src/modules/audio_coding/neteq/webrtc_neteq.c b/src/modules/audio_coding/neteq/webrtc_neteq.c
index 776f61f096..47355c041d 100644
--- a/src/modules/audio_coding/neteq/webrtc_neteq.c
+++ b/src/modules/audio_coding/neteq/webrtc_neteq.c
@@ -345,12 +345,7 @@ int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecod
     }
     *MaxNoOfPackets = (*MaxNoOfPackets) * multiplier;
     *sizeinbytes = (*sizeinbytes) * multiplier;
-    if (ok != 0)
-    {
-        NetEqMainInst->ErrorCode = -ok;
-        return (-1);
-    }
-    return (ok);
+    return 0;
 }
 
 int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr,