diff --git a/src/modules/audio_coding/codecs/G711/main/source/g711_interface.c b/src/modules/audio_coding/codecs/G711/main/source/g711_interface.c
index 5acd431642..73646d16d0 100644
--- a/src/modules/audio_coding/codecs/G711/main/source/g711_interface.c
+++ b/src/modules/audio_coding/codecs/G711/main/source/g711_interface.c
@@ -21,7 +21,7 @@ WebRtc_Word16 WebRtcG711_EncodeA(void *state,
     WebRtc_UWord16 tempVal,  tempVal2;
 
     // Set to avoid getting warnings
-    state = state;
+    state = NULL;
 
     // Sanity check of input length
     if (len < 0) {
@@ -51,7 +51,7 @@ WebRtc_Word16 WebRtcG711_EncodeA(void *state,
     return (len);
 }
 
-WebRtc_Word16 WebRtcG711_EncodeU(void *state,
+WebRtc_Word16 WebRtcG711_EncodeU(void  *state,
                                  WebRtc_Word16 *speechIn,
                                  WebRtc_Word16 len,
                                  WebRtc_Word16 *encoded)
@@ -60,7 +60,7 @@ WebRtc_Word16 WebRtcG711_EncodeU(void *state,
     WebRtc_UWord16 tempVal;
 
     // Set to avoid getting warnings
-    state = state;
+    state = NULL;
 
     // Sanity check of input length
     if (len < 0) {
@@ -98,7 +98,7 @@ WebRtc_Word16 WebRtcG711_DecodeA(void *state,
     WebRtc_UWord16 tempVal;
 
     // Set to avoid getting warnings
-    state = state;
+    state = NULL;
 
     // Sanity check of input length
     if (len < 0) {
@@ -136,7 +136,7 @@ WebRtc_Word16 WebRtcG711_DecodeU(void *state,
     WebRtc_UWord16 tempVal;
 
     // Set to avoid getting warnings
-    state = state;
+    state = NULL;
 
     // Sanity check of input length
     if (len < 0) {
diff --git a/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c b/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
index c188ba1b30..b3e7dc819a 100644
--- a/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
+++ b/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
@@ -261,7 +261,7 @@ int WebRtcIsacfix_EncodeImpl(WebRtc_Word16      *in,
     // then the limit at this point should be half of the assigned value
     payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1;
   }
-  else if ((frame_mode == 0))
+  else if (frame_mode == 0)
   {
     // it is a 30ms frame
     payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3;
@@ -492,7 +492,6 @@ int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t  *ISACenc_obj,
   int status;
   WebRtc_Word16 BWno = BWnumber;
   int stream_length = 0;
-  int usefulstr_len;
 
   WebRtc_Word16 model;
   const WebRtc_UWord16 *Q_PitchGain_cdf_ptr[1];
diff --git a/src/modules/audio_coding/codecs/iSAC/main/source/isac.c b/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
index 3e364bcf53..f793cfd1e7 100644
--- a/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
+++ b/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
@@ -1104,7 +1104,7 @@ WebRtc_Word16 WebRtcIsac_DecoderInit(
 	}
     }
 
-  if((instISAC->initFlag && BIT_MASK_ENC_INIT) !=
+  if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
      BIT_MASK_ENC_INIT)
     {
       WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,