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 4959d57..961fd3f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -252,7 +252,7 @@
    */
 
   int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                               const uint16_t *encoded,
+                               const uint8_t* encoded,
                                int16_t len,
                                int16_t *decoded,
                                int16_t *speechType);
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 33bf373..1c1a720 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -805,7 +805,7 @@
 
 
 int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                             const uint16_t   *encoded,
+                             const uint8_t* encoded,
                              int16_t          len,
                              int16_t          *decoded,
                              int16_t     *speechType)
@@ -844,10 +844,13 @@
   /* convert bitstream from int16_t to bytes */
 #ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<(len>>1); k++) {
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    uint16_t ek = ((const uint16_t*)encoded)[k];
+    ISAC_inst->ISACdec_obj.bitstr_obj.stream[k] =
+        (uint16_t)((ek >> 8) | ((ek & 0xff) << 8));
   }
   if (len & 0x0001)
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] & 0xFF)<<8);
+    ISAC_inst->ISACdec_obj.bitstr_obj.stream[k] =
+        (uint16_t)((((const uint16_t*)encoded)[k] & 0xff) << 8);
 #else
   memcpy(ISAC_inst->ISACdec_obj.bitstr_obj.stream, encoded, len);
 #endif
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 207ee8c..d93fae3 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
@@ -86,7 +86,7 @@
   int16_t audio_type;
   clock_t clocks = clock();
   value = WebRtcIsacfix_Decode(ISACFIX_main_inst_,
-                               reinterpret_cast<const uint16_t*>(bit_stream),
+                               bit_stream,
                                encoded_bytes, out_data, &audio_type);
   clocks = clock() - clocks;
   EXPECT_EQ(output_length_sample_, value);
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 8e379c6..ba50b0c 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -746,8 +746,12 @@
           /* Call getFramelen, only used here for function test */
           err = WebRtcIsacfix_ReadFrameLen(
               reinterpret_cast<const uint8_t*>(streamdata), stream_len, &FL);
-          declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
-                                         decoded, speechType );
+          declen = WebRtcIsacfix_Decode(
+              ISAC_main_inst,
+              reinterpret_cast<const uint8_t*>(streamdata),
+              stream_len,
+              decoded,
+              speechType);
           /* Error check */
           if (err<0 || declen<0 || FL!=declen) {
             errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
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 81b257c..6d0c32d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -216,7 +216,7 @@
 
   int16_t WebRtcIsac_Decode(
       ISACStruct*           ISAC_main_inst,
-      const uint16_t* encoded,
+      const uint8_t* encoded,
       int16_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
@@ -703,7 +703,7 @@
    */
   int16_t WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
-      const uint16_t* encoded,
+      const uint8_t* encoded,
       int16_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
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 05bdf2a..d2260e2 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -1046,7 +1046,7 @@
 }
 
 static int16_t Decode(ISACStruct* ISAC_main_inst,
-                      const uint16_t* encoded,
+                      const uint8_t* encoded,
                       int16_t lenEncodedBytes,
                       int16_t* decoded,
                       int16_t* speechType,
@@ -1065,7 +1065,6 @@
   int16_t lenEncodedLBBytes;
   int16_t validChecksum = 1;
   int16_t k;
-  uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
   uint16_t numLayer;
   int16_t totSizeBytes;
   int16_t err;
@@ -1094,7 +1093,7 @@
       STREAM_SIZE_MAX : lenEncodedBytes;
 
   /* Copy to lower-band bit-stream structure. */
-  memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, ptrEncodedUW8,
+  memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, encoded,
          lenEncodedLBBytes);
 
   /* Regardless of that the current codec is setup to work in
@@ -1116,12 +1115,12 @@
   totSizeBytes = numDecodedBytesLB;
   while (totSizeBytes != lenEncodedBytes) {
     if ((totSizeBytes > lenEncodedBytes) ||
-        (ptrEncodedUW8[totSizeBytes] == 0) ||
+        (encoded[totSizeBytes] == 0) ||
         (numLayer > MAX_NUM_LAYERS)) {
       instISAC->errorCode = ISAC_LENGTH_MISMATCH;
       return -1;
     }
-    totSizeBytes += ptrEncodedUW8[totSizeBytes];
+    totSizeBytes += encoded[totSizeBytes];
     numLayer++;
   }
 
@@ -1160,7 +1159,7 @@
       instISAC->resetFlag_8kHz = 2;
     } else {
       /* This includes the checksum and the bytes that stores the length. */
-      int16_t lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
+      int16_t lenNextStream = encoded[numDecodedBytesLB];
 
       /* Is this garbage or valid super-wideband bit-stream?
        * Check if checksum is valid. */
@@ -1170,14 +1169,13 @@
         validChecksum = 0;
       } else {
         /* Run CRC to see if the checksum match. */
-        WebRtcIsac_GetCrc((int16_t*)(
-                            &ptrEncodedUW8[numDecodedBytesLB + 1]),
+        WebRtcIsac_GetCrc((int16_t*)(&encoded[numDecodedBytesLB + 1]),
                           lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
 
         validChecksum = 1;
         for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
           validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) ==
-                            ptrEncodedUW8[numDecodedBytesLB + lenNextStream -
+                            encoded[numDecodedBytesLB + lenNextStream -
                                           LEN_CHECK_SUM_WORD8 + k]);
         }
       }
@@ -1209,7 +1207,7 @@
         lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1);
 
         memcpy(decInstUB->bitstr_obj.stream,
-               &ptrEncodedUW8[numDecodedBytesLB + 1], lenNextStream);
+               &encoded[numDecodedBytesLB + 1], lenNextStream);
 
         /* Reset bit-stream object, this is the first decoding. */
         WebRtcIsac_ResetBitstream(&(decInstUB->bitstr_obj));
@@ -1283,7 +1281,7 @@
         /* It might be less due to garbage. */
         if ((numDecodedBytesUB != lenNextStream) &&
             (numDecodedBytesUB != (lenNextStream -
-                ptrEncodedUW8[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
+                encoded[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
           instISAC->errorCode = ISAC_LENGTH_MISMATCH;
           return -1;
         }
@@ -1346,7 +1344,7 @@
  */
 
 int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
-                          const uint16_t* encoded,
+                          const uint8_t* encoded,
                           int16_t lenEncodedBytes,
                           int16_t* decoded,
                           int16_t* speechType) {
@@ -1378,7 +1376,7 @@
 
 
 int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
-                             const uint16_t* encoded,
+                             const uint8_t* encoded,
                              int16_t lenEncodedBytes,
                              int16_t* decoded,
                              int16_t* speechType) {
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 e7e729e..b9fec4b 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
@@ -909,26 +909,39 @@
 
             if(lostFrame)
             {
-                declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, streamdata,
-                    stream_len, decoded, speechType);
+                declen = WebRtcIsac_DecodeRcu(
+                    ISAC_main_inst,
+                    reinterpret_cast<const uint8_t*>(streamdata),
+                    stream_len,
+                    decoded,
+                    speechType);
 
                 if(doTransCoding)
                 {
-                    declenTC = WebRtcIsac_DecodeRcu(decoderTransCoding,
-                        streamDataTransCoding, streamLenTransCoding,
-                        decodedTC, speechType);
+                    declenTC = WebRtcIsac_DecodeRcu(
+                        decoderTransCoding,
+                        reinterpret_cast<const uint8_t*>(streamDataTransCoding),
+                        streamLenTransCoding,
+                        decodedTC,
+                        speechType);
                 }
             }
             else
             {
-                declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata,
-                    stream_len, decoded, speechType);
-
+                declen = WebRtcIsac_Decode(
+                    ISAC_main_inst,
+                    reinterpret_cast<const uint8_t*>(streamdata),
+                    stream_len,
+                    decoded,
+                    speechType);
                 if(doTransCoding)
                 {
-                    declenTC = WebRtcIsac_Decode(decoderTransCoding,
-                        streamDataTransCoding, streamLenTransCoding,
-                        decodedTC, speechType);
+                    declenTC = WebRtcIsac_Decode(
+                        decoderTransCoding,
+                        reinterpret_cast<const uint8_t*>(streamDataTransCoding),
+                        streamLenTransCoding,
+                        decodedTC,
+                        speechType);
                 }
             }
 
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 88b0944..6ec818e 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
@@ -424,8 +424,11 @@
         /**/
         // Decode
         lenDecodedAudio = WebRtcIsac_Decode(
-            codecInstance[receiverIdx], bitStream, streamLen,
-            audioBuff60ms, speechType);
+            codecInstance[receiverIdx],
+            reinterpret_cast<const uint8_t*>(bitStream),
+            streamLen,
+            audioBuff60ms,
+            speechType);
         if(lenDecodedAudio < 0)
         {
           printf(" Decoder error in client %d \n", receiverIdx + 1);
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 e63e1e0..ce759a4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -458,16 +458,24 @@
 
 			if((rand() % 100) < packetLossPercent)
 			{
-				declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU,
-					rcuStreamLen, decoded, speechType);
+                                declen = WebRtcIsac_DecodeRcu(
+                                    ISAC_main_inst,
+                                    (const uint8_t*)payloadRCU,
+                                    rcuStreamLen,
+                                    decoded,
+                                    speechType);
 				lostPacketCntr++;
 			}
 			else
 			{
-				declen = WebRtcIsac_Decode(ISAC_main_inst, payload,
-					stream_len, decoded, speechType);
-			}
-			if(declen <= 0)
+                                declen = WebRtcIsac_Decode(
+                                    ISAC_main_inst,
+                                    (const uint8_t*)payload,
+                                    stream_len,
+                                    decoded,
+                                    speechType);
+                        }
+                        if(declen <= 0)
 			{
 				//errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
 				fprintf(stderr,"\nError in decoder.\n");
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index 89e52f4..6ee2682 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -730,7 +730,7 @@
   CriticalSectionScoped lock(codec_inst_crit_sect_.get());
   int ret =
       ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
-                        reinterpret_cast<const uint16_t*>(encoded),
+                        encoded,
                         static_cast<int16_t>(encoded_len),
                         decoded,
                         &temp_type);
@@ -769,7 +769,7 @@
   CriticalSectionScoped lock(codec_inst_crit_sect_.get());
   int16_t ret =
       ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
-                         reinterpret_cast<const uint16_t*>(encoded),
+                         encoded,
                          static_cast<int16_t>(encoded_len),
                          decoded,
                          &temp_type);
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 028320a..661f2b1 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -165,7 +165,7 @@
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
   int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
-                                  reinterpret_cast<const uint16_t*>(encoded),
+                                  encoded,
                                   static_cast<int16_t>(encoded_len), decoded,
                                   &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -177,7 +177,7 @@
                                       SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
   int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
-                                     reinterpret_cast<const uint16_t*>(encoded),
+                                     encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -236,7 +236,7 @@
                                 int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
   int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
-                                     reinterpret_cast<const uint16_t*>(encoded),
+                                     encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   *speech_type = ConvertSpeechType(temp_type);