diff --git a/modules/audio_coding/codecs/g722/g722_decode.c b/modules/audio_coding/codecs/g722/g722_decode.c
index 499cc8f..e62af98 100644
--- a/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/modules/audio_coding/codecs/g722/g722_decode.c
@@ -49,12 +49,12 @@
 #define TRUE (!FALSE)
 #endif
 
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+static __inline int16_t saturate(int32_t amp)
 {
-    WebRtc_Word16 amp16;
+    int16_t amp16;
 
     /* Hopefully this is optimised for the common case - not clipping */
-    amp16 = (WebRtc_Word16) amp;
+    amp16 = (int16_t) amp;
     if (amp == amp16)
         return amp16;
     if (amp > WEBRTC_INT16_MAX)
@@ -190,8 +190,8 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
-                       const WebRtc_UWord8 g722_data[], int len)
+int WebRtc_g722_decode(g722_decode_state_t *s, int16_t amp[],
+                       const uint8_t g722_data[], int len)
 {
     static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
     static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
@@ -372,14 +372,14 @@
 
         if (s->itu_test_mode)
         {
-            amp[outlen++] = (WebRtc_Word16) (rlow << 1);
-            amp[outlen++] = (WebRtc_Word16) (rhigh << 1);
+            amp[outlen++] = (int16_t) (rlow << 1);
+            amp[outlen++] = (int16_t) (rhigh << 1);
         }
         else
         {
             if (s->eight_k)
             {
-                amp[outlen++] = (WebRtc_Word16) (rlow << 1);
+                amp[outlen++] = (int16_t) (rlow << 1);
             }
             else
             {
diff --git a/modules/audio_coding/codecs/g722/g722_enc_dec.h b/modules/audio_coding/codecs/g722/g722_enc_dec.h
index d2d19b0..ef279ac 100644
--- a/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ b/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -138,8 +138,8 @@
                                              int options);
 int WebRtc_g722_encode_release(g722_encode_state_t *s);
 int WebRtc_g722_encode(g722_encode_state_t *s,
-                       WebRtc_UWord8 g722_data[],
-                       const WebRtc_Word16 amp[],
+                       uint8_t g722_data[],
+                       const int16_t amp[],
                        int len);
 
 g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
@@ -147,8 +147,8 @@
                                              int options);
 int WebRtc_g722_decode_release(g722_decode_state_t *s);
 int WebRtc_g722_decode(g722_decode_state_t *s,
-                       WebRtc_Word16 amp[],
-                       const WebRtc_UWord8 g722_data[],
+                       int16_t amp[],
+                       const uint8_t g722_data[],
                        int len);
 
 #ifdef __cplusplus
diff --git a/modules/audio_coding/codecs/g722/g722_encode.c b/modules/audio_coding/codecs/g722/g722_encode.c
index 7487b64..5b07615 100644
--- a/modules/audio_coding/codecs/g722/g722_encode.c
+++ b/modules/audio_coding/codecs/g722/g722_encode.c
@@ -48,12 +48,12 @@
 #define TRUE (!FALSE)
 #endif
 
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+static __inline int16_t saturate(int32_t amp)
 {
-    WebRtc_Word16 amp16;
+    int16_t amp16;
 
     /* Hopefully this is optimised for the common case - not clipping */
-    amp16 = (WebRtc_Word16) amp;
+    amp16 = (int16_t) amp;
     if (amp == amp16)
         return amp16;
     if (amp > WEBRTC_INT16_MAX)
@@ -191,10 +191,10 @@
  */
 //#define RUN_LIKE_REFERENCE_G722
 #ifdef RUN_LIKE_REFERENCE_G722
-WebRtc_Word16 limitValues (WebRtc_Word16 rl)
+int16_t limitValues (int16_t rl)
 {
 
-    WebRtc_Word16 yl;
+    int16_t yl;
 
     yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
 
@@ -202,8 +202,8 @@
 }
 #endif
 
-int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
-                       const WebRtc_Word16 amp[], int len)
+int WebRtc_g722_encode(g722_encode_state_t *s, uint8_t g722_data[],
+                       const int16_t amp[], int len)
 {
     static const int q6[32] =
     {
@@ -418,14 +418,14 @@
             s->out_bits += s->bits_per_sample;
             if (s->out_bits >= 8)
             {
-                g722_data[g722_bytes++] = (WebRtc_UWord8) (s->out_buffer & 0xFF);
+                g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
                 s->out_bits -= 8;
                 s->out_buffer >>= 8;
             }
         }
         else
         {
-            g722_data[g722_bytes++] = (WebRtc_UWord8) code;
+            g722_data[g722_bytes++] = (uint8_t) code;
         }
     }
     return g722_bytes;
diff --git a/modules/audio_coding/codecs/g722/g722_interface.c b/modules/audio_coding/codecs/g722/g722_interface.c
index d559014..7075669 100644
--- a/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/modules/audio_coding/codecs/g722/g722_interface.c
@@ -17,7 +17,7 @@
 #include "typedefs.h"
 
 
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
 {
     *G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
     if (*G722enc_inst!=NULL) {
@@ -27,7 +27,7 @@
     }
 }
 
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
 {
     // Create and/or reset the G.722 encoder
     // Bitrate 64 kbps and wideband mode (2)
@@ -40,16 +40,16 @@
     }
 }
 
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
 {
     // Free encoder memory
     return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
 }
 
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
-                                WebRtc_Word16 *speechIn,
-                                WebRtc_Word16 len,
-                                WebRtc_Word16 *encoded)
+int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                          int16_t *speechIn,
+                          int16_t len,
+                          int16_t *encoded)
 {
     unsigned char *codechar = (unsigned char*) encoded;
     // Encode the input speech vector
@@ -57,7 +57,7 @@
                        codechar, speechIn, len);
 }
 
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
 {
     *G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
     if (*G722dec_inst!=NULL) {
@@ -67,7 +67,7 @@
     }
 }
 
-WebRtc_Word16 WebRtcG722_DecoderInit(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)
@@ -80,25 +80,25 @@
     }
 }
 
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
 {
     // Free encoder memory
     return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
 }
 
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
-                                WebRtc_Word16 *encoded,
-                                WebRtc_Word16 len,
-                                WebRtc_Word16 *decoded,
-                                WebRtc_Word16 *speechType)
+int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                          int16_t *encoded,
+                          int16_t len,
+                          int16_t *decoded,
+                          int16_t *speechType)
 {
     // Decode the G.722 encoder stream
     *speechType=G722_WEBRTC_SPEECH;
     return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
-                              decoded, (WebRtc_UWord8*) encoded, len);
+                              decoded, (uint8_t*) encoded, len);
 }
 
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len)
+int16_t WebRtcG722_Version(char *versionStr, short len)
 {
     // Get version string
     char version[30] = "2.0.0\n";
diff --git a/modules/audio_coding/codecs/g722/include/g722_interface.h b/modules/audio_coding/codecs/g722/include/g722_interface.h
index e50d66f..0948a18 100644
--- a/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -43,7 +43,7 @@
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
 
 
 /****************************************************************************
@@ -59,7 +59,7 @@
  *                              -1 - Error
  */
 
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
 
 
 /****************************************************************************
@@ -73,7 +73,7 @@
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
 
 
 
@@ -95,10 +95,10 @@
  *                              -1 - Error
  */
 
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
-                                WebRtc_Word16 *speechIn,
-                                WebRtc_Word16 len,
-                                WebRtc_Word16 *encoded);
+int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                          int16_t *speechIn,
+                          int16_t len,
+                          int16_t *encoded);
 
 
 /****************************************************************************
@@ -112,7 +112,7 @@
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
 
 
 /****************************************************************************
@@ -128,7 +128,7 @@
  *                           -1 - Error
  */
 
-WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
+int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
 
 
 /****************************************************************************
@@ -143,7 +143,7 @@
  *                              -1 - Error
  */
 
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
 
 
 /****************************************************************************
@@ -167,11 +167,11 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
-                                WebRtc_Word16 *encoded,
-                                WebRtc_Word16 len,
-                                WebRtc_Word16 *decoded,
-                                WebRtc_Word16 *speechType);
+int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                          int16_t *encoded,
+                          int16_t len,
+                          int16_t *decoded,
+                          int16_t *speechType);
 
 /****************************************************************************
  * WebRtcG722_Version(...)
@@ -179,7 +179,7 @@
  * Get a string with the current version of the codec
  */
 
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len);
+int16_t WebRtcG722_Version(char *versionStr, short len);
 
 
 #ifdef __cplusplus
diff --git a/modules/audio_coding/codecs/g722/test/testG722.cc b/modules/audio_coding/codecs/g722/test/testG722.cc
index d2fdca3..d51301d 100644
--- a/modules/audio_coding/codecs/g722/test/testG722.cc
+++ b/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -29,11 +29,11 @@
 typedef struct WebRtcG722DecInst    G722DecInst;
 
 /* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length)
+int readframe(int16_t *data, FILE *inp, int length)
 {
     short k, rlen, status = 0;
 
-    rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
+    rlen = (short)fread(data, sizeof(int16_t), length, inp);
     if (rlen < length) {
         for (k = rlen; k < length; k++)
             data[k] = 0;
@@ -49,7 +49,7 @@
     FILE *inp, *outbitp, *outp;
 
     int framecnt, endfile;
-    WebRtc_Word16 framelength = 160;
+    int16_t framelength = 160;
     G722EncInst *G722enc_inst;
     G722DecInst *G722dec_inst;
     int err;
@@ -59,11 +59,11 @@
     double runtime = 0;
     double length_file;
 
-    WebRtc_Word16 stream_len = 0;
-    WebRtc_Word16 shortdata[960];
-    WebRtc_Word16 decoded[960];
-    WebRtc_Word16 streamdata[80*3];
-    WebRtc_Word16 speechType[1];
+    int16_t stream_len = 0;
+    int16_t shortdata[960];
+    int16_t decoded[960];
+    int16_t streamdata[80*3];
+    int16_t speechType[1];
 
     /* handling wrong input arguments in the command line */
     if (argc!=5)  {
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant.c b/modules/audio_coding/codecs/ilbc/abs_quant.c
index 4a70c8b..8a8b133 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant.c
+++ b/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -32,16 +32,16 @@
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
                                    and idxVec, uses state_first as
                                    input) */
-    WebRtc_Word16 *in,     /* (i) vector to encode */
-    WebRtc_Word16 *weightDenum   /* (i) denominator of synthesis filter */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
                             ) {
-  WebRtc_Word16 *syntOut;
-  WebRtc_Word16 quantLen[2];
+  int16_t *syntOut;
+  int16_t quantLen[2];
 
   /* Stack based */
-  WebRtc_Word16 syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
-  WebRtc_Word16 in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
-  WebRtc_Word16 *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+  int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+  int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER];
 
   /* Initialize the buffers */
   WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant.h b/modules/audio_coding/codecs/ilbc/abs_quant.h
index fa59593..c061ff4 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant.h
+++ b/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -32,8 +32,8 @@
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
                                    and idxVec, uses state_first as
                                    input) */
-    WebRtc_Word16 *in,     /* (i) vector to encode */
-    WebRtc_Word16 *weightDenum   /* (i) denominator of synthesis filter */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
index 4eebc3e..1a18a1d 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
+++ b/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -20,26 +20,21 @@
 #include "constants.h"
 #include "sort_sq.h"
 
-void WebRtcIlbcfix_AbsQuantLoop(
-    WebRtc_Word16 *syntOutIN,
-    WebRtc_Word16 *in_weightedIN,
-    WebRtc_Word16 *weightDenumIN,
-    WebRtc_Word16 *quantLenIN,
-    WebRtc_Word16 *idxVecIN
-                                )
-{
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, int16_t *quantLenIN,
+                                int16_t *idxVecIN ) {
   int n, k1, k2;
-  WebRtc_Word16 index;
-  WebRtc_Word32 toQW32;
-  WebRtc_Word32 toQ32;
-  WebRtc_Word16 tmp16a;
-  WebRtc_Word16 xq;
+  int16_t index;
+  int32_t toQW32;
+  int32_t toQ32;
+  int16_t tmp16a;
+  int16_t xq;
 
-  WebRtc_Word16 *syntOut   = syntOutIN;
-  WebRtc_Word16 *in_weighted  = in_weightedIN;
-  WebRtc_Word16 *weightDenum  = weightDenumIN;
-  WebRtc_Word16 *quantLen  = quantLenIN;
-  WebRtc_Word16 *idxVec   = idxVecIN;
+  int16_t *syntOut   = syntOutIN;
+  int16_t *in_weighted  = in_weightedIN;
+  int16_t *weightDenum  = weightDenumIN;
+  int16_t *quantLen  = quantLenIN;
+  int16_t *idxVec   = idxVecIN;
 
   n=0;
 
@@ -52,14 +47,14 @@
           weightDenum, LPC_FILTERORDER+1, 1);
 
       /* the quantizer */
-      toQW32 = (WebRtc_Word32)(*in_weighted) - (WebRtc_Word32)(*syntOut);
+      toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut);
 
-      toQ32 = (((WebRtc_Word32)toQW32)<<2);
+      toQ32 = (((int32_t)toQW32)<<2);
 
       if (toQ32 > 32767) {
-        toQ32 = (WebRtc_Word32) 32767;
+        toQ32 = (int32_t) 32767;
       } else if (toQ32 < -32768) {
-        toQ32 = (WebRtc_Word32) -32768;
+        toQ32 = (int32_t) -32768;
       }
 
       /* Quantize the state */
@@ -74,7 +69,7 @@
            (state_sq3Tbl is in Q13 and toQ is in Q11)
         */
         WebRtcIlbcfix_SortSq(&xq, &index,
-                             (WebRtc_Word16)toQ32,
+                             (int16_t)toQ32,
                              WebRtcIlbcfix_kStateSq3, 8);
       }
 
@@ -84,7 +79,7 @@
       /* Compute decoded sample and update of the prediction filter */
       tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
 
-      *syntOut     = (WebRtc_Word16) (tmp16a + (WebRtc_Word32)(*in_weighted) - toQW32);
+      *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
 
       n++;
       syntOut++; in_weighted++;
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
index f506e8e..50c6ffe 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
+++ b/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -26,12 +26,8 @@
  *  (subrutine for WebRtcIlbcfix_StateSearch)
  *---------------------------------------------------------------*/
 
-void WebRtcIlbcfix_AbsQuantLoop(
-    WebRtc_Word16 *syntOutIN,
-    WebRtc_Word16 *in_weightedIN,
-    WebRtc_Word16 *weightDenumIN,
-    WebRtc_Word16 *quantLenIN,
-    WebRtc_Word16 *idxVecIN
-                                );
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, int16_t *quantLenIN,
+                                int16_t *idxVecIN);
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
index 6011e92..d8f8c93 100644
--- a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
+++ b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -21,24 +21,24 @@
 #include "augmented_cb_corr.h"
 
 void WebRtcIlbcfix_AugmentedCbCorr(
-    WebRtc_Word16 *target,   /* (i) Target vector */
-    WebRtc_Word16 *buffer,   /* (i) Memory buffer */
-    WebRtc_Word16 *interpSamples, /* (i) buffer with
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
                                      interpolated samples */
-    WebRtc_Word32 *crossDot,  /* (o) The cross correlation between
+    int32_t *crossDot,  /* (o) The cross correlation between
                                  the target and the Augmented
                                  vector */
-    WebRtc_Word16 low,    /* (i) Lag to start from (typically
+    int16_t low,    /* (i) Lag to start from (typically
                              20) */
-    WebRtc_Word16 high,   /* (i) Lag to end at (typically 39) */
-    WebRtc_Word16 scale)   /* (i) Scale factor to use for
+    int16_t high,   /* (i) Lag to end at (typically 39) */
+    int16_t scale)   /* (i) Scale factor to use for
                               the crossDot */
 {
   int lagcount;
-  WebRtc_Word16 ilow;
-  WebRtc_Word16 *targetPtr;
-  WebRtc_Word32 *crossDotPtr;
-  WebRtc_Word16 *iSPtr=interpSamples;
+  int16_t ilow;
+  int16_t *targetPtr;
+  int32_t *crossDotPtr;
+  int16_t *iSPtr=interpSamples;
 
   /* Calculate the correlation between the target and the
      interpolated codebook. The correlation is calculated in
@@ -46,7 +46,7 @@
   crossDotPtr=crossDot;
   for (lagcount=low; lagcount<=high; lagcount++) {
 
-    ilow = (WebRtc_Word16) (lagcount-4);
+    ilow = (int16_t) (lagcount-4);
 
     /* Compute dot product for the first (lagcount-4) samples */
     (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
diff --git a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
index 8e097fe..533d0a4 100644
--- a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -26,17 +26,17 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_AugmentedCbCorr(
-    WebRtc_Word16 *target,   /* (i) Target vector */
-    WebRtc_Word16 *buffer,   /* (i) Memory buffer */
-    WebRtc_Word16 *interpSamples, /* (i) buffer with
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
                                            interpolated samples */
-    WebRtc_Word32 *crossDot,  /* (o) The cross correlation between
+    int32_t *crossDot,  /* (o) The cross correlation between
                                            the target and the Augmented
                                            vector */
-    WebRtc_Word16 low,    /* (i) Lag to start from (typically
+    int16_t low,    /* (i) Lag to start from (typically
                                                    20) */
-    WebRtc_Word16 high,   /* (i) Lag to end at (typically 39 */
-    WebRtc_Word16 scale);   /* (i) Scale factor to use for
+    int16_t high,   /* (i) Lag to end at (typically 39 */
+    int16_t scale);   /* (i) Scale factor to use for
                                                    the crossDot */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/bw_expand.c b/modules/audio_coding/codecs/ilbc/bw_expand.c
index a2287aa..4c29bb1 100644
--- a/modules/audio_coding/codecs/ilbc/bw_expand.c
+++ b/modules/audio_coding/codecs/ilbc/bw_expand.c
@@ -24,11 +24,11 @@
 
 /* The output is in the same domain as the input */
 void WebRtcIlbcfix_BwExpand(
-    WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
-    WebRtc_Word16 *in,  /* (i) the lpc coefficients before bandwidth
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
                                    expansion */
-    WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
-    WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
                             ) {
   int i;
 
@@ -37,6 +37,6 @@
     /* out[i] = coef[i] * in[i] with rounding.
        in[] and out[] are in Q12 and coef[] is in Q15
     */
-    out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(coef[i], in[i])+16384)>>15);
+    out[i] = (int16_t)((WEBRTC_SPL_MUL_16_16(coef[i], in[i])+16384)>>15);
   }
 }
diff --git a/modules/audio_coding/codecs/ilbc/bw_expand.h b/modules/audio_coding/codecs/ilbc/bw_expand.h
index c9f3fab..b3b16d5 100644
--- a/modules/audio_coding/codecs/ilbc/bw_expand.h
+++ b/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_BwExpand(
-    WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
-    WebRtc_Word16 *in,  /* (i) the lpc coefficients before bandwidth
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
                                    expansion */
-    WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
-    WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_construct.c b/modules/audio_coding/codecs/ilbc/cb_construct.c
index 094a7e4..808451f 100644
--- a/modules/audio_coding/codecs/ilbc/cb_construct.c
+++ b/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -25,21 +25,21 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CbConstruct(
-    WebRtc_Word16 *decvector,  /* (o) Decoded vector */
-    WebRtc_Word16 *index,   /* (i) Codebook indices */
-    WebRtc_Word16 *gain_index,  /* (i) Gain quantization indices */
-    WebRtc_Word16 *mem,   /* (i) Buffer for codevector construction */
-    WebRtc_Word16 lMem,   /* (i) Length of buffer */
-    WebRtc_Word16 veclen   /* (i) Length of vector */
+    int16_t *decvector,  /* (o) Decoded vector */
+    int16_t *index,   /* (i) Codebook indices */
+    int16_t *gain_index,  /* (i) Gain quantization indices */
+    int16_t *mem,   /* (i) Buffer for codevector construction */
+    int16_t lMem,   /* (i) Length of buffer */
+    int16_t veclen   /* (i) Length of vector */
                                ){
   int j;
-  WebRtc_Word16 gain[CB_NSTAGES];
+  int16_t gain[CB_NSTAGES];
   /* Stack based */
-  WebRtc_Word16 cbvec0[SUBL];
-  WebRtc_Word16 cbvec1[SUBL];
-  WebRtc_Word16 cbvec2[SUBL];
-  WebRtc_Word32 a32;
-  WebRtc_Word16 *gainPtr;
+  int16_t cbvec0[SUBL];
+  int16_t cbvec1[SUBL];
+  int16_t cbvec2[SUBL];
+  int32_t a32;
+  int16_t *gainPtr;
 
   /* gain de-quantization */
 
@@ -60,7 +60,7 @@
     a32 += WEBRTC_SPL_MUL_16_16(*gainPtr++, cbvec1[j]);
     a32 += WEBRTC_SPL_MUL_16_16(*gainPtr, cbvec2[j]);
     gainPtr -= 2;
-    decvector[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(a32 + 8192, 14);
+    decvector[j] = (int16_t) WEBRTC_SPL_RSHIFT_W32(a32 + 8192, 14);
   }
 
   return;
diff --git a/modules/audio_coding/codecs/ilbc/cb_construct.h b/modules/audio_coding/codecs/ilbc/cb_construct.h
index bec759f..2e9080f 100644
--- a/modules/audio_coding/codecs/ilbc/cb_construct.h
+++ b/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -26,12 +26,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CbConstruct(
-    WebRtc_Word16 *decvector,  /* (o) Decoded vector */
-    WebRtc_Word16 *index,   /* (i) Codebook indices */
-    WebRtc_Word16 *gain_index,  /* (i) Gain quantization indices */
-    WebRtc_Word16 *mem,   /* (i) Buffer for codevector construction */
-    WebRtc_Word16 lMem,   /* (i) Length of buffer */
-    WebRtc_Word16 veclen   /* (i) Length of vector */
+    int16_t *decvector,  /* (o) Decoded vector */
+    int16_t *index,   /* (i) Codebook indices */
+    int16_t *gain_index,  /* (i) Gain quantization indices */
+    int16_t *mem,   /* (i) Buffer for codevector construction */
+    int16_t lMem,   /* (i) Length of buffer */
+    int16_t veclen   /* (i) Length of vector */
                                );
 
 
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
index 8613fa2..f883287 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -27,18 +27,18 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CbMemEnergy(
-    WebRtc_Word16 range,
-    WebRtc_Word16 *CB,   /* (i) The CB memory (1:st section) */
-    WebRtc_Word16 *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
-    WebRtc_Word16 lMem,   /* (i) Length of the CB memory */
-    WebRtc_Word16 lTarget,   /* (i) Length of the target vector */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size  /* (i) Index to where the energy values should be stored */
+    int16_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    int16_t lMem,   /* (i) Length of the CB memory */
+    int16_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size  /* (i) Index to where the energy values should be stored */
                                ) {
-  WebRtc_Word16 *ppi, *ppo, *pp;
-  WebRtc_Word32 energy, tmp32;
+  int16_t *ppi, *ppo, *pp;
+  int32_t energy, tmp32;
 
   /* Compute the energy and store it in a vector. Also the
    * corresponding shift values are stored. The energy values
@@ -52,9 +52,9 @@
   energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
 
   /* Normalize the energy and store the number of shifts */
-  energyShifts[0] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+  energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy);
   tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[0]);
-  energyW16[0] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+  energyW16[0] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
 
   /* Compute the energy of the rest of the cb memory
    * by step wise adding and subtracting the next
@@ -68,9 +68,9 @@
   energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
 
   /* Normalize the energy and store the number of shifts */
-  energyShifts[base_size] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+  energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy);
   tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[base_size]);
-  energyW16[base_size] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+  energyW16[base_size] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
 
   ppi = filteredCB + lMem - 1 - lTarget;
   ppo = filteredCB + lMem - 1;
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 1aa2b7b..1b50c0b 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -20,15 +20,15 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
 
 void WebRtcIlbcfix_CbMemEnergy(
-    WebRtc_Word16 range,
-    WebRtc_Word16 *CB,   /* (i) The CB memory (1:st section) */
-    WebRtc_Word16 *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
-    WebRtc_Word16 lMem,   /* (i) Length of the CB memory */
-    WebRtc_Word16 lTarget,   /* (i) Length of the target vector */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size  /* (i) Index to where the energy values should be stored */
+    int16_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    int16_t lMem,   /* (i) Length of the CB memory */
+    int16_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size  /* (i) Index to where the energy values should be stored */
                                );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
index 0c6f479..29f499f 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -20,19 +20,19 @@
 #include "constants.h"
 
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
-    WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
-    WebRtc_Word16 *CBmem,   /* (i) The CB memory */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size,  /* (i) Index to where the energy values should be stored */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
+    int16_t *interpSamples, /* (i) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size,  /* (i) Index to where the energy values should be stored */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts /* (o) Shift value of the energy */
                                            ){
-  WebRtc_Word32 energy, tmp32;
-  WebRtc_Word16 *ppe, *pp, *interpSamplesPtr;
-  WebRtc_Word16 *CBmemPtr, lagcount;
-  WebRtc_Word16 *enPtr=&energyW16[base_size-20];
-  WebRtc_Word16 *enShPtr=&energyShifts[base_size-20];
-  WebRtc_Word32 nrjRecursive;
+  int32_t energy, tmp32;
+  int16_t *ppe, *pp, *interpSamplesPtr;
+  int16_t *CBmemPtr, lagcount;
+  int16_t *enPtr=&energyW16[base_size-20];
+  int16_t *enShPtr=&energyShifts[base_size-20];
+  int32_t nrjRecursive;
 
   CBmemPtr = CBmem+147;
   interpSamplesPtr = interpSamples;
@@ -58,9 +58,9 @@
     energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
 
     /* Normalize the energy and store the number of shifts */
-    (*enShPtr) = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+    (*enShPtr) = (int16_t)WebRtcSpl_NormW32(energy);
     tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, (*enShPtr));
-    (*enPtr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+    (*enPtr) = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
     enShPtr++;
     enPtr++;
   }
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
index 938b87e..6c181bd 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -20,12 +20,12 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
 
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
-    WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
-    WebRtc_Word16 *CBmem,   /* (i) The CB memory */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size,  /* (i) Index to where the energy values should be stored */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
+    int16_t *interpSamples, /* (i) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size,  /* (i) Index to where the energy values should be stored */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts /* (o) Shift value of the energy */
                                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
index 40bb708..a2bc9b8 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -22,20 +22,20 @@
  * by step wise adding and subtracting the next
  * sample and the last sample respectively */
 void WebRtcIlbcfix_CbMemEnergyCalc(
-    WebRtc_Word32 energy,   /* (i) input start energy */
-    WebRtc_Word16 range,   /* (i) number of iterations */
-    WebRtc_Word16 *ppi,   /* (i) input pointer 1 */
-    WebRtc_Word16 *ppo,   /* (i) input pointer 2 */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size  /* (i) Index to where the energy values should be stored */
+    int32_t energy,   /* (i) input start energy */
+    int16_t range,   /* (i) number of iterations */
+    int16_t *ppi,   /* (i) input pointer 1 */
+    int16_t *ppo,   /* (i) input pointer 2 */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size  /* (i) Index to where the energy values should be stored */
                                    )
 {
-  WebRtc_Word16 j,shft;
-  WebRtc_Word32 tmp;
-  WebRtc_Word16 *eSh_ptr;
-  WebRtc_Word16 *eW16_ptr;
+  int16_t j,shft;
+  int32_t tmp;
+  int16_t *eSh_ptr;
+  int16_t *eW16_ptr;
 
 
   eSh_ptr  = &energyShifts[1+base_size];
@@ -53,13 +53,13 @@
     ppi--;
     ppo--;
 
-    /* Normalize the energy into a WebRtc_Word16 and store
+    /* Normalize the energy into a int16_t and store
        the number of shifts */
 
-    shft = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+    shft = (int16_t)WebRtcSpl_NormW32(energy);
     *eSh_ptr++ = shft;
 
     tmp = WEBRTC_SPL_LSHIFT_W32(energy, shft);
-    *eW16_ptr++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp, 16);
+    *eW16_ptr++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp, 16);
   }
 }
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
index ee2e285..c7e1e54 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -20,14 +20,14 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
 
 void WebRtcIlbcfix_CbMemEnergyCalc(
-    WebRtc_Word32 energy,   /* (i) input start energy */
-    WebRtc_Word16 range,   /* (i) number of iterations */
-    WebRtc_Word16 *ppi,   /* (i) input pointer 1 */
-    WebRtc_Word16 *ppo,   /* (i) input pointer 2 */
-    WebRtc_Word16 *energyW16,  /* (o) Energy in the CB vectors */
-    WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
-    WebRtc_Word16 scale,   /* (i) The scaling of all energy values */
-    WebRtc_Word16 base_size  /* (i) Index to where the energy values should be stored */
+    int32_t energy,   /* (i) input start energy */
+    int16_t range,   /* (i) number of iterations */
+    int16_t *ppi,   /* (i) input pointer 1 */
+    int16_t *ppo,   /* (i) input pointer 2 */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int16_t scale,   /* (i) The scaling of all energy values */
+    int16_t base_size  /* (i) Index to where the energy values should be stored */
                                    );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_search.c b/modules/audio_coding/codecs/ilbc/cb_search.c
index 551a9a2..667240a 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -36,51 +36,51 @@
 void WebRtcIlbcfix_CbSearch(
     iLBC_Enc_Inst_t *iLBCenc_inst,
     /* (i) the encoder state structure */
-    WebRtc_Word16 *index,  /* (o) Codebook indices */
-    WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
-    WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
-    WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
-    WebRtc_Word16 lMem,  /* (i) Length of buffer */
-    WebRtc_Word16 lTarget,  /* (i) Length of vector */
-    WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
-    WebRtc_Word16 block  /* (i) the subblock number */
+    int16_t *index,  /* (o) Codebook indices */
+    int16_t *gain_index, /* (o) Gain quantization indices */
+    int16_t *intarget, /* (i) Target vector for encoding */
+    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
+    int16_t lMem,  /* (i) Length of buffer */
+    int16_t lTarget,  /* (i) Length of vector */
+    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
+    int16_t block  /* (i) the subblock number */
                             ) {
-  WebRtc_Word16 i, j, stage, range;
-  WebRtc_Word16 *pp, scale, tmp;
-  WebRtc_Word16 bits, temp1, temp2;
-  WebRtc_Word16 base_size;
-  WebRtc_Word32 codedEner, targetEner;
-  WebRtc_Word16 gains[CB_NSTAGES+1];
-  WebRtc_Word16 *cb_vecPtr;
-  WebRtc_Word16 indexOffset, sInd, eInd;
-  WebRtc_Word32 CritMax=0;
-  WebRtc_Word16 shTotMax=WEBRTC_SPL_WORD16_MIN;
-  WebRtc_Word16 bestIndex=0;
-  WebRtc_Word16 bestGain=0;
-  WebRtc_Word16 indexNew, CritNewSh;
-  WebRtc_Word32 CritNew;
-  WebRtc_Word32 *cDotPtr;
-  WebRtc_Word16 noOfZeros;
-  WebRtc_Word16 *gainPtr;
-  WebRtc_Word32 t32, tmpW32;
-  WebRtc_Word16 *WebRtcIlbcfix_kGainSq5_ptr;
+  int16_t i, j, stage, range;
+  int16_t *pp, scale, tmp;
+  int16_t bits, temp1, temp2;
+  int16_t base_size;
+  int32_t codedEner, targetEner;
+  int16_t gains[CB_NSTAGES+1];
+  int16_t *cb_vecPtr;
+  int16_t indexOffset, sInd, eInd;
+  int32_t CritMax=0;
+  int16_t shTotMax=WEBRTC_SPL_WORD16_MIN;
+  int16_t bestIndex=0;
+  int16_t bestGain=0;
+  int16_t indexNew, CritNewSh;
+  int32_t CritNew;
+  int32_t *cDotPtr;
+  int16_t noOfZeros;
+  int16_t *gainPtr;
+  int32_t t32, tmpW32;
+  int16_t *WebRtcIlbcfix_kGainSq5_ptr;
   /* Stack based */
-  WebRtc_Word16 CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
-  WebRtc_Word32 cDot[128];
-  WebRtc_Word32 Crit[128];
-  WebRtc_Word16 targetVec[SUBL+LPC_FILTERORDER];
-  WebRtc_Word16 cbvectors[CB_MEML + 1];  /* Adding one extra position for
+  int16_t CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
+  int32_t cDot[128];
+  int32_t Crit[128];
+  int16_t targetVec[SUBL+LPC_FILTERORDER];
+  int16_t cbvectors[CB_MEML + 1];  /* Adding one extra position for
                                             Coverity warnings. */
-  WebRtc_Word16 codedVec[SUBL];
-  WebRtc_Word16 interpSamples[20*4];
-  WebRtc_Word16 interpSamplesFilt[20*4];
-  WebRtc_Word16 energyW16[CB_EXPAND*128];
-  WebRtc_Word16 energyShifts[CB_EXPAND*128];
-  WebRtc_Word16 *inverseEnergy=energyW16;   /* Reuse memory */
-  WebRtc_Word16 *inverseEnergyShifts=energyShifts; /* Reuse memory */
-  WebRtc_Word16 *buf = &CBbuf[LPC_FILTERORDER];
-  WebRtc_Word16 *target = &targetVec[LPC_FILTERORDER];
-  WebRtc_Word16 *aug_vec = (WebRtc_Word16*)cDot;   /* length [SUBL], reuse memory */
+  int16_t codedVec[SUBL];
+  int16_t interpSamples[20*4];
+  int16_t interpSamplesFilt[20*4];
+  int16_t energyW16[CB_EXPAND*128];
+  int16_t energyShifts[CB_EXPAND*128];
+  int16_t *inverseEnergy=energyW16;   /* Reuse memory */
+  int16_t *inverseEnergyShifts=energyShifts; /* Reuse memory */
+  int16_t *buf = &CBbuf[LPC_FILTERORDER];
+  int16_t *target = &targetVec[LPC_FILTERORDER];
+  int16_t *aug_vec = (int16_t*)cDot;   /* length [SUBL], reuse memory */
 
   /* Determine size of codebook sections */
 
@@ -108,8 +108,8 @@
 
   /* Find the highest absolute value to calculate proper
      vector scale factor (so that it uses 12 bits) */
-  temp1 = WebRtcSpl_MaxAbsValueW16(buf, (WebRtc_Word16)lMem);
-  temp2 = WebRtcSpl_MaxAbsValueW16(target, (WebRtc_Word16)lTarget);
+  temp1 = WebRtcSpl_MaxAbsValueW16(buf, (int16_t)lMem);
+  temp2 = WebRtcSpl_MaxAbsValueW16(target, (int16_t)lTarget);
 
   if ((temp1>0)&&(temp2>0)) {
     temp1 = WEBRTC_SPL_MAX(temp1, temp2);
@@ -147,7 +147,7 @@
 
     /* Compute the CB vectors' energies for the second cb section (filtered cb) */
     WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors,
-                                          scale, (WebRtc_Word16)(base_size+20), energyW16, energyShifts);
+                                          scale, (int16_t)(base_size+20), energyW16, energyShifts);
 
     /* Compute the CB vectors' energies and store them in the vector
      * energyW16. Also the corresponding shift values are stored. The
@@ -221,11 +221,11 @@
 
     /* Update the global best index and the corresponding gain */
     WebRtcIlbcfix_CbUpdateBestIndex(
-        CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew+indexOffset],
+        CritNew, CritNewSh, (int16_t)(indexNew+indexOffset), cDot[indexNew+indexOffset],
         inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
         &CritMax, &shTotMax, &bestIndex, &bestGain);
 
-    sInd=bestIndex-(WebRtc_Word16)(CB_RESRANGE>>1);
+    sInd=bestIndex-(int16_t)(CB_RESRANGE>>1);
     eInd=sInd+CB_RESRANGE;
     if (sInd<0) {
       eInd-=sInd;
@@ -243,7 +243,7 @@
       if (sInd<20) {
         WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors+lMem,
                                       interpSamplesFilt, cDot,
-                                      (WebRtc_Word16)(sInd+20), (WebRtc_Word16)(WEBRTC_SPL_MIN(39, (eInd+20))), scale);
+                                      (int16_t)(sInd+20), (int16_t)(WEBRTC_SPL_MIN(39, (eInd+20))), scale);
         i=20;
       }
 
@@ -251,14 +251,14 @@
       cb_vecPtr = cbvectors+lMem-20-i;
 
       /* Calculate the cross correlations (main part of the filtered CB) */
-      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-i+1), scale, -1);
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (int16_t)(eInd-i+1), scale, -1);
 
     } else {
       cDotPtr = cDot;
       cb_vecPtr = cbvectors+lMem-lTarget-sInd;
 
       /* Calculate the cross correlations (main part of the filtered CB) */
-      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-sInd+1), scale, -1);
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (int16_t)(eInd-sInd+1), scale, -1);
 
     }
 
@@ -267,13 +267,13 @@
 
     /* Search for best index in this part of the vector */
     WebRtcIlbcfix_CbSearchCore(
-        cDot, (WebRtc_Word16)(eInd-sInd+1), stage, inverseEnergy+indexOffset,
+        cDot, (int16_t)(eInd-sInd+1), stage, inverseEnergy+indexOffset,
         inverseEnergyShifts+indexOffset, Crit,
         &indexNew, &CritNew, &CritNewSh);
 
     /* Update the global best index and the corresponding gain */
     WebRtcIlbcfix_CbUpdateBestIndex(
-        CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew],
+        CritNew, CritNewSh, (int16_t)(indexNew+indexOffset), cDot[indexNew],
         inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
         &CritMax, &shTotMax, &bestIndex, &bestGain);
 
@@ -281,7 +281,7 @@
 
 
     bestGain = WebRtcIlbcfix_GainQuant(bestGain,
-                                       (WebRtc_Word16)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
+                                       (int16_t)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
 
     /* Extract the best (according to measure) codebook vector
        Also adjust the index, so that the augmented vectors are last.
@@ -308,7 +308,7 @@
           /* Adjust index and extract vector */
           index[stage]+=(base_size-20);
 
-          WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-base_size+40),
+          WebRtcIlbcfix_CreateAugmentedVec((int16_t)(index[stage]-base_size+40),
                                            buf+lMem, aug_vec);
           pp = aug_vec;
 
@@ -323,7 +323,7 @@
         } else {
           /* Adjust index and extract vector */
           index[stage]+=(base_size-20);
-          WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-2*base_size+40),
+          WebRtcIlbcfix_CreateAugmentedVec((int16_t)(index[stage]-2*base_size+40),
                                            cbvectors+lMem, aug_vec);
           pp = aug_vec;
         }
@@ -333,7 +333,7 @@
     /* Subtract the best codebook vector, according
        to measure, from the target vector */
 
-    WebRtcSpl_AddAffineVectorToVector(target, pp, (WebRtc_Word16)(-bestGain), (WebRtc_Word32)8192, (WebRtc_Word16)14, (int)lTarget);
+    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain), (int32_t)8192, (int16_t)14, (int)lTarget);
 
     /* record quantized gain */
     gains[stage+1] = bestGain;
@@ -350,8 +350,8 @@
 
   j=gain_index[0];
 
-  temp1 = (WebRtc_Word16)WebRtcSpl_NormW32(codedEner);
-  temp2 = (WebRtc_Word16)WebRtcSpl_NormW32(targetEner);
+  temp1 = (int16_t)WebRtcSpl_NormW32(codedEner);
+  temp2 = (int16_t)WebRtcSpl_NormW32(targetEner);
 
   if(temp1 < temp2) {
     bits = 16 - temp1;
@@ -359,19 +359,19 @@
     bits = 16 - temp2;
   }
 
-  tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(gains[1],gains[1], 14);
+  tmp = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(gains[1],gains[1], 14);
 
   targetEner = WEBRTC_SPL_MUL_16_16(
       WEBRTC_SPL_SHIFT_W32(targetEner, -bits), tmp);
 
-  tmpW32 = ((WebRtc_Word32)(gains[1]-1))<<1;
+  tmpW32 = ((int32_t)(gains[1]-1))<<1;
 
   /* Pointer to the table that contains
      gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
-  gainPtr=(WebRtc_Word16*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
-  temp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
+  gainPtr=(int16_t*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
+  temp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
 
-  WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[j];
+  WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j];
 
   /* targetEner and codedEner are in Q(-2*scale) */
   for (i=gain_index[0];i<32;i++) {
@@ -386,7 +386,7 @@
     if (t32 < 0) {
       if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
         j=i;
-        WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[i];
+        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[i];
       }
     }
     gainPtr++;
diff --git a/modules/audio_coding/codecs/ilbc/cb_search.h b/modules/audio_coding/codecs/ilbc/cb_search.h
index e4ad4b5..fc62190 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search.h
+++ b/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -22,14 +22,14 @@
 void WebRtcIlbcfix_CbSearch(
     iLBC_Enc_Inst_t *iLBCenc_inst,
     /* (i) the encoder state structure */
-    WebRtc_Word16 *index,  /* (o) Codebook indices */
-    WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
-    WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
-    WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
-    WebRtc_Word16 lMem,  /* (i) Length of buffer */
-    WebRtc_Word16 lTarget,  /* (i) Length of vector */
-    WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
-    WebRtc_Word16 block  /* (i) the subblock number */
+    int16_t *index,  /* (o) Codebook indices */
+    int16_t *gain_index, /* (o) Gain quantization indices */
+    int16_t *intarget, /* (i) Target vector for encoding */
+    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
+    int16_t lMem,  /* (i) Length of buffer */
+    int16_t lTarget,  /* (i) Length of vector */
+    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
+    int16_t block  /* (i) the subblock number */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_search_core.c b/modules/audio_coding/codecs/ilbc/cb_search_core.c
index 711e2df..c2299d5 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search_core.c
+++ b/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -20,29 +20,29 @@
 #include "constants.h"
 
 void WebRtcIlbcfix_CbSearchCore(
-    WebRtc_Word32 *cDot,    /* (i) Cross Correlation */
-    WebRtc_Word16 range,    /* (i) Search range */
-    WebRtc_Word16 stage,    /* (i) Stage of this search */
-    WebRtc_Word16 *inverseEnergy,  /* (i) Inversed energy */
-    WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
+    int32_t *cDot,    /* (i) Cross Correlation */
+    int16_t range,    /* (i) Search range */
+    int16_t stage,    /* (i) Stage of this search */
+    int16_t *inverseEnergy,  /* (i) Inversed energy */
+    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
                                            with the offset 2*16-29 */
-    WebRtc_Word32 *Crit,    /* (o) The criteria */
-    WebRtc_Word16 *bestIndex,   /* (o) Index that corresponds to
+    int32_t *Crit,    /* (o) The criteria */
+    int16_t *bestIndex,   /* (o) Index that corresponds to
                                                    maximum criteria (in this
                                                    vector) */
-    WebRtc_Word32 *bestCrit,   /* (o) Value of critera for the
+    int32_t *bestCrit,   /* (o) Value of critera for the
                                                    chosen index */
-    WebRtc_Word16 *bestCritSh)   /* (o) The domain of the chosen
+    int16_t *bestCritSh)   /* (o) The domain of the chosen
                                                    criteria */
 {
-  WebRtc_Word32 maxW32, tmp32;
-  WebRtc_Word16 max, sh, tmp16;
+  int32_t maxW32, tmp32;
+  int16_t max, sh, tmp16;
   int i;
-  WebRtc_Word32 *cDotPtr;
-  WebRtc_Word16 cDotSqW16;
-  WebRtc_Word16 *inverseEnergyPtr;
-  WebRtc_Word32 *critPtr;
-  WebRtc_Word16 *inverseEnergyShiftPtr;
+  int32_t *cDotPtr;
+  int16_t cDotSqW16;
+  int16_t *inverseEnergyPtr;
+  int32_t *critPtr;
+  int16_t *inverseEnergyShiftPtr;
 
   /* Don't allow negative values for stage 0 */
   if (stage==0) {
@@ -53,10 +53,10 @@
     }
   }
 
-  /* Normalize cDot to WebRtc_Word16, calculate the square of cDot and store the upper WebRtc_Word16 */
+  /* Normalize cDot to int16_t, calculate the square of cDot and store the upper int16_t */
   maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
 
-  sh = (WebRtc_Word16)WebRtcSpl_NormW32(maxW32);
+  sh = (int16_t)WebRtcSpl_NormW32(maxW32);
   cDotPtr = cDot;
   inverseEnergyPtr = inverseEnergy;
   critPtr = Crit;
@@ -64,10 +64,10 @@
   max=WEBRTC_SPL_WORD16_MIN;
 
   for (i=0;i<range;i++) {
-    /* Calculate cDot*cDot and put the result in a WebRtc_Word16 */
+    /* Calculate cDot*cDot and put the result in a int16_t */
     tmp32 = WEBRTC_SPL_LSHIFT_W32(*cDotPtr,sh);
-    tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32,16);
-    cDotSqW16 = (WebRtc_Word16)(((WebRtc_Word32)(tmp16)*(tmp16))>>16);
+    tmp16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32,16);
+    cDotSqW16 = (int16_t)(((int32_t)(tmp16)*(tmp16))>>16);
 
     /* Calculate the criteria (cDot*cDot/energy) */
     *critPtr=WEBRTC_SPL_MUL_16_16(cDotSqW16, (*inverseEnergyPtr));
diff --git a/modules/audio_coding/codecs/ilbc/cb_search_core.h b/modules/audio_coding/codecs/ilbc/cb_search_core.h
index e074c52..e4f2e92 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search_core.h
+++ b/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -22,19 +22,19 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_CbSearchCore(
-    WebRtc_Word32 *cDot,    /* (i) Cross Correlation */
-    WebRtc_Word16 range,    /* (i) Search range */
-    WebRtc_Word16 stage,    /* (i) Stage of this search */
-    WebRtc_Word16 *inverseEnergy,  /* (i) Inversed energy */
-    WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
+    int32_t *cDot,    /* (i) Cross Correlation */
+    int16_t range,    /* (i) Search range */
+    int16_t stage,    /* (i) Stage of this search */
+    int16_t *inverseEnergy,  /* (i) Inversed energy */
+    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
                                           with the offset 2*16-29 */
-    WebRtc_Word32 *Crit,    /* (o) The criteria */
-    WebRtc_Word16 *bestIndex,   /* (o) Index that corresponds to
+    int32_t *Crit,    /* (o) The criteria */
+    int16_t *bestIndex,   /* (o) Index that corresponds to
                                    maximum criteria (in this
                                    vector) */
-    WebRtc_Word32 *bestCrit,   /* (o) Value of critera for the
+    int32_t *bestCrit,   /* (o) Value of critera for the
                                   chosen index */
-    WebRtc_Word16 *bestCritSh);  /* (o) The domain of the chosen
+    int16_t *bestCritSh);  /* (o) The domain of the chosen
                                     criteria */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
index bf85408..88ea199 100644
--- a/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
+++ b/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -21,22 +21,22 @@
 #include "constants.h"
 
 void WebRtcIlbcfix_CbUpdateBestIndex(
-    WebRtc_Word32 CritNew,    /* (i) New Potentially best Criteria */
-    WebRtc_Word16 CritNewSh,   /* (i) Shift value of above Criteria */
-    WebRtc_Word16 IndexNew,   /* (i) Index of new Criteria */
-    WebRtc_Word32 cDotNew,    /* (i) Cross dot of new index */
-    WebRtc_Word16 invEnergyNew,  /* (i) Inversed energy new index */
-    WebRtc_Word16 energyShiftNew,  /* (i) Energy shifts of new index */
-    WebRtc_Word32 *CritMax,   /* (i/o) Maximum Criteria (so far) */
-    WebRtc_Word16 *shTotMax,   /* (i/o) Shifts of maximum criteria */
-    WebRtc_Word16 *bestIndex,   /* (i/o) Index that corresponds to
+    int32_t CritNew,    /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
+    int16_t IndexNew,   /* (i) Index of new Criteria */
+    int32_t cDotNew,    /* (i) Cross dot of new index */
+    int16_t invEnergyNew,  /* (i) Inversed energy new index */
+    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
+    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
+    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
+    int16_t *bestIndex,   /* (i/o) Index that corresponds to
                                                    maximum criteria */
-    WebRtc_Word16 *bestGain)   /* (i/o) Gain in Q14 that corresponds
+    int16_t *bestGain)   /* (i/o) Gain in Q14 that corresponds
                                                    to maximum criteria */
 {
-  WebRtc_Word16 shOld, shNew, tmp16;
-  WebRtc_Word16 scaleTmp;
-  WebRtc_Word32 gainW32;
+  int16_t shOld, shNew, tmp16;
+  int16_t scaleTmp;
+  int32_t gainW32;
 
   /* Normalize the new and old Criteria to the same domain */
   if (CritNewSh>(*shTotMax)) {
@@ -54,19 +54,19 @@
   if (WEBRTC_SPL_RSHIFT_W32(CritNew, shNew)>
       WEBRTC_SPL_RSHIFT_W32((*CritMax),shOld)) {
 
-    tmp16 = (WebRtc_Word16)WebRtcSpl_NormW32(cDotNew);
+    tmp16 = (int16_t)WebRtcSpl_NormW32(cDotNew);
     tmp16 = 16 - tmp16;
 
     /* Calculate the gain in Q14
        Compensate for inverseEnergyshift in Q29 and that the energy
-       value was stored in a WebRtc_Word16 (shifted down 16 steps)
+       value was stored in a int16_t (shifted down 16 steps)
        => 29-14+16 = 31 */
 
     scaleTmp = -energyShiftNew-tmp16+31;
     scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
 
     gainW32 = WEBRTC_SPL_MUL_16_16_RSFT(
-        ((WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16)), invEnergyNew, scaleTmp);
+        ((int16_t)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16)), invEnergyNew, scaleTmp);
 
     /* Check if criteria satisfies Gain criteria (max 1.3)
        if it is larger set the gain to 1.3
@@ -77,7 +77,7 @@
     } else if (gainW32<-21299) {
       *bestGain=-21299;
     } else {
-      *bestGain=(WebRtc_Word16)gainW32;
+      *bestGain=(int16_t)gainW32;
     }
 
     *CritMax=CritNew;
diff --git a/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
index 9015187..e8519d4 100644
--- a/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
+++ b/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -22,17 +22,17 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_CbUpdateBestIndex(
-    WebRtc_Word32 CritNew,    /* (i) New Potentially best Criteria */
-    WebRtc_Word16 CritNewSh,   /* (i) Shift value of above Criteria */
-    WebRtc_Word16 IndexNew,   /* (i) Index of new Criteria */
-    WebRtc_Word32 cDotNew,    /* (i) Cross dot of new index */
-    WebRtc_Word16 invEnergyNew,  /* (i) Inversed energy new index */
-    WebRtc_Word16 energyShiftNew,  /* (i) Energy shifts of new index */
-    WebRtc_Word32 *CritMax,   /* (i/o) Maximum Criteria (so far) */
-    WebRtc_Word16 *shTotMax,   /* (i/o) Shifts of maximum criteria */
-    WebRtc_Word16 *bestIndex,   /* (i/o) Index that corresponds to
+    int32_t CritNew,    /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
+    int16_t IndexNew,   /* (i) Index of new Criteria */
+    int32_t cDotNew,    /* (i) Cross dot of new index */
+    int16_t invEnergyNew,  /* (i) Inversed energy new index */
+    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
+    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
+    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
+    int16_t *bestIndex,   /* (i/o) Index that corresponds to
                                    maximum criteria */
-    WebRtc_Word16 *bestGain);   /* (i/o) Gain in Q14 that corresponds
+    int16_t *bestGain);   /* (i/o) Gain in Q14 that corresponds
                                    to maximum criteria */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/chebyshev.c b/modules/audio_coding/codecs/ilbc/chebyshev.c
index 90108ff..b49dd79 100644
--- a/modules/audio_coding/codecs/ilbc/chebyshev.c
+++ b/modules/audio_coding/codecs/ilbc/chebyshev.c
@@ -26,57 +26,57 @@
  *   T_i(x) is the i:th order Chebyshev polynomial
  *------------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
+int16_t WebRtcIlbcfix_Chebyshev(
     /* (o) Result of C(x) */
-    WebRtc_Word16 x,  /* (i) Value to the Chevyshev polynomial */
-    WebRtc_Word16 *f  /* (i) The coefficients in the polynomial */
+    int16_t x,  /* (i) Value to the Chevyshev polynomial */
+    int16_t *f  /* (i) The coefficients in the polynomial */
                                       ) {
-  WebRtc_Word16 b1_high, b1_low; /* Use the high, low format to increase the accuracy */
-  WebRtc_Word32 b2;
-  WebRtc_Word32 tmp1W32;
-  WebRtc_Word32 tmp2W32;
+  int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */
+  int32_t b2;
+  int32_t tmp1W32;
+  int32_t tmp2W32;
   int i;
 
-  b2 = (WebRtc_Word32)0x1000000; /* b2 = 1.0 (Q23) */
+  b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */
   /* Calculate b1 = 2*x + f[1] */
-  tmp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x, 10);
-  tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[1], 14);
+  tmp1W32 = WEBRTC_SPL_LSHIFT_W32((int32_t)x, 10);
+  tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[1], 14);
 
   for (i = 2; i < 5; i++) {
     tmp2W32 = tmp1W32;
 
     /* Split b1 (in tmp1W32) into a high and low part */
-    b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
-    b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
+    b1_high = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
+    b1_low = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((int32_t)b1_high),16), 1);
 
     /* Calculate 2*x*b1-b2+f[i] */
     tmp1W32 = WEBRTC_SPL_LSHIFT_W32( (WEBRTC_SPL_MUL_16_16(b1_high, x) +
                                       WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15)), 2);
 
     tmp1W32 -= b2;
-    tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 14);
+    tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[i], 14);
 
     /* Update b2 for next round */
     b2 = tmp2W32;
   }
 
   /* Split b1 (in tmp1W32) into a high and low part */
-  b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
-  b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
+  b1_high = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
+  b1_low = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((int32_t)b1_high),16), 1);
 
   /* tmp1W32 = x*b1 - b2 + f[i]/2 */
   tmp1W32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(b1_high, x), 1) +
       WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15), 1);
 
   tmp1W32 -= b2;
-  tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 13);
+  tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[i], 13);
 
-  /* Handle overflows and set to maximum or minimum WebRtc_Word16 instead */
-  if (tmp1W32>((WebRtc_Word32)33553408)) {
+  /* Handle overflows and set to maximum or minimum int16_t instead */
+  if (tmp1W32>((int32_t)33553408)) {
     return(WEBRTC_SPL_WORD16_MAX);
-  } else if (tmp1W32<((WebRtc_Word32)-33554432)) {
+  } else if (tmp1W32<((int32_t)-33554432)) {
     return(WEBRTC_SPL_WORD16_MIN);
   } else {
-    return((WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 10));
+    return((int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 10));
   }
 }
diff --git a/modules/audio_coding/codecs/ilbc/chebyshev.h b/modules/audio_coding/codecs/ilbc/chebyshev.h
index 57aab99..bf10132 100644
--- a/modules/audio_coding/codecs/ilbc/chebyshev.h
+++ b/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -28,10 +28,10 @@
  *   T_i(x) is the i:th order Chebyshev polynomial
  *------------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
+int16_t WebRtcIlbcfix_Chebyshev(
     /* (o) Result of C(x) */
-    WebRtc_Word16 x,  /* (i) Value to the Chevyshev polynomial */
-    WebRtc_Word16 *f  /* (i) The coefficients in the polynomial */
+    int16_t x,  /* (i) Value to the Chevyshev polynomial */
+    int16_t *f  /* (i) The coefficients in the polynomial */
                                       );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/comp_corr.c b/modules/audio_coding/codecs/ilbc/comp_corr.c
index 3d7f93e..a53e8a7 100644
--- a/modules/audio_coding/codecs/ilbc/comp_corr.c
+++ b/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -24,15 +24,15 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CompCorr(
-    WebRtc_Word32 *corr, /* (o) cross correlation */
-    WebRtc_Word32 *ener, /* (o) energy */
-    WebRtc_Word16 *buffer, /* (i) signal buffer */
-    WebRtc_Word16 lag,  /* (i) pitch lag */
-    WebRtc_Word16 bLen, /* (i) length of buffer */
-    WebRtc_Word16 sRange, /* (i) correlation search length */
-    WebRtc_Word16 scale /* (i) number of rightshifts to use */
+    int32_t *corr, /* (o) cross correlation */
+    int32_t *ener, /* (o) energy */
+    int16_t *buffer, /* (i) signal buffer */
+    int16_t lag,  /* (i) pitch lag */
+    int16_t bLen, /* (i) length of buffer */
+    int16_t sRange, /* (i) correlation search length */
+    int16_t scale /* (i) number of rightshifts to use */
                             ){
-  WebRtc_Word16 *w16ptr;
+  int16_t *w16ptr;
 
   w16ptr=&buffer[bLen-sRange-lag];
 
diff --git a/modules/audio_coding/codecs/ilbc/comp_corr.h b/modules/audio_coding/codecs/ilbc/comp_corr.h
index cd46532..4ff80aa 100644
--- a/modules/audio_coding/codecs/ilbc/comp_corr.h
+++ b/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -27,13 +27,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CompCorr(
-    WebRtc_Word32 *corr, /* (o) cross correlation */
-    WebRtc_Word32 *ener, /* (o) energy */
-    WebRtc_Word16 *buffer, /* (i) signal buffer */
-    WebRtc_Word16 lag,  /* (i) pitch lag */
-    WebRtc_Word16 bLen, /* (i) length of buffer */
-    WebRtc_Word16 sRange, /* (i) correlation search length */
-    WebRtc_Word16 scale /* (i) number of rightshifts to use */
+    int32_t *corr, /* (o) cross correlation */
+    int32_t *ener, /* (o) energy */
+    int16_t *buffer, /* (i) signal buffer */
+    int16_t lag,  /* (i) pitch lag */
+    int16_t bLen, /* (i) length of buffer */
+    int16_t sRange, /* (i) correlation search length */
+    int16_t scale /* (i) number of rightshifts to use */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/constants.c b/modules/audio_coding/codecs/ilbc/constants.c
index 5ebe9be..1d384b7 100644
--- a/modules/audio_coding/codecs/ilbc/constants.c
+++ b/modules/audio_coding/codecs/ilbc/constants.c
@@ -21,25 +21,25 @@
 
 /* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
 
-const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
-const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
+const int16_t WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
 
 /* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
    the 80 sample start state
 */
-const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
+const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
   1638, 1843, 2048, 1843, 1638
 };
 
 /* LP Filter coeffs used for downsampling */
-const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
+const int16_t WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
   -273, 512, 1297, 1696, 1297, 512, -273
 };
 
 /* Constants used in the LPC calculations */
 
 /* Hanning LPC window (in Q15) */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
+const int16_t WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
   6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
   932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
   3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
@@ -63,7 +63,7 @@
 };
 
 /* Asymmetric LPC window (in Q15)*/
-const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
+const int16_t WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
   2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
   280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
   1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
@@ -87,7 +87,7 @@
 };
 
 /* Lag window for LPC (Q31) */
-const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
+const int32_t WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
   2147483647,   2144885453,   2137754373,   2125918626,   2109459810,
   2088483140,   2063130336,   2033564590,   1999977009,   1962580174,
   1921610283};
@@ -95,7 +95,7 @@
 /* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
  * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
  */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
+const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
   32767, 29573, 26690, 24087,
   21739, 19619, 17707, 15980,
   14422, 13016, 11747};
@@ -103,12 +103,12 @@
 /* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
  * floating point vector {1 0.4222 0.4222^2... }
  */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
+const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
   32767, 13835, 5841, 2466, 1041, 440,
   186, 78,  33,  14,  6};
 
 /* LSF quantization Q13 domain */
-const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
+const int16_t WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
   1273,       2238,       3696,
   3199,       5309,       8209,
   3606,       5671,       7829,
@@ -431,22 +431,22 @@
   15878,      18550,      20718,      22906
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
-const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
+const int16_t WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
+const int16_t WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
 
-const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
+const int16_t WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
   2308,       3652,       5434,       7885,
   10255,      12559,      15160,      17513,
   20328,      22752};
 
-const WebRtc_Word16 WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
+const int16_t WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
   31476, 29565, 25819, 18725, 10276,
   1236, -9049, -17600, -25884, -30618
 };
 
 /* Q14 */
-const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
-const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
+const int16_t WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
+const int16_t WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
 
 /*
    cos(x) in Q15
@@ -454,7 +454,7 @@
    used in WebRtcIlbcfix_Lsp2Lsf()
 */
 
-const WebRtc_Word16 WebRtcIlbcfix_kCos[64] = {
+const int16_t WebRtcIlbcfix_kCos[64] = {
   32767,  32729,  32610,  32413,  32138,  31786,  31357,  30853,
   30274,  29622,  28899,  28106,  27246,  26320,  25330,  24279,
   23170,  22006,  20788,  19520,  18205,  16846,  15447,  14010,
@@ -469,7 +469,7 @@
    Derivative in Q19, used to interpolate between the
    WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
 */
-const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[64] = {
+const int16_t WebRtcIlbcfix_kCosDerivative[64] = {
   -632,  -1893,  -3150,  -4399,  -5638,  -6863,  -8072,  -9261,
   -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
   -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
@@ -484,7 +484,7 @@
   WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
 */
 
-const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
+const int16_t WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
   32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
   29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
   21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
@@ -500,7 +500,7 @@
    used in WebRtcIlbcfix_Lsp2Lsf()
 */
 
-const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[64] = {
+const int16_t WebRtcIlbcfix_kAcosDerivative[64] = {
   -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
   -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
   -901, -861, -827, -797, -772, -750, -730, -713,
@@ -515,7 +515,7 @@
 /* Tables for quantization of start state */
 
 /* State quantization tables */
-const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
+const int16_t WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
   -30473, -17838, -9257, -2537,
   3639, 10893, 19958, 32636
 };
@@ -524,7 +524,7 @@
    less or equal than value 0 => index = 0
    less or equal than value k => index = k
 */
-const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[64] = {
+const int32_t WebRtcIlbcfix_kChooseFrgQuant[64] = {
   118, 163, 222, 305, 425, 604,
   851, 1174, 1617, 2222, 3080, 4191,
   5525, 7215, 9193, 11540, 14397, 17604,
@@ -538,7 +538,7 @@
   37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kScale[64] = {
+const int16_t WebRtcIlbcfix_kScale[64] = {
   /* Values in Q16 */
   29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
   6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
@@ -560,7 +560,7 @@
   37:58 in Q5
   59:63 in Q3
 */
-const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[64] = {
+const int16_t WebRtcIlbcfix_kFrgQuantMod[64] = {
   /* First 37 values in Q8 */
   569, 671, 786, 916, 1077, 1278,
   1529, 1802, 2109, 2481, 2898, 3440,
@@ -583,36 +583,36 @@
 /* Expansion filter to get additional cb section.
  * Q12 and reversed compared to flp
  */
-const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
+const int16_t WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
   -140, 446, -755, 3302, 2922, -590, 343, -138};
 
 /* Weighting coefficients for short lags.
  * [0.2 0.4 0.6 0.8] in Q15 */
-const WebRtc_Word16 WebRtcIlbcfix_kAlpha[4]={
+const int16_t WebRtcIlbcfix_kAlpha[4]={
   6554, 13107, 19661, 26214};
 
 /* Ranges for search and filters at different subframes */
 
-const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+const int16_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
   {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
 
-const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+const int16_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
 
 /* Gain Quantization for the codebook gains of the 3 stages */
 
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[9]={
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq3[9]={
   -16384, -10813, -5407, 0, 4096, 8192,
   12288, 16384, 32767};
 
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[17]={
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq4[17]={
   -17203, -14746, -12288, -9830, -7373, -4915,
   -2458, 0, 2458, 4915, 7373, 9830,
   12288, 14746, 17203, 19661, 32767};
 
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[33]={
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq5[33]={
   614,        1229,        1843,        2458,        3072,       3686,
   4301,        4915,        5530,        6144,        6758,        7373,
   7987,        8602,        9216,        9830,       10445,       11059,
@@ -621,7 +621,7 @@
   19046,       19661,    32767};
 
 /* Q14 gain_sq5Tbl squared in Q14 */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[32] = {
+const int16_t WebRtcIlbcfix_kGainSq5Sq[32] = {
   23,   92,    207,  368,  576,  829,
   1129,  1474,   1866,  2304,  2787,  3317,
   3893,  4516,   5184,  5897,  6658,  7464,
@@ -630,37 +630,37 @@
   22140,  23593
 };
 
-const WebRtc_Word16* const WebRtcIlbcfix_kGain[3] =
+const int16_t* const WebRtcIlbcfix_kGain[3] =
 {WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
 
 
 /* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
 
-const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
+const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
   {0,    0,    0, 4096,    0,  0,   0},
   {64, -315, 1181, 3531, -436, 77, -64},
   {97, -509, 2464, 2464, -509, 97, -97},
   {77, -436, 3531, 1181, -315, 64, -77}
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[3] = {
+const int16_t WebRtcIlbcfix_kEnhWt[3] = {
   4800, 16384, 27968 /* Q16 */
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+const int16_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
   160, 480, 800, 1120, 1440, 1760, 2080, 2400  /* Q(-2) */
 };
 
 /* PLC table */
 
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
+const int16_t WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
   839, 1343, 2048, 2998, 4247, 5849
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
+const int16_t WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
   0, 5462, 10922, 16384, 21846, 27306
 };
 
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
+const int16_t WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
   26667, 18729, 13653, 10258, 7901, 6214
 };
diff --git a/modules/audio_coding/codecs/ilbc/constants.h b/modules/audio_coding/codecs/ilbc/constants.h
index f787f74..cdc8a9c 100644
--- a/modules/audio_coding/codecs/ilbc/constants.h
+++ b/modules/audio_coding/codecs/ilbc/constants.h
@@ -24,69 +24,69 @@
 
 /* high pass filters */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[];
+extern const int16_t WebRtcIlbcfix_kHpInCoefs[];
+extern const int16_t WebRtcIlbcfix_kHpOutCoefs[];
 
 /* Window for start state decision */
-extern const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[];
+extern const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[];
 
 /* low pass filter used for downsampling */
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[];
+extern const int16_t WebRtcIlbcfix_kLpFiltCoefs[];
 
 /* LPC analysis and quantization */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[];
-extern const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLspMean[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCos[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[];
+extern const int16_t WebRtcIlbcfix_kLpcWin[];
+extern const int16_t WebRtcIlbcfix_kLpcAsymWin[];
+extern const int32_t WebRtcIlbcfix_kLpcLagWin[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[];
+extern const int16_t WebRtcIlbcfix_kLsfDimCb[];
+extern const int16_t WebRtcIlbcfix_kLsfSizeCb[];
+extern const int16_t WebRtcIlbcfix_kLsfCb[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight20ms[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight30ms[];
+extern const int16_t WebRtcIlbcfix_kLsfMean[];
+extern const int16_t WebRtcIlbcfix_kLspMean[];
+extern const int16_t WebRtcIlbcfix_kCos[];
+extern const int16_t WebRtcIlbcfix_kCosDerivative[];
+extern const int16_t WebRtcIlbcfix_kCosGrid[];
+extern const int16_t WebRtcIlbcfix_kAcosDerivative[];
 
 /* state quantization tables */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[];
-extern const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kScale[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[];
+extern const int16_t WebRtcIlbcfix_kStateSq3[];
+extern const int32_t WebRtcIlbcfix_kChooseFrgQuant[];
+extern const int16_t WebRtcIlbcfix_kScale[];
+extern const int16_t WebRtcIlbcfix_kFrgQuantMod[];
 
 /* Ranges for search and filters at different subframes */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
-extern const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[];
+extern const int16_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const int16_t WebRtcIlbcfix_kFilterRange[];
 
 /* gain quantization tables */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[];
-extern const WebRtc_Word16* const WebRtcIlbcfix_kGain[];
+extern const int16_t WebRtcIlbcfix_kGainSq3[];
+extern const int16_t WebRtcIlbcfix_kGainSq4[];
+extern const int16_t WebRtcIlbcfix_kGainSq5[];
+extern const int16_t WebRtcIlbcfix_kGainSq5Sq[];
+extern const int16_t* const WebRtcIlbcfix_kGain[];
 
 /* adaptive codebook definitions */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kAlpha[];
+extern const int16_t WebRtcIlbcfix_kCbFiltersRev[];
+extern const int16_t WebRtcIlbcfix_kAlpha[];
 
 /* enhancer definitions */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[];
+extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
+extern const int16_t WebRtcIlbcfix_kEnhWt[];
+extern const int16_t WebRtcIlbcfix_kEnhPlocs[];
 
 /* PLC tables */
 
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[];
+extern const int16_t WebRtcIlbcfix_kPlcPerSqr[];
+extern const int16_t WebRtcIlbcfix_kPlcPitchFact[];
+extern const int16_t WebRtcIlbcfix_kPlcPfSlope[];
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
index f021c4d..965cbe0 100644
--- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -25,14 +25,14 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CreateAugmentedVec(
-    WebRtc_Word16 index,  /* (i) Index for the augmented vector to be created */
-    WebRtc_Word16 *buffer,  /* (i) Pointer to the end of the codebook memory that
+    int16_t index,  /* (i) Index for the augmented vector to be created */
+    int16_t *buffer,  /* (i) Pointer to the end of the codebook memory that
                                            is used for creation of the augmented codebook */
-    WebRtc_Word16 *cbVec  /* (o) The construced codebook vector */
+    int16_t *cbVec  /* (o) The construced codebook vector */
                                       ) {
-  WebRtc_Word16 ilow;
-  WebRtc_Word16 *ppo, *ppi;
-  WebRtc_Word16 cbVecTmp[4];
+  int16_t ilow;
+  int16_t *ppo, *ppi;
+  int16_t cbVecTmp[4];
 
   ilow = index-4;
 
diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
index 970a9be..e3c3c7b 100644
--- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
+++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -27,10 +27,10 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CreateAugmentedVec(
-    WebRtc_Word16 index,  /* (i) Index for the augmented vector to be created */
-    WebRtc_Word16 *buffer,  /* (i) Pointer to the end of the codebook memory that
+    int16_t index,  /* (i) Index for the augmented vector to be created */
+    int16_t *buffer,  /* (i) Pointer to the end of the codebook memory that
                                            is used for creation of the augmented codebook */
-    WebRtc_Word16 *cbVec  /* (o) The construced codebook vector */
+    int16_t *cbVec  /* (o) The construced codebook vector */
                                       );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/decode.c b/modules/audio_coding/codecs/ilbc/decode.c
index 3bca764..5da9685 100644
--- a/modules/audio_coding/codecs/ilbc/decode.c
+++ b/modules/audio_coding/codecs/ilbc/decode.c
@@ -37,25 +37,25 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecodeImpl(
-    WebRtc_Word16 *decblock,    /* (o) decoded signal block */
-    const WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
+    int16_t *decblock,    /* (o) decoded signal block */
+    const uint16_t *bytes, /* (i) encoded signal bits */
     iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
                                            structure */
-    WebRtc_Word16 mode      /* (i) 0: bad packet, PLC,
+    int16_t mode      /* (i) 0: bad packet, PLC,
                                                                    1: normal */
                            ) {
   int i;
-  WebRtc_Word16 order_plus_one;
+  int16_t order_plus_one;
 
-  WebRtc_Word16 last_bit;
-  WebRtc_Word16 *data;
+  int16_t last_bit;
+  int16_t *data;
   /* Stack based */
-  WebRtc_Word16 decresidual[BLOCKL_MAX];
-  WebRtc_Word16 PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
-  WebRtc_Word16 syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
-  WebRtc_Word16 PLClpc[LPC_FILTERORDER + 1];
+  int16_t decresidual[BLOCKL_MAX];
+  int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+  int16_t PLClpc[LPC_FILTERORDER + 1];
 #ifndef WEBRTC_BIG_ENDIAN
-  WebRtc_UWord16 swapped[NO_OF_WORDS_30MS];
+  uint16_t swapped[NO_OF_WORDS_30MS];
 #endif
   iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
 
@@ -87,14 +87,14 @@
 
     if (mode) { /* No bit errors was detected, continue decoding */
       /* Stack based */
-      WebRtc_Word16 lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
-      WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+      int16_t lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+      int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
 
       /* adjust index */
       WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
 
       /* decode the lsf */
-      WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (WebRtc_Word16*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (int16_t*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
       WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
       WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
                                           lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
@@ -105,7 +105,7 @@
       /* preparing the plc for a future loss! */
       WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 0,
                               decresidual, syntdenum + (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
-                              (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
+                              (int16_t)(iLBCdec_inst->last_lag), iLBCdec_inst);
 
       /* Use the output from doThePLC */
       WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
@@ -121,7 +121,7 @@
     /* packet loss conceal */
 
     WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 1,
-                            decresidual, syntdenum, (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
+                            decresidual, syntdenum, (int16_t)(iLBCdec_inst->last_lag), iLBCdec_inst);
 
     WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
 
@@ -187,18 +187,18 @@
     WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
 
   } else { /* Enhancer not activated */
-    WebRtc_Word16 lag;
+    int16_t lag;
 
     /* Find last lag (since the enhancer is not called to give this info) */
     lag = 20;
     if (iLBCdec_inst->mode==20) {
-      lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
+      lag = (int16_t)WebRtcIlbcfix_XcorrCoef(
           &decresidual[iLBCdec_inst->blockl-60],
           &decresidual[iLBCdec_inst->blockl-60-lag],
           60,
           80, lag, -1);
     } else {
-      lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
+      lag = (int16_t)WebRtcIlbcfix_XcorrCoef(
           &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
           &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
           ENH_BLOCKL,
@@ -228,7 +228,7 @@
   WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
 
   /* High pass filter the signal (with upscaling a factor 2 and saturation) */
-  WebRtcIlbcfix_HpOutput(decblock, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+  WebRtcIlbcfix_HpOutput(decblock, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
                          iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
                          iLBCdec_inst->blockl);
 
diff --git a/modules/audio_coding/codecs/ilbc/decode.h b/modules/audio_coding/codecs/ilbc/decode.h
index 2c9b5a2..71e3802 100644
--- a/modules/audio_coding/codecs/ilbc/decode.h
+++ b/modules/audio_coding/codecs/ilbc/decode.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecodeImpl(
-    WebRtc_Word16 *decblock,    /* (o) decoded signal block */
-    const WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
+    int16_t *decblock,    /* (o) decoded signal block */
+    const uint16_t *bytes, /* (i) encoded signal bits */
     iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
                                            structure */
-    WebRtc_Word16 mode      /* (i) 0: bad packet, PLC,
+    int16_t mode      /* (i) 0: bad packet, PLC,
                                                                    1: normal */
                            );
 
diff --git a/modules/audio_coding/codecs/ilbc/decode_residual.c b/modules/audio_coding/codecs/ilbc/decode_residual.c
index da56db4..d70fd38 100644
--- a/modules/audio_coding/codecs/ilbc/decode_residual.c
+++ b/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -35,15 +35,15 @@
     /* (i/o) the decoder state structure */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
                                 for the decoding  */
-    WebRtc_Word16 *decresidual,  /* (o) decoded residual frame */
-    WebRtc_Word16 *syntdenum   /* (i) the decoded synthesis filter
+    int16_t *decresidual,  /* (o) decoded residual frame */
+    int16_t *syntdenum   /* (i) the decoded synthesis filter
                                   coefficients */
                                   ) {
-  WebRtc_Word16 meml_gotten, Nfor, Nback, diff, start_pos;
-  WebRtc_Word16 subcount, subframe;
-  WebRtc_Word16 *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
-  WebRtc_Word16 *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
-  WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
+  int16_t meml_gotten, Nfor, Nback, diff, start_pos;
+  int16_t subcount, subframe;
+  int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
+  int16_t *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
 
   diff = STATE_LEN - iLBCdec_inst->state_short_len;
 
@@ -64,7 +64,7 @@
 
     /* setup memory */
 
-    WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCdec_inst->state_short_len));
+    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCdec_inst->state_short_len));
     WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
                           iLBCdec_inst->state_short_len);
 
@@ -74,7 +74,7 @@
         &decresidual[start_pos+iLBCdec_inst->state_short_len],
         iLBC_encbits->cb_index, iLBC_encbits->gain_index,
         mem+CB_MEML-ST_MEM_L_TBL,
-        ST_MEM_L_TBL, (WebRtc_Word16)diff
+        ST_MEM_L_TBL, (int16_t)diff
                               );
 
   }
@@ -85,7 +85,7 @@
     meml_gotten = iLBCdec_inst->state_short_len;
     WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
                                   decresidual+start_pos, meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
 
     /* construct decoded vector */
 
@@ -154,7 +154,7 @@
 
     WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
                                   decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
 
     /* loop over subframes to decode */
 
diff --git a/modules/audio_coding/codecs/ilbc/decode_residual.h b/modules/audio_coding/codecs/ilbc/decode_residual.h
index ea7208a..04d6227 100644
--- a/modules/audio_coding/codecs/ilbc/decode_residual.h
+++ b/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -30,8 +30,8 @@
     /* (i/o) the decoder state structure */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
                                    for the decoding  */
-    WebRtc_Word16 *decresidual,  /* (o) decoded residual frame */
-    WebRtc_Word16 *syntdenum   /* (i) the decoded synthesis filter
+    int16_t *decresidual,  /* (o) decoded residual frame */
+    int16_t *syntdenum   /* (i) the decoded synthesis filter
                                                    coefficients */
                                   );
 
diff --git a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
index eee3105..3557665 100644
--- a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
+++ b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -26,16 +26,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecoderInterpolateLsp(
-    WebRtc_Word16 *syntdenum,  /* (o) synthesis filter coefficients */
-    WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
+    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
+    int16_t *weightdenum, /* (o) weighting denumerator
                                    coefficients */
-    WebRtc_Word16 *lsfdeq,   /* (i) dequantized lsf coefficients */
-    WebRtc_Word16 length,   /* (i) length of lsf coefficient vector */
+    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
+    int16_t length,   /* (i) length of lsf coefficient vector */
     iLBC_Dec_Inst_t *iLBCdec_inst
     /* (i) the decoder state structure */
                                           ){
   int  i, pos, lp_length;
-  WebRtc_Word16  lp[LPC_FILTERORDER + 1], *lsfdeq2;
+  int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
 
   lsfdeq2 = lsfdeq + length;
   lp_length = length + 1;
@@ -46,7 +46,7 @@
     WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
                                          WebRtcIlbcfix_kLsfWeight30ms[0], length);
     WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
-    WebRtcIlbcfix_BwExpand(weightdenum, lp, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
 
     /* subframes 2 to 6: interpolation between first and last LSF */
 
@@ -56,7 +56,7 @@
                                            WebRtcIlbcfix_kLsfWeight30ms[i], length);
       WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
       WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
-                             (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
       pos += lp_length;
     }
   } else { /* iLBCdec_inst->mode=20 */
@@ -67,7 +67,7 @@
                                            WebRtcIlbcfix_kLsfWeight20ms[i], length);
       WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
       WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
-                             (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
       pos += lp_length;
     }
   }
diff --git a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
index 3896ca9..02c653a 100644
--- a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecoderInterpolateLsp(
-    WebRtc_Word16 *syntdenum,  /* (o) synthesis filter coefficients */
-    WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
+    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
+    int16_t *weightdenum, /* (o) weighting denumerator
                                    coefficients */
-    WebRtc_Word16 *lsfdeq,   /* (i) dequantized lsf coefficients */
-    WebRtc_Word16 length,   /* (i) length of lsf coefficient vector */
+    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
+    int16_t length,   /* (i) length of lsf coefficient vector */
     iLBC_Dec_Inst_t *iLBCdec_inst
     /* (i) the decoder state structure */
                                           );
diff --git a/modules/audio_coding/codecs/ilbc/defines.h b/modules/audio_coding/codecs/ilbc/defines.h
index bdeba01..b60eaf4 100644
--- a/modules/audio_coding/codecs/ilbc/defines.h
+++ b/modules/audio_coding/codecs/ilbc/defines.h
@@ -118,53 +118,53 @@
 
 /* Struct for the bits */
 typedef struct iLBC_bits_t_ {
-  WebRtc_Word16 lsf[LSF_NSPLIT*LPC_N_MAX];
-  WebRtc_Word16 cb_index[CB_NSTAGES*(NASUB_MAX+1)];  /* First CB_NSTAGES values contains extra CB index */
-  WebRtc_Word16 gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
-  WebRtc_Word16 idxForMax;
-  WebRtc_Word16 state_first;
-  WebRtc_Word16 idxVec[STATE_SHORT_LEN_30MS];
-  WebRtc_Word16 firstbits;
-  WebRtc_Word16 startIdx;
+  int16_t lsf[LSF_NSPLIT*LPC_N_MAX];
+  int16_t cb_index[CB_NSTAGES*(NASUB_MAX+1)];  /* First CB_NSTAGES values contains extra CB index */
+  int16_t gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
+  int16_t idxForMax;
+  int16_t state_first;
+  int16_t idxVec[STATE_SHORT_LEN_30MS];
+  int16_t firstbits;
+  int16_t startIdx;
 } iLBC_bits;
 
 /* type definition encoder instance */
 typedef struct iLBC_Enc_Inst_t_ {
 
   /* flag for frame size mode */
-  WebRtc_Word16 mode;
+  int16_t mode;
 
   /* basic parameters for different frame sizes */
-  WebRtc_Word16 blockl;
-  WebRtc_Word16 nsub;
-  WebRtc_Word16 nasub;
-  WebRtc_Word16 no_of_bytes, no_of_words;
-  WebRtc_Word16 lpc_n;
-  WebRtc_Word16 state_short_len;
+  int16_t blockl;
+  int16_t nsub;
+  int16_t nasub;
+  int16_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  int16_t state_short_len;
 
   /* analysis filter state */
-  WebRtc_Word16 anaMem[LPC_FILTERORDER];
+  int16_t anaMem[LPC_FILTERORDER];
 
   /* Fix-point old lsf parameters for interpolation */
-  WebRtc_Word16 lsfold[LPC_FILTERORDER];
-  WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
+  int16_t lsfold[LPC_FILTERORDER];
+  int16_t lsfdeqold[LPC_FILTERORDER];
 
   /* signal buffer for LP analysis */
-  WebRtc_Word16 lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+  int16_t lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
 
   /* state of input HP filter */
-  WebRtc_Word16 hpimemx[2];
-  WebRtc_Word16 hpimemy[4];
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
 
 #ifdef SPLIT_10MS
-  WebRtc_Word16 weightdenumbuf[66];
-  WebRtc_Word16 past_samples[160];
-  WebRtc_UWord16 bytes[25];
-  WebRtc_Word16 section;
-  WebRtc_Word16 Nfor_flag;
-  WebRtc_Word16 Nback_flag;
-  WebRtc_Word16 start_pos;
-  WebRtc_Word16 diff;
+  int16_t weightdenumbuf[66];
+  int16_t past_samples[160];
+  uint16_t bytes[25];
+  int16_t section;
+  int16_t Nfor_flag;
+  int16_t Nback_flag;
+  int16_t start_pos;
+  int16_t diff;
 #endif
 
 } iLBC_Enc_Inst_t;
@@ -173,46 +173,46 @@
 typedef struct iLBC_Dec_Inst_t_ {
 
   /* flag for frame size mode */
-  WebRtc_Word16 mode;
+  int16_t mode;
 
   /* basic parameters for different frame sizes */
-  WebRtc_Word16 blockl;
-  WebRtc_Word16 nsub;
-  WebRtc_Word16 nasub;
-  WebRtc_Word16 no_of_bytes, no_of_words;
-  WebRtc_Word16 lpc_n;
-  WebRtc_Word16 state_short_len;
+  int16_t blockl;
+  int16_t nsub;
+  int16_t nasub;
+  int16_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  int16_t state_short_len;
 
   /* synthesis filter state */
-  WebRtc_Word16 syntMem[LPC_FILTERORDER];
+  int16_t syntMem[LPC_FILTERORDER];
 
   /* old LSF for interpolation */
-  WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
+  int16_t lsfdeqold[LPC_FILTERORDER];
 
   /* pitch lag estimated in enhancer and used in PLC */
   int last_lag;
 
   /* PLC state information */
   int consPLICount, prev_enh_pl;
-  WebRtc_Word16 perSquare;
+  int16_t perSquare;
 
-  WebRtc_Word16 prevScale, prevPLI;
-  WebRtc_Word16 prevLag, prevLpc[LPC_FILTERORDER+1];
-  WebRtc_Word16 prevResidual[NSUB_MAX*SUBL];
-  WebRtc_Word16 seed;
+  int16_t prevScale, prevPLI;
+  int16_t prevLag, prevLpc[LPC_FILTERORDER+1];
+  int16_t prevResidual[NSUB_MAX*SUBL];
+  int16_t seed;
 
   /* previous synthesis filter parameters */
 
-  WebRtc_Word16 old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+  int16_t old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
 
   /* state of output HP filter */
-  WebRtc_Word16 hpimemx[2];
-  WebRtc_Word16 hpimemy[4];
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
 
   /* enhancer state information */
   int use_enhancer;
-  WebRtc_Word16 enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
-  WebRtc_Word16 enh_period[ENH_NBLOCKS_TOT];
+  int16_t enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
+  int16_t enh_period[ENH_NBLOCKS_TOT];
 
 } iLBC_Dec_Inst_t;
 
diff --git a/modules/audio_coding/codecs/ilbc/do_plc.c b/modules/audio_coding/codecs/ilbc/do_plc.c
index 0dfae2b..c0f5368 100644
--- a/modules/audio_coding/codecs/ilbc/do_plc.c
+++ b/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -27,36 +27,36 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DoThePlc(
-    WebRtc_Word16 *PLCresidual,  /* (o) concealed residual */
-    WebRtc_Word16 *PLClpc,    /* (o) concealed LP parameters */
-    WebRtc_Word16 PLI,     /* (i) packet loss indicator
+    int16_t *PLCresidual,  /* (o) concealed residual */
+    int16_t *PLClpc,    /* (o) concealed LP parameters */
+    int16_t PLI,     /* (i) packet loss indicator
                                                            0 - no PL, 1 = PL */
-    WebRtc_Word16 *decresidual,  /* (i) decoded residual */
-    WebRtc_Word16 *lpc,    /* (i) decoded LPC (only used for no PL) */
-    WebRtc_Word16 inlag,    /* (i) pitch lag */
+    int16_t *decresidual,  /* (i) decoded residual */
+    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
+    int16_t inlag,    /* (i) pitch lag */
     iLBC_Dec_Inst_t *iLBCdec_inst
     /* (i/o) decoder instance */
                             ){
-  WebRtc_Word16 i, pick;
-  WebRtc_Word32 cross, ener, cross_comp, ener_comp = 0;
-  WebRtc_Word32 measure, maxMeasure, energy;
-  WebRtc_Word16 max, crossSquareMax, crossSquare;
-  WebRtc_Word16 j, lag, tmp1, tmp2, randlag;
-  WebRtc_Word16 shift1, shift2, shift3, shiftMax;
-  WebRtc_Word16 scale3;
-  WebRtc_Word16 corrLen;
-  WebRtc_Word32 tmpW32, tmp2W32;
-  WebRtc_Word16 use_gain;
-  WebRtc_Word16 tot_gain;
-  WebRtc_Word16 max_perSquare;
-  WebRtc_Word16 scale1, scale2;
-  WebRtc_Word16 totscale;
-  WebRtc_Word32 nom;
-  WebRtc_Word16 denom;
-  WebRtc_Word16 pitchfact;
-  WebRtc_Word16 use_lag;
+  int16_t i, pick;
+  int32_t cross, ener, cross_comp, ener_comp = 0;
+  int32_t measure, maxMeasure, energy;
+  int16_t max, crossSquareMax, crossSquare;
+  int16_t j, lag, tmp1, tmp2, randlag;
+  int16_t shift1, shift2, shift3, shiftMax;
+  int16_t scale3;
+  int16_t corrLen;
+  int32_t tmpW32, tmp2W32;
+  int16_t use_gain;
+  int16_t tot_gain;
+  int16_t max_perSquare;
+  int16_t scale1, scale2;
+  int16_t totscale;
+  int32_t nom;
+  int16_t denom;
+  int16_t pitchfact;
+  int16_t use_lag;
   int ind;
-  WebRtc_Word16 randvec[BLOCKL_MAX];
+  int16_t randvec[BLOCKL_MAX];
 
   /* Packet Loss */
   if (PLI == 1) {
@@ -70,7 +70,7 @@
 
       /* Maximum 60 samples are correlated, preserve as high accuracy
          as possible without getting overflow */
-      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (WebRtc_Word16)iLBCdec_inst->blockl);
+      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (int16_t)iLBCdec_inst->blockl);
       scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
       if (scale3 < 0) {
         scale3 = 0;
@@ -92,7 +92,7 @@
 
       /* Normalize and store cross^2 and the number of shifts */
       shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
-      crossSquareMax = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross, -shiftMax),
+      crossSquareMax = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross, -shiftMax),
                                                                 WEBRTC_SPL_SHIFT_W32(cross, -shiftMax), 15);
 
       for (j=inlag-2;j<=inlag+3;j++) {
@@ -103,7 +103,7 @@
            this lag is better or not. To avoid the division,
            do a cross multiplication */
         shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
-        crossSquare = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1),
+        crossSquare = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1),
                                                                WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1), 15);
 
         shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
@@ -147,25 +147,25 @@
                                             corrLen, scale3);
 
       if ((tmp2W32>0)&&(ener_comp>0)) {
-        /* norm energies to WebRtc_Word16, compute the product of the energies and
-           use the upper WebRtc_Word16 as the denominator */
+        /* norm energies to int16_t, compute the product of the energies and
+           use the upper int16_t as the denominator */
 
-        scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp2W32)-16;
-        tmp1=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp2W32)-16;
+        tmp1=(int16_t)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
 
-        scale2=(WebRtc_Word16)WebRtcSpl_NormW32(ener)-16;
-        tmp2=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, scale2);
-        denom=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp1, tmp2, 16); /* denom in Q(scale1+scale2-16) */
+        scale2=(int16_t)WebRtcSpl_NormW32(ener)-16;
+        tmp2=(int16_t)WEBRTC_SPL_SHIFT_W32(ener, scale2);
+        denom=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp1, tmp2, 16); /* denom in Q(scale1+scale2-16) */
 
         /* Square the cross correlation and norm it such that max_perSquare
            will be in Q15 after the division */
 
         totscale = scale1+scale2-1;
-        tmp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
-        tmp2 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
+        tmp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
+        tmp2 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
 
         nom = WEBRTC_SPL_MUL_16_16(tmp1, tmp2);
-        max_perSquare = (WebRtc_Word16)WebRtcSpl_DivW32W16(nom, denom);
+        max_perSquare = (int16_t)WebRtcSpl_DivW32W16(nom, denom);
 
       } else {
         max_perSquare = 0;
@@ -209,10 +209,10 @@
         ind--;
       }
       /* pitch fact is approximated by first order */
-      tmpW32 = (WebRtc_Word32)WebRtcIlbcfix_kPlcPitchFact[ind] +
+      tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] +
           WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kPlcPfSlope[ind], (max_perSquare-WebRtcIlbcfix_kPlcPerSqr[ind]), 11);
 
-      pitchfact = (WebRtc_Word16)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
+      pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
 
     } else { /* periodicity < 0.4 */
       pitchfact = 0;
@@ -230,8 +230,8 @@
     for (i=0; i<iLBCdec_inst->blockl; i++) {
 
       /* noise component -  52 < randlagFIX < 117 */
-      iLBCdec_inst->seed = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iLBCdec_inst->seed, 31821)+(WebRtc_Word32)13849);
-      randlag = 53 + (WebRtc_Word16)(iLBCdec_inst->seed & 63);
+      iLBCdec_inst->seed = (int16_t)(WEBRTC_SPL_MUL_16_16(iLBCdec_inst->seed, 31821)+(int32_t)13849);
+      randlag = 53 + (int16_t)(iLBCdec_inst->seed & 63);
 
       pick = i - randlag;
 
@@ -254,16 +254,16 @@
       if (i<80) {
         tot_gain=use_gain;
       } else if (i<160) {
-        tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(31130, use_gain, 15); /* 0.95*use_gain */
+        tot_gain=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(31130, use_gain, 15); /* 0.95*use_gain */
       } else {
-        tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(29491, use_gain, 15); /* 0.9*use_gain */
+        tot_gain=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(29491, use_gain, 15); /* 0.9*use_gain */
       }
 
 
       /* mix noise and pitch repeatition */
 
-      PLCresidual[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tot_gain,
-                                                                (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( (WEBRTC_SPL_MUL_16_16(pitchfact, PLCresidual[i]) +
+      PLCresidual[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tot_gain,
+                                                                (int16_t)WEBRTC_SPL_RSHIFT_W32( (WEBRTC_SPL_MUL_16_16(pitchfact, PLCresidual[i]) +
                                                                                                        WEBRTC_SPL_MUL_16_16((32767-pitchfact), randvec[i]) + 16384),
                                                                                                       15),
                                                                 15);
@@ -276,7 +276,7 @@
     }
 
     /* less than 30 dB, use only noise */
-    if (energy < (WEBRTC_SPL_SHIFT_W32(((WebRtc_Word32)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) {
+    if (energy < (WEBRTC_SPL_SHIFT_W32(((int32_t)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) {
       energy = 0;
       for (i=0; i<iLBCdec_inst->blockl; i++) {
         PLCresidual[i] = randvec[i];
diff --git a/modules/audio_coding/codecs/ilbc/do_plc.h b/modules/audio_coding/codecs/ilbc/do_plc.h
index c5bcc52..15e68ec 100644
--- a/modules/audio_coding/codecs/ilbc/do_plc.h
+++ b/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -27,13 +27,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DoThePlc(
-    WebRtc_Word16 *PLCresidual,  /* (o) concealed residual */
-    WebRtc_Word16 *PLClpc,    /* (o) concealed LP parameters */
-    WebRtc_Word16 PLI,     /* (i) packet loss indicator
+    int16_t *PLCresidual,  /* (o) concealed residual */
+    int16_t *PLClpc,    /* (o) concealed LP parameters */
+    int16_t PLI,     /* (i) packet loss indicator
                                                            0 - no PL, 1 = PL */
-    WebRtc_Word16 *decresidual,  /* (i) decoded residual */
-    WebRtc_Word16 *lpc,    /* (i) decoded LPC (only used for no PL) */
-    WebRtc_Word16 inlag,    /* (i) pitch lag */
+    int16_t *decresidual,  /* (i) decoded residual */
+    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
+    int16_t inlag,    /* (i) pitch lag */
     iLBC_Dec_Inst_t *iLBCdec_inst
     /* (i/o) decoder instance */
                             );
diff --git a/modules/audio_coding/codecs/ilbc/encode.c b/modules/audio_coding/codecs/ilbc/encode.c
index d3d3ba8..75d1672 100644
--- a/modules/audio_coding/codecs/ilbc/encode.c
+++ b/modules/audio_coding/codecs/ilbc/encode.c
@@ -41,38 +41,38 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_EncodeImpl(
-    WebRtc_UWord16 *bytes,     /* (o) encoded data bits iLBC */
-    const WebRtc_Word16 *block, /* (i) speech vector to encode */
+    uint16_t *bytes,     /* (o) encoded data bits iLBC */
+    const int16_t *block, /* (i) speech vector to encode */
     iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
                                      state */
                           ){
   int n, meml_gotten, Nfor, Nback;
-  WebRtc_Word16 diff, start_pos;
+  int16_t diff, start_pos;
   int index;
   int subcount, subframe;
-  WebRtc_Word16 start_count, end_count;
-  WebRtc_Word16 *residual;
-  WebRtc_Word32 en1, en2;
-  WebRtc_Word16 scale, max;
-  WebRtc_Word16 *syntdenum;
-  WebRtc_Word16 *decresidual;
-  WebRtc_Word16 *reverseResidual;
-  WebRtc_Word16 *reverseDecresidual;
+  int16_t start_count, end_count;
+  int16_t *residual;
+  int32_t en1, en2;
+  int16_t scale, max;
+  int16_t *syntdenum;
+  int16_t *decresidual;
+  int16_t *reverseResidual;
+  int16_t *reverseDecresidual;
   /* Stack based */
-  WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
-  WebRtc_Word16 dataVec[BLOCKL_MAX + LPC_FILTERORDER];
-  WebRtc_Word16 memVec[CB_MEML+CB_FILTERLEN];
-  WebRtc_Word16 bitsMemory[sizeof(iLBC_bits)/sizeof(WebRtc_Word16)];
+  int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+  int16_t dataVec[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t memVec[CB_MEML+CB_FILTERLEN];
+  int16_t bitsMemory[sizeof(iLBC_bits)/sizeof(int16_t)];
   iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
 
 
 #ifdef SPLIT_10MS
-  WebRtc_Word16 *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
-  WebRtc_Word16 last_bit;
+  int16_t *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
+  int16_t last_bit;
 #endif
 
-  WebRtc_Word16 *data = &dataVec[LPC_FILTERORDER];
-  WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN];
+  int16_t *data = &dataVec[LPC_FILTERORDER];
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];
 
   /* Reuse som buffers to save stack memory */
   residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
@@ -83,8 +83,8 @@
 
 #ifdef SPLIT_10MS
 
-  WebRtcSpl_MemSetW16 (  (WebRtc_Word16 *) iLBCbits_inst, 0,
-                         (WebRtc_Word16) (sizeof(iLBC_bits) / sizeof(WebRtc_Word16))  );
+  WebRtcSpl_MemSetW16 (  (int16_t *) iLBCbits_inst, 0,
+                         (int16_t) (sizeof(iLBC_bits) / sizeof(int16_t))  );
 
   start_pos = iLBCenc_inst->start_pos;
   diff = iLBCenc_inst->diff;
@@ -121,7 +121,7 @@
 #endif
 
     /* high pass filtering of input signal and scale down the residual (*0.5) */
-    WebRtcIlbcfix_HpInput(data, (WebRtc_Word16*)WebRtcIlbcfix_kHpInCoefs,
+    WebRtcIlbcfix_HpInput(data, (int16_t*)WebRtcIlbcfix_kHpInCoefs,
                           iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
                           iLBCenc_inst->blockl);
 
@@ -191,7 +191,7 @@
 
       /* setup memory */
 
-      WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
+      WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
       WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
                             decresidual+start_pos, iLBCenc_inst->state_short_len);
 
@@ -222,7 +222,7 @@
 
       meml_gotten = iLBCenc_inst->state_short_len;
       WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
-      WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
+      WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
 
       /* encode subframes */
       WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
@@ -325,7 +325,7 @@
     }
 #else
     start_count = 0;
-    end_count = (WebRtc_Word16)Nfor;
+    end_count = (int16_t)Nfor;
 #endif
 
     /* loop over subframes to encode */
@@ -339,7 +339,7 @@
                              &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
                              mem, MEM_LF_TBL, SUBL,
                              &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
-                             (WebRtc_Word16)subcount);
+                             (int16_t)subcount);
 
       /* construct decoded vector */
 
@@ -396,7 +396,7 @@
     }
 
     WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
 
 #ifdef SPLIT_10MS
     if (iLBCenc_inst->Nback_flag > 0)
@@ -433,7 +433,7 @@
     }
 #else
     start_count = 0;
-    end_count = (WebRtc_Word16)Nback;
+    end_count = (int16_t)Nback;
 #endif
 
     /* loop over subframes to encode */
@@ -446,7 +446,7 @@
                              iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
                              mem, MEM_LF_TBL, SUBL,
                              &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
-                             (WebRtc_Word16)subcount);
+                             (int16_t)subcount);
 
       /* construct decoded vector */
 
diff --git a/modules/audio_coding/codecs/ilbc/encode.h b/modules/audio_coding/codecs/ilbc/encode.h
index b7d93d7..5a47c58 100644
--- a/modules/audio_coding/codecs/ilbc/encode.h
+++ b/modules/audio_coding/codecs/ilbc/encode.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_EncodeImpl(
-    WebRtc_UWord16 *bytes,     /* (o) encoded data bits iLBC */
-    const WebRtc_Word16 *block, /* (i) speech vector to encode */
+    uint16_t *bytes,     /* (o) encoded data bits iLBC */
+    const int16_t *block, /* (i) speech vector to encode */
     iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
                                            state */
                           );
diff --git a/modules/audio_coding/codecs/ilbc/energy_inverse.c b/modules/audio_coding/codecs/ilbc/energy_inverse.c
index d56069b..a6b1c75 100644
--- a/modules/audio_coding/codecs/ilbc/energy_inverse.c
+++ b/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -21,13 +21,13 @@
 #include "energy_inverse.h"
 
 void WebRtcIlbcfix_EnergyInverse(
-    WebRtc_Word16 *energy,    /* (i/o) Energy and inverse
+    int16_t *energy,    /* (i/o) Energy and inverse
                                                            energy (in Q29) */
     int noOfEnergies)  /* (i)   The length of the energy
                                    vector */
 {
-  WebRtc_Word32 Nom=(WebRtc_Word32)0x1FFFFFFF;
-  WebRtc_Word16 *energyPtr;
+  int32_t Nom=(int32_t)0x1FFFFFFF;
+  int16_t *energyPtr;
   int i;
 
   /* Set the minimum energy value to 16384 to avoid overflow */
@@ -40,7 +40,7 @@
   /* Calculate inverse energy in Q29 */
   energyPtr=energy;
   for (i=0; i<noOfEnergies; i++) {
-    (*energyPtr) = (WebRtc_Word16)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
+    (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
     energyPtr++;
   }
 }
diff --git a/modules/audio_coding/codecs/ilbc/energy_inverse.h b/modules/audio_coding/codecs/ilbc/energy_inverse.h
index db13589..7bb6721 100644
--- a/modules/audio_coding/codecs/ilbc/energy_inverse.h
+++ b/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -24,7 +24,7 @@
 /* Inverses the in vector in into Q29 domain */
 
 void WebRtcIlbcfix_EnergyInverse(
-    WebRtc_Word16 *energy,     /* (i/o) Energy and inverse
+    int16_t *energy,     /* (i/o) Energy and inverse
                                                                    energy (in Q29) */
     int noOfEnergies);   /* (i)   The length of the energy
                                    vector */
diff --git a/modules/audio_coding/codecs/ilbc/enh_upsample.c b/modules/audio_coding/codecs/ilbc/enh_upsample.c
index 3343816..91f3970 100644
--- a/modules/audio_coding/codecs/ilbc/enh_upsample.c
+++ b/modules/audio_coding/codecs/ilbc/enh_upsample.c
@@ -24,13 +24,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_EnhUpsample(
-    WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
-    WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
+    int32_t *useq1, /* (o) upsampled output sequence */
+    int16_t *seq1 /* (i) unupsampled sequence */
                                 ){
   int j;
-  WebRtc_Word32 *pu1, *pu11;
-  WebRtc_Word16 *ps, *w16tmp;
-  const WebRtc_Word16 *pp;
+  int32_t *pu1, *pu11;
+  int16_t *ps, *w16tmp;
+  const int16_t *pp;
 
   /* filtering: filter overhangs left side of sequence */
   pu1=useq1;
diff --git a/modules/audio_coding/codecs/ilbc/enh_upsample.h b/modules/audio_coding/codecs/ilbc/enh_upsample.h
index 53534cc..00bb28b 100644
--- a/modules/audio_coding/codecs/ilbc/enh_upsample.h
+++ b/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_EnhUpsample(
-    WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
-    WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
+    int32_t *useq1, /* (o) upsampled output sequence */
+    int16_t *seq1 /* (i) unupsampled sequence */
                                 );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/enhancer.c b/modules/audio_coding/codecs/ilbc/enhancer.c
index b8f3335..38c3de3 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer.c
+++ b/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -27,16 +27,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Enhancer(
-    WebRtc_Word16 *odata,   /* (o) smoothed block, dimension blockl */
-    WebRtc_Word16 *idata,   /* (i) data buffer used for enhancing */
-    WebRtc_Word16 idatal,   /* (i) dimension idata */
-    WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
-    WebRtc_Word16 *period,   /* (i) pitch period array (pitch bward-in time) */
-    WebRtc_Word16 *plocs,   /* (i) locations where period array values valid */
-    WebRtc_Word16 periodl   /* (i) dimension of period and plocs */
+    int16_t *odata,   /* (o) smoothed block, dimension blockl */
+    int16_t *idata,   /* (i) data buffer used for enhancing */
+    int16_t idatal,   /* (i) dimension idata */
+    int16_t centerStartPos, /* (i) first sample current block within idata */
+    int16_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    int16_t *plocs,   /* (i) locations where period array values valid */
+    int16_t periodl   /* (i) dimension of period and plocs */
                             ){
   /* Stack based */
-  WebRtc_Word16 surround[ENH_BLOCKL];
+  int16_t surround[ENH_BLOCKL];
 
   WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
 
diff --git a/modules/audio_coding/codecs/ilbc/enhancer.h b/modules/audio_coding/codecs/ilbc/enhancer.h
index e14f559..83f48b0 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer.h
+++ b/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -27,13 +27,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Enhancer(
-    WebRtc_Word16 *odata,   /* (o) smoothed block, dimension blockl */
-    WebRtc_Word16 *idata,   /* (i) data buffer used for enhancing */
-    WebRtc_Word16 idatal,   /* (i) dimension idata */
-    WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
-    WebRtc_Word16 *period,   /* (i) pitch period array (pitch bward-in time) */
-    WebRtc_Word16 *plocs,   /* (i) locations where period array values valid */
-    WebRtc_Word16 periodl   /* (i) dimension of period and plocs */
+    int16_t *odata,   /* (o) smoothed block, dimension blockl */
+    int16_t *idata,   /* (i) data buffer used for enhancing */
+    int16_t idatal,   /* (i) dimension idata */
+    int16_t centerStartPos, /* (i) first sample current block within idata */
+    int16_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    int16_t *plocs,   /* (i) locations where period array values valid */
+    int16_t periodl   /* (i) dimension of period and plocs */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index 61b71d1..809f48f 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -29,41 +29,41 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
-    WebRtc_Word16 *out,     /* (o) enhanced signal */
-    WebRtc_Word16 *in,      /* (i) unenhanced signal */
+    int16_t *out,     /* (o) enhanced signal */
+    int16_t *in,      /* (i) unenhanced signal */
     iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
                                         ){
   int iblock;
   int lag=20, tlag=20;
   int inLen=iLBCdec_inst->blockl+120;
-  WebRtc_Word16 scale, scale1, plc_blockl;
-  WebRtc_Word16 *enh_buf, *enh_period;
-  WebRtc_Word32 tmp1, tmp2, max, new_blocks;
-  WebRtc_Word16 *enh_bufPtr1;
+  int16_t scale, scale1, plc_blockl;
+  int16_t *enh_buf, *enh_period;
+  int32_t tmp1, tmp2, max, new_blocks;
+  int16_t *enh_bufPtr1;
   int i, k;
-  WebRtc_Word16 EnChange;
-  WebRtc_Word16 SqrtEnChange;
-  WebRtc_Word16 inc;
-  WebRtc_Word16 win;
-  WebRtc_Word16 *tmpW16ptr;
-  WebRtc_Word16 startPos;
-  WebRtc_Word16 *plc_pred;
-  WebRtc_Word16 *target, *regressor;
-  WebRtc_Word16 max16;
+  int16_t EnChange;
+  int16_t SqrtEnChange;
+  int16_t inc;
+  int16_t win;
+  int16_t *tmpW16ptr;
+  int16_t startPos;
+  int16_t *plc_pred;
+  int16_t *target, *regressor;
+  int16_t max16;
   int shifts;
-  WebRtc_Word32 ener;
-  WebRtc_Word16 enerSh;
-  WebRtc_Word16 corrSh;
-  WebRtc_Word16 ind, sh;
-  WebRtc_Word16 start, stop;
+  int32_t ener;
+  int16_t enerSh;
+  int16_t corrSh;
+  int16_t ind, sh;
+  int16_t start, stop;
   /* Stack based */
-  WebRtc_Word16 totsh[3];
-  WebRtc_Word16 downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
-  WebRtc_Word32 corr32[50];
-  WebRtc_Word32 corrmax[3];
-  WebRtc_Word16 corr16[3];
-  WebRtc_Word16 en16[3];
-  WebRtc_Word16 lagmax[3];
+  int16_t totsh[3];
+  int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
+  int32_t corr32[50];
+  int32_t corrmax[3];
+  int16_t corr16[3];
+  int16_t en16[3];
+  int16_t lagmax[3];
 
   plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
                               downsampled are non overlapping */
@@ -97,10 +97,10 @@
 
   k=WebRtcSpl_DownsampleFast(
       enh_buf+ENH_BUFL-inLen,    /* Input samples */
-      (WebRtc_Word16)(inLen+ENH_BUFL_FILTEROVERHEAD),
+      (int16_t)(inLen+ENH_BUFL_FILTEROVERHEAD),
       downsampled,
-      (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(inLen, 1),
-      (WebRtc_Word16*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
+      (int16_t)WEBRTC_SPL_RSHIFT_W16(inLen, 1),
+      (int16_t*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
       FILTERORDER_DS_PLUS1,    /* Length of filter (order-1) */
       FACTOR_DS,
       DELAY_DS);
@@ -115,13 +115,13 @@
 
     /* scaling */
     max16=WebRtcSpl_MaxAbsValueW16(&regressor[-50],
-                                   (WebRtc_Word16)(ENH_BLOCKL_HALF+50-1));
+                                   (int16_t)(ENH_BLOCKL_HALF+50-1));
     shifts = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max16, max16)) - 25;
     shifts = WEBRTC_SPL_MAX(0, shifts);
 
     /* compute cross correlation */
     WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               ENH_BLOCKL_HALF, 50, (WebRtc_Word16)shifts, -1);
+                               ENH_BLOCKL_HALF, 50, (int16_t)shifts, -1);
 
     /* Find 3 highest correlations that should be compared for the
        highest (corr*corr)/ener */
@@ -147,10 +147,10 @@
                                            &regressor[-lagmax[i]],
                                            ENH_BLOCKL_HALF, shifts);
       enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
-      corr16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
-      corr16[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(corr16[i],
+      corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
+      corr16[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(corr16[i],
                                                            corr16[i], 16);
-      en16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
+      en16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
       totsh[i] = enerSh - WEBRTC_SPL_LSHIFT_W32(corrSh, 1);
     }
 
@@ -176,7 +176,7 @@
 
     /* Store the estimated lag in the non-downsampled domain */
     enh_period[ENH_NBLOCKS_TOT-new_blocks+iblock] =
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(lag, 8);
+        (int16_t)WEBRTC_SPL_MUL_16_16(lag, 8);
 
     /* Store the estimated lag for backward PLC */
     if (iLBCdec_inst->prev_enh_pl==1) {
@@ -203,7 +203,7 @@
     regressor=in+tlag-1;
 
     /* scaling */
-    max16=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(plc_blockl+3-1));
+    max16=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(plc_blockl+3-1));
     if (max16>5000)
       shifts=2;
     else
@@ -211,7 +211,7 @@
 
     /* compute cross correlation */
     WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               plc_blockl, 3, (WebRtc_Word16)shifts, 1);
+                               plc_blockl, 3, (int16_t)shifts, 1);
 
     /* find lag */
     lag=WebRtcSpl_MaxIndexW32(corr32, 3);
@@ -257,7 +257,7 @@
           &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
       max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
       max = WEBRTC_SPL_MAX(max, max16);
-      scale=22-(WebRtc_Word16)WebRtcSpl_NormW32(max);
+      scale=22-(int16_t)WebRtcSpl_NormW32(max);
       scale=WEBRTC_SPL_MAX(scale,0);
 
       tmp2 = WebRtcSpl_DotProductWithScale(
@@ -273,21 +273,21 @@
            Calculate EnChange=tmp2/tmp1 in Q16
         */
 
-        scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp1);
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp1);
         tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
 
         tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
-        EnChange = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp2,
-                                                      (WebRtc_Word16)tmp1);
+        EnChange = (int16_t)WebRtcSpl_DivW32W16(tmp2,
+                                                      (int16_t)tmp1);
 
         /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
-        SqrtEnChange = (WebRtc_Word16)WebRtcSpl_SqrtFloor(
-            WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)EnChange, 14));
+        SqrtEnChange = (int16_t)WebRtcSpl_SqrtFloor(
+            WEBRTC_SPL_LSHIFT_W32((int32_t)EnChange, 14));
 
 
         /* Multiply first part of vector with 2*SqrtEnChange */
         WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange,
-                              (WebRtc_Word16)(plc_blockl-16), 14);
+                              (int16_t)(plc_blockl-16), 14);
 
         /* Calculate increase parameter for window part (16 last samples) */
         /* (1-2*SqrtEnChange)/16 in Q15 */
@@ -297,7 +297,7 @@
         tmpW16ptr=&plc_pred[plc_blockl-16];
 
         for (i=16;i>0;i--) {
-          (*tmpW16ptr)=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+          (*tmpW16ptr)=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
               (*tmpW16ptr), (SqrtEnChange+(win>>1)), 14);
           /* multiply by (2.0*SqrtEnChange+win) */
 
@@ -320,13 +320,13 @@
       for (i=0; i<plc_blockl; i++) {
         win+=inc;
         *enh_bufPtr1 =
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*enh_bufPtr1), win, 14);
-        *enh_bufPtr1 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT((*enh_bufPtr1), win, 14);
+        *enh_bufPtr1 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
                 (16384-win), plc_pred[plc_blockl-1-i], 14);
         enh_bufPtr1--;
       }
     } else {
-      WebRtc_Word16 *synt = &downsampled[LPC_FILTERORDER];
+      int16_t *synt = &downsampled[LPC_FILTERORDER];
 
       enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
       WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
@@ -344,24 +344,24 @@
           synt,
           &iLBCdec_inst->old_syntdenum[
                                        (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
-                                       LPC_FILTERORDER+1, (WebRtc_Word16)lag);
+                                       LPC_FILTERORDER+1, (int16_t)lag);
 
       WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
                             LPC_FILTERORDER);
-      WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
                              iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
-                             (WebRtc_Word16)lag);
+                             (int16_t)lag);
       WebRtcSpl_FilterARFastQ12(
           enh_bufPtr1, synt,
           &iLBCdec_inst->old_syntdenum[
                                        (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
-                                       LPC_FILTERORDER+1, (WebRtc_Word16)lag);
+                                       LPC_FILTERORDER+1, (int16_t)lag);
 
       WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
                             LPC_FILTERORDER);
-      WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
                              iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
-                             (WebRtc_Word16)lag);
+                             (int16_t)lag);
     }
   }
 
@@ -372,9 +372,9 @@
     WebRtcIlbcfix_Enhancer(out+WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL),
                            enh_buf,
                            ENH_BUFL,
-                           (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL)+startPos),
+                           (int16_t)(WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL)+startPos),
                            enh_period,
-                           (WebRtc_Word16*)WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+                           (int16_t*)WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
   }
 
   return (lag);
diff --git a/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/modules/audio_coding/codecs/ilbc/enhancer_interface.h
index 37b27e2..f6ebed6 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer_interface.h
+++ b/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
-    WebRtc_Word16 *out,     /* (o) enhanced signal */
-    WebRtc_Word16 *in,      /* (i) unenhanced signal */
+    int16_t *out,     /* (o) enhanced signal */
+    int16_t *in,      /* (i) unenhanced signal */
     iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
                                         );
 
diff --git a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
index 7cece26..aa8170c 100644
--- a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
+++ b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_FilteredCbVecs(
-    WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
-    WebRtc_Word16 *CBmem,  /* (i) Codebook memory that is filtered to create a
+    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
                                            second CB section */
     int lMem,  /* (i) Length of codebook memory */
-    WebRtc_Word16 samples    /* (i) Number of samples to filter */
+    int16_t samples    /* (i) Number of samples to filter */
                                   ) {
 
   /* Set up the memory, start with zero state */
@@ -42,7 +42,7 @@
 
   WebRtcSpl_FilterMAFastQ12(
       CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
-      (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
+      (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
 
   return;
 }
diff --git a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
index c502e8f..99e89a0 100644
--- a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
+++ b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -28,11 +28,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_FilteredCbVecs(
-    WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
-    WebRtc_Word16 *CBmem,  /* (i) Codebook memory that is filtered to create a
+    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
                                            second CB section */
     int lMem,  /* (i) Length of codebook memory */
-    WebRtc_Word16 samples    /* (i) Number of samples to filter */
+    int16_t samples    /* (i) Number of samples to filter */
                                   );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/frame_classify.c b/modules/audio_coding/codecs/ilbc/frame_classify.c
index ea3675e..c2c6f75 100644
--- a/modules/audio_coding/codecs/ilbc/frame_classify.c
+++ b/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -23,19 +23,19 @@
  *  Classification of subframes to localize start state
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
+int16_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
     iLBC_Enc_Inst_t *iLBCenc_inst,
     /* (i/o) the encoder state structure */
-    WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
+    int16_t *residualFIX /* (i) lpc residual signal */
                                                 ){
-  WebRtc_Word16 max, scale;
-  WebRtc_Word32 ssqEn[NSUB_MAX-1];
-  WebRtc_Word16 *ssqPtr;
-  WebRtc_Word32 *seqEnPtr;
-  WebRtc_Word32 maxW32;
-  WebRtc_Word16 scale1;
-  WebRtc_Word16 pos;
+  int16_t max, scale;
+  int32_t ssqEn[NSUB_MAX-1];
+  int16_t *ssqPtr;
+  int32_t *seqEnPtr;
+  int32_t maxW32;
+  int16_t scale1;
+  int16_t pos;
   int n;
 
   /*
@@ -62,7 +62,7 @@
   }
 
   /* Scale to maximum 20 bits in order to allow for the 11 bit window */
-  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1));
+  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1));
   scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
   scale1 = WEBRTC_SPL_MAX(0, scale);
 
@@ -71,9 +71,9 @@
   */
   seqEnPtr=ssqEn;
   if (iLBCenc_inst->mode==20) {
-    ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
   } else {
-    ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin;
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin;
   }
   for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
     (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
@@ -82,7 +82,7 @@
   }
 
   /* Extract the best choise of start state */
-  pos = WebRtcSpl_MaxIndexW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1)) + 1;
+  pos = WebRtcSpl_MaxIndexW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1)) + 1;
 
   return(pos);
 }
diff --git a/modules/audio_coding/codecs/ilbc/frame_classify.h b/modules/audio_coding/codecs/ilbc/frame_classify.h
index faf4666..ba9b9fe 100644
--- a/modules/audio_coding/codecs/ilbc/frame_classify.h
+++ b/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -19,11 +19,11 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
 
-WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
+int16_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
     iLBC_Enc_Inst_t *iLBCenc_inst,
     /* (i/o) the encoder state structure */
-    WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
+    int16_t *residualFIX /* (i) lpc residual signal */
                                                 );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/gain_dequant.c b/modules/audio_coding/codecs/ilbc/gain_dequant.c
index 9450a80..8570c8b 100644
--- a/modules/audio_coding/codecs/ilbc/gain_dequant.c
+++ b/modules/audio_coding/codecs/ilbc/gain_dequant.c
@@ -24,14 +24,14 @@
  *  residual
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_GainDequant(
+int16_t WebRtcIlbcfix_GainDequant(
     /* (o) quantized gain value (Q14) */
-    WebRtc_Word16 index, /* (i) quantization index */
-    WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
-    WebRtc_Word16 stage /* (i) The stage of the search */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage /* (i) The stage of the search */
                                                 ){
-  WebRtc_Word16 scale;
-  const WebRtc_Word16 *gain;
+  int16_t scale;
+  const int16_t *gain;
 
   /* obtain correct scale factor */
 
@@ -41,5 +41,5 @@
   /* select the quantization table and return the decoded value */
   gain = WebRtcIlbcfix_kGain[stage];
 
-  return((WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, gain[index])+8192)>>14));
+  return((int16_t)((WEBRTC_SPL_MUL_16_16(scale, gain[index])+8192)>>14));
 }
diff --git a/modules/audio_coding/codecs/ilbc/gain_dequant.h b/modules/audio_coding/codecs/ilbc/gain_dequant.h
index 28f2ceb..6bda066 100644
--- a/modules/audio_coding/codecs/ilbc/gain_dequant.h
+++ b/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -26,11 +26,11 @@
  *  residual
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_GainDequant(
+int16_t WebRtcIlbcfix_GainDequant(
     /* (o) quantized gain value (Q14) */
-    WebRtc_Word16 index, /* (i) quantization index */
-    WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
-    WebRtc_Word16 stage /* (i) The stage of the search */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage /* (i) The stage of the search */
                                          );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/gain_quant.c b/modules/audio_coding/codecs/ilbc/gain_quant.c
index bdf88a5..f7a8083 100644
--- a/modules/audio_coding/codecs/ilbc/gain_quant.c
+++ b/modules/audio_coding/codecs/ilbc/gain_quant.c
@@ -23,16 +23,16 @@
  *  quantizer for the gain in the gain-shape coding of residual
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
-    WebRtc_Word16 gain, /* (i) gain value Q14 */
-    WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
-    WebRtc_Word16 stage, /* (i) The stage of the search */
-    WebRtc_Word16 *index /* (o) quantization index */
+int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+    int16_t gain, /* (i) gain value Q14 */
+    int16_t maxIn, /* (i) maximum of gain value Q14 */
+    int16_t stage, /* (i) The stage of the search */
+    int16_t *index /* (o) quantization index */
                                         ) {
 
-  WebRtc_Word16 scale, returnVal, cblen;
-  WebRtc_Word32 gainW32, measure1, measure2;
-  const WebRtc_Word16 *cbPtr, *cb;
+  int16_t scale, returnVal, cblen;
+  int32_t gainW32, measure1, measure2;
+  const int16_t *cbPtr, *cb;
   int loc, noMoves, noChecks, i;
 
   /* ensure a lower bound (0.1) on the scaling factor */
@@ -48,7 +48,7 @@
 
   /* Multiply the gain with 2^14 to make the comparison
      easier and with higher precision */
-  gainW32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)gain, 14);
+  gainW32 = WEBRTC_SPL_LSHIFT_W32((int32_t)gain, 14);
 
   /* Do a binary search, starting in the middle of the CB
      loc - defines the current position in the table
@@ -99,7 +99,7 @@
   *index=loc;
 
   /* Calculate the quantized gain value (in Q14) */
-  returnVal=(WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, cb[loc])+8192)>>14);
+  returnVal=(int16_t)((WEBRTC_SPL_MUL_16_16(scale, cb[loc])+8192)>>14);
 
   /* return the quantized value */
   return(returnVal);
diff --git a/modules/audio_coding/codecs/ilbc/gain_quant.h b/modules/audio_coding/codecs/ilbc/gain_quant.h
index a2f0596..3954364 100644
--- a/modules/audio_coding/codecs/ilbc/gain_quant.h
+++ b/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -25,11 +25,11 @@
  *  quantizer for the gain in the gain-shape coding of residual
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
-    WebRtc_Word16 gain, /* (i) gain value Q14 */
-    WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
-    WebRtc_Word16 stage, /* (i) The stage of the search */
-    WebRtc_Word16 *index /* (o) quantization index */
+int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+    int16_t gain, /* (i) gain value Q14 */
+    int16_t maxIn, /* (i) maximum of gain value Q14 */
+    int16_t stage, /* (i) The stage of the search */
+    int16_t *index /* (o) quantization index */
                                        );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/modules/audio_coding/codecs/ilbc/get_cd_vec.c
index aba3e31..ca07b0a 100644
--- a/modules/audio_coding/codecs/ilbc/get_cd_vec.c
+++ b/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -25,16 +25,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_GetCbVec(
-    WebRtc_Word16 *cbvec,   /* (o) Constructed codebook vector */
-    WebRtc_Word16 *mem,   /* (i) Codebook buffer */
-    WebRtc_Word16 index,   /* (i) Codebook index */
-    WebRtc_Word16 lMem,   /* (i) Length of codebook buffer */
-    WebRtc_Word16 cbveclen   /* (i) Codebook vector length */
+    int16_t *cbvec,   /* (o) Constructed codebook vector */
+    int16_t *mem,   /* (i) Codebook buffer */
+    int16_t index,   /* (i) Codebook index */
+    int16_t lMem,   /* (i) Length of codebook buffer */
+    int16_t cbveclen   /* (i) Codebook vector length */
                             ){
-  WebRtc_Word16 k, base_size;
-  WebRtc_Word16 lag;
+  int16_t k, base_size;
+  int16_t lag;
   /* Stack based */
-  WebRtc_Word16 tempbuff2[SUBL+5];
+  int16_t tempbuff2[SUBL+5];
 
   /* Determine size of codebook sections */
 
@@ -58,7 +58,7 @@
 
     /* Calculate lag */
 
-    k=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (index-(lMem-cbveclen+1)))+cbveclen;
+    k=(int16_t)WEBRTC_SPL_MUL_16_16(2, (index-(lMem-cbveclen+1)))+cbveclen;
 
     lag=WEBRTC_SPL_RSHIFT_W16(k, 1);
 
@@ -70,7 +70,7 @@
 
   else {
 
-    WebRtc_Word16 memIndTest;
+    int16_t memIndTest;
 
     /* first non-interpolated vectors */
 
@@ -86,7 +86,7 @@
       /* do filtering to get the codebook vector */
 
       WebRtcSpl_FilterMAFastQ12(
-          &mem[memIndTest+4], cbvec, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
+          &mem[memIndTest+4], cbvec, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
           CB_FILTERLEN, cbveclen);
     }
 
@@ -99,8 +99,8 @@
 
       /* do filtering */
       WebRtcSpl_FilterMAFastQ12(
-          &mem[memIndTest+7], tempbuff2, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
-          CB_FILTERLEN, (WebRtc_Word16)(cbveclen+5));
+          &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, (int16_t)(cbveclen+5));
 
       /* Calculate lag index */
       lag = (cbveclen<<1)-20+index-base_size-lMem-1;
diff --git a/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/modules/audio_coding/codecs/ilbc/get_cd_vec.h
index 99b5d4e..1c5ac8f 100644
--- a/modules/audio_coding/codecs/ilbc/get_cd_vec.h
+++ b/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -20,11 +20,11 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
 
 void WebRtcIlbcfix_GetCbVec(
-    WebRtc_Word16 *cbvec,   /* (o) Constructed codebook vector */
-    WebRtc_Word16 *mem,   /* (i) Codebook buffer */
-    WebRtc_Word16 index,   /* (i) Codebook index */
-    WebRtc_Word16 lMem,   /* (i) Length of codebook buffer */
-    WebRtc_Word16 cbveclen   /* (i) Codebook vector length */
+    int16_t *cbvec,   /* (o) Constructed codebook vector */
+    int16_t *mem,   /* (i) Codebook buffer */
+    int16_t index,   /* (i) Codebook index */
+    int16_t lMem,   /* (i) Length of codebook buffer */
+    int16_t cbveclen   /* (i) Codebook vector length */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
index c55e918..d44380f 100644
--- a/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
+++ b/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -38,19 +38,19 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_GetLspPoly(
-    WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
-    WebRtc_Word32 *f)  /* (o) polonymial in Q24 */
+    int16_t *lsp, /* (i) LSP in Q15 */
+    int32_t *f)  /* (o) polonymial in Q24 */
 {
-  WebRtc_Word32 tmpW32;
+  int32_t tmpW32;
   int i, j;
-  WebRtc_Word16 high, low;
-  WebRtc_Word16 *lspPtr;
-  WebRtc_Word32 *fPtr;
+  int16_t high, low;
+  int16_t *lspPtr;
+  int32_t *fPtr;
 
   lspPtr = lsp;
   fPtr = f;
   /* f[0] = 1.0 (Q24) */
-  (*fPtr) = (WebRtc_Word32)16777216;
+  (*fPtr) = (int32_t)16777216;
   fPtr++;
 
   (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
@@ -64,8 +64,8 @@
     for(j=i; j>1; j--)
     {
       /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
-      high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1], 16);
-      low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1]-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)high),16), 1);
+      high = (int16_t)WEBRTC_SPL_RSHIFT_W32(fPtr[-1], 16);
+      low = (int16_t)WEBRTC_SPL_RSHIFT_W32(fPtr[-1]-WEBRTC_SPL_LSHIFT_W32(((int32_t)high),16), 1);
 
       tmpW32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(high, (*lspPtr)), 2) +
           WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(low, (*lspPtr), 15), 2);
@@ -74,7 +74,7 @@
       (*fPtr) -= tmpW32;
       fPtr--;
     }
-    (*fPtr) -= (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(*lspPtr), 10);
+    (*fPtr) -= (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)(*lspPtr), 10);
 
     fPtr+=i;
     lspPtr+=2;
diff --git a/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
index b0520b4..46ade48 100644
--- a/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
+++ b/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -41,7 +41,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_GetLspPoly(
-    WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
-    WebRtc_Word32 *f);  /* (o) polonymial in Q24 */
+    int16_t *lsp, /* (i) LSP in Q15 */
+    int32_t *f);  /* (o) polonymial in Q24 */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/modules/audio_coding/codecs/ilbc/get_sync_seq.c
index ce72865..695631a 100644
--- a/modules/audio_coding/codecs/ilbc/get_sync_seq.c
+++ b/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -26,31 +26,31 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_GetSyncSeq(
-    WebRtc_Word16 *idata,   /* (i) original data */
-    WebRtc_Word16 idatal,   /* (i) dimension of data */
-    WebRtc_Word16 centerStartPos, /* (i) where current block starts */
-    WebRtc_Word16 *period,   /* (i) rough-pitch-period array       (Q-2) */
-    WebRtc_Word16 *plocs,   /* (i) where periods of period array are taken (Q-2) */
-    WebRtc_Word16 periodl,   /* (i) dimension period array */
-    WebRtc_Word16 hl,    /* (i) 2*hl+1 is the number of sequences */
-    WebRtc_Word16 *surround  /* (i/o) The contribution from this sequence
+    int16_t *idata,   /* (i) original data */
+    int16_t idatal,   /* (i) dimension of data */
+    int16_t centerStartPos, /* (i) where current block starts */
+    int16_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    int16_t *plocs,   /* (i) where periods of period array are taken (Q-2) */
+    int16_t periodl,   /* (i) dimension period array */
+    int16_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    int16_t *surround  /* (i/o) The contribution from this sequence
                                 summed with earlier contributions */
                               ){
-  WebRtc_Word16 i,centerEndPos,q;
+  int16_t i,centerEndPos,q;
   /* Stack based */
-  WebRtc_Word16 lagBlock[2*ENH_HL+1];
-  WebRtc_Word16 blockStartPos[2*ENH_HL+1]; /* Defines the position to search around (Q2) */
-  WebRtc_Word16 plocs2[ENH_PLOCSL];
+  int16_t lagBlock[2*ENH_HL+1];
+  int16_t blockStartPos[2*ENH_HL+1]; /* Defines the position to search around (Q2) */
+  int16_t plocs2[ENH_PLOCSL];
 
   centerEndPos=centerStartPos+ENH_BLOCKL-1;
 
   /* present (find predicted lag from this position) */
 
   WebRtcIlbcfix_NearestNeighbor(lagBlock+hl,plocs,
-                                (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (centerStartPos+centerEndPos)),
+                                (int16_t)WEBRTC_SPL_MUL_16_16(2, (centerStartPos+centerEndPos)),
                                 periodl);
 
-  blockStartPos[hl]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, centerStartPos);
+  blockStartPos[hl]=(int16_t)WEBRTC_SPL_MUL_16_16(4, centerStartPos);
 
   /* past (find predicted position and perform a refined
      search to find the best sequence) */
@@ -59,10 +59,10 @@
     blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
 
     WebRtcIlbcfix_NearestNeighbor(lagBlock+q, plocs,
-                                  (WebRtc_Word16)(blockStartPos[q] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)-period[lagBlock[q+1]]),
+                                  (int16_t)(blockStartPos[q] + (int16_t)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)-period[lagBlock[q+1]]),
                                   periodl);
 
-    if((blockStartPos[q]-(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_OVERHANG))>=0) {
+    if((blockStartPos[q]-(int16_t)WEBRTC_SPL_MUL_16_16(4, ENH_OVERHANG))>=0) {
 
       /* Find the best possible sequence in the 4 times upsampled
          domain around blockStartPos+q */
@@ -85,14 +85,14 @@
   for(q=hl+1;q<=WEBRTC_SPL_MUL_16_16(2, hl);q++) {
 
     WebRtcIlbcfix_NearestNeighbor(lagBlock+q,plocs2,
-                                  (WebRtc_Word16)(blockStartPos[q-1]+
-                                                  (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)),periodl);
+                                  (int16_t)(blockStartPos[q-1]+
+                                                  (int16_t)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)),periodl);
 
     blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
 
-    if( (blockStartPos[q]+(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, (ENH_BLOCKL+ENH_OVERHANG)))
+    if( (blockStartPos[q]+(int16_t)WEBRTC_SPL_MUL_16_16(4, (ENH_BLOCKL+ENH_OVERHANG)))
         <
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, idatal)) {
+        (int16_t)WEBRTC_SPL_MUL_16_16(4, idatal)) {
 
       /* Find the best possible sequence in the 4 times upsampled
          domain around blockStartPos+q */
diff --git a/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/modules/audio_coding/codecs/ilbc/get_sync_seq.h
index a0ffd39..f9b08b7 100644
--- a/modules/audio_coding/codecs/ilbc/get_sync_seq.h
+++ b/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -26,14 +26,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_GetSyncSeq(
-    WebRtc_Word16 *idata,   /* (i) original data */
-    WebRtc_Word16 idatal,   /* (i) dimension of data */
-    WebRtc_Word16 centerStartPos, /* (i) where current block starts */
-    WebRtc_Word16 *period,   /* (i) rough-pitch-period array       (Q-2) */
-    WebRtc_Word16 *plocs,   /* (i) where periods of period array are taken (Q-2) */
-    WebRtc_Word16 periodl,   /* (i) dimension period array */
-    WebRtc_Word16 hl,    /* (i) 2*hl+1 is the number of sequences */
-    WebRtc_Word16 *surround  /* (i/o) The contribution from this sequence
+    int16_t *idata,   /* (i) original data */
+    int16_t idatal,   /* (i) dimension of data */
+    int16_t centerStartPos, /* (i) where current block starts */
+    int16_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    int16_t *plocs,   /* (i) where periods of period array are taken (Q-2) */
+    int16_t periodl,   /* (i) dimension period array */
+    int16_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    int16_t *surround  /* (i/o) The contribution from this sequence
                                 summed with earlier contributions */
                               );
 
diff --git a/modules/audio_coding/codecs/ilbc/hp_input.c b/modules/audio_coding/codecs/ilbc/hp_input.c
index f202f62..48bd7c4 100644
--- a/modules/audio_coding/codecs/ilbc/hp_input.c
+++ b/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -23,18 +23,18 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_HpInput(
-    WebRtc_Word16 *signal,     /* (i/o) signal vector */
-    WebRtc_Word16 *ba,      /* (i)   B- and A-coefficients (2:nd order)
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
                                                                    {b[0] b[1] b[2] -a[1] -a[2]} a[0]
                                                                    is assumed to be 1.0 */
-    WebRtc_Word16 *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
-    WebRtc_Word16 *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    WebRtc_Word16 len)      /* (i)   Number of samples to filter */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    int16_t len)      /* (i)   Number of samples to filter */
 {
   int i;
-  WebRtc_Word32 tmpW32;
-  WebRtc_Word32 tmpW32b;
+  int32_t tmpW32;
+  int32_t tmpW32b;
 
   for (i=0; i<len; i++) {
 
@@ -62,10 +62,10 @@
     tmpW32b = tmpW32 + 4096;
 
     /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
-    tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)268435455, tmpW32b, (WebRtc_Word32)-268435456);
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456);
 
     /* Convert back to Q0 and multiply with 0.5 */
-    signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 13);
+    signal[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 13);
 
     /* Update state (filtered part) */
     y[2] = y[0];
@@ -80,8 +80,8 @@
       tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
     }
 
-    y[0] = (WebRtc_Word16)(tmpW32 >> 16);
-    y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((int32_t)y[0], 16))>>1);
   }
 
   return;
diff --git a/modules/audio_coding/codecs/ilbc/hp_input.h b/modules/audio_coding/codecs/ilbc/hp_input.h
index f56c4f7..a30f703 100644
--- a/modules/audio_coding/codecs/ilbc/hp_input.h
+++ b/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -22,13 +22,13 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_HpInput(
-    WebRtc_Word16 *signal,     /* (i/o) signal vector */
-    WebRtc_Word16 *ba,      /* (i)   B- and A-coefficients (2:nd order)
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
                                                                    {b[0] b[1] b[2] -a[1] -a[2]} a[0]
                                                                    is assumed to be 1.0 */
-    WebRtc_Word16 *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
-    WebRtc_Word16 *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    WebRtc_Word16 len);     /* (i)   Number of samples to filter */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    int16_t len);     /* (i)   Number of samples to filter */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/hp_output.c b/modules/audio_coding/codecs/ilbc/hp_output.c
index 8e1c919..432fdee 100644
--- a/modules/audio_coding/codecs/ilbc/hp_output.c
+++ b/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -23,18 +23,18 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_HpOutput(
-    WebRtc_Word16 *signal,     /* (i/o) signal vector */
-    WebRtc_Word16 *ba,      /* (i)   B- and A-coefficients (2:nd order)
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
                                                                    {b[0] b[1] b[2] -a[1] -a[2]} a[0]
                                                                    is assumed to be 1.0 */
-    WebRtc_Word16 *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
-    WebRtc_Word16 *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    WebRtc_Word16 len)      /* (i)   Number of samples to filter */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    int16_t len)      /* (i)   Number of samples to filter */
 {
   int i;
-  WebRtc_Word32 tmpW32;
-  WebRtc_Word32 tmpW32b;
+  int32_t tmpW32;
+  int32_t tmpW32b;
 
   for (i=0; i<len; i++) {
 
@@ -62,10 +62,10 @@
     tmpW32b = tmpW32 + 1024;
 
     /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
-    tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)67108863, tmpW32b, (WebRtc_Word32)-67108864);
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
 
     /* Convert back to Q0 and multiply with 2 */
-    signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11);
+    signal[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11);
 
     /* Update state (filtered part) */
     y[2] = y[0];
@@ -80,8 +80,8 @@
       tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
     }
 
-    y[0] = (WebRtc_Word16)(tmpW32 >> 16);
-    y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((int32_t)y[0], 16))>>1);
 
   }
 
diff --git a/modules/audio_coding/codecs/ilbc/hp_output.h b/modules/audio_coding/codecs/ilbc/hp_output.h
index c9a7426..7937ba0 100644
--- a/modules/audio_coding/codecs/ilbc/hp_output.h
+++ b/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -22,13 +22,13 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_HpOutput(
-    WebRtc_Word16 *signal,     /* (i/o) signal vector */
-    WebRtc_Word16 *ba,      /* (i)   B- and A-coefficients (2:nd order)
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
                                {b[0] b[1] b[2] -a[1] -a[2]} a[0]
                                is assumed to be 1.0 */
-    WebRtc_Word16 *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                               yhi[n-2] ylow[n-2] */
-    WebRtc_Word16 *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    WebRtc_Word16 len);      /* (i)   Number of samples to filter */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    int16_t len);      /* (i)   Number of samples to filter */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/ilbc.c b/modules/audio_coding/codecs/ilbc/ilbc.c
index 9a93d61..21d159f 100644
--- a/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -25,9 +25,9 @@
 #include <stdlib.h>
 
 
-WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst, WebRtc_Word16 *ILBCENC_inst_Addr, WebRtc_Word16 *size) {
+int16_t WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst, int16_t *ILBCENC_inst_Addr, int16_t *size) {
   *iLBC_encinst=(iLBC_encinst_t*)ILBCENC_inst_Addr;
-  *size=sizeof(iLBC_Enc_Inst_t)/sizeof(WebRtc_Word16);
+  *size=sizeof(iLBC_Enc_Inst_t)/sizeof(int16_t);
   if (*iLBC_encinst!=NULL) {
     return(0);
   } else {
@@ -35,9 +35,9 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst, WebRtc_Word16 *ILBCDEC_inst_Addr, WebRtc_Word16 *size) {
+int16_t WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst, int16_t *ILBCDEC_inst_Addr, int16_t *size) {
   *iLBC_decinst=(iLBC_decinst_t*)ILBCDEC_inst_Addr;
-  *size=sizeof(iLBC_Dec_Inst_t)/sizeof(WebRtc_Word16);
+  *size=sizeof(iLBC_Dec_Inst_t)/sizeof(int16_t);
   if (*iLBC_decinst!=NULL) {
     return(0);
   } else {
@@ -45,7 +45,7 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst) {
+int16_t WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst) {
   *iLBC_encinst=(iLBC_encinst_t*)malloc(sizeof(iLBC_Enc_Inst_t));
   if (*iLBC_encinst!=NULL) {
     WebRtcSpl_Init();
@@ -55,7 +55,7 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst) {
+int16_t WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst) {
   *iLBC_decinst=(iLBC_decinst_t*)malloc(sizeof(iLBC_Dec_Inst_t));
   if (*iLBC_decinst!=NULL) {
     WebRtcSpl_Init();
@@ -65,18 +65,18 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst) {
+int16_t WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst) {
   free(iLBC_encinst);
   return(0);
 }
 
-WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst) {
+int16_t WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst) {
   free(iLBC_decinst);
   return(0);
 }
 
 
-WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst, WebRtc_Word16 mode)
+int16_t WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst, int16_t mode)
 {
   if ((mode==20)||(mode==30)) {
     WebRtcIlbcfix_InitEncode((iLBC_Enc_Inst_t*) iLBCenc_inst, mode);
@@ -86,10 +86,10 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, const WebRtc_Word16 *speechIn, WebRtc_Word16 len, WebRtc_Word16 *encoded) {
+int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, const int16_t *speechIn, int16_t len, int16_t *encoded) {
 
-  WebRtc_Word16 pos = 0;
-  WebRtc_Word16 encpos = 0;
+  int16_t pos = 0;
+  int16_t encpos = 0;
 
   if ((len != ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
 #ifdef SPLIT_10MS
@@ -104,7 +104,7 @@
 
     /* call encoder */
     while (pos<len) {
-      WebRtcIlbcfix_EncodeImpl((WebRtc_UWord16*) &encoded[encpos], &speechIn[pos], (iLBC_Enc_Inst_t*) iLBCenc_inst);
+      WebRtcIlbcfix_EncodeImpl((uint16_t*) &encoded[encpos], &speechIn[pos], (iLBC_Enc_Inst_t*) iLBCenc_inst);
 #ifdef SPLIT_10MS
       pos += 80;
       if(((iLBC_Enc_Inst_t*)iLBCenc_inst)->section == 0)
@@ -117,7 +117,7 @@
   }
 }
 
-WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 mode) {
+int16_t WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst, int16_t mode) {
   if ((mode==20)||(mode==30)) {
     WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, mode, 1);
     return(0);
@@ -125,21 +125,21 @@
     return(-1);
   }
 }
-WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst) {
+int16_t WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 20, 1);
   return(0);
 }
-WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst) {
+int16_t WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 30, 1);
   return(0);
 }
 
 
-WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
-                                  const WebRtc_Word16 *encoded,
-                                  WebRtc_Word16 len,
-                                  WebRtc_Word16 *decoded,
-                                  WebRtc_Word16 *speechType)
+int16_t WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
+                             const int16_t *encoded,
+                             int16_t len,
+                             int16_t *decoded,
+                             int16_t *speechType)
 {
   int i=0;
   /* Allow for automatic switching between the frame sizes
@@ -172,7 +172,7 @@
   }
 
   while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
-    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const uint16_t*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
@@ -180,11 +180,11 @@
   return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
 }
 
-WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
-                                       const WebRtc_Word16 *encoded,
-                                       WebRtc_Word16 len,
-                                       WebRtc_Word16 *decoded,
-                                       WebRtc_Word16 *speechType)
+int16_t WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
+                                 const int16_t *encoded,
+                                 int16_t len,
+                                 int16_t *decoded,
+                                 int16_t *speechType)
 {
   int i=0;
   if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
@@ -196,7 +196,7 @@
   }
 
   while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
-    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const uint16_t*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
@@ -204,11 +204,11 @@
   return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
 }
 
-WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
-                                       const WebRtc_Word16 *encoded,
-                                       WebRtc_Word16 len,
-                                       WebRtc_Word16 *decoded,
-                                       WebRtc_Word16 *speechType)
+int16_t WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
+                                 const int16_t *encoded,
+                                 int16_t len,
+                                 int16_t *decoded,
+                                 int16_t *speechType)
 {
   int i=0;
   if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
@@ -220,7 +220,7 @@
   }
 
   while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
-    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+    WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (const uint16_t*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
@@ -228,9 +228,9 @@
   return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
 }
 
-WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
+int16_t WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst, int16_t *decoded, int16_t noOfLostFrames) {
   int i;
-  WebRtc_UWord16 dummy;
+  uint16_t dummy;
 
   for (i=0;i<noOfLostFrames;i++) {
     /* call decoder */
@@ -239,7 +239,7 @@
   return (noOfLostFrames*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
 }
 
-WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
+int16_t WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst, int16_t *decoded, int16_t noOfLostFrames) {
 
   /* Two input parameters not used, but needed for function pointers in NetEQ */
   (void)(decoded = NULL);
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/modules/audio_coding/codecs/ilbc/index_conv_dec.c
index 0d6346a..033a0a3 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_dec.c
+++ b/modules/audio_coding/codecs/ilbc/index_conv_dec.c
@@ -19,7 +19,7 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_IndexConvDec(
-    WebRtc_Word16 *index   /* (i/o) Codebook indexes */
+    int16_t *index   /* (i/o) Codebook indexes */
                                 ){
   int k;
 
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/modules/audio_coding/codecs/ilbc/index_conv_dec.h
index f29ee23..354c5b8 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_dec.h
+++ b/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -22,7 +22,7 @@
 #include "defines.h"
 
 void WebRtcIlbcfix_IndexConvDec(
-    WebRtc_Word16 *index   /* (i/o) Codebook indexes */
+    int16_t *index   /* (i/o) Codebook indexes */
                                 );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/modules/audio_coding/codecs/ilbc/index_conv_enc.c
index cbc04b6..0e7701b 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_enc.c
+++ b/modules/audio_coding/codecs/ilbc/index_conv_enc.c
@@ -22,7 +22,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_IndexConvEnc(
-    WebRtc_Word16 *index   /* (i/o) Codebook indexes */
+    int16_t *index   /* (i/o) Codebook indexes */
                                 ){
   int k;
 
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/modules/audio_coding/codecs/ilbc/index_conv_enc.h
index d28a6e2..d686331 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_enc.h
+++ b/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_IndexConvEnc(
-    WebRtc_Word16 *index   /* (i/o) Codebook indexes */
+    int16_t *index   /* (i/o) Codebook indexes */
                                 );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/init_decode.c b/modules/audio_coding/codecs/ilbc/init_decode.c
index ea16a32..f76ade0 100644
--- a/modules/audio_coding/codecs/ilbc/init_decode.c
+++ b/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -23,9 +23,9 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     iLBC_Dec_Inst_t *iLBCdec_inst,  /* (i/o) Decoder instance */
-    WebRtc_Word16 mode,  /* (i) frame size mode */
+    int16_t mode,  /* (i) frame size mode */
     int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
   int i;
 
diff --git a/modules/audio_coding/codecs/ilbc/init_decode.h b/modules/audio_coding/codecs/ilbc/init_decode.h
index 3452f34..2938c74 100644
--- a/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -25,9 +25,9 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
-    WebRtc_Word16 mode,     /* (i) frame size mode */
+    int16_t mode,     /* (i) frame size mode */
     int use_enhancer           /* (i) 1 to use enhancer
                                   0 to run without enhancer */
                                          );
diff --git a/modules/audio_coding/codecs/ilbc/init_encode.c b/modules/audio_coding/codecs/ilbc/init_encode.c
index e1c0fc6..f6da158 100644
--- a/modules/audio_coding/codecs/ilbc/init_encode.c
+++ b/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -23,9 +23,9 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     iLBC_Enc_Inst_t *iLBCenc_inst,  /* (i/o) Encoder instance */
-    WebRtc_Word16 mode) {  /* (i) frame size mode */
+    int16_t mode) {  /* (i) frame size mode */
   iLBCenc_inst->mode = mode;
 
   /* Set all the variables that are dependent on the frame size mode */
diff --git a/modules/audio_coding/codecs/ilbc/init_encode.h b/modules/audio_coding/codecs/ilbc/init_encode.h
index f1d1858..562efb1 100644
--- a/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -25,9 +25,9 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
-    WebRtc_Word16 mode     /* (i) frame size mode */
+    int16_t mode     /* (i) frame size mode */
                                          );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/modules/audio_coding/codecs/ilbc/interface/ilbc.h
index 6208962..9ab2e86 100644
--- a/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/interface/ilbc.h
@@ -60,12 +60,12 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst,
-					    WebRtc_Word16 *ILBCENC_inst_Addr,
-					    WebRtc_Word16 *size);
-  WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst,
-					    WebRtc_Word16 *ILBCDEC_inst_Addr,
-					    WebRtc_Word16 *size);
+  int16_t WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst,
+                                      int16_t *ILBCENC_inst_Addr,
+                                      int16_t *size);
+  int16_t WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst,
+                                      int16_t *ILBCDEC_inst_Addr,
+                                      int16_t *size);
 
 
   /****************************************************************************
@@ -80,8 +80,8 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst);
-  WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst);
+  int16_t WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst);
+  int16_t WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst);
 
   /****************************************************************************
    * WebRtcIlbcfix_XxxFree(...)
@@ -95,8 +95,8 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst);
-  WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst);
+  int16_t WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst);
+  int16_t WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst);
 
 
   /****************************************************************************
@@ -113,8 +113,8 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst,
-					  WebRtc_Word16 frameLen);
+  int16_t WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst,
+                                    int16_t frameLen);
 
   /****************************************************************************
    * WebRtcIlbcfix_Encode(...)
@@ -135,10 +135,10 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst,
-                                     const WebRtc_Word16 *speechIn,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *encoded);
+  int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst,
+                               const int16_t *speechIn,
+                               int16_t len,
+                               int16_t *encoded);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecoderInit(...)
@@ -156,10 +156,10 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst,
-					  WebRtc_Word16 frameLen);
-  WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst);
-  WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst);
+  int16_t WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst,
+                                    int16_t frameLen);
+  int16_t WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst);
+  int16_t WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst);
 
   /****************************************************************************
    * WebRtcIlbcfix_Decode(...)
@@ -181,21 +181,21 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
-                                     const WebRtc_Word16* encoded,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *decoded,
-                                     WebRtc_Word16 *speechType);
-  WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
-                                         const WebRtc_Word16 *encoded,
-                                         WebRtc_Word16 len,
-                                         WebRtc_Word16 *decoded,
-                                         WebRtc_Word16 *speechType);
-  WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
-                                         const WebRtc_Word16 *encoded,
-                                         WebRtc_Word16 len,
-                                         WebRtc_Word16 *decoded,
-                                         WebRtc_Word16 *speechType);
+  int16_t WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
+                               const int16_t* encoded,
+                               int16_t len,
+                               int16_t *decoded,
+                               int16_t *speechType);
+  int16_t WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
+                                   const int16_t *encoded,
+                                   int16_t len,
+                                   int16_t *decoded,
+                                   int16_t *speechType);
+  int16_t WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
+                                   const int16_t *encoded,
+                                   int16_t len,
+                                   int16_t *decoded,
+                                   int16_t *speechType);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecodePlc(...)
@@ -215,9 +215,9 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst,
-					WebRtc_Word16 *decoded,
-					WebRtc_Word16 noOfLostFrames);
+  int16_t WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst,
+                                  int16_t *decoded,
+                                  int16_t noOfLostFrames);
 
   /****************************************************************************
    * WebRtcIlbcfix_NetEqPlc(...)
@@ -237,9 +237,9 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst,
-				       WebRtc_Word16 *decoded,
-				       WebRtc_Word16 noOfLostFrames);
+  int16_t WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst,
+                                 int16_t *decoded,
+                                 int16_t noOfLostFrames);
 
   /****************************************************************************
    * WebRtcIlbcfix_version(...)
diff --git a/modules/audio_coding/codecs/ilbc/interpolate.c b/modules/audio_coding/codecs/ilbc/interpolate.c
index 11cb33c..b6ea201 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate.c
+++ b/modules/audio_coding/codecs/ilbc/interpolate.c
@@ -24,14 +24,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Interpolate(
-    WebRtc_Word16 *out, /* (o) output vector */
-    WebRtc_Word16 *in1, /* (i) first input vector */
-    WebRtc_Word16 *in2, /* (i) second input vector */
-    WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
-    WebRtc_Word16 length)  /* (i) number of sample is vectors */
+    int16_t *out, /* (o) output vector */
+    int16_t *in1, /* (i) first input vector */
+    int16_t *in2, /* (i) second input vector */
+    int16_t coef, /* (i) weight coefficient in Q14 */
+    int16_t length)  /* (i) number of sample is vectors */
 {
   int i;
-  WebRtc_Word16 invcoef;
+  int16_t invcoef;
 
   /*
     Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
@@ -39,7 +39,7 @@
 
   invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
   for (i = 0; i < length; i++) {
-    out[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+    out[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
         (WEBRTC_SPL_MUL_16_16(coef, in1[i]) + WEBRTC_SPL_MUL_16_16(invcoef, in2[i]))+8192,
         14);
   }
diff --git a/modules/audio_coding/codecs/ilbc/interpolate.h b/modules/audio_coding/codecs/ilbc/interpolate.h
index a12021c..0483232 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate.h
+++ b/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -26,10 +26,10 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Interpolate(
-    WebRtc_Word16 *out, /* (o) output vector */
-    WebRtc_Word16 *in1, /* (i) first input vector */
-    WebRtc_Word16 *in2, /* (i) second input vector */
-    WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
-    WebRtc_Word16 length); /* (i) number of sample is vectors */
+    int16_t *out, /* (o) output vector */
+    int16_t *in1, /* (i) first input vector */
+    int16_t *in2, /* (i) second input vector */
+    int16_t coef, /* (i) weight coefficient in Q14 */
+    int16_t length); /* (i) number of sample is vectors */
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/modules/audio_coding/codecs/ilbc/interpolate_samples.c
index 31eb52e..219eda7 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate_samples.c
+++ b/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -20,12 +20,12 @@
 #include "constants.h"
 
 void WebRtcIlbcfix_InterpolateSamples(
-    WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
-    WebRtc_Word16 *CBmem,   /* (i) The CB memory */
-    WebRtc_Word16 lMem    /* (i) Length of the CB memory */
+    int16_t *interpSamples, /* (o) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int16_t lMem    /* (i) Length of the CB memory */
                                       ) {
-  WebRtc_Word16 *ppi, *ppo, i, j, temp1, temp2;
-  WebRtc_Word16 *tmpPtr;
+  int16_t *ppi, *ppo, i, j, temp1, temp2;
+  int16_t *tmpPtr;
 
   /* Calculate the 20 vectors of interpolated samples (4 samples each)
      that are used in the codebooks for lag 20 to 39 */
@@ -37,8 +37,8 @@
     ppi = CBmem+lMem-j-24;
     for (i=0; i<4; i++) {
 
-      *tmpPtr++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
+      *tmpPtr++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
 
       ppo++;
       ppi++;
diff --git a/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/modules/audio_coding/codecs/ilbc/interpolate_samples.h
index 5c98aaf..586c27d 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate_samples.h
+++ b/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -26,9 +26,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_InterpolateSamples(
-    WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
-    WebRtc_Word16 *CBmem,   /* (i) The CB memory */
-    WebRtc_Word16 lMem    /* (i) Length of the CB memory */
+    int16_t *interpSamples, /* (o) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int16_t lMem    /* (i) Length of the CB memory */
                                       );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/lpc_encode.c b/modules/audio_coding/codecs/ilbc/lpc_encode.c
index 73d67a0..8f9a3c0 100644
--- a/modules/audio_coding/codecs/ilbc/lpc_encode.c
+++ b/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -28,18 +28,18 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LpcEncode(
-    WebRtc_Word16 *syntdenum,  /* (i/o) synthesis filter coefficients
+    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
                                            before/after encoding */
-    WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
+    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
                                    before/after encoding */
-    WebRtc_Word16 *lsf_index,  /* (o) lsf quantization index */
-    WebRtc_Word16 *data,   /* (i) Speech to do LPC analysis on */
+    int16_t *lsf_index,  /* (o) lsf quantization index */
+    int16_t *data,   /* (i) Speech to do LPC analysis on */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                               ) {
   /* Stack based */
-  WebRtc_Word16 lsf[LPC_FILTERORDER * LPC_N_MAX];
-  WebRtc_Word16 lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+  int16_t lsf[LPC_FILTERORDER * LPC_N_MAX];
+  int16_t lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
 
   /* Calculate LSF's from the input speech */
   WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
diff --git a/modules/audio_coding/codecs/ilbc/lpc_encode.h b/modules/audio_coding/codecs/ilbc/lpc_encode.h
index 36967a3..8d87b0a 100644
--- a/modules/audio_coding/codecs/ilbc/lpc_encode.h
+++ b/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -26,12 +26,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LpcEncode(
-    WebRtc_Word16 *syntdenum,  /* (i/o) synthesis filter coefficients
+    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
                                   before/after encoding */
-    WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
+    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
                                    before/after encoding */
-    WebRtc_Word16 *lsf_index,  /* (o) lsf quantization index */
-    WebRtc_Word16 *data,   /* (i) Speech to do LPC analysis on */
+    int16_t *lsf_index,  /* (o) lsf quantization index */
+    int16_t *data,   /* (i) Speech to do LPC analysis on */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                              );
diff --git a/modules/audio_coding/codecs/ilbc/lsf_check.c b/modules/audio_coding/codecs/ilbc/lsf_check.c
index 7097d74..0e3bd84 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_check.c
+++ b/modules/audio_coding/codecs/ilbc/lsf_check.c
@@ -24,15 +24,15 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_LsfCheck(
-    WebRtc_Word16 *lsf, /* LSF parameters */
+    int16_t *lsf, /* LSF parameters */
     int dim, /* dimension of LSF */
     int NoAn)  /* No of analysis per frame */
 {
   int k,n,m, Nit=2, change=0,pos;
-  const WebRtc_Word16 eps=319;  /* 0.039 in Q13 (50 Hz)*/
-  const WebRtc_Word16 eps2=160;  /* eps/2.0 in Q13;*/
-  const WebRtc_Word16 maxlsf=25723; /* 3.14; (4000 Hz)*/
-  const WebRtc_Word16 minlsf=82;  /* 0.01; (0 Hz)*/
+  const int16_t eps=319;  /* 0.039 in Q13 (50 Hz)*/
+  const int16_t eps2=160;  /* eps/2.0 in Q13;*/
+  const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/
+  const int16_t minlsf=82;  /* 0.01; (0 Hz)*/
 
   /* LSF separation check*/
   for (n=0;n<Nit;n++) {  /* Run through a 2 times */
diff --git a/modules/audio_coding/codecs/ilbc/lsf_check.h b/modules/audio_coding/codecs/ilbc/lsf_check.h
index 830bbed..2f4ac8c 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_check.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_LsfCheck(
-    WebRtc_Word16 *lsf, /* LSF parameters */
+    int16_t *lsf, /* LSF parameters */
     int dim, /* dimension of LSF */
     int NoAn); /* No of analysis per frame */
 
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
index 3bb23d0..66bbde8 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
@@ -25,14 +25,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LspInterpolate2PolyDec(
-    WebRtc_Word16 *a,   /* (o) lpc coefficients Q12 */
-    WebRtc_Word16 *lsf1,  /* (i) first set of lsf coefficients Q13 */
-    WebRtc_Word16 *lsf2,  /* (i) second set of lsf coefficients Q13 */
-    WebRtc_Word16 coef,  /* (i) weighting coefficient to use between
+    int16_t *a,   /* (o) lpc coefficients Q12 */
+    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
                                    lsf1 and lsf2 Q14 */
-    WebRtc_Word16 length  /* (i) length of coefficient vectors */
+    int16_t length  /* (i) length of coefficient vectors */
                                           ){
-  WebRtc_Word16 lsftmp[LPC_FILTERORDER];
+  int16_t lsftmp[LPC_FILTERORDER];
 
   /* interpolate LSF */
   WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
index 23fe3a7..3540c1c 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -26,12 +26,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LspInterpolate2PolyDec(
-    WebRtc_Word16 *a,   /* (o) lpc coefficients Q12 */
-    WebRtc_Word16 *lsf1,  /* (i) first set of lsf coefficients Q13 */
-    WebRtc_Word16 *lsf2,  /* (i) second set of lsf coefficients Q13 */
-    WebRtc_Word16 coef,  /* (i) weighting coefficient to use between
+    int16_t *a,   /* (o) lpc coefficients Q12 */
+    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
                                    lsf1 and lsf2 Q14 */
-    WebRtc_Word16 length  /* (i) length of coefficient vectors */
+    int16_t length  /* (i) length of coefficient vectors */
                                           );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
index 3b0a34d..cf67ecc 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
@@ -26,15 +26,15 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
-    WebRtc_Word16 *a,  /* (o) lpc coefficients Q12 */
-    WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
-    WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
-    WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+    int16_t *a,  /* (o) lpc coefficients Q12 */
+    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef, /* (i) weighting coefficient to use between
                            lsf1 and lsf2 Q14 */
-    WebRtc_Word16 length /* (i) length of coefficient vectors */
+    int16_t length /* (i) length of coefficient vectors */
                                           ) {
   /* Stack based */
-  WebRtc_Word16 lsftmp[LPC_FILTERORDER];
+  int16_t lsftmp[LPC_FILTERORDER];
 
   /* interpolate LSF */
   WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
index 1bbbb80..799c100 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -27,12 +27,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
-    WebRtc_Word16 *a,  /* (o) lpc coefficients Q12 */
-    WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
-    WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
-    WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+    int16_t *a,  /* (o) lpc coefficients Q12 */
+    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef, /* (i) weighting coefficient to use between
                            lsf1 and lsf2 Q14 */
-    WebRtc_Word16 length /* (i) length of coefficient vectors */
+    int16_t length /* (i) length of coefficient vectors */
                                           );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
index 84278a4..40737bb 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
@@ -24,19 +24,19 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Lsf2Lsp(
-    WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
-    WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
-    WebRtc_Word16 m  /* (i) number of coefficients */
+    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m  /* (i) number of coefficients */
                            ) {
-  WebRtc_Word16 i, k;
-  WebRtc_Word16 diff; /* difference, which is used for the
+  int16_t i, k;
+  int16_t diff; /* difference, which is used for the
                            linear approximation (Q8) */
-  WebRtc_Word16 freq; /* normalized frequency in Q15 (0..1) */
-  WebRtc_Word32 tmpW32;
+  int16_t freq; /* normalized frequency in Q15 (0..1) */
+  int32_t tmpW32;
 
   for(i=0; i<m; i++)
   {
-    freq = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15);
+    freq = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15);
     /* 20861: 1.0/(2.0*PI) in Q17 */
     /*
        Upper 8 bits give the index k and
@@ -54,7 +54,7 @@
 
     /* Calculate linear approximation */
     tmpW32 = WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kCosDerivative[k], diff);
-    lsp[i] = WebRtcIlbcfix_kCos[k]+(WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
+    lsp[i] = WebRtcIlbcfix_kCos[k]+(int16_t)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
   }
 
   return;
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
index db6549b..b2104d7 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -26,9 +26,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Lsf2Lsp(
-    WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
-    WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
-    WebRtc_Word16 m     /* (i) number of coefficients */
+    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m     /* (i) number of coefficients */
                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
index f1c4a9e..acc5ac8 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
@@ -22,15 +22,15 @@
 #include "constants.h"
 
 void WebRtcIlbcfix_Lsf2Poly(
-    WebRtc_Word16 *a,     /* (o) predictor coefficients (order = 10) in Q12 */
-    WebRtc_Word16 *lsf    /* (i) line spectral frequencies in Q13 */
+    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
                             ) {
-  WebRtc_Word32 f[2][6]; /* f[0][] and f[1][] corresponds to
+  int32_t f[2][6]; /* f[0][] and f[1][] corresponds to
                             F1(z) and F2(z) respectivly */
-  WebRtc_Word32 *f1ptr, *f2ptr;
-  WebRtc_Word16 *a1ptr, *a2ptr;
-  WebRtc_Word32 tmpW32;
-  WebRtc_Word16 lsp[10];
+  int32_t *f1ptr, *f2ptr;
+  int16_t *a1ptr, *a2ptr;
+  int32_t tmpW32;
+  int16_t lsp[10];
   int i;
 
   /* Convert lsf to lsp */
@@ -71,10 +71,10 @@
   for (i=5; i>0; i--)
   {
     tmpW32 = (*f1ptr) + (*f2ptr);
-    (*a1ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
+    (*a1ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
 
     tmpW32 = (*f1ptr) - (*f2ptr);
-    (*a2ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
+    (*a2ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
 
     a1ptr++;
     a2ptr--;
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
index a00693b..d85f510 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Lsf2Poly(
-    WebRtc_Word16 *a,     /* (o) predictor coefficients (order = 10) in Q12 */
-    WebRtc_Word16 *lsf    /* (i) line spectral frequencies in Q13 */
+    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
index 134afbb..7afa5af 100644
--- a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
+++ b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
@@ -24,17 +24,17 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Lsp2Lsf(
-    WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
-    WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
+    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
                            (ordered, so that lsf[i]<lsf[i+1]) */
-    WebRtc_Word16 m  /* (i) Number of coefficients */
+    int16_t m  /* (i) Number of coefficients */
                            )
 {
-  WebRtc_Word16 i, k;
-  WebRtc_Word16 diff; /* diff between table value and desired value (Q15) */
-  WebRtc_Word16 freq; /* lsf/(2*pi) (Q16) */
-  WebRtc_Word16 *lspPtr, *lsfPtr, *cosTblPtr;
-  WebRtc_Word16 tmp;
+  int16_t i, k;
+  int16_t diff; /* diff between table value and desired value (Q15) */
+  int16_t freq; /* lsf/(2*pi) (Q16) */
+  int16_t *lspPtr, *lsfPtr, *cosTblPtr;
+  int16_t tmp;
 
   /* set the index to maximum index value in WebRtcIlbcfix_kCos */
   k = 63;
@@ -46,14 +46,14 @@
   */
   lspPtr = &lsp[9];
   lsfPtr = &lsf[9];
-  cosTblPtr=(WebRtc_Word16*)&WebRtcIlbcfix_kCos[k];
+  cosTblPtr=(int16_t*)&WebRtcIlbcfix_kCos[k];
   for(i=m-1; i>=0; i--)
   {
     /*
        locate value in the table, which is just above lsp[i],
        basically an approximation to acos(x)
     */
-    while( (((WebRtc_Word32)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
+    while( (((int32_t)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
     {
       k-=1;
       cosTblPtr--;
@@ -68,13 +68,13 @@
     */
 
     /* tmp (linear offset) in Q16 */
-    tmp = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAcosDerivative[k],diff, 11);
+    tmp = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAcosDerivative[k],diff, 11);
 
     /* freq in Q16 */
-    freq = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W16(k,9)+tmp;
+    freq = (int16_t)WEBRTC_SPL_LSHIFT_W16(k,9)+tmp;
 
     /* lsf = freq*2*pi */
-    (*lsfPtr) = (WebRtc_Word16)(((WebRtc_Word32)freq*25736)>>15);
+    (*lsfPtr) = (int16_t)(((int32_t)freq*25736)>>15);
 
     lsfPtr--;
     lspPtr--;
diff --git a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
index 97ba7e4..a2bcaff 100644
--- a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -26,10 +26,10 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Lsp2Lsf(
-    WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
-    WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
+    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
                            (ordered, so that lsf[i]<lsf[i+1]) */
-    WebRtc_Word16 m  /* (i) Number of coefficients */
+    int16_t m  /* (i) Number of coefficients */
                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/my_corr.c b/modules/audio_coding/codecs/ilbc/my_corr.c
index 2162205..c6cd834 100644
--- a/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -23,13 +23,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_MyCorr(
-    WebRtc_Word32 *corr,  /* (o) correlation of seq1 and seq2 */
-    WebRtc_Word16 *seq1,  /* (i) first sequence */
-    WebRtc_Word16 dim1,  /* (i) dimension first seq1 */
-    const WebRtc_Word16 *seq2, /* (i) second sequence */
-    WebRtc_Word16 dim2   /* (i) dimension seq2 */
+    int32_t *corr,  /* (o) correlation of seq1 and seq2 */
+    int16_t *seq1,  /* (i) first sequence */
+    int16_t dim1,  /* (i) dimension first seq1 */
+    const int16_t *seq2, /* (i) second sequence */
+    int16_t dim2   /* (i) dimension seq2 */
                           ){
-  WebRtc_Word16 max, scale, loops;
+  int16_t max, scale, loops;
 
   /* Calculate correlation between the two sequences. Scale the
      result of the multiplcication to maximum 26 bits in order
@@ -37,7 +37,7 @@
   max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
   scale=WebRtcSpl_GetSizeInBits(max);
 
-  scale = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(2,scale)-26);
+  scale = (int16_t)(WEBRTC_SPL_MUL_16_16(2,scale)-26);
   if (scale<0) {
     scale=0;
   }
@@ -45,7 +45,7 @@
   loops=dim1-dim2+1;
 
   /* Calculate the cross correlations */
-  WebRtcSpl_CrossCorrelation(corr, (WebRtc_Word16*)seq2, seq1, dim2, loops, scale, 1);
+  WebRtcSpl_CrossCorrelation(corr, (int16_t*)seq2, seq1, dim2, loops, scale, 1);
 
   return;
 }
diff --git a/modules/audio_coding/codecs/ilbc/my_corr.h b/modules/audio_coding/codecs/ilbc/my_corr.h
index f588c53..ee66998 100644
--- a/modules/audio_coding/codecs/ilbc/my_corr.h
+++ b/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_MyCorr(
-    WebRtc_Word32 *corr,  /* (o) correlation of seq1 and seq2 */
-    WebRtc_Word16 *seq1,  /* (i) first sequence */
-    WebRtc_Word16 dim1,  /* (i) dimension first seq1 */
-    const WebRtc_Word16 *seq2, /* (i) second sequence */
-    WebRtc_Word16 dim2   /* (i) dimension seq2 */
+    int32_t *corr,  /* (o) correlation of seq1 and seq2 */
+    int16_t *seq1,  /* (i) first sequence */
+    int16_t dim1,  /* (i) dimension first seq1 */
+    const int16_t *seq2, /* (i) second sequence */
+    int16_t dim2   /* (i) dimension seq2 */
                           );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
index ea9e1eb..8d1272f 100644
--- a/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
+++ b/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -25,15 +25,15 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_NearestNeighbor(
-    WebRtc_Word16 *index, /* (o) index of array element closest to value */
-    WebRtc_Word16 *array, /* (i) data array (Q2) */
-    WebRtc_Word16 value, /* (i) value (Q2) */
-    WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
+    int16_t *index, /* (o) index of array element closest to value */
+    int16_t *array, /* (i) data array (Q2) */
+    int16_t value, /* (i) value (Q2) */
+    int16_t arlength /* (i) dimension of data array (==8) */
                                    ){
   int i;
-  WebRtc_Word16 diff;
+  int16_t diff;
   /* Stack based */
-  WebRtc_Word32 crit[8];
+  int32_t crit[8];
 
   /* Calculate square distance */
   for(i=0;i<arlength;i++){
@@ -42,5 +42,5 @@
   }
 
   /* Find the minimum square distance */
-  *index=WebRtcSpl_MinIndexW32(crit, (WebRtc_Word16)arlength);
+  *index=WebRtcSpl_MinIndexW32(crit, (int16_t)arlength);
 }
diff --git a/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
index 705e17a..0c03470 100644
--- a/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
+++ b/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -28,10 +28,10 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_NearestNeighbor(
-    WebRtc_Word16 *index, /* (o) index of array element closest to value */
-    WebRtc_Word16 *array, /* (i) data array (Q2) */
-    WebRtc_Word16 value, /* (i) value (Q2) */
-    WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
+    int16_t *index, /* (o) index of array element closest to value */
+    int16_t *array, /* (i) data array (Q2) */
+    int16_t value, /* (i) value (Q2) */
+    int16_t arlength /* (i) dimension of data array (==8) */
                                    );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/pack_bits.c b/modules/audio_coding/codecs/ilbc/pack_bits.c
index 3990fbe..9475ecb 100644
--- a/modules/audio_coding/codecs/ilbc/pack_bits.c
+++ b/modules/audio_coding/codecs/ilbc/pack_bits.c
@@ -23,24 +23,24 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_PackBits(
-    WebRtc_UWord16 *bitstream,   /* (o) The packetized bitstream */
+    uint16_t *bitstream,   /* (o) The packetized bitstream */
     iLBC_bits *enc_bits,  /* (i) Encoded bits */
-    WebRtc_Word16 mode     /* (i) Codec mode (20 or 30) */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
                              ){
-  WebRtc_UWord16 *bitstreamPtr;
+  uint16_t *bitstreamPtr;
   int i, k;
-  WebRtc_Word16 *tmpPtr;
+  int16_t *tmpPtr;
 
   bitstreamPtr=bitstream;
 
   /* Class 1 bits of ULP */
-  /* First WebRtc_Word16 */
-  (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->lsf[0])<<10;   /* Bit 0..5  */
+  /* First int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[0])<<10;   /* Bit 0..5  */
   (*bitstreamPtr) |= (enc_bits->lsf[1])<<3;     /* Bit 6..12 */
   (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4;    /* Bit 13..15 */
   bitstreamPtr++;
-  /* Second WebRtc_Word16 */
-  (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->lsf[2]&0xF)<<12;  /* Bit 0..3  */
+  /* Second int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[2]&0xF)<<12;  /* Bit 0..3  */
 
   if (mode==20) {
     (*bitstreamPtr) |= (enc_bits->startIdx)<<10;    /* Bit 4..5  */
@@ -48,7 +48,7 @@
     (*bitstreamPtr) |= (enc_bits->idxForMax)<<3;    /* Bit 7..12 */
     (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4;  /* Bit 13..15 */
     bitstreamPtr++;
-    /* Third WebRtc_Word16 */
+    /* Third int16_t */
     (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12;  /* Bit 0..2  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8;  /* Bit 3..4  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7;  /* Bit 5  */
@@ -60,15 +60,15 @@
     (*bitstreamPtr) |= (enc_bits->lsf[3])<<6;     /* Bit 4..9  */
     (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1;    /* Bit 10..15 */
     bitstreamPtr++;
-    /* Third WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->lsf[4]&0x1)<<15;  /* Bit 0  */
+    /* Third int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[4]&0x1)<<15;  /* Bit 0  */
     (*bitstreamPtr) |= (enc_bits->lsf[5])<<8;     /* Bit 1..7  */
     (*bitstreamPtr) |= (enc_bits->startIdx)<<5;     /* Bit 8..10 */
     (*bitstreamPtr) |= (enc_bits->state_first)<<4;    /* Bit 11  */
     (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2;   /* Bit 12..15 */
     bitstreamPtr++;
-    /* 4:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1  */
+    /* 4:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1  */
     (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7;   /* Bit 2..5  */
     (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5;  /* Bit 6  */
     (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5;  /* Bit 7  */
@@ -77,14 +77,14 @@
     (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3;  /* Bit 15  */
   }
   /* Class 2 bits of ULP */
-  /* 4:th to 6:th WebRtc_Word16 for 20 ms case
-     5:th to 7:th WebRtc_Word16 for 30 ms case */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
   bitstreamPtr++;
   tmpPtr=enc_bits->idxVec;
   for (k=0; k<3; k++) {
     (*bitstreamPtr) = 0;
     for (i=15; i>=0; i--) {
-      (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
       /* Bit 15-i  */
       tmpPtr++;
     }
@@ -92,10 +92,10 @@
   }
 
   if (mode==20) {
-    /* 7:th WebRtc_Word16 */
+    /* 7:th int16_t */
     (*bitstreamPtr) = 0;
     for (i=15; i>6; i--) {
-      (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
       /* Bit 15-i  */
       tmpPtr++;
     }
@@ -106,10 +106,10 @@
     (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2;  /* Bit 14..15 */
 
   } else { /* mode==30 */
-    /* 8:th WebRtc_Word16 */
+    /* 8:th int16_t */
     (*bitstreamPtr) = 0;
     for (i=15; i>5; i--) {
-      (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
       /* Bit 15-i  */
       tmpPtr++;
     }
@@ -119,13 +119,13 @@
     (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2);    /* Bit 14  */
     (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7;   /* Bit 15  */
     bitstreamPtr++;
-    /* 9:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5  */
+    /* 9:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5  */
     (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2;   /* Bit 6..12 */
     (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5;  /* Bit 13..15 */
     bitstreamPtr++;
-    /* 10:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
+    /* 10:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
     (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8;  /* Bit 4..5  */
     (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7;  /* Bit 6..7  */
     (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3;  /* Bit 8..9  */
@@ -137,27 +137,27 @@
   }
   bitstreamPtr++;
   /* Class 3 bits of ULP */
-  /*  8:th to 14:th WebRtc_Word16 for 20 ms case
-      11:th to 17:th WebRtc_Word16 for 30 ms case */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
   tmpPtr=enc_bits->idxVec;
   for (k=0; k<7; k++) {
     (*bitstreamPtr) = 0;
     for (i=14; i>=0; i-=2) {
-      (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
       tmpPtr++;
     }
     bitstreamPtr++;
   }
 
   if (mode==20) {
-    /* 15:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    /* 15:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
     (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13;  /* Bit 2  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6;   /* Bit 3..9  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1;  /* Bit 10..15 */
     bitstreamPtr++;
-    /* 16:th WebRtc_Word16 */
-    (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[2])&0x1))<<15;
+    /* 16:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[2])&0x1))<<15;
     /* Bit 0  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12;  /* Bit 1..3  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10;  /* Bit 4..5  */
@@ -165,18 +165,18 @@
     (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6;  /* Bit 9  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1;  /* Bit 10..15 */
     bitstreamPtr++;
-    /* 17:th WebRtc_Word16 */
-    (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[4])&0x1))<<15;
+    /* 17:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[4])&0x1))<<15;
     /* Bit 0  */
     (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8;    /* Bit 1..7  */
     (*bitstreamPtr) |= (enc_bits->cb_index[6]);     /* Bit 8..15 */
     bitstreamPtr++;
-    /* 18:th WebRtc_Word16 */
-    (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[7]))<<8; /* Bit 0..7  */
+    /* 18:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7]))<<8; /* Bit 0..7  */
     (*bitstreamPtr) |= (enc_bits->cb_index[8]);     /* Bit 8..15 */
     bitstreamPtr++;
-    /* 19:th WebRtc_Word16 */
-    (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->gain_index[3])&0x3))<<14;
+    /* 19:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->gain_index[3])&0x3))<<14;
     /* Bit 0..1  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12;  /* Bit 2..3  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9;   /* Bit 4..6  */
@@ -184,15 +184,15 @@
     (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4;  /* Bit 10..11 */
     (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1;   /* Bit 12..14 */
   } else { /* mode==30 */
-    /* 18:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    /* 18:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
     (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12;  /* Bit 2..3  */
     (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11;  /* Bit 4  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4;   /* Bit 5..11 */
     (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3;  /* Bit 12..15 */
     bitstreamPtr++;
-    /* 19:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->cb_index[2])&0x7)<<13;
+    /* 19:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[2])&0x7)<<13;
     /* Bit 0..2  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10;  /* Bit 3..5  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8;  /* Bit 6..7  */
@@ -200,36 +200,36 @@
     (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4;  /* Bit 11  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3;  /* Bit 12..15 */
     bitstreamPtr++;
-    /* 20:th WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->cb_index[4])&0x7)<<13;
+    /* 20:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[4])&0x7)<<13;
     /* Bit 0..2  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6;   /* Bit 3..9  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5;  /* Bit 10  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3;  /* Bit 11..15 */
     bitstreamPtr++;
-    /* 21:st WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->cb_index[7])&0x7)<<13;
+    /* 21:st int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[7])&0x7)<<13;
     /* Bit 0..2  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5;   /* Bit 3..10 */
     (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4;  /* Bit 11  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4;  /* Bit 12..15 */
     bitstreamPtr++;
-    /* 22:nd WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->cb_index[10])&0xF)<<12;
+    /* 22:nd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[10])&0xF)<<12;
     /* Bit 0..3  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4;   /* Bit 4..11 */
     (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3;  /* Bit 12  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5;  /* Bit 13..15 */
     bitstreamPtr++;
-    /* 23:rd WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->cb_index[13])&0x1F)<<11;
+    /* 23:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[13])&0x1F)<<11;
     /* Bit 0..4  */
     (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3;   /* Bit 5..12 */
     (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1;  /* Bit 13..14 */
     (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1);   /* Bit 15  */
     bitstreamPtr++;
-    /* 24:rd WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->gain_index[5]))<<13;
+    /* 24:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[5]))<<13;
     /* Bit 0..2  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10;  /* Bit 3..5  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8;  /* Bit 6..7  */
@@ -237,8 +237,8 @@
     (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1;  /* Bit 11..14 */
     (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2;  /* Bit 15  */
     bitstreamPtr++;
-    /* 25:rd WebRtc_Word16 */
-    (*bitstreamPtr)  = ((WebRtc_UWord16)(enc_bits->gain_index[10])&0x3)<<14;
+    /* 25:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[10])&0x3)<<14;
     /* Bit 0..1  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11;  /* Bit 2..4  */
     (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7;  /* Bit 5..8  */
diff --git a/modules/audio_coding/codecs/ilbc/pack_bits.h b/modules/audio_coding/codecs/ilbc/pack_bits.h
index ed3f224..603ddd4 100644
--- a/modules/audio_coding/codecs/ilbc/pack_bits.h
+++ b/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -26,9 +26,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_PackBits( 
-    WebRtc_UWord16 *bitstream,   /* (o) The packetized bitstream */
+    uint16_t *bitstream,   /* (o) The packetized bitstream */
     iLBC_bits *enc_bits,  /* (i) Encoded bits */
-    WebRtc_Word16 mode     /* (i) Codec mode (20 or 30) */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
                              );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
index fe91851..df8a78b 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
@@ -22,10 +22,10 @@
 #include "lsp_to_lsf.h"
 
 void WebRtcIlbcfix_Poly2Lsf(
-    WebRtc_Word16 *lsf,   /* (o) lsf coefficients (Q13) */
-    WebRtc_Word16 *a    /* (i) A coefficients (Q12) */
+    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
+    int16_t *a    /* (i) A coefficients (Q12) */
                             ) {
-  WebRtc_Word16 lsp[10];
-  WebRtcIlbcfix_Poly2Lsp(a, lsp, (WebRtc_Word16*)WebRtcIlbcfix_kLspMean);
+  int16_t lsp[10];
+  WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean);
   WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
 }
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
index 0ea595e..5a7f7bb 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Poly2Lsf(
-    WebRtc_Word16 *lsf,   /* (o) lsf coefficients (Q13) */
-    WebRtc_Word16 *a    /* (i) A coefficients (Q12) */
+    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
+    int16_t *a    /* (i) A coefficients (Q12) */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
index 29b4213..3add966 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
@@ -26,17 +26,17 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Poly2Lsp(
-    WebRtc_Word16 *a,  /* (o) A coefficients in Q12 */
-    WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
-    WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
+    int16_t *a,  /* (o) A coefficients in Q12 */
+    int16_t *lsp, /* (i) LSP coefficients in Q15 */
+    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
                               coefficients turn out to be unstable */
                             ) {
-  WebRtc_Word16 f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
-  WebRtc_Word16 *a_i_ptr, *a_10mi_ptr;
-  WebRtc_Word16 *f1ptr, *f2ptr;
-  WebRtc_Word32 tmpW32;
-  WebRtc_Word16 x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
-  WebRtc_Word16 shifts, sign;
+  int16_t f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
+  int16_t *a_i_ptr, *a_10mi_ptr;
+  int16_t *f1ptr, *f2ptr;
+  int32_t tmpW32;
+  int16_t x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
+  int16_t shifts, sign;
   int i, j;
   int foundFreqs;
   int fi_select;
@@ -56,8 +56,8 @@
   (*f1ptr) = 1024; /* 1.0 in Q10 */
   (*f2ptr) = 1024; /* 1.0 in Q10 */
   for (i = 0; i < 5; i++) {
-    (*(f1ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)+(*a_10mi_ptr)), 2) - (*f1ptr));
-    (*(f2ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)-(*a_10mi_ptr)), 2) + (*f2ptr));
+    (*(f1ptr+1)) = (int16_t)(WEBRTC_SPL_RSHIFT_W32(((int32_t)(*a_i_ptr)+(*a_10mi_ptr)), 2) - (*f1ptr));
+    (*(f2ptr+1)) = (int16_t)(WEBRTC_SPL_RSHIFT_W32(((int32_t)(*a_i_ptr)-(*a_10mi_ptr)), 2) + (*f2ptr));
     a_i_ptr++;
     a_10mi_ptr--;
     f1ptr++;
@@ -116,25 +116,25 @@
       } else {
         sign = y;
         y = WEBRTC_SPL_ABS_W16(y);
-        shifts = (WebRtc_Word16)WebRtcSpl_NormW32(y)-16;
+        shifts = (int16_t)WebRtcSpl_NormW32(y)-16;
         y = WEBRTC_SPL_LSHIFT_W16(y, shifts);
-        y = (WebRtc_Word16)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
+        y = (int16_t)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
 
         tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(x, y, (19-shifts));
 
         /* y=(xhigh-xlow)/(yhigh-ylow) */
-        y = (WebRtc_Word16)(tmpW32&0xFFFF);
+        y = (int16_t)(tmpW32&0xFFFF);
 
         if (sign < 0) {
           y = -y;
         }
         /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
         tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(ylow, y, 10);
-        xint = xlow-(WebRtc_Word16)(tmpW32&0xFFFF);
+        xint = xlow-(int16_t)(tmpW32&0xFFFF);
       }
 
       /* Store the calculated lsp */
-      lsp[foundFreqs] = (WebRtc_Word16)xint;
+      lsp[foundFreqs] = (int16_t)xint;
       foundFreqs++;
 
       /* if needed, set xlow and ylow for next recursion */
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
index 7eebb25..ed20fd9 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -27,9 +27,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Poly2Lsp(
-    WebRtc_Word16 *a,  /* (o) A coefficients in Q12 */
-    WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
-    WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
+    int16_t *a,  /* (o) A coefficients in Q12 */
+    int16_t *lsp, /* (i) LSP coefficients in Q15 */
+    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
                               coefficients turn out to be unstable */
                             );
 
diff --git a/modules/audio_coding/codecs/ilbc/refiner.c b/modules/audio_coding/codecs/ilbc/refiner.c
index 9210092..50d30c9 100644
--- a/modules/audio_coding/codecs/ilbc/refiner.c
+++ b/modules/audio_coding/codecs/ilbc/refiner.c
@@ -30,26 +30,26 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Refiner(
-    WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
-    WebRtc_Word16 *idata,   /* (i) original data buffer */
-    WebRtc_Word16 idatal,   /* (i) dimension of idata */
-    WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
-    WebRtc_Word16 estSegPos,  /* (i) estimated beginning other segment (Q-2) */
-    WebRtc_Word16 *surround,  /* (i/o) The contribution from this sequence
+    int16_t *updStartPos, /* (o) updated start point (Q-2) */
+    int16_t *idata,   /* (i) original data buffer */
+    int16_t idatal,   /* (i) dimension of idata */
+    int16_t centerStartPos, /* (i) beginning center segment */
+    int16_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    int16_t *surround,  /* (i/o) The contribution from this sequence
                                            summed with earlier contributions */
-    WebRtc_Word16 gain    /* (i) Gain to use for this sequence */
+    int16_t gain    /* (i) Gain to use for this sequence */
                            ){
-  WebRtc_Word16 estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
-  WebRtc_Word16 tloc,tloc2,i,st,en,fraction;
+  int16_t estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
+  int16_t tloc,tloc2,i,st,en,fraction;
 
-  WebRtc_Word32 maxtemp, scalefact;
-  WebRtc_Word16 *filtStatePtr, *polyPtr;
+  int32_t maxtemp, scalefact;
+  int16_t *filtStatePtr, *polyPtr;
   /* Stack based */
-  WebRtc_Word16 filt[7];
-  WebRtc_Word32 corrVecUps[ENH_CORRDIM*ENH_UPS0];
-  WebRtc_Word32 corrVecTemp[ENH_CORRDIM];
-  WebRtc_Word16 vect[ENH_VECTL];
-  WebRtc_Word16 corrVec[ENH_CORRDIM];
+  int16_t filt[7];
+  int32_t corrVecUps[ENH_CORRDIM*ENH_UPS0];
+  int32_t corrVecTemp[ENH_CORRDIM];
+  int16_t vect[ENH_VECTL];
+  int16_t corrVec[ENH_CORRDIM];
 
   /* defining array bounds */
 
@@ -71,21 +71,21 @@
      location of max */
 
   WebRtcIlbcfix_MyCorr(corrVecTemp,idata+searchSegStartPos,
-                       (WebRtc_Word16)(corrdim+ENH_BLOCKL-1),idata+centerStartPos,ENH_BLOCKL);
+                       (int16_t)(corrdim+ENH_BLOCKL-1),idata+centerStartPos,ENH_BLOCKL);
 
   /* Calculate the rescaling factor for the correlation in order to
-     put the correlation in a WebRtc_Word16 vector instead */
-  maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (WebRtc_Word16)corrdim);
+     put the correlation in a int16_t vector instead */
+  maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (int16_t)corrdim);
 
   scalefact=WebRtcSpl_GetSizeInBits(maxtemp)-15;
 
   if (scalefact>0) {
     for (i=0;i<corrdim;i++) {
-      corrVec[i]=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(corrVecTemp[i], scalefact);
+      corrVec[i]=(int16_t)WEBRTC_SPL_RSHIFT_W32(corrVecTemp[i], scalefact);
     }
   } else {
     for (i=0;i<corrdim;i++) {
-      corrVec[i]=(WebRtc_Word16)corrVecTemp[i];
+      corrVec[i]=(int16_t)corrVecTemp[i];
     }
   }
   /* In order to guarantee that all values are initialized */
@@ -97,11 +97,11 @@
   WebRtcIlbcfix_EnhUpsample(corrVecUps,corrVec);
 
   /* Find maximum */
-  tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (WebRtc_Word16) (ENH_UPS0*corrdim));
+  tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (int16_t) (ENH_UPS0*corrdim));
 
   /* make vector can be upsampled without ever running outside
      bounds */
-  *updStartPos = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(searchSegStartPos,4) + tloc + 4;
+  *updStartPos = (int16_t)WEBRTC_SPL_MUL_16_16(searchSegStartPos,4) + tloc + 4;
 
   tloc2 = WEBRTC_SPL_RSHIFT_W16((tloc+3), 2);
 
@@ -110,7 +110,7 @@
   /* initialize the vector to be filtered, stuff with zeros
      when data is outside idata buffer */
   if(st<0){
-    WebRtcSpl_MemSetW16(vect, 0, (WebRtc_Word16)(-st));
+    WebRtcSpl_MemSetW16(vect, 0, (int16_t)(-st));
     WEBRTC_SPL_MEMCPY_W16(&vect[-st], idata, (ENH_VECTL+st));
   }
   else{
@@ -120,19 +120,19 @@
       WEBRTC_SPL_MEMCPY_W16(vect, &idata[st],
                             (ENH_VECTL-(en-idatal)));
       WebRtcSpl_MemSetW16(&vect[ENH_VECTL-(en-idatal)], 0,
-                          (WebRtc_Word16)(en-idatal));
+                          (int16_t)(en-idatal));
     }
     else {
       WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
     }
   }
   /* Calculate which of the 4 fractions to use */
-  fraction=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(tloc2,ENH_UPS0)-tloc;
+  fraction=(int16_t)WEBRTC_SPL_MUL_16_16(tloc2,ENH_UPS0)-tloc;
 
   /* compute the segment (this is actually a convolution) */
 
   filtStatePtr = filt + 6;
-  polyPtr = (WebRtc_Word16*)WebRtcIlbcfix_kEnhPolyPhaser[fraction];
+  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[fraction];
   for (i=0;i<7;i++) {
     *filtStatePtr-- = *polyPtr++;
   }
@@ -144,7 +144,7 @@
   /* Add the contribution from this vector (scaled with gain) to the total surround vector */
   WebRtcSpl_AddAffineVectorToVector(
       surround, vect, gain,
-      (WebRtc_Word32)32768, 16, ENH_BLOCKL);
+      (int32_t)32768, 16, ENH_BLOCKL);
 
   return;
 }
diff --git a/modules/audio_coding/codecs/ilbc/refiner.h b/modules/audio_coding/codecs/ilbc/refiner.h
index 559555c..d139961 100644
--- a/modules/audio_coding/codecs/ilbc/refiner.h
+++ b/modules/audio_coding/codecs/ilbc/refiner.h
@@ -30,14 +30,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Refiner(
-    WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
-    WebRtc_Word16 *idata,   /* (i) original data buffer */
-    WebRtc_Word16 idatal,   /* (i) dimension of idata */
-    WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
-    WebRtc_Word16 estSegPos,  /* (i) estimated beginning other segment (Q-2) */
-    WebRtc_Word16 *surround,  /* (i/o) The contribution from this sequence
+    int16_t *updStartPos, /* (o) updated start point (Q-2) */
+    int16_t *idata,   /* (i) original data buffer */
+    int16_t idatal,   /* (i) dimension of idata */
+    int16_t centerStartPos, /* (i) beginning center segment */
+    int16_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    int16_t *surround,  /* (i/o) The contribution from this sequence
                                  summed with earlier contributions */
-    WebRtc_Word16 gain    /* (i) Gain to use for this sequence */
+    int16_t gain    /* (i) Gain to use for this sequence */
                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
index ee5e643..9055493 100644
--- a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
+++ b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -26,27 +26,27 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleInterpolateLsf(
-    WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
+    int16_t *syntdenum, /* (o) the synthesis filter denominator
                                    resulting from the quantized
                                    interpolated lsf Q12 */
-    WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
+    int16_t *weightdenum, /* (o) the weighting filter denominator
                                    resulting from the unquantized
                                    interpolated lsf Q12 */
-    WebRtc_Word16 *lsf,  /* (i) the unquantized lsf coefficients Q13 */
-    WebRtc_Word16 *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
-    WebRtc_Word16 *lsfold,  /* (i) the unquantized lsf coefficients of
+    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
+    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
+    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
                                            the previous signal frame Q13 */
-    WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
                                    previous signal frame Q13 */
-    WebRtc_Word16 length,  /* (i) should equate FILTERORDER */
+    int16_t length,  /* (i) should equate FILTERORDER */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         ) {
   int i, pos, lp_length;
 
-  WebRtc_Word16 *lsf2, *lsfdeq2;
+  int16_t *lsf2, *lsfdeq2;
   /* Stack based */
-  WebRtc_Word16 lp[LPC_FILTERORDER + 1];
+  int16_t lp[LPC_FILTERORDER + 1];
 
   lsf2 = lsf + length;
   lsfdeq2 = lsfdeq + length;
@@ -67,8 +67,8 @@
                                          WebRtcIlbcfix_kLsfWeight30ms[0],
                                          length);
     WebRtcIlbcfix_BwExpand(weightdenum, lp,
-                           (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
-                           (WebRtc_Word16)lp_length);
+                           (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                           (int16_t)lp_length);
 
     /* subframe 2 to 6: Interpolation between first and second
        set of lsf coefficients */
@@ -87,8 +87,8 @@
                                            WebRtcIlbcfix_kLsfWeight30ms[i],
                                            length);
       WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
-                             (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
-                             (WebRtc_Word16)lp_length);
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
 
       pos += lp_length;
     }
@@ -113,8 +113,8 @@
                                            WebRtcIlbcfix_kLsfWeight20ms[i],
                                            length);
       WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
-                             (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
-                             (WebRtc_Word16)lp_length);
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
 
       pos += lp_length;
     }
diff --git a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
index 8cdd7da..b11f5d8 100644
--- a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -26,19 +26,19 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleInterpolateLsf(
-    WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
+    int16_t *syntdenum, /* (o) the synthesis filter denominator
                                    resulting from the quantized
                                    interpolated lsf Q12 */
-    WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
+    int16_t *weightdenum, /* (o) the weighting filter denominator
                                    resulting from the unquantized
                                    interpolated lsf Q12 */
-    WebRtc_Word16 *lsf,  /* (i) the unquantized lsf coefficients Q13 */
-    WebRtc_Word16 *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
-    WebRtc_Word16 *lsfold,  /* (i) the unquantized lsf coefficients of
+    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
+    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
+    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
                                            the previous signal frame Q13 */
-    WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
                                    previous signal frame Q13 */
-    WebRtc_Word16 length,  /* (i) should equate FILTERORDER */
+    int16_t length,  /* (i) should equate FILTERORDER */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         );
diff --git a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
index 2d19edd..f8125e1 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
+++ b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -27,20 +27,20 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLpcAnalysis(
-    WebRtc_Word16 *lsf,   /* (o) lsf coefficients */
-    WebRtc_Word16 *data,   /* (i) new block of speech */
+    int16_t *lsf,   /* (o) lsf coefficients */
+    int16_t *data,   /* (i) new block of speech */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                      ) {
   int k;
   int scale;
-  WebRtc_Word16 is;
-  WebRtc_Word16 stability;
+  int16_t is;
+  int16_t stability;
   /* Stack based */
-  WebRtc_Word16 A[LPC_FILTERORDER + 1];
-  WebRtc_Word32 R[LPC_FILTERORDER + 1];
-  WebRtc_Word16 windowedData[BLOCKL_MAX];
-  WebRtc_Word16 rc[LPC_FILTERORDER];
+  int16_t A[LPC_FILTERORDER + 1];
+  int32_t R[LPC_FILTERORDER + 1];
+  int16_t windowedData[BLOCKL_MAX];
+  int16_t rc[LPC_FILTERORDER];
 
   is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
   WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
@@ -80,7 +80,7 @@
     }
 
     /* Bandwidth expand the filter coefficients */
-    WebRtcIlbcfix_BwExpand(A, A, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
+    WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
 
     /* Convert from A to LSF representation */
     WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
diff --git a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
index 83c1e5b..a3d1985 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -26,8 +26,8 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLpcAnalysis(
-    WebRtc_Word16 *lsf,   /* (o) lsf coefficients */
-    WebRtc_Word16 *data,   /* (i) new block of speech */
+    int16_t *lsf,   /* (o) lsf coefficients */
+    int16_t *data,   /* (i) new block of speech */
     iLBC_Enc_Inst_t *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                      );
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
index 7b5efa0..ef54883 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
@@ -24,9 +24,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLsfDeQ(
-    WebRtc_Word16 *lsfdeq,  /* (o) dequantized lsf coefficients */
-    WebRtc_Word16 *index,  /* (i) quantization index */
-    WebRtc_Word16 lpc_n  /* (i) number of LPCs */
+    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
+    int16_t *index,  /* (i) quantization index */
+    int16_t lpc_n  /* (i) number of LPCs */
                                 ){
   int i, j, pos, cb_pos;
 
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
index efd3103..353edb2 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -26,9 +26,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLsfDeQ(
-    WebRtc_Word16 *lsfdeq,  /* (o) dequantized lsf coefficients */
-    WebRtc_Word16 *index,  /* (i) quantization index */
-    WebRtc_Word16 lpc_n  /* (i) number of LPCs */
+    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
+    int16_t *index,  /* (i) quantization index */
+    int16_t lpc_n  /* (i) number of LPCs */
                                 );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
index aa27fb4..8daba50 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
@@ -25,23 +25,23 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLsfQ(
-    WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
+    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
                                    (dimension FILTERORDER) Q13 */
-    WebRtc_Word16 *index, /* (o) quantization index */
-    WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
+    int16_t *index, /* (o) quantization index */
+    int16_t *lsf, /* (i) the lsf coefficient vector to be
                            quantized (dimension FILTERORDER) Q13 */
-    WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
+    int16_t lpc_n /* (i) number of lsf sets to quantize */
                               ){
 
   /* Quantize first LSF with memoryless split VQ */
   WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
-                         (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
+                         (int16_t*)WebRtcIlbcfix_kLsfCb, (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
 
   if (lpc_n==2) {
     /* Quantize second LSF with memoryless split VQ */
     WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
-                           lsf + LPC_FILTERORDER, (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb,
-                           (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
+                           lsf + LPC_FILTERORDER, (int16_t*)WebRtcIlbcfix_kLsfCb,
+                           (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
   }
   return;
 }
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
index fd17b2e..94f804b 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -26,12 +26,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SimpleLsfQ(
-    WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
+    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
                                    (dimension FILTERORDER) Q13 */
-    WebRtc_Word16 *index, /* (o) quantization index */
-    WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
+    int16_t *index, /* (o) quantization index */
+    int16_t *lsf, /* (i) the lsf coefficient vector to be
                            quantized (dimension FILTERORDER) Q13 */
-    WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
+    int16_t lpc_n /* (i) number of lsf sets to quantize */
                               );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/smooth.c b/modules/audio_coding/codecs/ilbc/smooth.c
index b606077..c975098 100644
--- a/modules/audio_coding/codecs/ilbc/smooth.c
+++ b/modules/audio_coding/codecs/ilbc/smooth.c
@@ -25,22 +25,22 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Smooth(
-    WebRtc_Word16 *odata,   /* (o) smoothed output */
-    WebRtc_Word16 *current,  /* (i) the un enhanced residual for
+    int16_t *odata,   /* (o) smoothed output */
+    int16_t *current,  /* (i) the un enhanced residual for
                                 this block */
-    WebRtc_Word16 *surround  /* (i) The approximation from the
+    int16_t *surround  /* (i) The approximation from the
                                 surrounding sequences */
                           ) {
-  WebRtc_Word16 maxtot, scale, scale1, scale2;
-  WebRtc_Word16 A, B, C, denomW16;
-  WebRtc_Word32 B_W32, denom, num;
-  WebRtc_Word32 errs;
-  WebRtc_Word32 w00,w10,w11, endiff, crit;
-  WebRtc_Word32 w00prim, w10prim, w11_div_w00;
-  WebRtc_Word16 w11prim;
-  WebRtc_Word16 bitsw00, bitsw10, bitsw11;
-  WebRtc_Word32 w11w00, w10w10, w00w00;
-  WebRtc_Word16 max1, max2;
+  int16_t maxtot, scale, scale1, scale2;
+  int16_t A, B, C, denomW16;
+  int32_t B_W32, denom, num;
+  int32_t errs;
+  int32_t w00,w10,w11, endiff, crit;
+  int32_t w00prim, w10prim, w11_div_w00;
+  int16_t w11prim;
+  int16_t bitsw00, bitsw10, bitsw11;
+  int32_t w11w00, w10w10, w00w00;
+  int16_t max1, max2;
 
   /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
 
@@ -51,7 +51,7 @@
   maxtot=WEBRTC_SPL_MAX(max1, max2);
 
   scale=WebRtcSpl_GetSizeInBits(maxtot);
-  scale = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2,scale)-26;
+  scale = (int16_t)WEBRTC_SPL_MUL_16_16(2,scale)-26;
   scale=WEBRTC_SPL_MAX(0, scale);
 
   w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
@@ -77,13 +77,13 @@
   }
 
   w00prim = WEBRTC_SPL_LSHIFT_W32(w00, scale1);
-  w11prim = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w11, scale2);
+  w11prim = (int16_t) WEBRTC_SPL_SHIFT_W32(w11, scale2);
 
   /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
   if (w11prim>64) {
     endiff = WEBRTC_SPL_LSHIFT_W32(
-        (WebRtc_Word32)WebRtcSpl_DivW32W16(w00prim, w11prim), 6);
-    C = (WebRtc_Word16)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
+        (int32_t)WebRtcSpl_DivW32W16(w00prim, w11prim), 6);
+    C = (int16_t)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
   } else {
     C = 1;
   }
@@ -123,23 +123,23 @@
     }
 
     w11w00 = WEBRTC_SPL_MUL_16_16(
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w11, -scale),
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w11, -scale),
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale));
 
     w10w10 = WEBRTC_SPL_MUL_16_16(
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale),
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale));
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale),
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale));
 
     w00w00 = WEBRTC_SPL_MUL_16_16(
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale),
-        (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale),
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale));
 
     /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
     if (w00w00>65536) {
       endiff = (w11w00-w10w10);
       endiff = WEBRTC_SPL_MAX(0, endiff);
       /* denom is in Q16 */
-      denom = WebRtcSpl_DivW32W16(endiff, (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(w00w00, 16));
+      denom = WebRtcSpl_DivW32W16(endiff, (int16_t)WEBRTC_SPL_RSHIFT_W32(w00w00, 16));
     } else {
       denom = 65536;
     }
@@ -151,20 +151,20 @@
 
       if (scale>0) {
         /* denomW16 is in Q(16+scale) */
-        denomW16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(denom, scale);
+        denomW16=(int16_t)WEBRTC_SPL_RSHIFT_W32(denom, scale);
 
         /* num in Q(34-scale) */
         num=WEBRTC_SPL_RSHIFT_W32(ENH_A0_MINUS_A0A0DIV4, scale);
       } else {
         /* denomW16 is in Q16 */
-        denomW16=(WebRtc_Word16)denom;
+        denomW16=(int16_t)denom;
 
         /* num in Q34 */
         num=ENH_A0_MINUS_A0A0DIV4;
       }
 
       /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
-      A = (WebRtc_Word16)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
+      A = (int16_t)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
 
       /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
       scale1 = 31-bitsw10;
@@ -179,15 +179,15 @@
       }
 
       if ((w00prim>0)&&(w10prim>0)) {
-        w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (WebRtc_Word16)w00prim);
+        w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (int16_t)w00prim);
 
         if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
           B_W32 = 0;
         } else {
-          B_W32 = (WebRtc_Word32)1073741824 - (WebRtc_Word32)ENH_A0DIV2 -
+          B_W32 = (int32_t)1073741824 - (int32_t)ENH_A0DIV2 -
               WEBRTC_SPL_MUL(A, w11_div_w00);
         }
-        B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B_W32, 16); /* B in Q14 */
+        B = (int16_t)WEBRTC_SPL_RSHIFT_W32(B_W32, 16); /* B in Q14 */
       } else {
         /* No smoothing */
         A = 0;
diff --git a/modules/audio_coding/codecs/ilbc/smooth.h b/modules/audio_coding/codecs/ilbc/smooth.h
index 88ce805..add0c7b 100644
--- a/modules/audio_coding/codecs/ilbc/smooth.h
+++ b/modules/audio_coding/codecs/ilbc/smooth.h
@@ -26,10 +26,10 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Smooth(
-    WebRtc_Word16 *odata,   /* (o) smoothed output */
-    WebRtc_Word16 *current,  /* (i) the un enhanced residual for
+    int16_t *odata,   /* (o) smoothed output */
+    int16_t *current,  /* (i) the un enhanced residual for
                                 this block */
-    WebRtc_Word16 *surround  /* (i) The approximation from the
+    int16_t *surround  /* (i) The approximation from the
                                 surrounding sequences */
                           );
 
diff --git a/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/modules/audio_coding/codecs/ilbc/smooth_out_data.c
index 9bacd85..df3a3b7 100644
--- a/modules/audio_coding/codecs/ilbc/smooth_out_data.c
+++ b/modules/audio_coding/codecs/ilbc/smooth_out_data.c
@@ -19,25 +19,25 @@
 #include "defines.h"
 #include "constants.h"
 
-WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
-    WebRtc_Word16 *odata,
-    WebRtc_Word16 *psseq,
-    WebRtc_Word16 *surround,
-    WebRtc_Word16 C)
+int32_t WebRtcIlbcfix_Smooth_odata(
+    int16_t *odata,
+    int16_t *psseq,
+    int16_t *surround,
+    int16_t C)
 {
   int i;
 
-  WebRtc_Word16 err;
-  WebRtc_Word32 errs;
+  int16_t err;
+  int32_t errs;
 
   for(i=0;i<80;i++) {
-    odata[i]= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+    odata[i]= (int16_t)WEBRTC_SPL_RSHIFT_W32(
         (WEBRTC_SPL_MUL_16_16(C, surround[i])+1024), 11);
   }
 
   errs=0;
   for(i=0;i<80;i++) {
-    err=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
+    err=(int16_t)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
     errs+=WEBRTC_SPL_MUL_16_16(err, err); /* errs in Q-6 */
   }
 
diff --git a/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/modules/audio_coding/codecs/ilbc/smooth_out_data.h
index 6fbe694..8324439 100644
--- a/modules/audio_coding/codecs/ilbc/smooth_out_data.h
+++ b/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -25,11 +25,11 @@
  * help function to WebRtcIlbcfix_Smooth()
  *---------------------------------------------------------------*/
 
-WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
-    WebRtc_Word16 *odata,
-    WebRtc_Word16 *psseq,
-    WebRtc_Word16 *surround,
-    WebRtc_Word16 C);
+int32_t WebRtcIlbcfix_Smooth_odata(
+    int16_t *odata,
+    int16_t *psseq,
+    int16_t *surround,
+    int16_t C);
 
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/sort_sq.c b/modules/audio_coding/codecs/ilbc/sort_sq.c
index 9276a7b..dcfd8bd 100644
--- a/modules/audio_coding/codecs/ilbc/sort_sq.c
+++ b/modules/audio_coding/codecs/ilbc/sort_sq.c
@@ -23,11 +23,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SortSq(
-    WebRtc_Word16 *xq,   /* (o) the quantized value */
-    WebRtc_Word16 *index,  /* (o) the quantization index */
-    WebRtc_Word16 x,   /* (i) the value to quantize */
-    const WebRtc_Word16 *cb, /* (i) the quantization codebook */
-    WebRtc_Word16 cb_size  /* (i) the size of the quantization codebook */
+    int16_t *xq,   /* (o) the quantized value */
+    int16_t *index,  /* (o) the quantization index */
+    int16_t x,   /* (i) the value to quantize */
+    const int16_t *cb, /* (i) the quantization codebook */
+    int16_t cb_size  /* (i) the size of the quantization codebook */
                           ){
   int i;
 
@@ -40,7 +40,7 @@
       i++;
     }
 
-    if (x > WEBRTC_SPL_RSHIFT_W32(( (WebRtc_Word32)cb[i] + cb[i - 1] + 1),1)) {
+    if (x > WEBRTC_SPL_RSHIFT_W32(( (int32_t)cb[i] + cb[i - 1] + 1),1)) {
       *index = i;
       *xq = cb[i];
     } else {
diff --git a/modules/audio_coding/codecs/ilbc/sort_sq.h b/modules/audio_coding/codecs/ilbc/sort_sq.h
index 2863dc5..eaf175b 100644
--- a/modules/audio_coding/codecs/ilbc/sort_sq.h
+++ b/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SortSq(
-    WebRtc_Word16 *xq,   /* (o) the quantized value */
-    WebRtc_Word16 *index,  /* (o) the quantization index */
-    WebRtc_Word16 x,   /* (i) the value to quantize */
-    const WebRtc_Word16 *cb, /* (i) the quantization codebook */
-    WebRtc_Word16 cb_size  /* (i) the size of the quantization codebook */
+    int16_t *xq,   /* (o) the quantized value */
+    int16_t *index,  /* (o) the quantization index */
+    int16_t x,   /* (i) the value to quantize */
+    const int16_t *cb, /* (i) the quantization codebook */
+    int16_t cb_size  /* (i) the size of the quantization codebook */
                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/split_vq.c b/modules/audio_coding/codecs/ilbc/split_vq.c
index d908fa2..39b6e1b 100644
--- a/modules/audio_coding/codecs/ilbc/split_vq.c
+++ b/modules/audio_coding/codecs/ilbc/split_vq.c
@@ -26,16 +26,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SplitVq(
-    WebRtc_Word16 *qX,  /* (o) the quantized vector in Q13 */
-    WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
+    int16_t *qX,  /* (o) the quantized vector in Q13 */
+    int16_t *index, /* (o) a vector of indexes for all vector
                                    codebooks in the split */
-    WebRtc_Word16 *X,  /* (i) the vector to quantize */
-    WebRtc_Word16 *CB,  /* (i) the quantizer codebook in Q13 */
-    WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
-    WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
+    int16_t *X,  /* (i) the vector to quantize */
+    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
+    int16_t *dim, /* (i) the dimension of X and qX */
+    int16_t *cbsize /* (i) the number of vectors in the codebook */
                            ) {
 
-  WebRtc_Word16 *qXPtr, *indexPtr, *CBPtr, *XPtr;
+  int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr;
 
   /* Quantize X with the 3 vectror quantization tables */
 
diff --git a/modules/audio_coding/codecs/ilbc/split_vq.h b/modules/audio_coding/codecs/ilbc/split_vq.h
index 7264a21..2ca98cb 100644
--- a/modules/audio_coding/codecs/ilbc/split_vq.h
+++ b/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -26,13 +26,13 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SplitVq(
-    WebRtc_Word16 *qX,  /* (o) the quantized vector in Q13 */
-    WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
+    int16_t *qX,  /* (o) the quantized vector in Q13 */
+    int16_t *index, /* (o) a vector of indexes for all vector
                                    codebooks in the split */
-    WebRtc_Word16 *X,  /* (i) the vector to quantize */
-    WebRtc_Word16 *CB,  /* (i) the quantizer codebook in Q13 */
-    WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
-    WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
+    int16_t *X,  /* (i) the vector to quantize */
+    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
+    int16_t *dim, /* (i) the dimension of X and qX */
+    int16_t *cbsize /* (i) the number of vectors in the codebook */
                            );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/state_construct.c b/modules/audio_coding/codecs/ilbc/state_construct.c
index 9d03cc3..492ad59 100644
--- a/modules/audio_coding/codecs/ilbc/state_construct.c
+++ b/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -24,23 +24,23 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateConstruct(
-    WebRtc_Word16 idxForMax,   /* (i) 6-bit index for the quantization of
+    int16_t idxForMax,   /* (i) 6-bit index for the quantization of
                                            max amplitude */
-    WebRtc_Word16 *idxVec,   /* (i) vector of quantization indexes */
-    WebRtc_Word16 *syntDenum,  /* (i) synthesis filter denumerator */
-    WebRtc_Word16 *Out_fix,  /* (o) the decoded state vector */
-    WebRtc_Word16 len    /* (i) length of a state vector */
+    int16_t *idxVec,   /* (i) vector of quantization indexes */
+    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
+    int16_t *Out_fix,  /* (o) the decoded state vector */
+    int16_t len    /* (i) length of a state vector */
                                   ) {
   int k;
-  WebRtc_Word16 maxVal;
-  WebRtc_Word16 *tmp1, *tmp2, *tmp3;
+  int16_t maxVal;
+  int16_t *tmp1, *tmp2, *tmp3;
   /* Stack based */
-  WebRtc_Word16 numerator[1+LPC_FILTERORDER];
-  WebRtc_Word16 sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
-  WebRtc_Word16 sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
-  WebRtc_Word16 *sampleVal = &sampleValVec[LPC_FILTERORDER];
-  WebRtc_Word16 *sampleMa = &sampleMaVec[LPC_FILTERORDER];
-  WebRtc_Word16 *sampleAr = &sampleValVec[LPC_FILTERORDER];
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *sampleVal = &sampleValVec[LPC_FILTERORDER];
+  int16_t *sampleMa = &sampleMaVec[LPC_FILTERORDER];
+  int16_t *sampleAr = &sampleValVec[LPC_FILTERORDER];
 
   /* initialization of coefficients */
 
@@ -60,7 +60,7 @@
     for(k=0; k<len; k++){
       /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
         maxVal is in Q8 and result is in Q(-1) */
-      (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)2097152) >> 22);
+      (*tmp1) = (int16_t) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(int32_t)2097152) >> 22);
       tmp1++;
       tmp2--;
     }
@@ -68,7 +68,7 @@
     for(k=0; k<len; k++){
       /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
         maxVal is in Q5 and result is in Q(-1) */
-      (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)262144) >> 19);
+      (*tmp1) = (int16_t) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(int32_t)262144) >> 19);
       tmp1++;
       tmp2--;
     }
@@ -76,7 +76,7 @@
     for(k=0; k<len; k++){
       /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
         maxVal is in Q3 and result is in Q(-1) */
-      (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)65536) >> 17);
+      (*tmp1) = (int16_t) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(int32_t)65536) >> 17);
       tmp1++;
       tmp2--;
     }
@@ -93,11 +93,11 @@
   /* Run MA filter + AR filter */
   WebRtcSpl_FilterMAFastQ12(
       sampleVal, sampleMa,
-      numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(len + LPC_FILTERORDER));
+      numerator, LPC_FILTERORDER+1, (int16_t)(len + LPC_FILTERORDER));
   WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
   WebRtcSpl_FilterARFastQ12(
       sampleMa, sampleAr,
-      syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*len));
+      syntDenum, LPC_FILTERORDER+1, (int16_t)(2*len));
 
   tmp1 = &sampleAr[len-1];
   tmp2 = &sampleAr[2*len-1];
diff --git a/modules/audio_coding/codecs/ilbc/state_construct.h b/modules/audio_coding/codecs/ilbc/state_construct.h
index 465699b..22d75e2 100644
--- a/modules/audio_coding/codecs/ilbc/state_construct.h
+++ b/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -24,12 +24,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateConstruct(
-    WebRtc_Word16 idxForMax,   /* (i) 6-bit index for the quantization of
+    int16_t idxForMax,   /* (i) 6-bit index for the quantization of
                                            max amplitude */
-    WebRtc_Word16 *idxVec,   /* (i) vector of quantization indexes */
-    WebRtc_Word16 *syntDenum,  /* (i) synthesis filter denumerator */
-    WebRtc_Word16 *Out_fix,  /* (o) the decoded state vector */
-    WebRtc_Word16 len    /* (i) length of a state vector */
+    int16_t *idxVec,   /* (i) vector of quantization indexes */
+    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
+    int16_t *Out_fix,  /* (o) the decoded state vector */
+    int16_t len    /* (i) length of a state vector */
                                   );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/state_search.c b/modules/audio_coding/codecs/ilbc/state_search.c
index 824a0ba..bf9c7a9 100644
--- a/modules/audio_coding/codecs/ilbc/state_search.c
+++ b/modules/audio_coding/codecs/ilbc/state_search.c
@@ -29,23 +29,23 @@
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
                                and idxVec, input state_first) */
-    WebRtc_Word16 *residual,   /* (i) target residual vector */
-    WebRtc_Word16 *syntDenum,  /* (i) lpc synthesis filter */
-    WebRtc_Word16 *weightDenum  /* (i) weighting filter denuminator */
+    int16_t *residual,   /* (i) target residual vector */
+    int16_t *syntDenum,  /* (i) lpc synthesis filter */
+    int16_t *weightDenum  /* (i) weighting filter denuminator */
                                ) {
-  WebRtc_Word16 k, index;
-  WebRtc_Word16 maxVal;
-  WebRtc_Word16 scale, shift;
-  WebRtc_Word32 maxValsq;
-  WebRtc_Word16 scaleRes;
-  WebRtc_Word16 max;
+  int16_t k, index;
+  int16_t maxVal;
+  int16_t scale, shift;
+  int32_t maxValsq;
+  int16_t scaleRes;
+  int16_t max;
   int i;
   /* Stack based */
-  WebRtc_Word16 numerator[1+LPC_FILTERORDER];
-  WebRtc_Word16 residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
-  WebRtc_Word16 sampleMa[2*STATE_SHORT_LEN_30MS];
-  WebRtc_Word16 *residualLong = &residualLongVec[LPC_FILTERORDER];
-  WebRtc_Word16 *sampleAr = residualLong;
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMa[2*STATE_SHORT_LEN_30MS];
+  int16_t *residualLong = &residualLongVec[LPC_FILTERORDER];
+  int16_t *sampleAr = residualLong;
 
   /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
   max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
@@ -66,12 +66,12 @@
   WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
   WebRtcSpl_FilterMAFastQ12(
       residualLong, sampleMa,
-      numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(iLBCenc_inst->state_short_len + LPC_FILTERORDER));
+      numerator, LPC_FILTERORDER+1, (int16_t)(iLBCenc_inst->state_short_len + LPC_FILTERORDER));
   WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
 
   WebRtcSpl_FilterARFastQ12(
       sampleMa, sampleAr,
-      syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*iLBCenc_inst->state_short_len));
+      syntDenum, LPC_FILTERORDER+1, (int16_t)(2*iLBCenc_inst->state_short_len));
 
   for(k=0;k<iLBCenc_inst->state_short_len;k++){
     sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
@@ -82,10 +82,10 @@
 
   /* Find the best index */
 
-  if ((((WebRtc_Word32)maxVal)<<scaleRes)<23170) {
-    maxValsq=((WebRtc_Word32)maxVal*maxVal)<<(2+2*scaleRes);
+  if ((((int32_t)maxVal)<<scaleRes)<23170) {
+    maxValsq=((int32_t)maxVal*maxVal)<<(2+2*scaleRes);
   } else {
-    maxValsq=(WebRtc_Word32)WEBRTC_SPL_WORD32_MAX;
+    maxValsq=(int32_t)WEBRTC_SPL_WORD32_MAX;
   }
 
   index=0;
@@ -110,7 +110,7 @@
 
   /* Set up vectors for AbsQuant and rescale it with the scale factor */
   WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
-                              iLBCenc_inst->state_short_len, (WebRtc_Word16)(shift-scaleRes));
+                              iLBCenc_inst->state_short_len, (int16_t)(shift-scaleRes));
 
   /* Quantize the values in fout[] */
   WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
diff --git a/modules/audio_coding/codecs/ilbc/state_search.h b/modules/audio_coding/codecs/ilbc/state_search.h
index 8b7f298..80a4f13 100644
--- a/modules/audio_coding/codecs/ilbc/state_search.h
+++ b/modules/audio_coding/codecs/ilbc/state_search.h
@@ -30,9 +30,9 @@
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
                                and idxVec, input state_first) */
-    WebRtc_Word16 *residual,   /* (i) target residual vector */
-    WebRtc_Word16 *syntDenum,  /* (i) lpc synthesis filter */
-    WebRtc_Word16 *weightDenum  /* (i) weighting filter denuminator */
+    int16_t *residual,   /* (i) target residual vector */
+    int16_t *syntDenum,  /* (i) lpc synthesis filter */
+    int16_t *weightDenum  /* (i) weighting filter denuminator */
                                );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/swap_bytes.c b/modules/audio_coding/codecs/ilbc/swap_bytes.c
index a48a066..8bbac42 100644
--- a/modules/audio_coding/codecs/ilbc/swap_bytes.c
+++ b/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -23,9 +23,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SwapBytes(
-    const WebRtc_UWord16* input,   /* (i) the sequence to swap */
-    WebRtc_Word16 wordLength,      /* (i) number or WebRtc_UWord16 to swap */
-    WebRtc_UWord16* output         /* (o) the swapped sequence */
+    const uint16_t* input,   /* (i) the sequence to swap */
+    int16_t wordLength,      /* (i) number or uint16_t to swap */
+    uint16_t* output         /* (o) the swapped sequence */
                               ) {
   int k;
   for (k = wordLength; k > 0; k--) {
diff --git a/modules/audio_coding/codecs/ilbc/swap_bytes.h b/modules/audio_coding/codecs/ilbc/swap_bytes.h
index 1632311..a909b2c 100644
--- a/modules/audio_coding/codecs/ilbc/swap_bytes.h
+++ b/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -26,9 +26,9 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_SwapBytes(
-    const WebRtc_UWord16* input,   /* (i) the sequence to swap */
-    WebRtc_Word16 wordLength,      /* (i) number or WebRtc_UWord16 to swap */
-    WebRtc_UWord16* output         /* (o) the swapped sequence */
+    const uint16_t* input,   /* (i) the sequence to swap */
+    int16_t wordLength,      /* (i) number or uint16_t to swap */
+    uint16_t* output         /* (o) the swapped sequence */
                               );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
index 19569ac..4b86b91 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -44,14 +44,14 @@
 {
 
   FILE *ifileid,*efileid,*ofileid, *cfileid;
-  WebRtc_Word16 data[BLOCKL_MAX];
-  WebRtc_Word16 encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
+  int16_t data[BLOCKL_MAX];
+  int16_t encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
   int len;
   short pli, mode;
   int blockcount = 0;
   int packetlosscount = 0;
   int frameLen;
-  WebRtc_Word16 speechType;
+  int16_t speechType;
   iLBC_encinst_t *Enc_Inst;
   iLBC_decinst_t *Dec_Inst;
 
@@ -155,7 +155,7 @@
 
   /* loop over input blocks */
 
-  while (((WebRtc_Word16)fread(data,sizeof(WebRtc_Word16),frameLen,ifileid))==
+  while (((int16_t)fread(data,sizeof(int16_t),frameLen,ifileid))==
          frameLen) {
 
     blockcount++;
@@ -163,20 +163,20 @@
     /* encoding */
 
     fprintf(stderr, "--- Encoding block %i --- ",blockcount);
-    len=WebRtcIlbcfix_Encode(Enc_Inst, data, (WebRtc_Word16)frameLen, encoded_data);
+    len=WebRtcIlbcfix_Encode(Enc_Inst, data, (int16_t)frameLen, encoded_data);
     fprintf(stderr, "\r");
 
     /* write byte file */
 
-    if (fwrite(encoded_data, sizeof(WebRtc_Word16),
-               ((len+1)/sizeof(WebRtc_Word16)), efileid) !=
-        (size_t)(((len+1)/sizeof(WebRtc_Word16)))) {
+    if (fwrite(encoded_data, sizeof(int16_t),
+               ((len+1)/sizeof(int16_t)), efileid) !=
+        (size_t)(((len+1)/sizeof(int16_t)))) {
       return -1;
     }
 
     /* get channel data if provided */
     if (argc==6) {
-      if (fread(&pli, sizeof(WebRtc_Word16), 1, cfileid)) {
+      if (fread(&pli, sizeof(int16_t), 1, cfileid)) {
         if ((pli!=0)&&(pli!=1)) {
           fprintf(stderr, "Error in channel file\n");
           exit(0);
@@ -184,7 +184,7 @@
         if (pli==0) {
           /* Packet loss -> remove info from frame */
           memset(encoded_data, 0,
-                 sizeof(WebRtc_Word16)*ILBCNOOFWORDS_MAX);
+                 sizeof(int16_t)*ILBCNOOFWORDS_MAX);
           packetlosscount++;
         }
       } else {
@@ -200,7 +200,7 @@
     fprintf(stderr, "--- Decoding block %i --- ",blockcount);
     if (pli==1) {
       len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
-                               (WebRtc_Word16)len, decoded_data,&speechType);
+                               (int16_t)len, decoded_data,&speechType);
     } else {
       len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
     }
@@ -208,7 +208,7 @@
 
     /* write output file */
 
-    if (fwrite(decoded_data, sizeof(WebRtc_Word16), len,
+    if (fwrite(decoded_data, sizeof(int16_t), len,
                ofileid) != (size_t)len) {
       return -1;
     }
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index ee5e484..87ec489 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -153,14 +153,14 @@
     if(len != 0){ //len may be 0 in 10ms split case
       /* get channel data if provided */
       if (argc==6) {
-        if (fread(&pli, sizeof(WebRtc_Word16), 1, chfileid)) {
+        if (fread(&pli, sizeof(int16_t), 1, chfileid)) {
           if ((pli!=0)&&(pli!=1)) {
             fprintf(stderr, "Error in channel file\n");
             exit(0);
           }
           if (pli==0) {
             /* Packet loss -> remove info from frame */
-            memset(encoded_data, 0, sizeof(WebRtc_Word16)*25);
+            memset(encoded_data, 0, sizeof(int16_t)*25);
           }
         } else {
           fprintf(stderr, "Error. Channel file too short\n");
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
index f67945e..934d4be 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -41,12 +41,12 @@
 
 short encode(                         /* (o) Number of bytes encoded */
     iLBC_Enc_Inst_t *iLBCenc_inst,    /* (i/o) Encoder instance */
-    WebRtc_Word16 *encoded_data,      /* (o) The encoded bytes */
-    WebRtc_Word16 *data               /* (i) The signal block to encode */
+    int16_t *encoded_data,      /* (o) The encoded bytes */
+    int16_t *data               /* (i) The signal block to encode */
                                                         ){
 
   /* do the actual encoding */
-  WebRtcIlbcfix_Encode((WebRtc_UWord16 *)encoded_data, data, iLBCenc_inst);
+  WebRtcIlbcfix_Encode((uint16_t *)encoded_data, data, iLBCenc_inst);
 
   return (iLBCenc_inst->no_of_bytes);
 }
@@ -69,7 +69,7 @@
 
   /* do actual decoding of block */
 
-  WebRtcIlbcfix_Decode(decoded_data, (WebRtc_UWord16 *)encoded_data,
+  WebRtcIlbcfix_Decode(decoded_data, (uint16_t *)encoded_data,
                        iLBCdec_inst, mode);
 
   return (iLBCdec_inst->blockl);
@@ -213,7 +213,7 @@
   frameLen = Enc_Inst.blockl;
 
   while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
-               Enc_Inst.blockl,ifileid)==(WebRtc_UWord16)Enc_Inst.blockl){
+               Enc_Inst.blockl,ifileid)==(uint16_t)Enc_Inst.blockl){
     noOfBlocks++;
   }
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/unpack_bits.c b/modules/audio_coding/codecs/ilbc/unpack_bits.c
index 6ed9265..fae06f6 100644
--- a/modules/audio_coding/codecs/ilbc/unpack_bits.c
+++ b/modules/audio_coding/codecs/ilbc/unpack_bits.c
@@ -22,23 +22,23 @@
  *  unpacking of bits from bitstream, i.e., vector of bytes
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
-    const WebRtc_UWord16 *bitstream,    /* (i) The packatized bitstream */
+int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+    const uint16_t *bitstream,    /* (i) The packatized bitstream */
     iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
-    WebRtc_Word16 mode     /* (i) Codec mode (20 or 30) */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
                                         ) {
-  const WebRtc_UWord16 *bitstreamPtr;
+  const uint16_t *bitstreamPtr;
   int i, k;
-  WebRtc_Word16 *tmpPtr;
+  int16_t *tmpPtr;
 
   bitstreamPtr=bitstream;
 
-  /* First WebRtc_Word16 */
+  /* First int16_t */
   enc_bits->lsf[0]  =  (*bitstreamPtr)>>10;       /* Bit 0..5  */
   enc_bits->lsf[1]  = ((*bitstreamPtr)>>3)&0x7F;      /* Bit 6..12 */
   enc_bits->lsf[2]  = ((*bitstreamPtr)&0x7)<<4;      /* Bit 13..15 */
   bitstreamPtr++;
-  /* Second WebRtc_Word16 */
+  /* Second int16_t */
   enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF;      /* Bit 0..3  */
 
   if (mode==20) {
@@ -47,7 +47,7 @@
     enc_bits->idxForMax            = ((*bitstreamPtr)>>3)&0x3F;  /* Bit 7..12 */
     enc_bits->cb_index[0]          = ((*bitstreamPtr)&0x7)<<4;  /* Bit 13..15 */
     bitstreamPtr++;
-    /* Third WebRtc_Word16 */
+    /* Third int16_t */
     enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>12)&0xE;  /* Bit 0..2  */
     enc_bits->gain_index[0]        = ((*bitstreamPtr)>>8)&0x18;  /* Bit 3..4  */
     enc_bits->gain_index[1]        = ((*bitstreamPtr)>>7)&0x8;  /* Bit 5  */
@@ -59,14 +59,14 @@
     enc_bits->lsf[3]               = ((*bitstreamPtr)>>6)&0x3F;  /* Bit 4..9  */
     enc_bits->lsf[4]               = ((*bitstreamPtr)<<1)&0x7E;  /* Bit 10..15 */
     bitstreamPtr++;
-    /* Third WebRtc_Word16 */
+    /* Third int16_t */
     enc_bits->lsf[4]              |= ((*bitstreamPtr)>>15)&0x1;  /* Bit 0  */
     enc_bits->lsf[5]               = ((*bitstreamPtr)>>8)&0x7F;  /* Bit 1..7  */
     enc_bits->startIdx             = ((*bitstreamPtr)>>5)&0x7;  /* Bit 8..10 */
     enc_bits->state_first          = ((*bitstreamPtr)>>4)&0x1;  /* Bit 11  */
     enc_bits->idxForMax            = ((*bitstreamPtr)<<2)&0x3C;  /* Bit 12..15 */
     bitstreamPtr++;
-    /* 4:th WebRtc_Word16 */
+    /* 4:th int16_t */
     enc_bits->idxForMax           |= ((*bitstreamPtr)>>14)&0x3;  /* Bit 0..1  */
     enc_bits->cb_index[0]        = ((*bitstreamPtr)>>7)&0x78;  /* Bit 2..5  */
     enc_bits->gain_index[0]        = ((*bitstreamPtr)>>5)&0x10;  /* Bit 6  */
@@ -76,8 +76,8 @@
     enc_bits->gain_index[4]        = ((*bitstreamPtr)<<3)&0x8;  /* Bit 15  */
   }
   /* Class 2 bits of ULP */
-  /* 4:th to 6:th WebRtc_Word16 for 20 ms case
-     5:th to 7:th WebRtc_Word16 for 30 ms case */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
   bitstreamPtr++;
   tmpPtr=enc_bits->idxVec;
   for (k=0; k<3; k++) {
@@ -90,7 +90,7 @@
   }
 
   if (mode==20) {
-    /* 7:th WebRtc_Word16 */
+    /* 7:th int16_t */
     for (i=15; i>6; i--) {
       (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
       /* Bit 15-i  */
@@ -103,7 +103,7 @@
     enc_bits->gain_index[7]        = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
 
   } else { /* mode==30 */
-    /* 8:th WebRtc_Word16 */
+    /* 8:th int16_t */
     for (i=15; i>5; i--) {
       (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
       /* Bit 15-i  */
@@ -115,12 +115,12 @@
     enc_bits->cb_index[3]         |= ((*bitstreamPtr))&0x2;  /* Bit 14  */
     enc_bits->cb_index[6]          = ((*bitstreamPtr)<<7)&0x80; /* Bit 15  */
     bitstreamPtr++;
-    /* 9:th WebRtc_Word16 */
+    /* 9:th int16_t */
     enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5  */
     enc_bits->cb_index[9]          = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
     enc_bits->cb_index[12]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
     bitstreamPtr++;
-    /* 10:th WebRtc_Word16 */
+    /* 10:th int16_t */
     enc_bits->cb_index[12]         |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
     enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5  */
     enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7  */
@@ -133,8 +133,8 @@
   }
   bitstreamPtr++;
   /* Class 3 bits of ULP */
-  /*  8:th to 14:th WebRtc_Word16 for 20 ms case
-      11:th to 17:th WebRtc_Word16 for 30 ms case */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
   tmpPtr=enc_bits->idxVec;
   for (k=0; k<7; k++) {
     for (i=14; i>=0; i-=2) {
@@ -145,13 +145,13 @@
   }
 
   if (mode==20) {
-    /* 15:th WebRtc_Word16 */
+    /* 15:th int16_t */
     enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
     enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2  */
     enc_bits->cb_index[1]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
     enc_bits->cb_index[2]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
     bitstreamPtr++;
-    /* 16:th WebRtc_Word16 */
+    /* 16:th int16_t */
     enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
     enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3  */
     enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5  */
@@ -159,16 +159,16 @@
     enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9  */
     enc_bits->cb_index[4]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
     bitstreamPtr++;
-    /* 17:th WebRtc_Word16 */
+    /* 17:th int16_t */
     enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
     enc_bits->cb_index[5]          = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7  */
     enc_bits->cb_index[6]          = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
     bitstreamPtr++;
-    /* 18:th WebRtc_Word16 */
+    /* 18:th int16_t */
     enc_bits->cb_index[7]          = (*bitstreamPtr)>>8;  /* Bit 0..7  */
     enc_bits->cb_index[8]          = (*bitstreamPtr)&0xFF;  /* Bit 8..15 */
     bitstreamPtr++;
-    /* 19:th WebRtc_Word16 */
+    /* 19:th int16_t */
     enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
     enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
     enc_bits->gain_index[5]        = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6  */
@@ -176,14 +176,14 @@
     enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
     enc_bits->gain_index[8]        = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
   } else { /* mode==30 */
-    /* 18:th WebRtc_Word16 */
+    /* 18:th int16_t */
     enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
     enc_bits->idxVec[57]          |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
     enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>11)&1; /* Bit 4  */
     enc_bits->cb_index[1]          = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
     enc_bits->cb_index[2]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
     bitstreamPtr++;
-    /* 19:th WebRtc_Word16 */
+    /* 19:th int16_t */
     enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
     enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
     enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
@@ -191,31 +191,31 @@
     enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
     enc_bits->cb_index[4]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
     bitstreamPtr++;
-    /* 20:th WebRtc_Word16 */
+    /* 20:th int16_t */
     enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
     enc_bits->cb_index[5]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
     enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10  */
     enc_bits->cb_index[7]          = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
     bitstreamPtr++;
-    /* 21:st WebRtc_Word16 */
+    /* 21:st int16_t */
     enc_bits->cb_index[7]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
     enc_bits->cb_index[8]          = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
     enc_bits->cb_index[9]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
     enc_bits->cb_index[10]         = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
     bitstreamPtr++;
-    /* 22:nd WebRtc_Word16 */
+    /* 22:nd int16_t */
     enc_bits->cb_index[10]        |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3  */
     enc_bits->cb_index[11]         = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
     enc_bits->cb_index[12]        |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12  */
     enc_bits->cb_index[13]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
     bitstreamPtr++;
-    /* 23:rd WebRtc_Word16 */
+    /* 23:rd int16_t */
     enc_bits->cb_index[13]        |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4  */
     enc_bits->cb_index[14]         = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
     enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
     enc_bits->gain_index[4]       |= ((*bitstreamPtr)&0x1);  /* Bit 15  */
     bitstreamPtr++;
-    /* 24:rd WebRtc_Word16 */
+    /* 24:rd int16_t */
     enc_bits->gain_index[5]        = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
     enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
     enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
@@ -223,7 +223,7 @@
     enc_bits->gain_index[9]       |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
     enc_bits->gain_index[10]      |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15  */
     bitstreamPtr++;
-    /* 25:rd WebRtc_Word16 */
+    /* 25:rd int16_t */
     enc_bits->gain_index[10]      |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
     enc_bits->gain_index[11]       = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4  */
     enc_bits->gain_index[12]      |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8  */
diff --git a/modules/audio_coding/codecs/ilbc/unpack_bits.h b/modules/audio_coding/codecs/ilbc/unpack_bits.h
index 31c728e..9586a12 100644
--- a/modules/audio_coding/codecs/ilbc/unpack_bits.h
+++ b/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -25,10 +25,10 @@
  *  unpacking of bits from bitstream, i.e., vector of bytes
  *---------------------------------------------------------------*/
 
-WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
-    const WebRtc_UWord16 *bitstream,    /* (i) The packatized bitstream */
+int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+    const uint16_t *bitstream,    /* (i) The packatized bitstream */
     iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
-    WebRtc_Word16 mode     /* (i) Codec mode (20 or 30) */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
                                         );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/vq3.c b/modules/audio_coding/codecs/ilbc/vq3.c
index 81d1bfa..a6c6cdb 100644
--- a/modules/audio_coding/codecs/ilbc/vq3.c
+++ b/modules/audio_coding/codecs/ilbc/vq3.c
@@ -24,16 +24,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Vq3(
-    WebRtc_Word16 *Xq, /* quantized vector (Q13) */
-    WebRtc_Word16 *index,
-    WebRtc_Word16 *CB, /* codebook in Q13 */
-    WebRtc_Word16 *X,  /* vector to quantize (Q13) */
-    WebRtc_Word16 n_cb
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
                        ){
-  WebRtc_Word16 i, j;
-  WebRtc_Word16 pos, minindex=0;
-  WebRtc_Word16 tmp;
-  WebRtc_Word32 dist, mindist;
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
 
   pos = 0;
   mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
diff --git a/modules/audio_coding/codecs/ilbc/vq3.h b/modules/audio_coding/codecs/ilbc/vq3.h
index f2628e0..e2e2ab5 100644
--- a/modules/audio_coding/codecs/ilbc/vq3.h
+++ b/modules/audio_coding/codecs/ilbc/vq3.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Vq3(
-    WebRtc_Word16 *Xq,  /* (o) the quantized vector (Q13) */
-    WebRtc_Word16 *index, /* (o) the quantization index */
-    WebRtc_Word16 *CB,  /* (i) the vector quantization codebook (Q13) */
-    WebRtc_Word16 *X,  /* (i) the vector to quantize (Q13) */
-    WebRtc_Word16 n_cb  /* (i) the number of vectors in the codebook */
+    int16_t *Xq,  /* (o) the quantized vector (Q13) */
+    int16_t *index, /* (o) the quantization index */
+    int16_t *CB,  /* (i) the vector quantization codebook (Q13) */
+    int16_t *X,  /* (i) the vector to quantize (Q13) */
+    int16_t n_cb  /* (i) the number of vectors in the codebook */
                        );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/vq4.c b/modules/audio_coding/codecs/ilbc/vq4.c
index 3d4c26d..7776dfb 100644
--- a/modules/audio_coding/codecs/ilbc/vq4.c
+++ b/modules/audio_coding/codecs/ilbc/vq4.c
@@ -24,16 +24,16 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Vq4(
-    WebRtc_Word16 *Xq, /* quantized vector (Q13) */
-    WebRtc_Word16 *index,
-    WebRtc_Word16 *CB, /* codebook in Q13 */
-    WebRtc_Word16 *X,  /* vector to quantize (Q13) */
-    WebRtc_Word16 n_cb
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
                        ){
-  WebRtc_Word16 i, j;
-  WebRtc_Word16 pos, minindex=0;
-  WebRtc_Word16 tmp;
-  WebRtc_Word32 dist, mindist;
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
 
   pos = 0;
   mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
diff --git a/modules/audio_coding/codecs/ilbc/vq4.h b/modules/audio_coding/codecs/ilbc/vq4.h
index 1b8cff2..5ae4c87 100644
--- a/modules/audio_coding/codecs/ilbc/vq4.h
+++ b/modules/audio_coding/codecs/ilbc/vq4.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Vq4(
-    WebRtc_Word16 *Xq,  /* (o) the quantized vector (Q13) */
-    WebRtc_Word16 *index, /* (o) the quantization index */
-    WebRtc_Word16 *CB,  /* (i) the vector quantization codebook (Q13) */
-    WebRtc_Word16 *X,  /* (i) the vector to quantize (Q13) */
-    WebRtc_Word16 n_cb  /* (i) the number of vectors in the codebook */
+    int16_t *Xq,  /* (o) the quantized vector (Q13) */
+    int16_t *index, /* (o) the quantization index */
+    int16_t *CB,  /* (i) the vector quantization codebook (Q13) */
+    int16_t *X,  /* (i) the vector to quantize (Q13) */
+    int16_t n_cb  /* (i) the number of vectors in the codebook */
                        );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/window32_w32.c b/modules/audio_coding/codecs/ilbc/window32_w32.c
index b0e8406..9ff1be3 100644
--- a/modules/audio_coding/codecs/ilbc/window32_w32.c
+++ b/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -23,18 +23,18 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Window32W32(
-    WebRtc_Word32 *z,    /* Output */
-    WebRtc_Word32 *x,    /* Input (same domain as Output)*/
-    const WebRtc_Word32  *y,  /* Q31 Window */
-    WebRtc_Word16 N     /* length to process */
+    int32_t *z,    /* Output */
+    int32_t *x,    /* Input (same domain as Output)*/
+    const int32_t  *y,  /* Q31 Window */
+    int16_t N     /* length to process */
                                ) {
-  WebRtc_Word16 i;
-  WebRtc_Word16 x_low, x_hi, y_low, y_hi;
-  WebRtc_Word16 left_shifts;
-  WebRtc_Word32 temp;
+  int16_t i;
+  int16_t x_low, x_hi, y_low, y_hi;
+  int16_t left_shifts;
+  int32_t temp;
 
-  left_shifts = (WebRtc_Word16)WebRtcSpl_NormW32(x[0]);
-  WebRtcSpl_VectorBitShiftW32(x, N, x, (WebRtc_Word16)(-left_shifts));
+  left_shifts = (int16_t)WebRtcSpl_NormW32(x[0]);
+  WebRtcSpl_VectorBitShiftW32(x, N, x, (int16_t)(-left_shifts));
 
 
   /* The double precision numbers use a special representation:
@@ -42,15 +42,15 @@
    */
   for (i = 0; i < N; i++) {
     /* Extract higher bytes */
-    x_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(x[i], 16);
-    y_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(y[i], 16);
+    x_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(x[i], 16);
+    y_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(y[i], 16);
 
     /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
-    temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_hi, 16);
-    x_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((x[i] - temp), 1);
+    temp = WEBRTC_SPL_LSHIFT_W32((int32_t)x_hi, 16);
+    x_low = (int16_t) WEBRTC_SPL_RSHIFT_W32((x[i] - temp), 1);
 
-    temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y_hi, 16);
-    y_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((y[i] - temp), 1);
+    temp = WEBRTC_SPL_LSHIFT_W32((int32_t)y_hi, 16);
+    y_low = (int16_t) WEBRTC_SPL_RSHIFT_W32((y[i] - temp), 1);
 
     /* Calculate z by a 32 bit multiplication using both low and high from x and y */
     temp = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(x_hi, y_hi), 1);
diff --git a/modules/audio_coding/codecs/ilbc/window32_w32.h b/modules/audio_coding/codecs/ilbc/window32_w32.h
index 121188a..4ee6fce 100644
--- a/modules/audio_coding/codecs/ilbc/window32_w32.h
+++ b/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -26,10 +26,10 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Window32W32(
-    WebRtc_Word32 *z,    /* Output */
-    WebRtc_Word32 *x,    /* Input (same domain as Output)*/
-    const WebRtc_Word32  *y,  /* Q31 Window */
-    WebRtc_Word16 N     /* length to process */
+    int32_t *z,    /* Output */
+    int32_t *x,    /* Input (same domain as Output)*/
+    const int32_t  *y,  /* Q31 Window */
+    int16_t N     /* length to process */
                                );
 
 #endif
diff --git a/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/modules/audio_coding/codecs/ilbc/xcorr_coef.c
index 04170ad..eb7f828 100644
--- a/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+++ b/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -24,26 +24,26 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_XcorrCoef(
-    WebRtc_Word16 *target,  /* (i) first array */
-    WebRtc_Word16 *regressor, /* (i) second array */
-    WebRtc_Word16 subl,  /* (i) dimension arrays */
-    WebRtc_Word16 searchLen, /* (i) the search lenght */
-    WebRtc_Word16 offset,  /* (i) samples offset between arrays */
-    WebRtc_Word16 step   /* (i) +1 or -1 */
+    int16_t *target,  /* (i) first array */
+    int16_t *regressor, /* (i) second array */
+    int16_t subl,  /* (i) dimension arrays */
+    int16_t searchLen, /* (i) the search lenght */
+    int16_t offset,  /* (i) samples offset between arrays */
+    int16_t step   /* (i) +1 or -1 */
                             ){
   int k;
-  WebRtc_Word16 maxlag;
-  WebRtc_Word16 pos;
-  WebRtc_Word16 max;
-  WebRtc_Word16 crossCorrScale, Energyscale;
-  WebRtc_Word16 crossCorrSqMod, crossCorrSqMod_Max;
-  WebRtc_Word32 crossCorr, Energy;
-  WebRtc_Word16 crossCorrmod, EnergyMod, EnergyMod_Max;
-  WebRtc_Word16 *tp, *rp;
-  WebRtc_Word16 *rp_beg, *rp_end;
-  WebRtc_Word16 totscale, totscale_max;
-  WebRtc_Word16 scalediff;
-  WebRtc_Word32 newCrit, maxCrit;
+  int16_t maxlag;
+  int16_t pos;
+  int16_t max;
+  int16_t crossCorrScale, Energyscale;
+  int16_t crossCorrSqMod, crossCorrSqMod_Max;
+  int32_t crossCorr, Energy;
+  int16_t crossCorrmod, EnergyMod, EnergyMod_Max;
+  int16_t *tp, *rp;
+  int16_t *rp_beg, *rp_end;
+  int16_t totscale, totscale_max;
+  int16_t scalediff;
+  int32_t newCrit, maxCrit;
   int shifts;
 
   /* Initializations, to make sure that the first one is selected */
@@ -55,16 +55,16 @@
 
   /* Find scale value and start position */
   if (step==1) {
-    max=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(subl+searchLen-1));
+    max=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(subl+searchLen-1));
     rp_beg = regressor;
     rp_end = &regressor[subl];
   } else { /* step==-1 */
-    max=WebRtcSpl_MaxAbsValueW16(&regressor[-searchLen], (WebRtc_Word16)(subl+searchLen-1));
+    max=WebRtcSpl_MaxAbsValueW16(&regressor[-searchLen], (int16_t)(subl+searchLen-1));
     rp_beg = &regressor[-1];
     rp_end = &regressor[subl-1];
   }
 
-  /* Introduce a scale factor on the Energy in WebRtc_Word32 in
+  /* Introduce a scale factor on the Energy in int32_t in
      order to make sure that the calculation does not
      overflow */
 
@@ -86,13 +86,13 @@
     if ((Energy>0)&&(crossCorr>0)) {
 
       /* Put cross correlation and energy on 16 bit word */
-      crossCorrScale=(WebRtc_Word16)WebRtcSpl_NormW32(crossCorr)-16;
-      crossCorrmod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
-      Energyscale=(WebRtc_Word16)WebRtcSpl_NormW32(Energy)-16;
-      EnergyMod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
+      crossCorrScale=(int16_t)WebRtcSpl_NormW32(crossCorr)-16;
+      crossCorrmod=(int16_t)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
+      Energyscale=(int16_t)WebRtcSpl_NormW32(Energy)-16;
+      EnergyMod=(int16_t)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
 
-      /* Square cross correlation and store upper WebRtc_Word16 */
-      crossCorrSqMod=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(crossCorrmod, crossCorrmod, 16);
+      /* Square cross correlation and store upper int16_t */
+      crossCorrSqMod=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(crossCorrmod, crossCorrmod, 16);
 
       /* Calculate the total number of (dynamic) right shifts that have
          been performed on (crossCorr*crossCorr)/energy
@@ -111,11 +111,11 @@
          division */
 
       if (scalediff<0) {
-        newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
-        maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod);
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod);
       } else {
-        newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max);
-        maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod)>>scalediff;
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod)>>scalediff;
       }
 
       /* Store the new lag value if the new criteria is larger
@@ -132,7 +132,7 @@
 
     /* Do a +/- to get the next energy */
     Energy += step*(WEBRTC_SPL_RSHIFT_W32(
-        ((WebRtc_Word32)(*rp_end)*(*rp_end)) - ((WebRtc_Word32)(*rp_beg)*(*rp_beg)),
+        ((int32_t)(*rp_end)*(*rp_end)) - ((int32_t)(*rp_beg)*(*rp_beg)),
         shifts));
     rp_beg+=step;
     rp_end+=step;
diff --git a/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/modules/audio_coding/codecs/ilbc/xcorr_coef.h
index ac885c4..1f4c58d 100644
--- a/modules/audio_coding/codecs/ilbc/xcorr_coef.h
+++ b/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -27,12 +27,12 @@
  *---------------------------------------------------------------*/
 
 int WebRtcIlbcfix_XcorrCoef(
-    WebRtc_Word16 *target,  /* (i) first array */
-    WebRtc_Word16 *regressor, /* (i) second array */
-    WebRtc_Word16 subl,  /* (i) dimension arrays */
-    WebRtc_Word16 searchLen, /* (i) the search lenght */
-    WebRtc_Word16 offset,  /* (i) samples offset between arrays */
-    WebRtc_Word16 step   /* (i) +1 or -1 */
+    int16_t *target,  /* (i) first array */
+    int16_t *regressor, /* (i) second array */
+    int16_t subl,  /* (i) dimension arrays */
+    int16_t searchLen, /* (i) the search lenght */
+    int16_t offset,  /* (i) samples offset between arrays */
+    int16_t step   /* (i) +1 or -1 */
                             );
 
 #endif
diff --git a/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
index 28e9429..da7163e 100644
--- a/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -35,7 +35,7 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes);
+  int16_t WebRtcIsacfix_AssignSize(int *sizeinbytes);
 
   /**************************************************************************
    * WebRtcIsacfix_Assign(...)
@@ -52,7 +52,7 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst,
+  int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst,
                                      void *ISACFIX_inst_Addr);
 
   /****************************************************************************
@@ -68,7 +68,7 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst);
+  int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst);
 
 
   /****************************************************************************
@@ -83,7 +83,7 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -104,8 +104,8 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
-                                          WebRtc_Word16  CodingMode);
+  int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
+                                    int16_t  CodingMode);
 
 
   /****************************************************************************
@@ -129,9 +129,9 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                                     const WebRtc_Word16 *speechIn,
-                                     WebRtc_Word16 *encoded);
+  int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                               const int16_t *speechIn,
+                               int16_t *encoded);
 
 
 
@@ -161,9 +161,9 @@
 
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                       const WebRtc_Word16 *speechIn,
-                                       WebRtc_Word16 *encoded);
+  int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+                                 const int16_t *speechIn,
+                                 int16_t *encoded);
 #endif //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 
@@ -181,7 +181,7 @@
    *         -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -201,11 +201,11 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
-                                                const WebRtc_UWord16 *encoded,
-                                                WebRtc_Word32  packet_size,
-                                                WebRtc_UWord16 rtp_seq_number,
-                                                WebRtc_UWord32 arr_ts);
+  int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
+                                          const uint16_t *encoded,
+                                          int32_t  packet_size,
+                                          uint16_t rtp_seq_number,
+                                          uint32_t arr_ts);
 
   /****************************************************************************
    * WebRtcIsacfix_UpdateBwEstimate(...)
@@ -226,12 +226,12 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
-                                               const WebRtc_UWord16   *encoded,
-                                               WebRtc_Word32          packet_size,
-                                               WebRtc_UWord16         rtp_seq_number,
-                                               WebRtc_UWord32         send_ts,
-                                               WebRtc_UWord32         arr_ts);
+  int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
+                                         const uint16_t   *encoded,
+                                         int32_t          packet_size,
+                                         uint16_t         rtp_seq_number,
+                                         uint32_t         send_ts,
+                                         uint32_t         arr_ts);
 
   /****************************************************************************
    * WebRtcIsacfix_Decode(...)
@@ -252,11 +252,11 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                                     const WebRtc_UWord16 *encoded,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *decoded,
-                                     WebRtc_Word16 *speechType);
+  int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+                               const uint16_t *encoded,
+                               int16_t len,
+                               int16_t *decoded,
+                               int16_t *speechType);
 
 
   /****************************************************************************
@@ -281,11 +281,11 @@
    */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                       const WebRtc_UWord16 *encoded,
-                                       WebRtc_Word16 len,
-                                       WebRtc_Word16 *decoded,
-                                       WebRtc_Word16 *speechType);
+  int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+                                 const uint16_t *encoded,
+                                 int16_t len,
+                                 int16_t *decoded,
+                                 int16_t *speechType);
 #endif //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 
@@ -311,9 +311,9 @@
    */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                          WebRtc_Word16 *decoded,
-                                          WebRtc_Word16 noOfLostFrames );
+  int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
+                                    int16_t *decoded,
+                                    int16_t noOfLostFrames);
 #endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 
@@ -339,9 +339,9 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
-                                        WebRtc_Word16 *decoded,
-                                        WebRtc_Word16 noOfLostFrames );
+  int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
+                                  int16_t *decoded,
+                                  int16_t noOfLostFrames );
 
 
   /****************************************************************************
@@ -357,8 +357,8 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
-                                           WebRtc_Word16* frameLength);
+  int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
+                                     int16_t* frameLength);
 
   /****************************************************************************
    * WebRtcIsacfix_Control(...)
@@ -376,9 +376,9 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                                      WebRtc_Word16          rate,
-                                      WebRtc_Word16          framesize);
+  int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
+                                int16_t          rate,
+                                int16_t          framesize);
 
 
 
@@ -404,10 +404,10 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
-                                         WebRtc_Word16 rateBPS,
-                                         WebRtc_Word16 frameSizeMs,
-                                         WebRtc_Word16 enforceFrameSize);
+  int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
+                                   int16_t rateBPS,
+                                   int16_t frameSizeMs,
+                                   int16_t enforceFrameSize);
 
 
 
@@ -438,7 +438,7 @@
    * Return value             : Error code
    */
 
-  WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -453,7 +453,7 @@
    *                            else bitrate
    */
 
-  WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst);
+  int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -474,8 +474,8 @@
    *                           -1 if error happens
    */
 
-  WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
-                                                WebRtc_Word16 maxPayloadBytes);
+  int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
+                                          int16_t maxPayloadBytes);
 
 
   /****************************************************************************
@@ -501,8 +501,8 @@
    *                           -1 if error happens
    */
 
-  WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
-                                         WebRtc_Word32 maxRate);
+  int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
+                                   int32_t maxRate);
 
   /****************************************************************************
    * WebRtcIsacfix_CreateInternal(...)
@@ -516,7 +516,7 @@
    *                           -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -531,7 +531,7 @@
    *                                -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst);
 
 
   /****************************************************************************
@@ -553,10 +553,10 @@
    *                            -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
-                                              WebRtc_Word16          bweIndex,
-                                              float              scale,
-                                              WebRtc_Word16        *encoded);
+  int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
+                                        int16_t          bweIndex,
+                                        float              scale,
+                                        int16_t        *encoded);
 
 
   /****************************************************************************
@@ -573,8 +573,8 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
-                                                 WebRtc_Word16*     rateIndex);
+  int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
+                                           int16_t*     rateIndex);
 
 
   /****************************************************************************
@@ -589,8 +589,8 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
-                                             WebRtc_Word16     rateIndex);
+  int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
+                                       int16_t     rateIndex);
 
 
   /****************************************************************************
@@ -606,8 +606,8 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
-                                          WebRtc_Word16* rateIndex);
+  int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
+                                    int16_t* rateIndex);
 
 
   /****************************************************************************
@@ -621,7 +621,7 @@
    * Return value             : frame lenght in samples
    */
 
-  WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
+  int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
 
 
 #if defined(__cplusplus)
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
index ee62bad..38eecb7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
@@ -30,10 +30,10 @@
  *
  * Return value             : number of bytes in the stream
  */
-WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData)
+int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData)
 {
-  WebRtc_UWord16 *streamPtr;
-  WebRtc_UWord16 negCarry;
+  uint16_t *streamPtr;
+  uint16_t negCarry;
 
   /* point to the right place in the stream buffer */
   streamPtr = streamData->stream + streamData->stream_index;
@@ -72,10 +72,10 @@
     }
     /* write remaining data to bitstream, if "full == 0" first byte has data */
     if (streamData->full == 0) {
-      *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+      *streamPtr++ += (uint16_t) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
       streamData->full = 1;
     } else {
-      *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
+      *streamPtr = (uint16_t) WEBRTC_SPL_LSHIFT_W32(
           WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
       streamData->full = 0;
     }
@@ -111,10 +111,10 @@
     }
     /* write remaining data (2 bytes) to bitstream */
     if (streamData->full) {
-      *streamPtr++ = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 16);
+      *streamPtr++ = (uint16_t) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 16);
     } else {
-      *streamPtr++ |= (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
-      *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 8)
+      *streamPtr++ |= (uint16_t) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+      *streamPtr = (uint16_t) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 8)
           & 0xFF00;
     }
   }
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
index 14f1add..5c23f7a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
@@ -33,20 +33,20 @@
  *                            <0 if error detected
  */
 int WebRtcIsacfix_EncHistMulti(Bitstr_enc *streamData,
-                              const WebRtc_Word16 *data,
-                              const WebRtc_UWord16 **cdf,
-                              const WebRtc_Word16 lenData)
+                               const int16_t *data,
+                               const uint16_t **cdf,
+                               const int16_t lenData)
 {
-  WebRtc_UWord32 W_lower;
-  WebRtc_UWord32 W_upper;
-  WebRtc_UWord32 W_upper_LSB;
-  WebRtc_UWord32 W_upper_MSB;
-  WebRtc_UWord16 *streamPtr;
-  WebRtc_UWord16 negCarry;
-  WebRtc_UWord16 *maxStreamPtr;
-  WebRtc_UWord16 *streamPtrCarry;
-  WebRtc_UWord32 cdfLo;
-  WebRtc_UWord32 cdfHi;
+  uint32_t W_lower;
+  uint32_t W_upper;
+  uint32_t W_upper_LSB;
+  uint32_t W_upper_MSB;
+  uint16_t *streamPtr;
+  uint16_t negCarry;
+  uint16_t *maxStreamPtr;
+  uint16_t *streamPtrCarry;
+  uint32_t cdfLo;
+  uint32_t cdfHi;
   int k;
 
 
@@ -60,8 +60,8 @@
   for (k = lenData; k > 0; k--)
   {
     /* fetch cdf_lower and cdf_upper from cdf tables */
-    cdfLo = (WebRtc_UWord32) *(*cdf + (WebRtc_UWord32)*data);
-    cdfHi = (WebRtc_UWord32) *(*cdf++ + (WebRtc_UWord32)*data++ + 1);
+    cdfLo = (uint32_t) *(*cdf + (uint32_t)*data);
+    cdfHi = (uint32_t) *(*cdf++ + (uint32_t)*data++ + 1);
 
     /* update interval */
     W_upper_LSB = W_upper & 0x0000FFFF;
@@ -103,10 +103,10 @@
     {
       W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
       if (streamData->full == 0) {
-        *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+        *streamPtr++ += (uint16_t) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
         streamData->full = 1;
       } else {
-        *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
+        *streamPtr = (uint16_t) WEBRTC_SPL_LSHIFT_W32(
             WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
         streamData->full = 0;
       }
@@ -145,21 +145,21 @@
  * Return value             : number of bytes in the stream
  *                            <0 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(WebRtc_Word16 *data,
-                                              Bitstr_dec *streamData,
-                                              const WebRtc_UWord16 **cdf,
-                                              const WebRtc_UWord16 *cdfSize,
-                                              const WebRtc_Word16 lenData)
+int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t *data,
+                                         Bitstr_dec *streamData,
+                                         const uint16_t **cdf,
+                                         const uint16_t *cdfSize,
+                                         const int16_t lenData)
 {
-  WebRtc_UWord32    W_lower = 0;
-  WebRtc_UWord32    W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord32    W_upper_LSB;
-  WebRtc_UWord32    W_upper_MSB;
-  WebRtc_UWord32    streamval;
-  const WebRtc_UWord16 *streamPtr;
-  const WebRtc_UWord16 *cdfPtr;
-  WebRtc_Word16     sizeTmp;
+  uint32_t    W_lower = 0;
+  uint32_t    W_upper;
+  uint32_t    W_tmp;
+  uint32_t    W_upper_LSB;
+  uint32_t    W_upper_MSB;
+  uint32_t    streamval;
+  const uint16_t *streamPtr;
+  const uint16_t *cdfPtr;
+  int16_t     sizeTmp;
   int             k;
 
 
@@ -175,7 +175,7 @@
   if (streamData->stream_index == 0)
   {
     /* read first word from bytestream */
-    streamval = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)*streamPtr++, 16);
+    streamval = WEBRTC_SPL_LSHIFT_W32((uint32_t)*streamPtr++, 16);
     streamval |= *streamPtr++;
   } else {
     streamval = streamData->streamval;
@@ -282,20 +282,20 @@
  * Return value             : number of bytes in original stream
  *                            <0 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(WebRtc_Word16 *data,
-                                               Bitstr_dec *streamData,
-                                               const WebRtc_UWord16 **cdf,
-                                               const WebRtc_UWord16 *initIndex,
-                                               const WebRtc_Word16 lenData)
+int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t *data,
+                                          Bitstr_dec *streamData,
+                                          const uint16_t **cdf,
+                                          const uint16_t *initIndex,
+                                          const int16_t lenData)
 {
-  WebRtc_UWord32    W_lower;
-  WebRtc_UWord32    W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord32    W_upper_LSB;
-  WebRtc_UWord32    W_upper_MSB;
-  WebRtc_UWord32    streamval;
-  const WebRtc_UWord16 *streamPtr;
-  const WebRtc_UWord16 *cdfPtr;
+  uint32_t    W_lower;
+  uint32_t    W_upper;
+  uint32_t    W_tmp;
+  uint32_t    W_upper_LSB;
+  uint32_t    W_upper_MSB;
+  uint32_t    streamval;
+  const uint16_t *streamPtr;
+  const uint16_t *cdfPtr;
   int             k;
 
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index 39c437e..b540ed5 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -21,7 +21,7 @@
 /* Tables for piecewise linear cdf functions: y = k*x */
 
 /* x Points for function piecewise() in Q15 */
-static const WebRtc_Word32 kHistEdges[51] = {
+static const int32_t kHistEdges[51] = {
   -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
   -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858,  -91751,  -78644,
   -65536,  -52429,  -39322,  -26215,  -13108,       0,   13107,   26214,   39321,   52428,
@@ -32,7 +32,7 @@
 
 
 /* k Points for function piecewise() in Q0 */
-static const WebRtc_UWord16 kCdfSlope[51] = {
+static const uint16_t kCdfSlope[51] = {
   5,    5,     5,     5,     5,     5,     5,     5,    5,    5,
   5,    5,    13,    23,    47,    87,   154,   315,  700, 1088,
   2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312,
@@ -42,7 +42,7 @@
 };
 
 /* y Points for function piecewise() in Q0 */
-static const WebRtc_UWord16 kCdfLogistic[51] = {
+static const uint16_t kCdfLogistic[51] = {
   0,     2,     4,     6,     8,    10,    12,    14,    16,    18,
   20,    22,    24,    29,    38,    57,    92,   153,   279,   559,
   994,  1983,  4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636,
@@ -64,10 +64,10 @@
  */
 
 
-static __inline WebRtc_UWord16 WebRtcIsacfix_Piecewise(WebRtc_Word32 xinQ15) {
-  WebRtc_Word32 ind;
-  WebRtc_Word32 qtmp1;
-  WebRtc_UWord16 qtmp2;
+static __inline uint16_t WebRtcIsacfix_Piecewise(int32_t xinQ15) {
+  int32_t ind;
+  int32_t qtmp1;
+  uint16_t qtmp2;
 
   /* Find index for x-value */
   qtmp1 = WEBRTC_SPL_SAT(kHistEdges[50],xinQ15,kHistEdges[0]);
@@ -76,7 +76,7 @@
 
   /* Calculate corresponding y-value ans return*/
   qtmp1 = qtmp1 - kHistEdges[ind];
-  qtmp2 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(
+  qtmp2 = (uint16_t)WEBRTC_SPL_RSHIFT_U32(
       WEBRTC_SPL_UMUL_32_16(qtmp1,kCdfSlope[ind]), 15);
   return (kCdfLogistic[ind] + qtmp2);
 }
@@ -97,20 +97,20 @@
  *                            <0 otherwise.
  */
 int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
-                                   WebRtc_Word16 *dataQ7,
-                                   const WebRtc_UWord16 *envQ8,
-                                   const WebRtc_Word16 lenData)
+                                   int16_t *dataQ7,
+                                   const uint16_t *envQ8,
+                                   const int16_t lenData)
 {
-  WebRtc_UWord32 W_lower;
-  WebRtc_UWord32 W_upper;
-  WebRtc_UWord16 W_upper_LSB;
-  WebRtc_UWord16 W_upper_MSB;
-  WebRtc_UWord16 *streamPtr;
-  WebRtc_UWord16 *maxStreamPtr;
-  WebRtc_UWord16 *streamPtrCarry;
-  WebRtc_UWord16 negcarry;
-  WebRtc_UWord32 cdfLo;
-  WebRtc_UWord32 cdfHi;
+  uint32_t W_lower;
+  uint32_t W_upper;
+  uint16_t W_upper_LSB;
+  uint16_t W_upper_MSB;
+  uint16_t *streamPtr;
+  uint16_t *maxStreamPtr;
+  uint16_t *streamPtrCarry;
+  uint16_t negcarry;
+  uint32_t cdfLo;
+  uint32_t cdfHi;
   int k;
 
   /* point to beginning of stream buffer
@@ -148,8 +148,8 @@
 
 
     /* update interval */
-    W_upper_LSB = (WebRtc_UWord16)W_upper;
-    W_upper_MSB = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
+    W_upper_LSB = (uint16_t)W_upper;
+    W_upper_MSB = (uint16_t)WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
     W_lower = WEBRTC_SPL_UMUL_32_16(cdfLo, W_upper_MSB);
     W_lower += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfLo, W_upper_LSB);
     W_upper = WEBRTC_SPL_UMUL_32_16(cdfHi, W_upper_MSB);
@@ -187,11 +187,11 @@
     {
       W_upper = WEBRTC_SPL_LSHIFT_U32(W_upper, 8);
       if (streamData->full == 0) {
-        *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(
+        *streamPtr++ += (uint16_t) WEBRTC_SPL_RSHIFT_U32(
             streamData->streamval, 24);
         streamData->full = 1;
       } else {
-        *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_U32(
+        *streamPtr = (uint16_t) WEBRTC_SPL_LSHIFT_U32(
             WEBRTC_SPL_RSHIFT_U32(streamData->streamval, 24), 8);
         streamData->full = 0;
       }
@@ -228,25 +228,25 @@
  * Return value             : number of bytes in the stream so far
  *                            -1 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(WebRtc_Word16 *dataQ7,
-                                             Bitstr_dec *streamData,
-                                             const WebRtc_Word32 *envQ8,
-                                             const WebRtc_Word16 lenData)
+int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
+                                        Bitstr_dec *streamData,
+                                        const int32_t *envQ8,
+                                        const int16_t lenData)
 {
-  WebRtc_UWord32    W_lower;
-  WebRtc_UWord32    W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord16    W_upper_LSB;
-  WebRtc_UWord16    W_upper_MSB;
-  WebRtc_UWord32    streamVal;
-  WebRtc_UWord16    cdfTmp;
-  WebRtc_Word32     res;
-  WebRtc_Word32     inSqrt;
-  WebRtc_Word32     newRes;
-  const WebRtc_UWord16 *streamPtr;
-  WebRtc_Word16     candQ7;
-  WebRtc_Word16     envCount;
-  WebRtc_UWord16    tmpARSpecQ8 = 0;
+  uint32_t    W_lower;
+  uint32_t    W_upper;
+  uint32_t    W_tmp;
+  uint16_t    W_upper_LSB;
+  uint16_t    W_upper_MSB;
+  uint32_t    streamVal;
+  uint16_t    cdfTmp;
+  int32_t     res;
+  int32_t     inSqrt;
+  int32_t     newRes;
+  const uint16_t *streamPtr;
+  int16_t     candQ7;
+  int16_t     envCount;
+  uint16_t    tmpARSpecQ8 = 0;
   int             k, i;
 
 
@@ -266,7 +266,7 @@
   }
 
 
-  res = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,
+  res = WEBRTC_SPL_LSHIFT_W32((int32_t)1,
                                WEBRTC_SPL_RSHIFT_W16(WebRtcSpl_GetSizeInBits(envQ8[0]), 1));
   envCount = 0;
 
@@ -290,13 +290,13 @@
       newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
     } while (newRes != res && i-- > 0);
 
-    tmpARSpecQ8 = (WebRtc_UWord16)newRes;
+    tmpARSpecQ8 = (uint16_t)newRes;
 
     for(k4 = 0; k4 < 4; k4++)
     {
       /* find the integer *data for which streamVal lies in [W_lower+1, W_upper] */
-      W_upper_LSB = (WebRtc_UWord16) (W_upper & 0x0000FFFF);
-      W_upper_MSB = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
+      W_upper_LSB = (uint16_t) (W_upper & 0x0000FFFF);
+      W_upper_MSB = (uint16_t) WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
 
       /* find first candidate by inverting the logistic cdf
        * Input dither value collected from io-stream */
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
index 9aa49da..584bc47 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
@@ -38,9 +38,9 @@
  */
 int WebRtcIsacfix_EncLogisticMulti2(
     Bitstr_enc *streamData,
-    WebRtc_Word16 *dataQ7,
-    const WebRtc_UWord16 *env,
-    const WebRtc_Word16 lenData);
+    int16_t *dataQ7,
+    const uint16_t *env,
+    const int16_t lenData);
 
 
 /****************************************************************************
@@ -54,7 +54,7 @@
  *
  * Return value             : number of bytes in the stream
  */
-WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
+int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
 
 
 /****************************************************************************
@@ -74,11 +74,11 @@
  * Return value             : number of bytes in the stream so far
  *                            <0 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(
-    WebRtc_Word16 *data,
+int16_t WebRtcIsacfix_DecLogisticMulti2(
+    int16_t *data,
     Bitstr_dec *streamData,
-    const WebRtc_Word32 *env,
-    const WebRtc_Word16 lenData);
+    const int32_t *env,
+    const int16_t lenData);
 
 
 /****************************************************************************
@@ -97,9 +97,9 @@
  */
 int WebRtcIsacfix_EncHistMulti(
     Bitstr_enc *streamData,
-    const WebRtc_Word16 *data,
-    const WebRtc_UWord16 **cdf,
-    const WebRtc_Word16 lenData);
+    const int16_t *data,
+    const uint16_t **cdf,
+    const int16_t lenData);
 
 
 /****************************************************************************
@@ -122,12 +122,12 @@
  * Return value             : number of bytes in the stream
  *                            <0 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(
-    WebRtc_Word16 *data,
+int16_t WebRtcIsacfix_DecHistBisectMulti(
+    int16_t *data,
     Bitstr_dec *streamData,
-    const WebRtc_UWord16 **cdf,
-    const WebRtc_UWord16 *cdfSize,
-    const WebRtc_Word16 lenData);
+    const uint16_t **cdf,
+    const uint16_t *cdfSize,
+    const int16_t lenData);
 
 
 /****************************************************************************
@@ -150,11 +150,11 @@
  * Return value             : number of bytes in original stream
  *                            <0 if error detected
  */
-WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(
-    WebRtc_Word16 *data,
+int16_t WebRtcIsacfix_DecHistOneStepMulti(
+    int16_t *data,
     Bitstr_dec *streamData,
-    const WebRtc_UWord16 **cdf,
-    const WebRtc_UWord16 *initIndex,
-    const WebRtc_Word16 lenData);
+    const uint16_t **cdf,
+    const uint16_t *initIndex,
+    const int16_t lenData);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index 8e21b29..724a900 100644
--- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -24,14 +24,14 @@
 
 /* array of quantization levels for bottle neck info; Matlab code: */
 /* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
-static const WebRtc_Word16 kQRateTable[12] = {
+static const int16_t kQRateTable[12] = {
   10000, 11115, 12355, 13733, 15265, 16967,
   18860, 20963, 23301, 25900, 28789, 32000
 };
 
 /* 0.1 times the values in the table kQRateTable */
 /* values are in Q16                                         */
-static const WebRtc_Word32 KQRate01[12] = {
+static const int32_t KQRate01[12] = {
   65536000,  72843264,  80969728,  90000589,  100040704, 111194931,
   123600896, 137383117, 152705434, 169738240, 188671590, 209715200
 };
@@ -40,10 +40,10 @@
  * 8 bits/byte * 1000 msec/sec * 1/framelength (in msec)->bits/byte*sec
  * frame length will either be 30 or 60 msec. 8738 is 1/60 in Q19 and 1/30 in Q18
  * The following number is either in Q15 or Q14 depending on the current frame length */
-static const WebRtc_Word32 kBitsByteSec = 4369000;
+static const int32_t kBitsByteSec = 4369000;
 
 /* Received header rate. First value is for 30 ms packets and second for 60 ms */
-static const WebRtc_Word16 kRecHeaderRate[2] = {
+static const int16_t kRecHeaderRate[2] = {
   9333, 4666
 };
 
@@ -51,13 +51,13 @@
    minBwInv 30 ms, maxBwInv 30 ms,
    minBwInv 60 ms, maxBwInv 69 ms
 */
-static const WebRtc_Word32 kInvBandwidth[4] = {
+static const int32_t kInvBandwidth[4] = {
   55539, 25978,
   73213, 29284
 };
 
 /* Number of samples in 25 msec */
-static const WebRtc_Word32 kSamplesIn25msec = 400;
+static const int32_t kSamplesIn25msec = 400;
 
 
 /****************************************************************************
@@ -70,7 +70,7 @@
  *
  * Return value            : 0
  */
-WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr)
+int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr)
 {
   bweStr->prevFrameSizeMs       = INIT_FRAME_LEN;
   bweStr->prevRtpNumber         = 0;
@@ -93,15 +93,15 @@
   bweStr->recBw                 = INIT_BN_EST;
   bweStr->recBwAvgQ             = INIT_BN_EST_Q7;
   bweStr->recBwAvg              = INIT_REC_BN_EST_Q5;
-  bweStr->recJitter             = (WebRtc_Word32) 327680;   /* 10 in Q15 */
+  bweStr->recJitter             = (int32_t) 327680;   /* 10 in Q15 */
   bweStr->recJitterShortTerm    = 0;
-  bweStr->recJitterShortTermAbs = (WebRtc_Word32) 40960;    /* 5 in Q13 */
-  bweStr->recMaxDelay           = (WebRtc_Word32) 10;
-  bweStr->recMaxDelayAvgQ       = (WebRtc_Word32) 5120;     /* 10 in Q9 */
+  bweStr->recJitterShortTermAbs = (int32_t) 40960;    /* 5 in Q13 */
+  bweStr->recMaxDelay           = (int32_t) 10;
+  bweStr->recMaxDelayAvgQ       = (int32_t) 5120;     /* 10 in Q9 */
   bweStr->recHeaderRate         = INIT_HDR_RATE;
   bweStr->countRecPkts          = 0;
   bweStr->sendBwAvg             = INIT_BN_EST_Q7;
-  bweStr->sendMaxDelayAvg       = (WebRtc_Word32) 5120;     /* 10 in Q9 */
+  bweStr->sendMaxDelayAvg       = (int32_t) 5120;     /* 10 in Q9 */
 
   bweStr->countHighSpeedRec     = 0;
   bweStr->highSpeedRec          = 0;
@@ -139,42 +139,42 @@
  * Return value            : 0 if everything went fine,
  *                           -1 otherwise
  */
-WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
-                                               const WebRtc_UWord16 rtpNumber,
-                                               const WebRtc_Word16  frameSize,
-                                               const WebRtc_UWord32 sendTime,
-                                               const WebRtc_UWord32 arrivalTime,
-                                               const WebRtc_Word16  pksize,
-                                               const WebRtc_UWord16 Index)
+int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
+                                         const uint16_t rtpNumber,
+                                         const int16_t  frameSize,
+                                         const uint32_t sendTime,
+                                         const uint32_t arrivalTime,
+                                         const int16_t  pksize,
+                                         const uint16_t Index)
 {
-  WebRtc_UWord16  weight = 0;
-  WebRtc_UWord32  currBwInv = 0;
-  WebRtc_UWord16  recRtpRate;
-  WebRtc_UWord32  arrTimeProj;
-  WebRtc_Word32   arrTimeDiff;
-  WebRtc_Word32   arrTimeNoise;
-  WebRtc_Word32   arrTimeNoiseAbs;
-  WebRtc_Word32   sendTimeDiff;
+  uint16_t  weight = 0;
+  uint32_t  currBwInv = 0;
+  uint16_t  recRtpRate;
+  uint32_t  arrTimeProj;
+  int32_t   arrTimeDiff;
+  int32_t   arrTimeNoise;
+  int32_t   arrTimeNoiseAbs;
+  int32_t   sendTimeDiff;
 
-  WebRtc_Word32 delayCorrFactor = DELAY_CORRECTION_MED;
-  WebRtc_Word32 lateDiff = 0;
-  WebRtc_Word16 immediateSet = 0;
-  WebRtc_Word32 frameSizeSampl;
+  int32_t delayCorrFactor = DELAY_CORRECTION_MED;
+  int32_t lateDiff = 0;
+  int16_t immediateSet = 0;
+  int32_t frameSizeSampl;
 
-  WebRtc_Word32  temp;
-  WebRtc_Word32  msec;
-  WebRtc_UWord32 exponent;
-  WebRtc_UWord32 reductionFactor;
-  WebRtc_UWord32 numBytesInv;
-  WebRtc_Word32  sign;
+  int32_t  temp;
+  int32_t  msec;
+  uint32_t exponent;
+  uint32_t reductionFactor;
+  uint32_t numBytesInv;
+  int32_t  sign;
 
-  WebRtc_UWord32 byteSecondsPerBit;
-  WebRtc_UWord32 tempLower;
-  WebRtc_UWord32 tempUpper;
-  WebRtc_Word32 recBwAvgInv;
-  WebRtc_Word32 numPktsExpected;
+  uint32_t byteSecondsPerBit;
+  uint32_t tempLower;
+  uint32_t tempUpper;
+  int32_t recBwAvgInv;
+  int32_t numPktsExpected;
 
-  WebRtc_Word16 errCode;
+  int16_t errCode;
 
   /* UPDATE ESTIMATES FROM OTHER SIDE */
 
@@ -200,8 +200,8 @@
     }
 
     /* kBitsByteSec is in Q15 */
-    recRtpRate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
-                                                                     (WebRtc_Word32)pksize), 15) + bweStr->recHeaderRate;
+    recRtpRate = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
+                                                                     (int32_t)pksize), 15) + bweStr->recHeaderRate;
 
   } else {
     /* If frameSize changed since last call, from 60 to 30, recalculate some values */
@@ -215,8 +215,8 @@
     }
 
     /* kBitsByteSec is in Q14 */
-    recRtpRate = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
-                                                                      (WebRtc_Word32)pksize), 14) + bweStr->recHeaderRate;
+    recRtpRate = (uint16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
+                                                                      (int32_t)pksize), 14) + bweStr->recHeaderRate;
   }
 
 
@@ -243,7 +243,7 @@
   bweStr->countRecPkts++;
 
   /* Calculate framesize in msec */
-  frameSizeSampl = WEBRTC_SPL_MUL_16_16((WebRtc_Word16)SAMPLES_PER_MSEC, frameSize);
+  frameSizeSampl = WEBRTC_SPL_MUL_16_16((int16_t)SAMPLES_PER_MSEC, frameSize);
 
   /* Check that it's not one of the first 9 packets */
   if ( bweStr->countUpdates > 0 ) {
@@ -292,13 +292,13 @@
           reductionFactor = WEBRTC_SPL_RSHIFT_U32(reductionFactor, 11);
 
           if ( reductionFactor != 0 ) {
-            bweStr->recBwInv = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recBwInv, (WebRtc_Word32)reductionFactor);
-            bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwInv, 13);
+            bweStr->recBwInv = WEBRTC_SPL_MUL((int32_t)bweStr->recBwInv, (int32_t)reductionFactor);
+            bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((int32_t)bweStr->recBwInv, 13);
 
           } else {
             /* recBwInv = 1 / (INIT_BN_EST + INIT_HDR_RATE) in Q26 (Q30??)*/
             bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
-                                               WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
+                                               WEBRTC_SPL_LSHIFT_W32(((int32_t)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
           }
 
           /* reset time-since-update counter */
@@ -332,12 +332,12 @@
 
           /* 8000 is 1/2 second (in samples at FS) */
           if (lateDiff > 8000) {
-            delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MAX;
+            delayCorrFactor = (int32_t) DELAY_CORRECTION_MAX;
             bweStr->inWaitPeriod = 1;
             bweStr->startWaitPeriod = arrivalTime;
             immediateSet = 1;
           } else if (lateDiff > 5120) {
-            delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MED;
+            delayCorrFactor = (int32_t) DELAY_CORRECTION_MED;
             immediateSet = 1;
             bweStr->inWaitPeriod = 1;
             bweStr->startWaitPeriod = arrivalTime;
@@ -345,19 +345,19 @@
         }
       }
 
-      if ((bweStr->prevRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->recBwAvg, 5)) &&
-          (recRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwAvg, 5)) &&
+      if ((bweStr->prevRtpRate > WEBRTC_SPL_RSHIFT_W32((int32_t) bweStr->recBwAvg, 5)) &&
+          (recRtpRate > WEBRTC_SPL_RSHIFT_W32((int32_t)bweStr->recBwAvg, 5)) &&
           !bweStr->inWaitPeriod) {
 
         /* test if still in initiation period and increment counter */
         if (bweStr->countUpdates++ > 99) {
           /* constant weight after initiation part, 0.01 in Q13 */
-          weight = (WebRtc_UWord16) 82;
+          weight = (uint16_t) 82;
         } else {
           /* weight decreases with number of updates, 1/countUpdates in Q13  */
-          weight = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
-              (WebRtc_Word32)(8192 + WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->countUpdates, 1)),
-              (WebRtc_Word16)bweStr->countUpdates);
+          weight = (uint16_t) WebRtcSpl_DivW32W16(
+              (int32_t)(8192 + WEBRTC_SPL_RSHIFT_W32((int32_t) bweStr->countUpdates, 1)),
+              (int16_t)bweStr->countUpdates);
         }
 
         /* Bottle Neck Estimation */
@@ -373,9 +373,9 @@
         }
 
         /* compute inverse receiving rate for last packet, in Q19 */
-        numBytesInv = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
-            (WebRtc_Word32)(524288 + WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)pksize + HEADER_SIZE), 1)),
-            (WebRtc_Word16)(pksize + HEADER_SIZE));
+        numBytesInv = (uint16_t) WebRtcSpl_DivW32W16(
+            (int32_t)(524288 + WEBRTC_SPL_RSHIFT_W32(((int32_t)pksize + HEADER_SIZE), 1)),
+            (int16_t)(pksize + HEADER_SIZE));
 
         /* 8389 is  ~ 1/128000 in Q30 */
         byteSecondsPerBit = WEBRTC_SPL_MUL_16_16(arrTimeDiff, 8389);
@@ -402,7 +402,7 @@
 
         /* update bottle neck rate estimate */
         bweStr->recBwInv = WEBRTC_SPL_UMUL(weight, currBwInv) +
-            WEBRTC_SPL_UMUL((WebRtc_UWord32) 8192 - weight, bweStr->recBwInv);
+            WEBRTC_SPL_UMUL((uint32_t) 8192 - weight, bweStr->recBwInv);
 
         /* Shift back to Q30 from Q40 (actual used bits shouldn't be more than 27 based on minBwInv)
            up to 30 bits used with Q13 weight */
@@ -417,24 +417,24 @@
            and NOT right shifting recBwAvg 5 bits to an integer
            At max 13 bits are used
            shift to Q5 */
-        recBwAvgInv = WEBRTC_SPL_UDIV((WebRtc_UWord32)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
+        recBwAvgInv = WEBRTC_SPL_UDIV((uint32_t)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
                                       bweStr->recBwAvg);
 
         /* Calculate Projected arrival time difference */
 
         /* The numerator of the quotient can be 22 bits so right shift inv by 4 to avoid overflow
            result in Q22 */
-        arrTimeProj = WEBRTC_SPL_MUL((WebRtc_Word32)8000, recBwAvgInv);
+        arrTimeProj = WEBRTC_SPL_MUL((int32_t)8000, recBwAvgInv);
         /* shift to Q22 */
         arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 4);
         /* complete calulation */
-        arrTimeProj = WEBRTC_SPL_MUL(((WebRtc_Word32)pksize + HEADER_SIZE), arrTimeProj);
+        arrTimeProj = WEBRTC_SPL_MUL(((int32_t)pksize + HEADER_SIZE), arrTimeProj);
         /* shift to Q10 */
         arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 12);
 
         /* difference between projected and actual arrival time differences */
         /* Q9 (only shift arrTimeDiff by 5 to simulate divide by 16 (need to revisit if change sampling rate) DH */
-        if (WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) > (WebRtc_Word32)arrTimeProj) {
+        if (WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) > (int32_t)arrTimeProj) {
           arrTimeNoise = WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) -  arrTimeProj;
           sign = 1;
         } else {
@@ -454,8 +454,8 @@
         bweStr->recJitter = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitter, 10);
 
         /* Maximum jitter is 10 msec in Q15 */
-        if (bweStr->recJitter > (WebRtc_Word32)327680) {
-          bweStr->recJitter = (WebRtc_Word32)327680;
+        if (bweStr->recJitter > (int32_t)327680) {
+          bweStr->recJitter = (int32_t)327680;
         }
 
         /* short term averaged absolute jitter */
@@ -520,8 +520,8 @@
     bweStr->recBw = WEBRTC_SPL_UMUL(delayCorrFactor, bweStr->recBw);
     bweStr->recBw = WEBRTC_SPL_RSHIFT_U32(bweStr->recBw, 10);
 
-    if (bweStr->recBw < (WebRtc_Word32) MIN_ISAC_BW) {
-      bweStr->recBw = (WebRtc_Word32) MIN_ISAC_BW;
+    if (bweStr->recBw < (int32_t) MIN_ISAC_BW) {
+      bweStr->recBw = (int32_t) MIN_ISAC_BW;
     }
 
     bweStr->recBwAvg = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw + bweStr->recHeaderRate, 5);
@@ -542,10 +542,10 @@
 /* This function updates the send bottle neck rate                                                   */
 /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
 /* returns 0 if everything went fine, -1 otherwise                                                   */
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
-                                              const WebRtc_Word16 Index)
+int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
+                                        const int16_t Index)
 {
-  WebRtc_UWord16 RateInd;
+  uint16_t RateInd;
 
   if ( (Index < 0) || (Index > 23) ) {
     return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
@@ -558,7 +558,7 @@
     /* compute the jitter estimate as decoded on the other side in Q9 */
     /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MAX_ISAC_MD */
     bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
-        WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MAX_ISAC_MD, 9));
+        WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((int32_t)MAX_ISAC_MD, 9));
     bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
 
   } else {
@@ -566,7 +566,7 @@
     /* compute the jitter estimate as decoded on the other side in Q9 */
     /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MIN_ISAC_MD */
     bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
-        WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MIN_ISAC_MD,9));
+        WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((int32_t)MIN_ISAC_MD,9));
     bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
 
   }
@@ -605,18 +605,18 @@
  * Return:
  *      bandwith and jitter index (0..23)
  */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr)
+uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr)
 {
-  WebRtc_Word32  rate;
-  WebRtc_Word32  maxDelay;
-  WebRtc_UWord16 rateInd;
-  WebRtc_UWord16 maxDelayBit;
-  WebRtc_Word32  tempTerm1;
-  WebRtc_Word32  tempTerm2;
-  WebRtc_Word32  tempTermX;
-  WebRtc_Word32  tempTermY;
-  WebRtc_Word32  tempMin;
-  WebRtc_Word32  tempMax;
+  int32_t  rate;
+  int32_t  maxDelay;
+  uint16_t rateInd;
+  uint16_t maxDelayBit;
+  int32_t  tempTerm1;
+  int32_t  tempTerm2;
+  int32_t  tempTermX;
+  int32_t  tempTermY;
+  int32_t  tempMin;
+  int32_t  tempMax;
 
   /* Get Rate Index */
 
@@ -627,7 +627,7 @@
 
   /* recBwAvg = 0.9 * recBwAvg + 0.1 * (rate + bweStr->recHeaderRate), 0.9 and 0.1 in Q9 */
   bweStr->recBwAvg = WEBRTC_SPL_UMUL(922, bweStr->recBwAvg) +
-      WEBRTC_SPL_UMUL(102, WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)rate + bweStr->recHeaderRate, 5));
+      WEBRTC_SPL_UMUL(102, WEBRTC_SPL_LSHIFT_U32((uint32_t)rate + bweStr->recHeaderRate, 5));
   bweStr->recBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 10);
 
   /* Find quantization index that gives the closest rate after averaging.
@@ -649,7 +649,7 @@
   tempTermX = WEBRTC_SPL_UMUL(461, bweStr->recBwAvgQ) - tempTerm1;
 
   /* rate in Q16 */
-  tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)rate, 16);
+  tempTermY = WEBRTC_SPL_LSHIFT_W32((int32_t)rate, 16);
 
   /* 0.1 * kQRateTable[rateInd] = KQRate01[rateInd] */
   tempTerm1 = tempTermX + KQRate01[rateInd] - tempTermY;
@@ -690,8 +690,8 @@
   /* Update quantized max delay average */
   tempMax = 652800; /* MAX_ISAC_MD * 0.1 in Q18 */
   tempMin = 130560; /* MIN_ISAC_MD * 0.1 in Q18 */
-  tempTermX = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recMaxDelayAvgQ, (WebRtc_Word32)461);
-  tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)maxDelay, 18);
+  tempTermX = WEBRTC_SPL_MUL((int32_t)bweStr->recMaxDelayAvgQ, (int32_t)461);
+  tempTermY = WEBRTC_SPL_LSHIFT_W32((int32_t)maxDelay, 18);
 
   tempTerm1 = tempTermX + tempMax - tempTermY;
   tempTerm2 = tempTermY - tempTermX - tempMin;
@@ -711,24 +711,24 @@
   }
 
   /* Return bandwitdh and jitter index (0..23) */
-  return (WebRtc_UWord16)(rateInd + maxDelayBit);
+  return (uint16_t)(rateInd + maxDelayBit);
 }
 
 /* get the bottle neck rate from far side to here, as estimated on this side */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
+uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
 {
-  WebRtc_UWord32  recBw;
-  WebRtc_Word32   jitter_sign; /* Q8 */
-  WebRtc_Word32   bw_adjust;   /* Q16 */
-  WebRtc_Word32   rec_jitter_short_term_abs_inv; /* Q18 */
-  WebRtc_Word32   temp;
+  uint32_t  recBw;
+  int32_t   jitter_sign; /* Q8 */
+  int32_t   bw_adjust;   /* Q16 */
+  int32_t   rec_jitter_short_term_abs_inv; /* Q18 */
+  int32_t   temp;
 
   /* Q18  rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
      2^18 then needs to be shifted 13 bits to 2^31 */
   rec_jitter_short_term_abs_inv = WEBRTC_SPL_UDIV(0x80000000, bweStr->recJitterShortTermAbs);
 
   /* Q27 = 9 + 18 */
-  jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (WebRtc_Word32)rec_jitter_short_term_abs_inv);
+  jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (int32_t)rec_jitter_short_term_abs_inv);
 
   if (jitter_sign < 0) {
     temp = -jitter_sign;
@@ -748,9 +748,9 @@
     temp = WEBRTC_SPL_MUL(jitter_sign, temp);
     temp = -temp;
     temp = WEBRTC_SPL_RSHIFT_W32(temp, 8);
-    bw_adjust = (WebRtc_UWord32)65536 + temp; /* (1 << 16) + temp; */
+    bw_adjust = (uint32_t)65536 + temp; /* (1 << 16) + temp; */
   } else {
-    bw_adjust = (WebRtc_UWord32)65536 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(jitter_sign, temp), 8);/* (1 << 16) - ((jitter_sign * temp) >> 8); */
+    bw_adjust = (uint32_t)65536 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(jitter_sign, temp), 8);/* (1 << 16) - ((jitter_sign * temp) >> 8); */
   }
 
   //make sure following multiplication won't overflow
@@ -769,15 +769,15 @@
     recBw = MAX_ISAC_BW;
   }
 
-  return  (WebRtc_UWord16) recBw;
+  return  (uint16_t) recBw;
 }
 
 /* Returns the mmax delay (in ms) */
-WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
+int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
 {
-  WebRtc_Word16 recMaxDelay;
+  int16_t recMaxDelay;
 
-  recMaxDelay = (WebRtc_Word16)  WEBRTC_SPL_RSHIFT_W32(bweStr->recMaxDelay, 15);
+  recMaxDelay = (int16_t)  WEBRTC_SPL_RSHIFT_W32(bweStr->recMaxDelay, 15);
 
   /* limit range of jitter estimate */
   if (recMaxDelay < MIN_ISAC_MD) {
@@ -790,11 +790,11 @@
 }
 
 /* get the bottle neck rate from here to far side, as estimated by far side */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bweStr)
+int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bweStr)
 {
-  WebRtc_Word16 send_bw;
+  int16_t send_bw;
 
-  send_bw = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7);
+  send_bw = (int16_t) WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7);
 
   /* limit range of bottle neck rate */
   if (send_bw < MIN_ISAC_BW) {
@@ -809,11 +809,11 @@
 
 
 /* Returns the max delay value from the other side in ms */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bweStr)
+int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bweStr)
 {
-  WebRtc_Word16 send_max_delay;
+  int16_t send_max_delay;
 
-  send_max_delay = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
+  send_max_delay = (int16_t) WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
 
   /* limit range of jitter estimate */
   if (send_max_delay < MIN_ISAC_MD) {
@@ -832,17 +832,17 @@
  * update long-term average bitrate and amount of data in buffer
  * returns minimum payload size (bytes)
  */
-WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
-                                         WebRtc_Word16 StreamSize,                    /* bytes in bitstream */
-                                         const WebRtc_Word16 FrameSamples,            /* samples per frame */
-                                         const WebRtc_Word16 BottleNeck,        /* bottle neck rate; excl headers (bps) */
-                                         const WebRtc_Word16 DelayBuildUp)      /* max delay from bottle neck buffering (ms) */
+uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
+                                   int16_t StreamSize,                    /* bytes in bitstream */
+                                   const int16_t FrameSamples,            /* samples per frame */
+                                   const int16_t BottleNeck,        /* bottle neck rate; excl headers (bps) */
+                                   const int16_t DelayBuildUp)      /* max delay from bottle neck buffering (ms) */
 {
-  WebRtc_Word32 MinRate = 0;
-  WebRtc_UWord16    MinBytes;
-  WebRtc_Word16 TransmissionTime;
-  WebRtc_Word32 inv_Q12;
-  WebRtc_Word32 den;
+  int32_t MinRate = 0;
+  uint16_t    MinBytes;
+  int16_t TransmissionTime;
+  int32_t inv_Q12;
+  int32_t den;
 
 
   /* first 10 packets @ low rate, then INIT_BURST_LEN packets @ fixed rate of INIT_RATE bps */
@@ -887,7 +887,7 @@
   //round and shift before conversion
   MinRate += 256;
   MinRate = WEBRTC_SPL_RSHIFT_W32(MinRate, 9);
-  MinBytes = (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
+  MinBytes = (uint16_t)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
 
   /* StreamSize will be adjusted if less than MinBytes */
   if (StreamSize < MinBytes) {
@@ -904,12 +904,12 @@
         State->ExceedAgo = 0;
       }
     } else {
-      State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);       /* ms */
+      State->ExceedAgo += (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);       /* ms */
       State->PrevExceed = 1;
     }
   } else {
     State->PrevExceed = 0;
-    State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);           /* ms */
+    State->ExceedAgo += (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);           /* ms */
   }
 
   /* set burst flag if bottle neck not exceeded for long time */
@@ -923,9 +923,9 @@
 
 
   /* Update buffer delay */
-  TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck);    /* ms */
+  TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck);    /* ms */
   State->StillBuffered += TransmissionTime;
-  State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);  //>>4 =  SAMPLES_PER_MSEC        /* ms */
+  State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);  //>>4 =  SAMPLES_PER_MSEC        /* ms */
   if (State->StillBuffered < 0) {
     State->StillBuffered = 0;
   }
@@ -942,19 +942,19 @@
  * update long-term average bitrate and amount of data in buffer
  */
 void WebRtcIsacfix_UpdateRateModel(RateModel *State,
-                                   WebRtc_Word16 StreamSize,                    /* bytes in bitstream */
-                                   const WebRtc_Word16 FrameSamples,            /* samples per frame */
-                                   const WebRtc_Word16 BottleNeck)        /* bottle neck rate; excl headers (bps) */
+                                   int16_t StreamSize,                    /* bytes in bitstream */
+                                   const int16_t FrameSamples,            /* samples per frame */
+                                   const int16_t BottleNeck)        /* bottle neck rate; excl headers (bps) */
 {
-  WebRtc_Word16 TransmissionTime;
+  int16_t TransmissionTime;
 
   /* avoid the initial "high-rate" burst */
   State->InitCounter = 0;
 
   /* Update buffer delay */
-  TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck);    /* ms */
+  TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck);    /* ms */
   State->StillBuffered += TransmissionTime;
-  State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);            /* ms */
+  State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);            /* ms */
   if (State->StillBuffered < 0) {
     State->StillBuffered = 0;
   }
@@ -975,9 +975,9 @@
 
 
 
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framesamples)
+int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, int16_t current_framesamples)
 {
-  WebRtc_Word16 new_framesamples;
+  int16_t new_framesamples;
 
   new_framesamples = current_framesamples;
 
@@ -1000,20 +1000,20 @@
   return new_framesamples;
 }
 
-WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples)
+int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples)
 {
-  WebRtc_Word16 s2nr = 0;
+  int16_t s2nr = 0;
 
   /* find new SNR value */
   //consider BottleNeck to be in Q10 ( * 1 in Q10)
   switch(framesamples) {
     case 480:
       /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
       break;
     case 960:
       /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
       break;
     default:
       s2nr = -1; /* Error */
diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
index 76a50f8..acd5dd7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
+++ b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
@@ -33,7 +33,7 @@
  * Return value            : 0
  */
 
-WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
+int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
 
 
 /****************************************************************************
@@ -57,17 +57,17 @@
  *                           -1 otherwise
  */
 
-WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr            *bwest_str,
-                                               const WebRtc_UWord16        rtp_number,
-                                               const WebRtc_Word16         frameSize,
-                                               const WebRtc_UWord32    send_ts,
-                                               const WebRtc_UWord32        arr_ts,
-                                               const WebRtc_Word16         pksize,
-                                               const WebRtc_UWord16        Index);
+int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr       *bwest_str,
+                                         const uint16_t        rtp_number,
+                                         const int16_t         frameSize,
+                                         const uint32_t        send_ts,
+                                         const uint32_t        arr_ts,
+                                         const int16_t         pksize,
+                                         const uint16_t        Index);
 
 /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
-                                              const WebRtc_Word16 Index);
+int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
+                                        const int16_t Index);
 
 /****************************************************************************
  * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
@@ -81,47 +81,47 @@
  * Return:
  *      bandwith and jitter index (0..23)
  */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bwest_str);
+uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bwest_str);
 
 /* Returns the bandwidth estimation (in bps) */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bwest_str);
+uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bwest_str);
 
 /* Returns the bandwidth that iSAC should send with in bps */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str);
+int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str);
 
 /* Returns the max delay (in ms) */
-WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
+int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
 
 /* Returns the max delay value from the other side in ms */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
+int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
 
 /*
  * update amount of data in bottle neck buffer and burst handling
  * returns minimum payload size (bytes)
  */
-WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
-                                         WebRtc_Word16 StreamSize,     /* bytes in bitstream */
-                                         const WebRtc_Word16 FrameLen,    /* ms per frame */
-                                         const WebRtc_Word16 BottleNeck,        /* bottle neck rate; excl headers (bps) */
-                                         const WebRtc_Word16 DelayBuildUp);     /* max delay from bottle neck buffering (ms) */
+uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
+                                   int16_t StreamSize,     /* bytes in bitstream */
+                                   const int16_t FrameLen,    /* ms per frame */
+                                   const int16_t BottleNeck,        /* bottle neck rate; excl headers (bps) */
+                                   const int16_t DelayBuildUp);     /* max delay from bottle neck buffering (ms) */
 
 /*
  * update long-term average bitrate and amount of data in buffer
  */
 void WebRtcIsacfix_UpdateRateModel(RateModel *State,
-                                   WebRtc_Word16 StreamSize,    /* bytes in bitstream */
-                                   const WebRtc_Word16 FrameSamples,  /* samples per frame */
-                                   const WebRtc_Word16 BottleNeck);       /* bottle neck rate; excl headers (bps) */
+                                   int16_t StreamSize,    /* bytes in bitstream */
+                                   const int16_t FrameSamples,  /* samples per frame */
+                                   const int16_t BottleNeck);       /* bottle neck rate; excl headers (bps) */
 
 
 void WebRtcIsacfix_InitRateModel(RateModel *State);
 
 /* Returns the new framelength value (input argument: bottle_neck) */
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framelength);
+int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, int16_t current_framelength);
 
 /* Returns the new SNR value (input argument: bottle_neck) */
 //returns snr in Q10
-WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples);
+int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples);
 
 
 #endif /*  WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/codec.h b/modules/audio_coding/codecs/isac/fix/source/codec.h
index 689f329..88c7e1a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -27,23 +27,23 @@
 
 int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr* bwest_str,
                                     Bitstr_dec* streamdata,
-                                    WebRtc_Word32 packet_size,
-                                    WebRtc_UWord16 rtp_seq_number,
-                                    WebRtc_UWord32 send_ts,
-                                    WebRtc_UWord32 arr_ts);
+                                    int32_t packet_size,
+                                    uint16_t rtp_seq_number,
+                                    uint32_t send_ts,
+                                    uint32_t arr_ts);
 
-WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16* signal_out16,
+int16_t WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
                                        ISACFIX_DecInst_t* ISACdec_obj,
-                                       WebRtc_Word16* current_framesamples);
+                                       int16_t* current_framesamples);
 
-WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16* decoded,
+int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
                                           ISACFIX_DecInst_t* ISACdec_obj,
-                                          WebRtc_Word16* current_framesample );
+                                          int16_t* current_framesample );
 
-int WebRtcIsacfix_EncodeImpl(WebRtc_Word16* in,
+int WebRtcIsacfix_EncodeImpl(int16_t* in,
                              ISACFIX_EncInst_t* ISACenc_obj,
                              BwEstimatorstr* bw_estimatordata,
-                             WebRtc_Word16 CodingMode);
+                             int16_t CodingMode);
 
 int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t* ISACenc_obj,
                                    int BWnumber,
@@ -69,93 +69,93 @@
 
 void WebRtcIsacfix_InitTransform();
 
-typedef void (*Time2Spec)(WebRtc_Word16* inre1Q9,
-                          WebRtc_Word16* inre2Q9,
-                          WebRtc_Word16* outre,
-                          WebRtc_Word16* outim);
-typedef void (*Spec2Time)(WebRtc_Word16* inreQ7,
-                          WebRtc_Word16* inimQ7,
-                          WebRtc_Word32* outre1Q16,
-                          WebRtc_Word32* outre2Q16);
+typedef void (*Time2Spec)(int16_t* inre1Q9,
+                          int16_t* inre2Q9,
+                          int16_t* outre,
+                          int16_t* outim);
+typedef void (*Spec2Time)(int16_t* inreQ7,
+                          int16_t* inimQ7,
+                          int32_t* outre1Q16,
+                          int32_t* outre2Q16);
 
 extern Time2Spec WebRtcIsacfix_Time2Spec;
 extern Spec2Time WebRtcIsacfix_Spec2Time;
 
-void WebRtcIsacfix_Time2SpecC(WebRtc_Word16* inre1Q9,
-                              WebRtc_Word16* inre2Q9,
-                              WebRtc_Word16* outre,
-                              WebRtc_Word16* outim);
-void WebRtcIsacfix_Spec2TimeC(WebRtc_Word16* inreQ7,
-                              WebRtc_Word16* inimQ7,
-                              WebRtc_Word32* outre1Q16,
-                              WebRtc_Word32* outre2Q16);
+void WebRtcIsacfix_Time2SpecC(int16_t* inre1Q9,
+                              int16_t* inre2Q9,
+                              int16_t* outre,
+                              int16_t* outim);
+void WebRtcIsacfix_Spec2TimeC(int16_t* inreQ7,
+                              int16_t* inimQ7,
+                              int32_t* outre1Q16,
+                              int32_t* outre2Q16);
 
 #if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
-void WebRtcIsacfix_Time2SpecNeon(WebRtc_Word16* inre1Q9,
-                                 WebRtc_Word16* inre2Q9,
-                                 WebRtc_Word16* outre,
-                                 WebRtc_Word16* outim);
-void WebRtcIsacfix_Spec2TimeNeon(WebRtc_Word16* inreQ7,
-                                 WebRtc_Word16* inimQ7,
-                                 WebRtc_Word32* outre1Q16,
-                                 WebRtc_Word32* outre2Q16);
+void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9,
+                                 int16_t* inre2Q9,
+                                 int16_t* outre,
+                                 int16_t* outim);
+void WebRtcIsacfix_Spec2TimeNeon(int16_t* inreQ7,
+                                 int16_t* inimQ7,
+                                 int32_t* outre1Q16,
+                                 int32_t* outre2Q16);
 #endif
 
 
 /* filterbank functions */
 
-void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16* in,
-                                   WebRtc_Word16* LP16,
-                                   WebRtc_Word16* HP16,
+void WebRtcIsacfix_SplitAndFilter1(int16_t* in,
+                                   int16_t* LP16,
+                                   int16_t* HP16,
                                    PreFiltBankstr* prefiltdata);
 
-void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16* tempin_ch1,
-                                     WebRtc_Word16* tempin_ch2,
-                                     WebRtc_Word16* out16,
+void WebRtcIsacfix_FilterAndCombine1(int16_t* tempin_ch1,
+                                     int16_t* tempin_ch2,
+                                     int16_t* out16,
                                      PostFiltBankstr* postfiltdata);
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
-void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16* in,
-                                   WebRtc_Word16* LP16,
-                                   WebRtc_Word16* HP16,
+void WebRtcIsacfix_SplitAndFilter2(int16_t* in,
+                                   int16_t* LP16,
+                                   int16_t* HP16,
                                    PreFiltBankstr* prefiltdata);
 
-void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16* tempin_ch1,
-                                     WebRtc_Word16* tempin_ch2,
-                                     WebRtc_Word16* out16,
+void WebRtcIsacfix_FilterAndCombine2(int16_t* tempin_ch1,
+                                     int16_t* tempin_ch2,
+                                     int16_t* out16,
                                      PostFiltBankstr* postfiltdata,
-                                     WebRtc_Word16 len);
+                                     int16_t len);
 
 #endif
 
 /* normalized lattice filters */
 
-void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
-                                       WebRtc_Word32* stateGQ15,
-                                       WebRtc_Word16* lat_inQ0,
-                                       WebRtc_Word16* filt_coefQ15,
-                                       WebRtc_Word32* gain_lo_hiQ17,
-                                       WebRtc_Word16 lo_hi,
-                                       WebRtc_Word16* lat_outQ9);
+void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
+                                       int32_t* stateGQ15,
+                                       int16_t* lat_inQ0,
+                                       int16_t* filt_coefQ15,
+                                       int32_t* gain_lo_hiQ17,
+                                       int16_t lo_hi,
+                                       int16_t* lat_outQ9);
 
-void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
-                                       WebRtc_Word16* stateGQ0,
-                                       WebRtc_Word32* lat_inQ25,
-                                       WebRtc_Word16* filt_coefQ15,
-                                       WebRtc_Word32* gain_lo_hiQ17,
-                                       WebRtc_Word16 lo_hi,
-                                       WebRtc_Word16* lat_outQ0);
+void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
+                                       int16_t* stateGQ0,
+                                       int32_t* lat_inQ25,
+                                       int16_t* filt_coefQ15,
+                                       int32_t* gain_lo_hiQ17,
+                                       int16_t lo_hi,
+                                       int16_t* lat_outQ0);
 
 /* TODO(kma): Remove the following functions into individual header files. */
 
 /* Internal functions in both C and ARM Neon versions */
 
-int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
-                            const WebRtc_Word16* __restrict x,
-                            WebRtc_Word16 N,
-                            WebRtc_Word16 order,
-                            WebRtc_Word16* __restrict scale);
+int WebRtcIsacfix_AutocorrC(int32_t* __restrict r,
+                            const int16_t* __restrict x,
+                            int16_t N,
+                            int16_t order,
+                            int16_t* __restrict scale);
 
 void WebRtcIsacfix_FilterMaLoopC(int16_t input0,
                                  int16_t input1,
@@ -165,11 +165,11 @@
                                  int32_t* ptr2);
 
 #if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
-int WebRtcIsacfix_AutocorrNeon(WebRtc_Word32* __restrict r,
-                               const WebRtc_Word16* __restrict x,
-                               WebRtc_Word16 N,
-                               WebRtc_Word16 order,
-                               WebRtc_Word16* __restrict scale);
+int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r,
+                               const int16_t* __restrict x,
+                               int16_t N,
+                               int16_t order,
+                               int16_t* __restrict scale);
 
 void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
                                     int16_t input1,
@@ -181,11 +181,11 @@
 
 /* Function pointers associated with the above functions. */
 
-typedef int (*AutocorrFix)(WebRtc_Word32* __restrict r,
-                           const WebRtc_Word16* __restrict x,
-                           WebRtc_Word16 N,
-                           WebRtc_Word16 order,
-                           WebRtc_Word16* __restrict scale);
+typedef int (*AutocorrFix)(int32_t* __restrict r,
+                           const int16_t* __restrict x,
+                           int16_t N,
+                           int16_t order,
+                           int16_t* __restrict scale);
 extern AutocorrFix WebRtcIsacfix_AutocorrFix;
 
 typedef void (*FilterMaLoopFix)(int16_t input0,
diff --git a/modules/audio_coding/codecs/isac/fix/source/decode.c b/modules/audio_coding/codecs/isac/fix/source/decode.c
index 2e15e7a..ae0d687 100644
--- a/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -27,43 +27,43 @@
 
 
 
-WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16       *signal_out16,
-                                       ISACFIX_DecInst_t *ISACdec_obj,
-                                       WebRtc_Word16       *current_framesamples)
+int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
+                                 ISACFIX_DecInst_t *ISACdec_obj,
+                                 int16_t       *current_framesamples)
 {
   int k;
   int err;
-  WebRtc_Word16 BWno;
-  WebRtc_Word16 len = 0;
+  int16_t BWno;
+  int16_t len = 0;
 
-  WebRtc_Word16 model;
+  int16_t model;
 
 
-  WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES/2];
-  WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES/2];
+  int16_t Vector_Word16_1[FRAMESAMPLES/2];
+  int16_t Vector_Word16_2[FRAMESAMPLES/2];
 
-  WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES/2];
-  WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES/2];
+  int32_t Vector_Word32_1[FRAMESAMPLES/2];
+  int32_t Vector_Word32_2[FRAMESAMPLES/2];
 
-  WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
-  WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
-  WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+  int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
+  int16_t hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
+  int32_t gain_lo_hiQ17[2*SUBFRAMES];
 
-  WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
-  WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
-  WebRtc_Word16 AvgPitchGain_Q12;
+  int16_t PitchLags_Q7[PITCH_SUBFRAMES];
+  int16_t PitchGains_Q12[PITCH_SUBFRAMES];
+  int16_t AvgPitchGain_Q12;
 
-  WebRtc_Word16 tmp_1, tmp_2;
-  WebRtc_Word32 tmp32a, tmp32b;
-  WebRtc_Word16 gainQ13;
+  int16_t tmp_1, tmp_2;
+  int32_t tmp32a, tmp32b;
+  int16_t gainQ13;
 
 
-  WebRtc_Word16 frame_nb; /* counter */
-  WebRtc_Word16 frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
-  WebRtc_Word16 processed_samples;
+  int16_t frame_nb; /* counter */
+  int16_t frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
+  int16_t processed_samples;
 
   /* PLC */
-  WebRtc_Word16 overlapWin[ 240 ];
+  int16_t overlapWin[ 240 ];
 
   (ISACdec_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
   (ISACdec_obj->bitstr_obj).streamval = 0;
@@ -76,8 +76,8 @@
   if (err<0)  // error check
     return err;
 
-  frame_mode = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
-  processed_samples = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
+  frame_mode = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
+  processed_samples = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
 
   err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
   if (err<0)  // error check
@@ -95,7 +95,7 @@
     if (err<0)  // error check
       return err;
 
-    AvgPitchGain_Q12 = (WebRtc_Word16)(((WebRtc_Word32)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2);
+    AvgPitchGain_Q12 = (int16_t)(((int32_t)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2);
 
     /* decode & dequantize FiltCoef */
     err = WebRtcIsacfix_DecodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
@@ -113,7 +113,7 @@
     WebRtcIsacfix_Spec2Time(Vector_Word16_1, Vector_Word16_2, Vector_Word32_1, Vector_Word32_2);
 
     for (k=0; k<FRAMESAMPLES/2; k++) {
-      Vector_Word16_1[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(Vector_Word32_1[k]+64, 7); //Q16 -> Q9
+      Vector_Word16_1[k] = (int16_t)WEBRTC_SPL_RSHIFT_W32(Vector_Word32_1[k]+64, 7); //Q16 -> Q9
     }
 
     /* ----  If this is recovery frame ---- */
@@ -129,14 +129,14 @@
       ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX;    /* DECAY_RATE is in Q15 */
       ISACdec_obj->plcstr_obj.pitchCycles = 0;
 
-      PitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
+      PitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
 
       /* ---- Add-overlap ---- */
       WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
       for( k = 0; k < RECOVERY_OVERLAP; k++ )
         Vector_Word16_1[k] = WEBRTC_SPL_ADD_SAT_W16(
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
 
 
 
@@ -177,11 +177,11 @@
     /* gain = 1.0f - 0.45f * AvgPitchGain; */
     tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
     tmp32b = 262144 - tmp32a;  // Q18
-    gainQ13 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q13
+    gainQ13 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q13
 
     for (k = 0; k < FRAMESAMPLES/2; k++)
     {
-      Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(Vector_Word16_2[k], gainQ13), 3); // Q25
+      Vector_Word32_1[k] = (int32_t) WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(Vector_Word16_2[k], gainQ13), 3); // Q25
     }
 
 
@@ -204,8 +204,8 @@
 
     /* Form the polyphase signals, and compensate for DC offset */
     for (k=0;k<FRAMESAMPLES/2;k++) {
-      tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1)); /* Construct a new upper channel signal*/
-      tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k])); /* Construct a new lower channel signal*/
+      tmp_1 = (int16_t)WebRtcSpl_SatW32ToW16(((int32_t)Vector_Word16_1[k]+Vector_Word16_2[k] + 1)); /* Construct a new upper channel signal*/
+      tmp_2 = (int16_t)WebRtcSpl_SatW32ToW16(((int32_t)Vector_Word16_1[k]-Vector_Word16_2[k])); /* Construct a new lower channel signal*/
       Vector_Word16_1[k] = tmp_1;
       Vector_Word16_2[k] = tmp_2;
     }
diff --git a/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c b/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
index 68c6003..c1221e7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
+++ b/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
@@ -26,13 +26,13 @@
 
 int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
                                     Bitstr_dec  *streamdata,
-                                    WebRtc_Word32  packet_size,
-                                    WebRtc_UWord16 rtp_seq_number,
-                                    WebRtc_UWord32 send_ts,
-                                    WebRtc_UWord32 arr_ts)
+                                    int32_t  packet_size,
+                                    uint16_t rtp_seq_number,
+                                    uint32_t send_ts,
+                                    uint32_t arr_ts)
 {
-  WebRtc_Word16 index;
-  WebRtc_Word16 frame_samples;
+  int16_t index;
+  int16_t frame_samples;
   int err;
 
   /* decode framelength */
@@ -53,10 +53,10 @@
   err = WebRtcIsacfix_UpdateUplinkBwImpl(
       bwest_str,
       rtp_seq_number,
-      (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
+      (uint16_t)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
       send_ts,
       arr_ts,
-      (WebRtc_Word16) packet_size,  /* in bytes */
+      (int16_t) packet_size,  /* in bytes */
       index);
 
   /* error check */
diff --git a/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
index de51658..0021675 100644
--- a/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
+++ b/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
@@ -33,29 +33,29 @@
  * returns the total number of bytes in the stream
  */
 
-static WebRtc_Word16 plc_filterma_Fast(
-    WebRtc_Word16 *In,  /* (i)   Vector to be filtered. InOut[-orderCoef+1]
+static int16_t plc_filterma_Fast(
+    int16_t *In,  /* (i)   Vector to be filtered. InOut[-orderCoef+1]
                            to InOut[-1] contains state */
-    WebRtc_Word16 *Out,  /* (o)   Filtered vector */
-    WebRtc_Word16 *B,   /* (i)   The filter coefficients (in Q0) */
-    WebRtc_Word16 Blen,  /* (i)   Number of B coefficients */
-    WebRtc_Word16 len,   /* (i)  Number of samples to be filtered */
-    WebRtc_Word16 reduceDecay,
-    WebRtc_Word16 decay,
-    WebRtc_Word16 rshift )
+    int16_t *Out,  /* (o)   Filtered vector */
+    int16_t *B,   /* (i)   The filter coefficients (in Q0) */
+    int16_t Blen,  /* (i)   Number of B coefficients */
+    int16_t len,   /* (i)  Number of samples to be filtered */
+    int16_t reduceDecay,
+    int16_t decay,
+    int16_t rshift )
 {
   int i, j;
-  WebRtc_Word32 o;
-  WebRtc_Word32 lim;
+  int32_t o;
+  int32_t lim;
 
-  lim = WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 15 + rshift )-1;
+  lim = WEBRTC_SPL_LSHIFT_W32( (int32_t)1, 15 + rshift )-1;
 
   for (i = 0; i < len; i++)
   {
-    G_CONST WebRtc_Word16 *b_ptr = &B[0];
-    G_CONST WebRtc_Word16 *x_ptr = &In[i];
+    G_CONST int16_t *b_ptr = &B[0];
+    G_CONST int16_t *x_ptr = &In[i];
 
-    o = (WebRtc_Word32)0;
+    o = (int32_t)0;
 
     for (j = 0;j < Blen; j++)
     {
@@ -68,13 +68,13 @@
     o = WEBRTC_SPL_ADD_SAT_W32( o, WEBRTC_SPL_LSHIFT_W32( 1, (rshift-1) ) );
 
     /* saturate according to the domain of the filter coefficients */
-    o = WEBRTC_SPL_SAT((WebRtc_Word32)lim, o, (WebRtc_Word32)-lim);
+    o = WEBRTC_SPL_SAT((int32_t)lim, o, (int32_t)-lim);
 
-    /* o should be in the range of WebRtc_Word16 */
+    /* o should be in the range of int16_t */
     o = WEBRTC_SPL_RSHIFT_W32( o, rshift );
 
     /* decay the output signal; this is specific to plc */
-    *Out++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16)o, decay, 15); // ((o + (WebRtc_Word32)2048) >> 12);
+    *Out++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (int16_t)o, decay, 15); // ((o + (int32_t)2048) >> 12);
 
     /* change the decay */
     decay -= reduceDecay;
@@ -91,13 +91,13 @@
 
 
 
-static __inline WebRtc_Word32 log2_Q8_T( WebRtc_UWord32 x ) {
+static __inline int32_t log2_Q8_T( uint32_t x ) {
 
-  WebRtc_Word32 zeros, lg2;
-  WebRtc_Word16 frac;
+  int32_t zeros, lg2;
+  int16_t frac;
 
   zeros=WebRtcSpl_NormU32(x);
-  frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
+  frac=(int16_t)WEBRTC_SPL_RSHIFT_W32(((uint32_t)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
   /* log2(magn(i)) */
 
   lg2= (WEBRTC_SPL_LSHIFT_W16((31-zeros), 8)+frac);
@@ -105,16 +105,16 @@
 
 }
 
-static __inline WebRtc_Word16  exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
+static __inline int16_t  exp2_Q10_T(int16_t x) { // Both in and out in Q10
 
-  WebRtc_Word16 tmp16_1, tmp16_2;
+  int16_t tmp16_1, tmp16_2;
 
-  tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
-  tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+  tmp16_2=(int16_t)(0x0400|(x&0x03FF));
+  tmp16_1=-(int16_t)WEBRTC_SPL_RSHIFT_W16(x,10);
   if(tmp16_1>0)
-    return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+    return (int16_t) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
   else
-    return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+    return (int16_t) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
 
 }
 
@@ -126,9 +126,9 @@
   The function implements membership values for two sets. The mebership functions are
   of second orders corresponding to half-bell-shapped pulses.
 */
-static void MemshipValQ15( WebRtc_Word16 in, WebRtc_Word16 *A, WebRtc_Word16 *B )
+static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
 {
-  WebRtc_Word16 x;
+  int16_t x;
 
   in -= 700;    /* translate the lowLim to 0, limHigh = 5000 - 700, M = 2150 */
 
@@ -142,12 +142,12 @@
       /* x = in / 2150 {in Q15} = x * 15.2409 {in Q15} =
          x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999     */
 
-      /* we are sure that x is in the range of WebRtc_Word16            */
-      x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
+      /* we are sure that x is in the range of int16_t            */
+      x = (int16_t)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
                            WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
-      *B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
+      *B = (int16_t)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
       *A = WEBRTC_SPL_WORD16_MAX - *B;
     }
     else
@@ -162,11 +162,11 @@
     {
       /* This is a mirror case of the above */
       in = 4300 - in;
-      x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
+      x = (int16_t)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
                            WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
-      *A = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
+      *A = (int16_t)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
       *B = WEBRTC_SPL_WORD16_MAX - *A;
 
     }
@@ -181,11 +181,11 @@
 
 
 
-static void LinearResampler( WebRtc_Word16 *in, WebRtc_Word16 *out, WebRtc_Word16 lenIn, WebRtc_Word16 lenOut )
+static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t lenOut )
 {
-  WebRtc_Word32 n;
-  WebRtc_Word16 resOut, i, j, relativePos, diff; /* */
-  WebRtc_UWord16 udiff;
+  int32_t n;
+  int16_t resOut, i, j, relativePos, diff; /* */
+  uint16_t udiff;
 
   if( lenIn == lenOut )
   {
@@ -193,8 +193,8 @@
     return;
   }
 
-  n = WEBRTC_SPL_MUL_16_16( (WebRtc_Word16)(lenIn-1), RESAMP_RES );
-  resOut = WebRtcSpl_DivW32W16ResW16( n, (WebRtc_Word16)(lenOut-1) );
+  n = WEBRTC_SPL_MUL_16_16( (int16_t)(lenIn-1), RESAMP_RES );
+  resOut = WebRtcSpl_DivW32W16ResW16( n, (int16_t)(lenOut-1) );
 
   out[0] = in[0];
   for( i = 1, j = 0, relativePos = 0; i < lenOut; i++ )
@@ -214,20 +214,20 @@
 
     if( (in[ j ] > 0) && (in[j + 1] < 0) )
     {
-      udiff = (WebRtc_UWord16)(in[ j ] - in[j + 1]);
-      out[ i ] = in[ j ] - (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
+      udiff = (uint16_t)(in[ j ] - in[j + 1]);
+      out[ i ] = in[ j ] - (uint16_t)( ((int32_t)( udiff * relativePos )) >> RESAMP_RES_BIT);
     }
     else
     {
       if( (in[j] < 0) && (in[j+1] > 0) )
       {
-        udiff = (WebRtc_UWord16)( in[j + 1] - in[ j ] );
-        out[ i ] = in[ j ] + (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
+        udiff = (uint16_t)( in[j + 1] - in[ j ] );
+        out[ i ] = in[ j ] + (uint16_t)( ((int32_t)( udiff * relativePos )) >> RESAMP_RES_BIT);
       }
       else
       {
         diff = in[ j + 1 ] - in[ j ];
-        out[ i ] = in[ j ] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
+        out[ i ] = in[ j ] + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
       }
     }
   }
@@ -237,46 +237,46 @@
 
 
 
-WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16 *signal_out16,
-                                          ISACFIX_DecInst_t *ISACdec_obj,
-                                          WebRtc_Word16 *current_framesamples )
+int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
+                                    ISACFIX_DecInst_t *ISACdec_obj,
+                                    int16_t *current_framesamples )
 {
   int subframecnt;
-  WebRtc_Word16 len = 0;
+  int16_t len = 0;
 
-  WebRtc_Word16* Vector_Word16_1;
-  WebRtc_Word16  Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
-  WebRtc_Word16* Vector_Word16_2;
-  WebRtc_Word16  Vector_Word16_Extended_2[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
+  int16_t* Vector_Word16_1;
+  int16_t  Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
+  int16_t* Vector_Word16_2;
+  int16_t  Vector_Word16_Extended_2[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
 
-  WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES_HALF];
-  WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES_HALF];
+  int32_t Vector_Word32_1[FRAMESAMPLES_HALF];
+  int32_t Vector_Word32_2[FRAMESAMPLES_HALF];
 
-  WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
-  WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
+  int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
+  int16_t hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
 
-  WebRtc_Word16 pitchLags_Q7[PITCH_SUBFRAMES];
-  WebRtc_Word16 pitchGains_Q12[PITCH_SUBFRAMES];
+  int16_t pitchLags_Q7[PITCH_SUBFRAMES];
+  int16_t pitchGains_Q12[PITCH_SUBFRAMES];
 
-  WebRtc_Word16 tmp_1, tmp_2;
-  WebRtc_Word32 tmp32a, tmp32b;
-  WebRtc_Word16 gainQ13;
+  int16_t tmp_1, tmp_2;
+  int32_t tmp32a, tmp32b;
+  int16_t gainQ13;
 
-  WebRtc_Word16 myDecayRate;
+  int16_t myDecayRate;
 
   /* ---------- PLC variables ------------ */
-  WebRtc_Word16 lag0, i, k, noiseIndex;
-  WebRtc_Word16 stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10];
+  int16_t lag0, i, k, noiseIndex;
+  int16_t stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10];
 
-  WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+  int32_t gain_lo_hiQ17[2*SUBFRAMES];
 
-  WebRtc_Word16 nLP, pLP, wNoisyLP, wPriodicLP, tmp16, minIdx;
-  WebRtc_Word32 nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff;
-  WebRtc_Word16 noise1, rshift;
+  int16_t nLP, pLP, wNoisyLP, wPriodicLP, tmp16, minIdx;
+  int32_t nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff;
+  int16_t noise1, rshift;
 
 
-  WebRtc_Word16 ltpGain, pitchGain, myVoiceIndicator, myAbs, maxAbs;
-  WebRtc_Word32 varIn, varOut, logVarIn, logVarOut, Q, logMaxAbs;
+  int16_t ltpGain, pitchGain, myVoiceIndicator, myAbs, maxAbs;
+  int32_t varIn, varOut, logVarIn, logVarOut, Q, logMaxAbs;
   int rightShiftIn, rightShiftOut;
 
 
@@ -363,13 +363,13 @@
               PITCH_MAX_LAG + 10 - lag0 + i] );
       maxAbs = (myAbs > maxAbs)? myAbs:maxAbs;
     }
-    logVarIn = log2_Q8_T( (WebRtc_UWord32)( varIn ) ) +
-        (WebRtc_Word32)(rightShiftIn << 8);
-    logVarOut = log2_Q8_T( (WebRtc_UWord32)( varOut ) ) +
-        (WebRtc_Word32)(rightShiftOut << 8);
-    logMaxAbs = log2_Q8_T( (WebRtc_UWord32)( maxAbs ) );
+    logVarIn = log2_Q8_T( (uint32_t)( varIn ) ) +
+        (int32_t)(rightShiftIn << 8);
+    logVarOut = log2_Q8_T( (uint32_t)( varOut ) ) +
+        (int32_t)(rightShiftOut << 8);
+    logMaxAbs = log2_Q8_T( (uint32_t)( maxAbs ) );
 
-    ltpGain = (WebRtc_Word16)(logVarOut - logVarIn);
+    ltpGain = (int16_t)(logVarOut - logVarIn);
     Q = 2 * logMaxAbs - ( logVarOut - 1512 );
 
     /*
@@ -381,8 +381,8 @@
      * --
      */
 
-    logVarIn -= log2_Q8_T( (WebRtc_UWord32)( lag0 ) );
-    tmp16 = (WebRtc_Word16)((logVarIn<<1) - (4<<10) );
+    logVarIn -= log2_Q8_T( (uint32_t)( lag0 ) );
+    tmp16 = (int16_t)((logVarIn<<1) - (4<<10) );
     rightShiftIn = 0;
     if( tmp16 > 4096 )
     {
@@ -457,11 +457,11 @@
   pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
       ((ISACdec_obj->plcstr_obj).stretchLag<<7);
   pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
-  pitchGains_Q12[2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+  pitchGains_Q12[2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
       pitchGains_Q12[3], 1010, 10 );
-  pitchGains_Q12[1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+  pitchGains_Q12[1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
       pitchGains_Q12[2], 1010, 10 );
-  pitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+  pitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
       pitchGains_Q12[1], 1010, 10 );
 
 
@@ -497,9 +497,9 @@
     }
     plc_filterma_Fast(Vector_Word16_1, Vector_Word16_Extended_1,
                       &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF -
-                                                                NOISE_FILTER_LEN], (WebRtc_Word16) NOISE_FILTER_LEN,
-                      (WebRtc_Word16) FRAMESAMPLES_HALF, (WebRtc_Word16)(5),
-                      (ISACdec_obj->plcstr_obj).decayCoeffNoise, (WebRtc_Word16)(6));
+                                                                NOISE_FILTER_LEN], (int16_t) NOISE_FILTER_LEN,
+                      (int16_t) FRAMESAMPLES_HALF, (int16_t)(5),
+                      (ISACdec_obj->plcstr_obj).decayCoeffNoise, (int16_t)(6));
 
     maxCoeff = WebRtcSpl_MaxAbsValueW32(
         &(ISACdec_obj->plcstr_obj).prevHP[
@@ -513,7 +513,7 @@
     }
     for( i = 0; i < NOISE_FILTER_LEN; i++ ) {
       Vector_Word16_1[ FRAMESAMPLES_HALF - NOISE_FILTER_LEN + i] =
-          (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+          (int16_t)WEBRTC_SPL_RSHIFT_W32(
               (ISACdec_obj->plcstr_obj).prevHP[
                   PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN + i], rshift);
     }
@@ -521,15 +521,15 @@
         Vector_Word16_2,
         Vector_Word16_Extended_2,
         &Vector_Word16_1[FRAMESAMPLES_HALF - NOISE_FILTER_LEN],
-        (WebRtc_Word16) NOISE_FILTER_LEN,
-        (WebRtc_Word16) FRAMESAMPLES_HALF,
-        (WebRtc_Word16) (5),
+        (int16_t) NOISE_FILTER_LEN,
+        (int16_t) FRAMESAMPLES_HALF,
+        (int16_t) (5),
         (ISACdec_obj->plcstr_obj).decayCoeffNoise,
-        (WebRtc_Word16) (7) );
+        (int16_t) (7) );
 
     for( i = 0; i < FRAMESAMPLES_HALF; i++ )
       Vector_Word32_2[i] = WEBRTC_SPL_LSHIFT_W32(
-          (WebRtc_Word32)Vector_Word16_Extended_2[i], rshift );
+          (int32_t)Vector_Word16_Extended_2[i], rshift );
 
     Vector_Word16_1 = Vector_Word16_Extended_1;
   }
@@ -541,12 +541,12 @@
       for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
       {
         /* --- Lowpass                                               */
-        pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
             stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
             (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
 
         /* --- Highpass                                              */
-        pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+        pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
             (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
                                              (ISACdec_obj->plcstr_obj).stretchLag +
@@ -590,9 +590,9 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)((
-                      (WebRtc_Word32)stretchPitchLP[k]* 3 +
-                      (WebRtc_Word32)stretchPitchLP1[k])>>2);
+                  stretchPitchLP[k] = (int16_t)((
+                      (int32_t)stretchPitchLP[k]* 3 +
+                      (int32_t)stretchPitchLP1[k])>>2);
                 }
                 break;
               }
@@ -600,9 +600,9 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)((
-                      (WebRtc_Word32)stretchPitchLP[k] +
-                      (WebRtc_Word32)stretchPitchLP1[k] )>>1);
+                  stretchPitchLP[k] = (int16_t)((
+                      (int32_t)stretchPitchLP[k] +
+                      (int32_t)stretchPitchLP1[k] )>>1);
                 }
                 break;
               }
@@ -610,8 +610,8 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)((stretchPitchLP[k] +
-                                                       (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
+                  stretchPitchLP[k] = (int16_t)((stretchPitchLP[k] +
+                                                       (int32_t)stretchPitchLP1[k]*3 )>>2);
                 }
                 break;
               }
@@ -641,8 +641,8 @@
         noise1 = WEBRTC_SPL_RSHIFT_W16(
             (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
 
-        nLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
-            (WebRtc_Word16)((noise1)*(ISACdec_obj->plcstr_obj).std),
+        nLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
+            (int16_t)((noise1)*(ISACdec_obj->plcstr_obj).std),
             (ISACdec_obj->plcstr_obj).decayCoeffNoise, 15 );
 
         /* --- Highpass                                              */
@@ -651,9 +651,9 @@
         noise1 = WEBRTC_SPL_RSHIFT_W16(
             (ISACdec_obj->plcstr_obj).seed, 11 ) - 8;
 
-        nHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+        nHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).decayCoeffNoise,
-            (WebRtc_Word32)(noise1*(ISACdec_obj->plcstr_obj).std) );
+            (int32_t)(noise1*(ISACdec_obj->plcstr_obj).std) );
 
         /* --- lower the muliplier (more decay at next sample) --- */
         (ISACdec_obj->plcstr_obj).decayCoeffNoise -= (myDecayRate);
@@ -662,12 +662,12 @@
 
         /* ------ Periodic Vector ---                                */
         /* --- Lowpass                                               */
-        pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
             stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
             (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
 
         /* --- Highpass                                              */
-        pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+        pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
             (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
                                              (ISACdec_obj->plcstr_obj).stretchLag +
@@ -681,14 +681,14 @@
         }
 
         /* ------ Weighting the noisy and periodic vectors -------   */
-        wNoisyLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
+        wNoisyLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
             (ISACdec_obj->plcstr_obj).A, nLP, 15 ) );
-        wNoisyHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
+        wNoisyHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).A, (nHP) ) );
 
-        wPriodicLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
+        wPriodicLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
             (ISACdec_obj->plcstr_obj).B, pLP, 15));
-        wPriodicHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
+        wPriodicHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).B, pHP));
 
         (ISACdec_obj->plcstr_obj).pitchIndex++;
@@ -720,9 +720,9 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)((
-                      (WebRtc_Word32)stretchPitchLP[k]* 3 +
-                      (WebRtc_Word32)stretchPitchLP1[k] )>>2);
+                  stretchPitchLP[k] = (int16_t)((
+                      (int32_t)stretchPitchLP[k]* 3 +
+                      (int32_t)stretchPitchLP1[k] )>>2);
                 }
                 break;
               }
@@ -730,9 +730,9 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)((
-                      (WebRtc_Word32)stretchPitchLP[k] +
-                      (WebRtc_Word32)stretchPitchLP1[k])>>1);
+                  stretchPitchLP[k] = (int16_t)((
+                      (int32_t)stretchPitchLP[k] +
+                      (int32_t)stretchPitchLP1[k])>>1);
                 }
                 break;
               }
@@ -740,9 +740,9 @@
               {
                 for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
                 {
-                  stretchPitchLP[k] = (WebRtc_Word16)(
+                  stretchPitchLP[k] = (int16_t)(
                       (stretchPitchLP[k] +
-                       (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
+                       (int32_t)stretchPitchLP1[k]*3 )>>2);
                 }
                 break;
               }
@@ -756,9 +756,9 @@
         }
 
         /* ------ Sum the noisy and periodic signals  ------ */
-        Vector_Word16_1[i] = (WebRtc_Word16)WEBRTC_SPL_ADD_SAT_W16(
+        Vector_Word16_1[i] = (int16_t)WEBRTC_SPL_ADD_SAT_W16(
             wNoisyLP, wPriodicLP );
-        Vector_Word32_2[i] = (WebRtc_Word32)WEBRTC_SPL_ADD_SAT_W32(
+        Vector_Word32_2[i] = (int32_t)WEBRTC_SPL_ADD_SAT_W32(
             wNoisyHP, wPriodicHP );
       }
     }
@@ -770,7 +770,7 @@
 
   for( i = 0; i < RECOVERY_OVERLAP; i++ )
   {
-    (ISACdec_obj->plcstr_obj).overlapLP[i] = (WebRtc_Word16)(
+    (ISACdec_obj->plcstr_obj).overlapLP[i] = (int16_t)(
         WEBRTC_SPL_MUL_16_16_RSFT(stretchPitchLP[k],
                                   (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15) );
     k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
@@ -788,11 +788,11 @@
   tmp32a = WEBRTC_SPL_MUL_16_16_RSFT((ISACdec_obj->plcstr_obj).AvgPitchGain_Q12,
                                      29, 0); // Q18
   tmp32b = 262144 - tmp32a;  // Q18
-  gainQ13 = (WebRtc_Word16) (tmp32b >> 5); // Q13
+  gainQ13 = (int16_t) (tmp32b >> 5); // Q13
 
   /* perceptual post-filtering (using normalized lattice filter) */
   for (k = 0; k < FRAMESAMPLES_HALF; k++)
-    Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(
+    Vector_Word32_1[k] = (int32_t) WEBRTC_SPL_MUL_16_16(
         Vector_Word16_2[k], gainQ13) << 3; // Q25
 
 
@@ -810,11 +810,11 @@
   for (k=0;k<FRAMESAMPLES_HALF;k++)
   {
     /* Construct a new upper channel signal*/
-    tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
-                                           ((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1));
+    tmp_1 = (int16_t)WebRtcSpl_SatW32ToW16(
+                                           ((int32_t)Vector_Word16_1[k]+Vector_Word16_2[k] + 1));
     /* Construct a new lower channel signal*/
-    tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
-                                           ((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k]));
+    tmp_2 = (int16_t)WebRtcSpl_SatW32ToW16(
+                                           ((int32_t)Vector_Word16_1[k]-Vector_Word16_2[k]));
     Vector_Word16_1[k] = tmp_1;
     Vector_Word16_2[k] = tmp_2;
   }
diff --git a/modules/audio_coding/codecs/isac/fix/source/encode.c b/modules/audio_coding/codecs/isac/fix/source/encode.c
index cb531e5..e209c0e 100644
--- a/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -28,40 +28,40 @@
 #include <stdio.h>
 
 
-int WebRtcIsacfix_EncodeImpl(WebRtc_Word16      *in,
-                         ISACFIX_EncInst_t  *ISACenc_obj,
-                         BwEstimatorstr      *bw_estimatordata,
-                         WebRtc_Word16         CodingMode)
+int WebRtcIsacfix_EncodeImpl(int16_t      *in,
+                             ISACFIX_EncInst_t  *ISACenc_obj,
+                             BwEstimatorstr      *bw_estimatordata,
+                             int16_t         CodingMode)
 {
-  WebRtc_Word16 stream_length = 0;
-  WebRtc_Word16 usefulstr_len = 0;
+  int16_t stream_length = 0;
+  int16_t usefulstr_len = 0;
   int k;
-  WebRtc_Word16 BWno;
+  int16_t BWno;
 
-  WebRtc_Word16 lofilt_coefQ15[(ORDERLO)*SUBFRAMES];
-  WebRtc_Word16 hifilt_coefQ15[(ORDERHI)*SUBFRAMES];
-  WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+  int16_t lofilt_coefQ15[(ORDERLO)*SUBFRAMES];
+  int16_t hifilt_coefQ15[(ORDERHI)*SUBFRAMES];
+  int32_t gain_lo_hiQ17[2*SUBFRAMES];
 
-  WebRtc_Word16 LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD];
-  WebRtc_Word16 LP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
-  WebRtc_Word16 HP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
+  int16_t LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD];
+  int16_t LP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
+  int16_t HP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
 
-  WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
-  WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
-  WebRtc_Word16 AvgPitchGain_Q12;
+  int16_t PitchLags_Q7[PITCH_SUBFRAMES];
+  int16_t PitchGains_Q12[PITCH_SUBFRAMES];
+  int16_t AvgPitchGain_Q12;
 
-  WebRtc_Word16 frame_mode; /* 0 for 30ms, 1 for 60ms */
-  WebRtc_Word16 processed_samples;
+  int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */
+  int16_t processed_samples;
   int status;
 
-  WebRtc_Word32 bits_gainsQ11;
-  WebRtc_Word16 MinBytes;
-  WebRtc_Word16 bmodel;
+  int32_t bits_gainsQ11;
+  int16_t MinBytes;
+  int16_t bmodel;
 
   transcode_obj transcodingParam;
-  WebRtc_Word16 payloadLimitBytes;
-  WebRtc_Word16 arithLenBeforeEncodingDFT;
-  WebRtc_Word16 iterCntr;
+  int16_t payloadLimitBytes;
+  int16_t arithLenBeforeEncodingDFT;
+  int16_t iterCntr;
 
   /* copy new frame length and bottle neck rate only for the first 10 ms data */
   if (ISACenc_obj->buffer_index == 0) {
@@ -112,7 +112,7 @@
 
     // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
     // 901/1024 is 0.87988281250000
-    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
+    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
                                              ISACenc_obj->current_framesamples);
 
     /* encode frame length */
@@ -276,22 +276,22 @@
   while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) ||
         (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH))
   {
-    WebRtc_Word16 arithLenDFTByte;
-    WebRtc_Word16 bytesLeftQ5;
-    WebRtc_Word16 ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25};
+    int16_t arithLenDFTByte;
+    int16_t bytesLeftQ5;
+    int16_t ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25};
 
     // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs
     // such as DTMF, sweep-sine, ...
     //
     // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14
-    // WebRtc_Word16 scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755};
+    // int16_t scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755};
 
 
     // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit
     // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403
     //
-    WebRtc_Word16 scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500};
-    WebRtc_Word16 idx;
+    int16_t scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500};
+    int16_t idx;
 
     if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION)
     {
@@ -348,8 +348,8 @@
     // scale FFT coefficients to reduce the bit-rate
     for(k = 0; k < FRAMESAMPLES_HALF; k++)
     {
-      LP16a[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
-      LPandHP[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
+      LP16a[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
+      LPandHP[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
     }
 
     // Save data for multiple packets memory
@@ -421,7 +421,7 @@
   {
 
     /* update rate model and get minimum number of bytes in this packet */
-    MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
+    MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (int16_t) stream_length,
                                          ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay);
 
     /* if bitstream is too short, add garbage at the end */
@@ -452,7 +452,7 @@
     {
       if (stream_length & 0x0001){
         ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
-        ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] |= (WebRtc_UWord16)(ISACenc_obj->bitstr_seed & 0xFF);
+        ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] |= (uint16_t)(ISACenc_obj->bitstr_seed & 0xFF);
       } else {
         ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
         ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] = WEBRTC_SPL_LSHIFT_U16(ISACenc_obj->bitstr_seed, 8);
@@ -473,7 +473,7 @@
   else
   {
     /* update rate model */
-    WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
+    WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (int16_t) stream_length,
                                   ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck);
   }
   return stream_length;
@@ -489,17 +489,17 @@
 {
   int ii;
   int status;
-  WebRtc_Word16 BWno = BWnumber;
+  int16_t BWno = BWnumber;
   int stream_length = 0;
 
-  WebRtc_Word16 model;
-  const WebRtc_UWord16 *Q_PitchGain_cdf_ptr[1];
-  const WebRtc_UWord16 **cdf;
+  int16_t model;
+  const uint16_t *Q_PitchGain_cdf_ptr[1];
+  const uint16_t **cdf;
   const ISAC_SaveEncData_t *SaveEnc_str;
-  WebRtc_Word32 tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
-  WebRtc_Word16 tmpLPCindex_g[KLT_ORDER_GAIN<<1];
-  WebRtc_Word16 tmp_fre[FRAMESAMPLES];
-  WebRtc_Word16 tmp_fim[FRAMESAMPLES];
+  int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
+  int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1];
+  int16_t tmp_fre[FRAMESAMPLES];
+  int16_t tmp_fim[FRAMESAMPLES];
 
   SaveEnc_str = ISACenc_obj->SaveEnc_ptr;
 
@@ -537,13 +537,13 @@
   if ((0.0 < scale) && (scale < 1.0)) {
     /* Compensate LPC gain */
     for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
-      tmpLPCcoeffs_g[ii] = (WebRtc_Word32) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]);
+      tmpLPCcoeffs_g[ii] = (int32_t) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]);
     }
 
     /* Scale DFT */
     for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
-      tmp_fre[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fre[ii]) ;
-      tmp_fim[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fim[ii]) ;
+      tmp_fre[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fre[ii]) ;
+      tmp_fim[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fim[ii]) ;
     }
   } else {
     for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index c2716fc..8b46d68 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -65,26 +65,26 @@
 /*
   This function implements the fix-point correspondant function to lrint.
 
-  FLP: (WebRtc_Word32)floor(flt+.499999999999)
+  FLP: (int32_t)floor(flt+.499999999999)
   FIP: (fixVal+roundVal)>>qDomain
 
   where roundVal = 2^(qDomain-1) = 1<<(qDomain-1)
 
 */
-static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal, WebRtc_Word16 qDomain) {
-  WebRtc_Word32 intgr;
-  WebRtc_Word32 roundVal;
+static __inline int32_t CalcLrIntQ(int32_t fixVal, int16_t qDomain) {
+  int32_t intgr;
+  int32_t roundVal;
 
-  roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, qDomain-1);
+  roundVal = WEBRTC_SPL_LSHIFT_W32((int32_t)1, qDomain-1);
   intgr = WEBRTC_SPL_RSHIFT_W32(fixVal+roundVal, qDomain);
 
   return intgr;
 }
 
 /*
-  __inline WebRtc_UWord32 stepwise(WebRtc_Word32 dinQ10) {
+  __inline uint32_t stepwise(int32_t dinQ10) {
 
-  WebRtc_Word32 ind, diQ10, dtQ10;
+  int32_t ind, diQ10, dtQ10;
 
   diQ10 = dinQ10;
   if (diQ10 < DPMIN_Q10)
@@ -110,13 +110,13 @@
    177.445678 should be subtracted (since logN() returns a Q8 value).
    For a X value in Q17, the value 177.445678*17 = 3017 should be
    subtracted */
-static WebRtc_Word16 CalcLogN(WebRtc_Word32 arg) {
-  WebRtc_Word16 zeros, log2, frac, logN;
+static int16_t CalcLogN(int32_t arg) {
+  int16_t zeros, log2, frac, logN;
 
   zeros=WebRtcSpl_NormU32(arg);
-  frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_LSHIFT_W32(arg, zeros)&0x7FFFFFFF, 23);
-  log2=(WebRtc_Word16)(WEBRTC_SPL_LSHIFT_W32(31-zeros, 8)+frac); // log2(x) in Q8
-  logN=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
+  frac=(int16_t)WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_LSHIFT_W32(arg, zeros)&0x7FFFFFFF, 23);
+  log2=(int16_t)(WEBRTC_SPL_LSHIFT_W32(31-zeros, 8)+frac); // log2(x) in Q8
+  logN=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
   logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x.
 
   return logN;
@@ -126,22 +126,22 @@
 /*
   expN(x) = 2^(a*x), where a = log2(e) ~= 1.442695
 
-  Input:  Q8  (WebRtc_Word16)
-  Output: Q17 (WebRtc_Word32)
+  Input:  Q8  (int16_t)
+  Output: Q17 (int32_t)
 
   a = log2(e) = log2(exp(1)) ~= 1.442695  ==>  a = 23637 in Q14 (1.442688)
   To this value, 700 is added or subtracted in order to get an average error
   nearer zero, instead of always same-sign.
 */
 
-static WebRtc_Word32 CalcExpN(WebRtc_Word16 x) {
-  WebRtc_Word16 ax, axINT, axFRAC;
-  WebRtc_Word16 exp16;
-  WebRtc_Word32 exp;
+static int32_t CalcExpN(int16_t x) {
+  int16_t ax, axINT, axFRAC;
+  int16_t exp16;
+  int32_t exp;
 
   if (x>=0) {
-    //  ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
-    ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
+    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
+    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     axINT = WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
     axFRAC = ax&0x00FF;
     exp16 = WEBRTC_SPL_LSHIFT_W32(1, axINT); //Q0
@@ -149,12 +149,12 @@
     exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q0*Q8 = Q8
     exp = WEBRTC_SPL_LSHIFT_W32(exp, 9); //Q17
   } else {
-    //  ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
-    ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
+    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
+    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     ax = -ax;
     axINT = 1 + WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
     axFRAC = 0x00FF - (ax&0x00FF);
-    exp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(32768, axINT); //Q15
+    exp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(32768, axINT); //Q15
     axFRAC = axFRAC+256; //Q8
     exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q15*Q8 = Q23
     exp = WEBRTC_SPL_RSHIFT_W32(exp, 6); //Q17
@@ -165,11 +165,11 @@
 
 
 /* compute correlation from power spectrum */
-static void CalcCorrelation(WebRtc_Word32 *PSpecQ12, WebRtc_Word32 *CorrQ7)
+static void CalcCorrelation(int32_t *PSpecQ12, int32_t *CorrQ7)
 {
-  WebRtc_Word32 summ[FRAMESAMPLES/8];
-  WebRtc_Word32 diff[FRAMESAMPLES/8];
-  WebRtc_Word32 sum;
+  int32_t summ[FRAMESAMPLES/8];
+  int32_t diff[FRAMESAMPLES/8];
+  int32_t sum;
   int k, n;
 
   for (k = 0; k < FRAMESAMPLES/8; k++) {
@@ -199,16 +199,16 @@
 
 
 /* compute inverse AR power spectrum */
-static void CalcInvArSpec(const WebRtc_Word16 *ARCoefQ12,
-                          const WebRtc_Word32 gainQ10,
-                          WebRtc_Word32 *CurveQ16)
+static void CalcInvArSpec(const int16_t *ARCoefQ12,
+                          const int32_t gainQ10,
+                          int32_t *CurveQ16)
 {
-  WebRtc_Word32 CorrQ11[AR_ORDER+1];
-  WebRtc_Word32 sum, tmpGain;
-  WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
-  const WebRtc_Word16 *CS_ptrQ9;
+  int32_t CorrQ11[AR_ORDER+1];
+  int32_t sum, tmpGain;
+  int32_t diffQ16[FRAMESAMPLES/8];
+  const int16_t *CS_ptrQ9;
   int k, n;
-  WebRtc_Word16 round, shftVal = 0, sh;
+  int16_t round, shftVal = 0, sh;
 
   sum = 0;
   for (n = 0; n < AR_ORDER+1; n++)
@@ -269,19 +269,19 @@
   }
 }
 
-static void CalcRootInvArSpec(const WebRtc_Word16 *ARCoefQ12,
-                              const WebRtc_Word32 gainQ10,
-                              WebRtc_UWord16 *CurveQ8)
+static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
+                              const int32_t gainQ10,
+                              uint16_t *CurveQ8)
 {
-  WebRtc_Word32 CorrQ11[AR_ORDER+1];
-  WebRtc_Word32 sum, tmpGain;
-  WebRtc_Word32 summQ16[FRAMESAMPLES/8];
-  WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
+  int32_t CorrQ11[AR_ORDER+1];
+  int32_t sum, tmpGain;
+  int32_t summQ16[FRAMESAMPLES/8];
+  int32_t diffQ16[FRAMESAMPLES/8];
 
-  const WebRtc_Word16 *CS_ptrQ9;
+  const int16_t *CS_ptrQ9;
   int k, n, i;
-  WebRtc_Word16 round, shftVal = 0, sh;
-  WebRtc_Word32 res, in_sqrt, newRes;
+  int16_t round, shftVal = 0, sh;
+  int32_t res, in_sqrt, newRes;
 
   sum = 0;
   for (n = 0; n < AR_ORDER+1; n++)
@@ -357,7 +357,7 @@
       newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
     } while (newRes != res && i-- > 0);
 
-    CurveQ8[k] = (WebRtc_Word16)newRes;
+    CurveQ8[k] = (int16_t)newRes;
   }
   for (k = FRAMESAMPLES/8; k < FRAMESAMPLES/4; k++) {
 
@@ -375,7 +375,7 @@
       newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
     } while (newRes != res && i-- > 0);
 
-    CurveQ8[k] = (WebRtc_Word16)newRes;
+    CurveQ8[k] = (int16_t)newRes;
   }
 
 }
@@ -383,31 +383,31 @@
 
 
 /* generate array of dither samples in Q7 */
-static void GenerateDitherQ7(WebRtc_Word16 *bufQ7,
-                             WebRtc_UWord32 seed,
-                             WebRtc_Word16 length,
-                             WebRtc_Word16 AvgPitchGain_Q12)
+static void GenerateDitherQ7(int16_t *bufQ7,
+                             uint32_t seed,
+                             int16_t length,
+                             int16_t AvgPitchGain_Q12)
 {
   int   k;
-  WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14, shft;
+  int16_t dither1_Q7, dither2_Q7, dither_gain_Q14, shft;
 
   if (AvgPitchGain_Q12 < 614)  /* this threshold should be equal to that in decode_spec() */
   {
     for (k = 0; k < length-2; k += 3)
     {
-      /* new random unsigned WebRtc_Word32 */
+      /* new random unsigned int32_t */
       seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
 
       /* fixed-point dither sample between -64 and 64 (Q7) */
-      dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25); // * 128/4294967295
+      dither1_Q7 = (int16_t)WEBRTC_SPL_RSHIFT_W32((int32_t)seed + 16777216, 25); // * 128/4294967295
 
-      /* new random unsigned WebRtc_Word32 */
+      /* new random unsigned int32_t */
       seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
 
       /* fixed-point dither sample between -64 and 64 */
-      dither2_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(seed + 16777216, 25);
+      dither2_Q7 = (int16_t)WEBRTC_SPL_RSHIFT_W32(seed + 16777216, 25);
 
-      shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 15);
+      shft = (int16_t)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 15);
       if (shft < 5)
       {
         bufQ7[k]   = dither1_Q7;
@@ -430,21 +430,21 @@
   }
   else
   {
-    dither_gain_Q14 = (WebRtc_Word16)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12));
+    dither_gain_Q14 = (int16_t)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12));
 
     /* dither on half of the coefficients */
     for (k = 0; k < length-1; k += 2)
     {
-      /* new random unsigned WebRtc_Word32 */
+      /* new random unsigned int32_t */
       seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
 
       /* fixed-point dither sample between -64 and 64 */
-      dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25);
+      dither1_Q7 = (int16_t)WEBRTC_SPL_RSHIFT_W32((int32_t)seed + 16777216, 25);
 
       /* dither sample is placed in either even or odd index */
-      shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 1);     /* either 0 or 1 */
+      shft = (int16_t)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 1);     /* either 0 or 1 */
 
-      bufQ7[k + shft] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(dither_gain_Q14, dither1_Q7) + 8192, 14);
+      bufQ7[k + shft] = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(dither_gain_Q14, dither1_Q7) + 8192, 14);
       bufQ7[k + 1 - shft] = 0;
     }
   }
@@ -457,18 +457,18 @@
  * function to decode the complex spectrum from the bitstream
  * returns the total number of bytes in the stream
  */
-WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
-                                       WebRtc_Word16 *frQ7,
-                                       WebRtc_Word16 *fiQ7,
-                                       WebRtc_Word16 AvgPitchGain_Q12)
+int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
+                                 int16_t *frQ7,
+                                 int16_t *fiQ7,
+                                 int16_t AvgPitchGain_Q12)
 {
-  WebRtc_Word16  data[FRAMESAMPLES];
-  WebRtc_Word32  invARSpec2_Q16[FRAMESAMPLES/4];
-  WebRtc_Word16  ARCoefQ12[AR_ORDER+1];
-  WebRtc_Word16  RCQ15[AR_ORDER];
-  WebRtc_Word16  gainQ10;
-  WebRtc_Word32  gain2_Q10;
-  WebRtc_Word16  len;
+  int16_t  data[FRAMESAMPLES];
+  int32_t  invARSpec2_Q16[FRAMESAMPLES/4];
+  int16_t  ARCoefQ12[AR_ORDER+1];
+  int16_t  RCQ15[AR_ORDER];
+  int16_t  gainQ10;
+  int32_t  gain2_Q10;
+  int16_t  len;
   int          k;
 
   /* create dither signal */
@@ -489,7 +489,7 @@
 
   /* arithmetic decoding of spectrum */
   /* 'data' input and output. Input = Dither */
-  len = WebRtcIsacfix_DecLogisticMulti2(data, streamdata, invARSpec2_Q16, (WebRtc_Word16)FRAMESAMPLES);
+  len = WebRtcIsacfix_DecLogisticMulti2(data, streamdata, invARSpec2_Q16, (int16_t)FRAMESAMPLES);
 
   if (len<1)
     return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
@@ -499,24 +499,24 @@
   {
     for (k = 0; k < FRAMESAMPLES; k += 4)
     {
-      gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)30, 10),
-                                              (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2195456, 16));
-      *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
-      *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
-      *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
-      *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
+      gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)30, 10),
+                                              (int16_t)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (uint32_t)2195456, 16));
+      *frQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
+      *fiQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
+      *frQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
+      *fiQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
     }
   }
   else
   {
     for (k = 0; k < FRAMESAMPLES; k += 4)
     {
-      gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)36, 10),
-                                              (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2654208, 16));
-      *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
-      *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
-      *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
-      *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
+      gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)36, 10),
+                                              (int16_t)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (uint32_t)2654208, 16));
+      *frQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
+      *fiQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
+      *frQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
+      *fiQ7++ = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
     }
   }
 
@@ -524,24 +524,24 @@
 }
 
 
-int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
-                             const WebRtc_Word16 *fi,
+int WebRtcIsacfix_EncodeSpec(const int16_t *fr,
+                             const int16_t *fi,
                              Bitstr_enc *streamdata,
-                             WebRtc_Word16 AvgPitchGain_Q12)
+                             int16_t AvgPitchGain_Q12)
 {
-  WebRtc_Word16  dataQ7[FRAMESAMPLES];
-  WebRtc_Word32  PSpec[FRAMESAMPLES/4];
-  WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES/4];
-  WebRtc_Word32  CorrQ7[AR_ORDER+1];
-  WebRtc_Word32  CorrQ7_norm[AR_ORDER+1];
-  WebRtc_Word16  RCQ15[AR_ORDER];
-  WebRtc_Word16  ARCoefQ12[AR_ORDER+1];
-  WebRtc_Word32  gain2_Q10;
-  WebRtc_Word16  val;
-  WebRtc_Word32  nrg;
-  WebRtc_UWord32 sum;
-  WebRtc_Word16  lft_shft;
-  WebRtc_Word16  status;
+  int16_t  dataQ7[FRAMESAMPLES];
+  int32_t  PSpec[FRAMESAMPLES/4];
+  uint16_t invARSpecQ8[FRAMESAMPLES/4];
+  int32_t  CorrQ7[AR_ORDER+1];
+  int32_t  CorrQ7_norm[AR_ORDER+1];
+  int16_t  RCQ15[AR_ORDER];
+  int16_t  ARCoefQ12[AR_ORDER+1];
+  int32_t  gain2_Q10;
+  int16_t  val;
+  int32_t  nrg;
+  uint32_t sum;
+  int16_t  lft_shft;
+  int16_t  status;
   int          k, n, j;
 
 
@@ -627,7 +627,7 @@
 
 
   /* arithmetic coding of spectrum */
-  status = WebRtcIsacfix_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, (WebRtc_Word16)FRAMESAMPLES);
+  status = WebRtcIsacfix_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, (int16_t)FRAMESAMPLES);
   if ( status )
     return( status );
 
@@ -636,7 +636,7 @@
 
 
 /* Matlab's LAR definition */
-static void Rc2LarFix(const WebRtc_Word16 *rcQ15, WebRtc_Word32 *larQ17, WebRtc_Word16 order) {
+static void Rc2LarFix(const int16_t *rcQ15, int32_t *larQ17, int16_t order) {
 
   /*
 
@@ -670,8 +670,8 @@
   */
 
   int k;
-  WebRtc_Word16 rc;
-  WebRtc_Word32 larAbsQ17;
+  int16_t rc;
+  int32_t larAbsQ17;
 
   for (k = 0; k < order; k++) {
 
@@ -702,7 +702,7 @@
 }
 
 
-static void Lar2RcFix(const WebRtc_Word32 *larQ17, WebRtc_Word16 *rcQ15,  WebRtc_Word16 order) {
+static void Lar2RcFix(const int32_t *larQ17, int16_t *rcQ15,  int16_t order) {
 
   /*
     This is a piece-wise implemenetation of a lar2rc-function
@@ -710,12 +710,12 @@
   */
 
   int k;
-  WebRtc_Word16 larAbsQ11;
-  WebRtc_Word32 rc;
+  int16_t larAbsQ11;
+  int32_t rc;
 
   for (k = 0; k < order; k++) {
 
-    larAbsQ11 = (WebRtc_Word16) WEBRTC_SPL_ABS_W32(WEBRTC_SPL_RSHIFT_W32(larQ17[k]+32,6)); //Q11
+    larAbsQ11 = (int16_t) WEBRTC_SPL_ABS_W32(WEBRTC_SPL_RSHIFT_W32(larQ17[k]+32,6)); //Q11
 
     if (larAbsQ11<4097) { //2.000012018559 in Q11
       // Q11*Q16>>12 = Q15
@@ -735,21 +735,21 @@
       rc = -rc;
     }
 
-    rcQ15[k] = (WebRtc_Word16) rc;  // Q15
+    rcQ15[k] = (int16_t) rc;  // Q15
   }
 }
 
-static void Poly2LarFix(WebRtc_Word16 *lowbandQ15,
-                        WebRtc_Word16 orderLo,
-                        WebRtc_Word16 *hibandQ15,
-                        WebRtc_Word16 orderHi,
-                        WebRtc_Word16 Nsub,
-                        WebRtc_Word32 *larsQ17) {
+static void Poly2LarFix(int16_t *lowbandQ15,
+                        int16_t orderLo,
+                        int16_t *hibandQ15,
+                        int16_t orderHi,
+                        int16_t Nsub,
+                        int32_t *larsQ17) {
 
   int k, n;
-  WebRtc_Word32 *outpQ17;
-  WebRtc_Word16 orderTot;
-  WebRtc_Word32 larQ17[MAX_ORDER];   // Size 7+6 is enough
+  int32_t *outpQ17;
+  int16_t orderTot;
+  int32_t larQ17[MAX_ORDER];   // Size 7+6 is enough
 
   orderTot = (orderLo + orderHi);
   outpQ17 = larsQ17;
@@ -772,18 +772,18 @@
 }
 
 
-static void Lar2polyFix(WebRtc_Word32 *larsQ17,
-                        WebRtc_Word16 *lowbandQ15,
-                        WebRtc_Word16 orderLo,
-                        WebRtc_Word16 *hibandQ15,
-                        WebRtc_Word16 orderHi,
-                        WebRtc_Word16 Nsub) {
+static void Lar2polyFix(int32_t *larsQ17,
+                        int16_t *lowbandQ15,
+                        int16_t orderLo,
+                        int16_t *hibandQ15,
+                        int16_t orderHi,
+                        int16_t Nsub) {
 
   int k, n;
-  WebRtc_Word16 orderTot;
-  WebRtc_Word16 *outplQ15, *outphQ15;
-  WebRtc_Word32 *inpQ17;
-  WebRtc_Word16 rcQ15[7+6];
+  int16_t orderTot;
+  int16_t *outplQ15, *outphQ15;
+  int32_t *inpQ17;
+  int16_t rcQ15[7+6];
 
   orderTot = (orderLo + orderHi);
   outplQ15 = lowbandQ15;
@@ -918,13 +918,13 @@
   }
 }
 
-int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
-                            WebRtc_Word16 *LPCCoef_loQ15,
-                            WebRtc_Word16 *LPCCoef_hiQ15,
+int WebRtcIsacfix_DecodeLpc(int32_t *gain_lo_hiQ17,
+                            int16_t *LPCCoef_loQ15,
+                            int16_t *LPCCoef_hiQ15,
                             Bitstr_dec *streamdata,
-                            WebRtc_Word16 *outmodel) {
+                            int16_t *outmodel) {
 
-  WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_GAIN+KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
+  int32_t larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_GAIN+KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
   int err;
 
   err = WebRtcIsacfix_DecodeLpcCoef(streamdata, larsQ17, gain_lo_hiQ17, outmodel);
@@ -938,24 +938,24 @@
 
 /* decode & dequantize LPC Coef */
 int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
-                                WebRtc_Word32 *LPCCoefQ17,
-                                WebRtc_Word32 *gain_lo_hiQ17,
-                                WebRtc_Word16 *outmodel)
+                                int32_t *LPCCoefQ17,
+                                int32_t *gain_lo_hiQ17,
+                                int16_t *outmodel)
 {
   int j, k, n;
   int err;
-  WebRtc_Word16 pos, pos2, posg, poss;
-  WebRtc_Word16 gainpos;
-  WebRtc_Word16 model;
-  WebRtc_Word16 index_QQ[KLT_ORDER_SHAPE];
-  WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
-  WebRtc_Word16 tmpcoeffs_sQ10[KLT_ORDER_SHAPE];
-  WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
-  WebRtc_Word32 tmpcoeffs2_sQ18[KLT_ORDER_SHAPE];
-  WebRtc_Word32 sumQQ;
-  WebRtc_Word16 sumQQ16;
-  WebRtc_Word32 tmp32;
+  int16_t pos, pos2, posg, poss;
+  int16_t gainpos;
+  int16_t model;
+  int16_t index_QQ[KLT_ORDER_SHAPE];
+  int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+  int16_t tmpcoeffs_sQ10[KLT_ORDER_SHAPE];
+  int32_t tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
+  int32_t tmpcoeffs2_sQ18[KLT_ORDER_SHAPE];
+  int32_t sumQQ;
+  int16_t sumQQ16;
+  int32_t tmp32;
 
 
 
@@ -1020,14 +1020,14 @@
   for (k=0; k<SUBFRAMES; k++) {
 
     /* log gains */
-    sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+    sumQQ16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
     sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
     sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
     gain_lo_hiQ17[gainpos] = sumQQ; //Q17
     gainpos++;
     posg++;
 
-    sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+    sumQQ16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
     sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
     sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
     gain_lo_hiQ17[gainpos] = sumQQ; //Q17
@@ -1056,28 +1056,28 @@
 }
 
 /* estimate codel length of LPC Coef */
-static int EstCodeLpcCoef(WebRtc_Word32 *LPCCoefQ17,
-                          WebRtc_Word32 *gain_lo_hiQ17,
-                          WebRtc_Word16 *model,
-                          WebRtc_Word32 *sizeQ11,
+static int EstCodeLpcCoef(int32_t *LPCCoefQ17,
+                          int32_t *gain_lo_hiQ17,
+                          int16_t *model,
+                          int32_t *sizeQ11,
                           Bitstr_enc *streamdata,
                           ISAC_SaveEncData_t* encData,
                           transcode_obj *transcodingParam) {
   int j, k, n;
-  WebRtc_Word16 posQQ, pos2QQ, gainpos;
-  WebRtc_Word16  pos, poss, posg, offsg;
-  WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN], index_sQQ[KLT_ORDER_SHAPE];
-  WebRtc_Word16 index_ovr_gQQ[KLT_ORDER_GAIN], index_ovr_sQQ[KLT_ORDER_SHAPE];
-  WebRtc_Word32 BitsQQ;
+  int16_t posQQ, pos2QQ, gainpos;
+  int16_t  pos, poss, posg, offsg;
+  int16_t index_gQQ[KLT_ORDER_GAIN], index_sQQ[KLT_ORDER_SHAPE];
+  int16_t index_ovr_gQQ[KLT_ORDER_GAIN], index_ovr_sQQ[KLT_ORDER_SHAPE];
+  int32_t BitsQQ;
 
-  WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
-  WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs2_sQ17[KLT_ORDER_SHAPE];
-  WebRtc_Word32 sumQQ;
-  WebRtc_Word32 tmp32;
-  WebRtc_Word16 sumQQ16;
+  int16_t tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
+  int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs2_sQ17[KLT_ORDER_SHAPE];
+  int32_t sumQQ;
+  int32_t tmp32;
+  int16_t sumQQ16;
   int status = 0;
 
   /* write LAR coefficients to statistics file */
@@ -1170,7 +1170,7 @@
   for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
   {
     posQQ = WebRtcIsacfix_kSelIndGain[k];
-    pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+    pos2QQ= (int16_t)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
 
     index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
     if (index_gQQ[k] < 0) {
@@ -1194,7 +1194,7 @@
 
   for (k=0; k<KLT_ORDER_SHAPE; k++) //ATTN: ok?
   {
-    index_sQQ[k] = (WebRtc_Word16)(CalcLrIntQ(tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]], 17) + WebRtcIsacfix_kQuantMinShape[k]); //ATTN: ok?
+    index_sQQ[k] = (int16_t)(CalcLrIntQ(tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]], 17) + WebRtcIsacfix_kQuantMinShape[k]); //ATTN: ok?
 
     if (index_sQQ[k] < 0)
       index_sQQ[k] = 0;
@@ -1321,7 +1321,7 @@
   gainpos = 0;
   for (k=0; k<2*SUBFRAMES; k++) {
 
-    sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+    sumQQ16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
     sumQQ16 += WebRtcIsacfix_kMeansGainQ8[0][posg];
     sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
     gain_lo_hiQ17[gainpos] = sumQQ; //Q17
@@ -1333,18 +1333,18 @@
   return 0;
 }
 
-int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
+int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17,
                                  Bitstr_enc *streamdata,
                                  ISAC_SaveEncData_t* encData) {
   int j, k;
-  WebRtc_Word16 posQQ, pos2QQ, gainpos;
-  WebRtc_Word16 posg;
-  WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN];
+  int16_t posQQ, pos2QQ, gainpos;
+  int16_t posg;
+  int16_t index_gQQ[KLT_ORDER_GAIN];
 
-  WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
-  WebRtc_Word32 sumQQ;
+  int16_t tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+  int32_t sumQQ;
   int status = 0;
 
   /* write LAR coefficients to statistics file */
@@ -1409,7 +1409,7 @@
   for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
   {
     posQQ = WebRtcIsacfix_kSelIndGain[k];
-    pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+    pos2QQ= (int16_t)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
 
     index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
     if (index_gQQ[k] < 0) {
@@ -1435,17 +1435,17 @@
 }
 
 
-int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
-                            WebRtc_Word16 *LPCCoef_loQ15,
-                            WebRtc_Word16 *LPCCoef_hiQ15,
-                            WebRtc_Word16 *model,
-                            WebRtc_Word32 *sizeQ11,
+int WebRtcIsacfix_EncodeLpc(int32_t *gain_lo_hiQ17,
+                            int16_t *LPCCoef_loQ15,
+                            int16_t *LPCCoef_hiQ15,
+                            int16_t *model,
+                            int32_t *sizeQ11,
                             Bitstr_enc *streamdata,
                             ISAC_SaveEncData_t* encData,
                             transcode_obj *transcodeParam)
 {
   int status = 0;
-  WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
+  int32_t larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
   // = (6+12)*6 == 108
 
   Poly2LarFix(LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES, larsQ17);
@@ -1463,10 +1463,10 @@
 
 
 /* decode & dequantize RC */
-int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata, WebRtc_Word16 *RCQ15)
+int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata, int16_t *RCQ15)
 {
   int k, err;
-  WebRtc_Word16 index[AR_ORDER];
+  int16_t index[AR_ORDER];
 
   /* entropy decoding of quantization indices */
   err = WebRtcIsacfix_DecHistOneStepMulti(index, streamdata, WebRtcIsacfix_kRcCdfPtr, WebRtcIsacfix_kRcInitInd, AR_ORDER);
@@ -1485,10 +1485,10 @@
 
 
 /* quantize & code RC */
-int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15, Bitstr_enc *streamdata)
+int WebRtcIsacfix_EncodeRcCoef(int16_t *RCQ15, Bitstr_enc *streamdata)
 {
   int k;
-  WebRtc_Word16 index[AR_ORDER];
+  int16_t index[AR_ORDER];
   int status;
 
   /* quantize reflection coefficients (add noise feedback?) */
@@ -1519,10 +1519,10 @@
 
 
 /* decode & dequantize squared Gain */
-int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata, WebRtc_Word32 *gainQ10)
+int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata, int32_t *gainQ10)
 {
   int err;
-  WebRtc_Word16 index;
+  int16_t index;
 
   /* entropy decoding of quantization index */
   err = WebRtcIsacfix_DecHistOneStepMulti(
@@ -1545,9 +1545,9 @@
 
 
 /* quantize & code squared Gain */
-int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gainQ10, Bitstr_enc *streamdata)
+int WebRtcIsacfix_EncodeGain2(int32_t *gainQ10, Bitstr_enc *streamdata)
 {
-  WebRtc_Word16 index;
+  int16_t index;
   int status = 0;
 
   /* find quantization index */
@@ -1576,11 +1576,11 @@
 /* code and decode Pitch Gains and Lags functions */
 
 /* decode & dequantize Pitch Gains */
-int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata, WebRtc_Word16 *PitchGains_Q12)
+int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata, int16_t *PitchGains_Q12)
 {
   int err;
-  WebRtc_Word16 index_comb;
-  const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
+  int16_t index_comb;
+  const uint16_t *pitch_gain_cdf_ptr[1];
 
   /* entropy decoding of quantization indices */
   *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
@@ -1600,20 +1600,20 @@
 
 
 /* quantize & code Pitch Gains */
-int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12, Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
+int WebRtcIsacfix_EncodePitchGain(int16_t *PitchGains_Q12, Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
 {
   int k,j;
-  WebRtc_Word16 SQ15[PITCH_SUBFRAMES];
-  WebRtc_Word16 index[3];
-  WebRtc_Word16 index_comb;
-  const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
-  WebRtc_Word32 CQ17;
+  int16_t SQ15[PITCH_SUBFRAMES];
+  int16_t index[3];
+  int16_t index_comb;
+  const uint16_t *pitch_gain_cdf_ptr[1];
+  int32_t CQ17;
   int status = 0;
 
 
   /* get the approximate arcsine (almost linear)*/
   for (k=0; k<PITCH_SUBFRAMES; k++)
-    SQ15[k] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
+    SQ15[k] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
 
 
   /* find quantization index; only for the first three transform coefficients */
@@ -1625,7 +1625,7 @@
       CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], SQ15[j],10); // Q17
     }
 
-    index[k] = (WebRtc_Word16)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
+    index[k] = (int16_t)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
 
     /* check that the index is not outside the boundaries of the table */
     if (index[k] < WebRtcIsacfix_kLowerlimiGain[k]) index[k] = WebRtcIsacfix_kLowerlimiGain[k];
@@ -1634,7 +1634,7 @@
   }
 
   /* calculate unique overall index */
-  index_comb = (WebRtc_Word16)(WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[0], index[0]) +
+  index_comb = (int16_t)(WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[0], index[0]) +
                                WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[1], index[1]) + index[2]);
 
   /* unquantize back to pitch gains by table look-up */
@@ -1667,21 +1667,21 @@
 
 /* decode & dequantize Pitch Lags */
 int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
-                                 WebRtc_Word16 *PitchGain_Q12,
-                                 WebRtc_Word16 *PitchLags_Q7)
+                                 int16_t *PitchGain_Q12,
+                                 int16_t *PitchLags_Q7)
 {
   int k, err;
-  WebRtc_Word16 index[PITCH_SUBFRAMES];
-  const WebRtc_Word16 *mean_val2Q10, *mean_val4Q10;
+  int16_t index[PITCH_SUBFRAMES];
+  const int16_t *mean_val2Q10, *mean_val4Q10;
 
-  const WebRtc_Word16 *lower_limit;
-  const WebRtc_UWord16 *init_index;
-  const WebRtc_UWord16 *cdf_size;
-  const WebRtc_UWord16 **cdf;
+  const int16_t *lower_limit;
+  const uint16_t *init_index;
+  const uint16_t *cdf_size;
+  const uint16_t **cdf;
 
-  WebRtc_Word32 meangainQ12;
-  WebRtc_Word32 CQ11, CQ10,tmp32a,tmp32b;
-  WebRtc_Word16 shft,tmp16a,tmp16c;
+  int32_t meangainQ12;
+  int32_t CQ11, CQ10,tmp32a,tmp32b;
+  int16_t shft,tmp16a,tmp16c;
 
   meangainQ12=0;
   for (k = 0; k < 4; k++)
@@ -1727,25 +1727,25 @@
 
 
   /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
-  CQ11 = ((WebRtc_Word32)index[0] + lower_limit[0]);  // Q0
+  CQ11 = ((int32_t)index[0] + lower_limit[0]);  // Q0
   CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
   for (k=0; k<PITCH_SUBFRAMES; k++) {
     tmp32a =  WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11);
-    tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);
+    tmp16a = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);
     PitchLags_Q7[k] = tmp16a;
   }
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
-    tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
+    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
+    tmp16c = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
     PitchLags_Q7[k] += tmp16c;
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
-    tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
+    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
+    tmp16c = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
     PitchLags_Q7[k] += tmp16c;
   }
 
@@ -1755,19 +1755,19 @@
 
 
 /* quantize & code Pitch Lags */
-int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagsQ7,WebRtc_Word16 *PitchGain_Q12,
+int WebRtcIsacfix_EncodePitchLag(int16_t *PitchLagsQ7,int16_t *PitchGain_Q12,
                                  Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
 {
   int k, j;
-  WebRtc_Word16 index[PITCH_SUBFRAMES];
-  WebRtc_Word32 meangainQ12, CQ17;
-  WebRtc_Word32 CQ11, CQ10,tmp32a;
+  int16_t index[PITCH_SUBFRAMES];
+  int32_t meangainQ12, CQ17;
+  int32_t CQ11, CQ10,tmp32a;
 
-  const WebRtc_Word16 *mean_val2Q10,*mean_val4Q10;
-  const WebRtc_Word16 *lower_limit, *upper_limit;
-  const WebRtc_UWord16 **cdf;
-  WebRtc_Word16 shft, tmp16a, tmp16b, tmp16c;
-  WebRtc_Word32 tmp32b;
+  const int16_t *mean_val2Q10,*mean_val4Q10;
+  const int16_t *lower_limit, *upper_limit;
+  const uint16_t **cdf;
+  int16_t shft, tmp16a, tmp16b, tmp16c;
+  int32_t tmp32b;
   int status = 0;
 
   /* compute mean pitch gain */
@@ -1817,7 +1817,7 @@
     CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize
 
     /* quantize */
-    tmp16b = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(CQ17 + 65536, 17 );
+    tmp16b = (int16_t) WEBRTC_SPL_RSHIFT_W32(CQ17 + 65536, 17 );
     index[k] =  tmp16b;
 
     /* check that the index is not outside the boundaries of the table */
@@ -1837,21 +1837,21 @@
 
   for (k=0; k<PITCH_SUBFRAMES; k++) {
     tmp32a =  WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11); // Q12
-    tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);// Q7
+    tmp16a = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);// Q7
     PitchLagsQ7[k] = tmp16a;
   }
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
-    tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
+    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
+    tmp16c = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
     PitchLagsQ7[k] += tmp16c;
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
-    tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
+    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
+    tmp16c = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
     PitchLagsQ7[k] += tmp16c;
   }
 
@@ -1870,22 +1870,22 @@
 
 
 /* cdf array for frame length indicator */
-const WebRtc_UWord16 kFrameLenCdf[4] = {
+const uint16_t kFrameLenCdf[4] = {
   0, 21845, 43690, 65535};
 
 /* pointer to cdf array for frame length indicator */
-const WebRtc_UWord16 *kFrameLenCdfPtr[1] = {kFrameLenCdf};
+const uint16_t *kFrameLenCdfPtr[1] = {kFrameLenCdf};
 
 /* initial cdf index for decoder of frame length indicator */
-const WebRtc_UWord16 kFrameLenInitIndex[1] = {1};
+const uint16_t kFrameLenInitIndex[1] = {1};
 
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 WebRtc_Word16 *framesamples)
+                                 int16_t *framesamples)
 {
 
   int err;
-  WebRtc_Word16 frame_mode;
+  int16_t frame_mode;
 
   err = 0;
   /* entropy decoding of frame length [1:30ms,2:60ms] */
@@ -1908,10 +1908,10 @@
 }
 
 
-int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr_enc *streamdata) {
+int WebRtcIsacfix_EncodeFrameLen(int16_t framesamples, Bitstr_enc *streamdata) {
 
   int status;
-  WebRtc_Word16 frame_mode;
+  int16_t frame_mode;
 
   status = 0;
   frame_mode = 0;
@@ -1936,34 +1936,34 @@
 }
 
 /* cdf array for estimated bandwidth */
-const WebRtc_UWord16 kBwCdf[25] = {
+const uint16_t kBwCdf[25] = {
   0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
   32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
   62804, 65535};
 
 /* pointer to cdf array for estimated bandwidth */
-const WebRtc_UWord16 *kBwCdfPtr[1] = {kBwCdf};
+const uint16_t *kBwCdfPtr[1] = {kBwCdf};
 
 /* initial cdf index for decoder of estimated bandwidth*/
-const WebRtc_UWord16 kBwInitIndex[1] = {7};
+const uint16_t kBwInitIndex[1] = {7};
 
 
-int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata, WebRtc_Word16 *BWno) {
+int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata, int16_t *BWno) {
 
   int err;
-  WebRtc_Word16 BWno32;
+  int16_t BWno32;
 
   /* entropy decoding of sender's BW estimation [0..23] */
   err = WebRtcIsacfix_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1);
   if (err<0)  // error check
     return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
-  *BWno = (WebRtc_Word16)BWno32;
+  *BWno = (int16_t)BWno32;
   return err;
 
 }
 
 
-int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno, Bitstr_enc *streamdata)
+int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t *BWno, Bitstr_enc *streamdata)
 {
   int status = 0;
   /* entropy encoding of receiver's BW estimation [0..23] */
@@ -1973,15 +1973,15 @@
 }
 
 /* estimate codel length of LPC Coef */
-void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *gain_lo_hiQ17,
-                                    WebRtc_Word16 *index_gQQ) {
+void WebRtcIsacfix_TranscodeLpcCoef(int32_t *gain_lo_hiQ17,
+                                    int16_t *index_gQQ) {
   int j, k;
-  WebRtc_Word16 posQQ, pos2QQ;
-  WebRtc_Word16 posg, offsg, gainpos;
-  WebRtc_Word32 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
-  WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
-  WebRtc_Word32 sumQQ;
+  int16_t posQQ, pos2QQ;
+  int16_t posg, offsg, gainpos;
+  int32_t tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+  int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+  int32_t sumQQ;
 
 
   /* log gains, mean removal and scaling */
@@ -2036,7 +2036,7 @@
   for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
   {
     posQQ = WebRtcIsacfix_kSelIndGain[k];
-    pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+    pos2QQ= (int16_t)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
 
     index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
     if (index_gQQ[k] < 0) {
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index 941fd0b..dd85090 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -22,91 +22,91 @@
 #include "structs.h"
 
 /* decode complex spectrum (return number of bytes in stream) */
-WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
-                                       WebRtc_Word16 *frQ7,
-                                       WebRtc_Word16 *fiQ7,
-                                       WebRtc_Word16 AvgPitchGain_Q12);
+int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
+                                 int16_t *frQ7,
+                                 int16_t *fiQ7,
+                                 int16_t AvgPitchGain_Q12);
 
 /* encode complex spectrum */
-int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
-                             const WebRtc_Word16 *fi,
+int WebRtcIsacfix_EncodeSpec(const int16_t *fr,
+                             const int16_t *fi,
                              Bitstr_enc *streamdata,
-                             WebRtc_Word16 AvgPitchGain_Q12);
+                             int16_t AvgPitchGain_Q12);
 
 
 /* decode & dequantize LPC Coef */
 int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec  *streamdata,
-                                WebRtc_Word32 *LPCCoefQ17,
-                                WebRtc_Word32 *gain_lo_hiQ17,
-                                WebRtc_Word16 *outmodel);
+                                int32_t *LPCCoefQ17,
+                                int32_t *gain_lo_hiQ17,
+                                int16_t *outmodel);
 
-int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
-                            WebRtc_Word16 *LPCCoef_loQ15,
-                            WebRtc_Word16 *LPCCoef_hiQ15,
+int WebRtcIsacfix_DecodeLpc(int32_t *gain_lo_hiQ17,
+                            int16_t *LPCCoef_loQ15,
+                            int16_t *LPCCoef_hiQ15,
                             Bitstr_dec  *streamdata,
-                            WebRtc_Word16 *outmodel);
+                            int16_t *outmodel);
 
 /* quantize & code LPC Coef */
-int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
-                            WebRtc_Word16 *LPCCoef_loQ15,
-                            WebRtc_Word16 *LPCCoef_hiQ15,
-                            WebRtc_Word16 *model,
-                            WebRtc_Word32 *sizeQ11,
+int WebRtcIsacfix_EncodeLpc(int32_t *gain_lo_hiQ17,
+                            int16_t *LPCCoef_loQ15,
+                            int16_t *LPCCoef_hiQ15,
+                            int16_t *model,
+                            int32_t *sizeQ11,
                             Bitstr_enc *streamdata,
                             ISAC_SaveEncData_t* encData,
                             transcode_obj *transcodeParam);
 
-int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
+int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17,
                                  Bitstr_enc *streamdata,
                                  ISAC_SaveEncData_t* encData);
 /* decode & dequantize RC */
 int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata,
-                               WebRtc_Word16 *RCQ15);
+                               int16_t *RCQ15);
 
 /* quantize & code RC */
-int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15,
+int WebRtcIsacfix_EncodeRcCoef(int16_t *RCQ15,
                                Bitstr_enc *streamdata);
 
 /* decode & dequantize squared Gain */
 int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata,
-                              WebRtc_Word32 *Gain2);
+                              int32_t *Gain2);
 
 /* quantize & code squared Gain (input is squared gain) */
-int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gain2,
+int WebRtcIsacfix_EncodeGain2(int32_t *gain2,
                               Bitstr_enc *streamdata);
 
-int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12,
+int WebRtcIsacfix_EncodePitchGain(int16_t *PitchGains_Q12,
                                   Bitstr_enc *streamdata,
                                   ISAC_SaveEncData_t* encData);
 
-int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagQ7,
-                                 WebRtc_Word16 *PitchGain_Q12,
+int WebRtcIsacfix_EncodePitchLag(int16_t *PitchLagQ7,
+                                 int16_t *PitchGain_Q12,
                                  Bitstr_enc *streamdata,
                                  ISAC_SaveEncData_t* encData);
 
 int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
-                                  WebRtc_Word16 *PitchGain_Q12);
+                                  int16_t *PitchGain_Q12);
 
 int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
-                                 WebRtc_Word16 *PitchGain_Q12,
-                                 WebRtc_Word16 *PitchLagQ7);
+                                 int16_t *PitchGain_Q12,
+                                 int16_t *PitchLagQ7);
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 WebRtc_Word16 *framelength);
+                                 int16_t *framelength);
 
 
-int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framelength,
+int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
                                  Bitstr_enc *streamdata);
 
 int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata,
-                                      WebRtc_Word16 *BWno);
+                                      int16_t *BWno);
 
 
-int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno,
+int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t *BWno,
                                          Bitstr_enc *streamdata);
 
-void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *tmpcoeffs_gQ6,
-                                    WebRtc_Word16 *index_gQQ);
+void WebRtcIsacfix_TranscodeLpcCoef(int32_t *tmpcoeffs_gQ6,
+                                    int16_t *index_gQQ);
 
 // Pointer functions for LPC transforms.
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/fft.c b/modules/audio_coding/codecs/isac/fix/source/fft.c
index fff35c4..d9cf8f2 100644
--- a/modules/audio_coding/codecs/isac/fix/source/fft.c
+++ b/modules/audio_coding/codecs/isac/fix/source/fft.c
@@ -18,7 +18,7 @@
 
 #include "fft.h"
 
-const WebRtc_Word16 kSortTabFft[240] = {
+const int16_t kSortTabFft[240] = {
   0, 60, 120, 180, 20, 80, 140, 200, 40, 100, 160, 220,
   4, 64, 124, 184, 24, 84, 144, 204, 44, 104, 164, 224,
   8, 68, 128, 188, 28, 88, 148, 208, 48, 108, 168, 228,
@@ -42,7 +42,7 @@
 };
 
 /* Cosine table in Q14 */
-const WebRtc_Word16 kCosTabFfftQ14[240] = {
+const int16_t kCosTabFfftQ14[240] = {
   16384,  16378, 16362,   16333,  16294,  16244,  16182,  16110,  16026,  15931,  15826,  15709,
   15582,  15444, 15296,   15137,  14968,  14788,  14598,  14399,  14189,  13970,  13741,  13502,
   13255,  12998, 12733,   12458,  12176,  11885,  11585,  11278,  10963,  10641,  10311,   9974,
@@ -68,18 +68,18 @@
 
 
 /* Uses 16x16 mul, without rounding, which is faster. Uses WEBRTC_SPL_MUL_16_16_RSFT */
-WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign) {
+int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], int16_t ImxQx[], int16_t iSign) {
 
-  WebRtc_Word16 dd, ee, ff, gg, hh, ii;
-  WebRtc_Word16 k0, k1, k2, k3, k4, kk;
-  WebRtc_Word16 tmp116, tmp216;
+  int16_t dd, ee, ff, gg, hh, ii;
+  int16_t k0, k1, k2, k3, k4, kk;
+  int16_t tmp116, tmp216;
 
-  WebRtc_Word16 ccc1Q14, ccc2Q14, ccc3Q14, sss1Q14, sss2Q14, sss3Q14;
-  WebRtc_Word16 sss60Q14, ccc72Q14, sss72Q14;
-  WebRtc_Word16 aaQx, ajQx, akQx, ajmQx, ajpQx, akmQx, akpQx;
-  WebRtc_Word16 bbQx, bjQx, bkQx, bjmQx, bjpQx, bkmQx, bkpQx;
+  int16_t ccc1Q14, ccc2Q14, ccc3Q14, sss1Q14, sss2Q14, sss3Q14;
+  int16_t sss60Q14, ccc72Q14, sss72Q14;
+  int16_t aaQx, ajQx, akQx, ajmQx, ajpQx, akmQx, akpQx;
+  int16_t bbQx, bjQx, bkQx, bjmQx, bjpQx, bkmQx, bkpQx;
 
-  WebRtc_Word16 ReDATAQx[240],  ImDATAQx[240];
+  int16_t ReDATAQx[240],  ImDATAQx[240];
 
   sss60Q14 = kCosTabFfftQ14[20];
   ccc72Q14 = kCosTabFfftQ14[48];
@@ -145,18 +145,18 @@
     // ImxQ16[k2] = ajpQ16 * sss2Q14 + bjpQ16 * ccc2Q14;
     // ImxQ16[k3] = akmQ16 * sss3Q14 + bkmQ16 * ccc3Q14;
 
-    RexQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) -
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul)
-    RexQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) -
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14);
-    RexQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) -
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14);
-    ImxQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) +
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14);
-    ImxQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14) +
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14);
-    ImxQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, akmQx, 14) +
-        (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, bkmQx, 14);
+    RexQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) -
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul)
+    RexQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) -
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14);
+    RexQx[k3] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) -
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14);
+    ImxQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) +
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14);
+    ImxQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14) +
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14);
+    ImxQx[k3] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, akmQx, 14) +
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, bkmQx, 14);
     //This mul segment needs 6*8 = 48 cycles for 16x16 muls, but 6*20 = 120 cycles for 16x32 muls
 
 
@@ -183,8 +183,8 @@
       tmp116 = RexQx[k1] - RexQx[k2];
       tmp216 = ImxQx[k1] - ImxQx[k2];
 
-      ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp116, 14); // Q14*Qx>>14 = Qx
-      bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp216, 14); // Q14*Qx>>14 = Qx
+      ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp116, 14); // Q14*Qx>>14 = Qx
+      bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp216, 14); // Q14*Qx>>14 = Qx
       RexQx[k1] = akQx - bjQx;
       RexQx[k2] = akQx + bjQx;
       ImxQx[k1] = bkQx + ajQx;
@@ -211,7 +211,7 @@
     kk += 20;
     ff = ff+4;
     for (hh=0; hh<2; hh++) {
-      ee = ff + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(hh, ff);
+      ee = ff + (int16_t)WEBRTC_SPL_MUL_16_16(hh, ff);
       dd = ee + 60;
       ccc2Q14 = kCosTabFfftQ14[ee];
       sss2Q14 = kCosTabFfftQ14[dd];
@@ -221,10 +221,10 @@
       for (ii=0; ii<4; ii++) {
         akQx = RexQx[kk];
         bkQx = ImxQx[kk];
-        RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - // Q14*Qx>>14 = Qx
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
-        ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + // Q14*Qx>>14 = Qx
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
+        RexQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - // Q14*Qx>>14 = Qx
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
+        ImxQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + // Q14*Qx>>14 = Qx
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
 
 
         kk += 60;
@@ -264,14 +264,14 @@
       RexQx[kk] = aaQx + akpQx + ajpQx;
       ImxQx[kk] = bbQx + bkpQx + bjpQx;
 
-      akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, akpQx, 14) +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14)  + aaQx;
-      bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bkpQx, 14) +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14)  + bbQx;
-      ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, akmQx, 14) +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajmQx, 14);
-      bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bkmQx, 14) +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjmQx, 14);
+      akQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, akpQx, 14) +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14)  + aaQx;
+      bkQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bkpQx, 14) +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14)  + bbQx;
+      ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, akmQx, 14) +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajmQx, 14);
+      bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bkmQx, 14) +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjmQx, 14);
       // 32+4*8=64 or 32+4*20=112
 
       RexQx[k1] = akQx - bjQx;
@@ -279,14 +279,14 @@
       ImxQx[k1] = bkQx + ajQx;
       ImxQx[k4] = bkQx - ajQx;
 
-      akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akpQx, 14)  +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, ajpQx, 14) + aaQx;
-      bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkpQx, 14)  +
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bjpQx, 14) + bbQx;
-      ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akmQx, 14) -
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, ajmQx, 14);
-      bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkmQx, 14) -
-          (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bjmQx, 14);
+      akQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akpQx, 14)  +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, ajpQx, 14) + aaQx;
+      bkQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkpQx, 14)  +
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bjpQx, 14) + bbQx;
+      ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akmQx, 14) -
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, ajmQx, 14);
+      bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkmQx, 14) -
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bjmQx, 14);
       // 8+4*8=40 or 8+4*20=88
 
       RexQx[k2] = akQx - bjQx;
@@ -308,7 +308,7 @@
 
   for (gg=0; gg<3; gg++) {
     kk += 4;
-    dd = 12 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(12, gg);
+    dd = 12 + (int16_t)WEBRTC_SPL_MUL_16_16(12, gg);
     ff = 0;
     for (hh=0; hh<4; hh++) {
       ff = ff+dd;
@@ -324,10 +324,10 @@
           sss2Q14 = -sss2Q14;
         }
 
-        RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) -
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
-        ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) +
-            (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
+        RexQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) -
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
+        ImxQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) +
+            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
 
         kk += 20;
       }
diff --git a/modules/audio_coding/codecs/isac/fix/source/fft.h b/modules/audio_coding/codecs/isac/fix/source/fft.h
index efa116e..dc7cea8 100644
--- a/modules/audio_coding/codecs/isac/fix/source/fft.h
+++ b/modules/audio_coding/codecs/isac/fix/source/fft.h
@@ -34,7 +34,7 @@
 #include "structs.h"
 
 
-WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign);
+int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], int16_t ImxQx[], int16_t iSign);
 
 
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
index c7eb590..9c9d098 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
@@ -92,13 +92,13 @@
                                           int32_t *state)
 {
   int k;
-  WebRtc_Word32 a1 = 0, b1 = 0, c = 0, in = 0;
-  WebRtc_Word32 a2 = 0, b2 = 0;
-  WebRtc_Word32 state0 = state[0];
-  WebRtc_Word32 state1 = state[1];
+  int32_t a1 = 0, b1 = 0, c = 0, in = 0;
+  int32_t a2 = 0, b2 = 0;
+  int32_t state0 = state[0];
+  int32_t state1 = state[1];
 
   for (k=0; k<len; k++) {
-    in = (WebRtc_Word32)io[k];
+    in = (int32_t)io[k];
 
 #ifdef WEBRTC_ARCH_ARM_V7
     {
@@ -134,11 +134,11 @@
     b2 = WEBRTC_SPL_MUL_32_32_RSFT32(coefficient[3], coefficient[2], state1);
 #endif
 
-    c = ((WebRtc_Word32)in) + WEBRTC_SPL_RSHIFT_W32(a1+b1, 7);  // Q0
-    io[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(c);  // Write output as Q0.
+    c = ((int32_t)in) + WEBRTC_SPL_RSHIFT_W32(a1+b1, 7);  // Q0
+    io[k] = (int16_t)WebRtcSpl_SatW32ToW16(c);  // Write output as Q0.
 
-    c = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in, 2) - a2 - b2;  // In Q2.
-    c = (WebRtc_Word32)WEBRTC_SPL_SAT(536870911, c, -536870912);
+    c = WEBRTC_SPL_LSHIFT_W32((int32_t)in, 2) - a2 - b2;  // In Q2.
+    c = (int32_t)WEBRTC_SPL_SAT(536870911, c, -536870912);
 
     state1 = state0;
     state0 = WEBRTC_SPL_LSHIFT_W32(c, 2);  // Write state as Q4
@@ -148,9 +148,9 @@
 }
 
 
-void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16 *pin,
-                                   WebRtc_Word16 *LP16,
-                                   WebRtc_Word16 *HP16,
+void WebRtcIsacfix_SplitAndFilter1(int16_t *pin,
+                                   int16_t *LP16,
+                                   int16_t *HP16,
                                    PreFiltBankstr *prefiltdata)
 {
   /* Function WebRtcIsacfix_SplitAndFilter */
@@ -159,10 +159,10 @@
 
   int k;
 
-  WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2 + QLOOKAHEAD];
-  WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2 + QLOOKAHEAD];
-  WebRtc_Word32 tmpState_ch1[2 * (QORDER-1)]; /* 4 */
-  WebRtc_Word32 tmpState_ch2[2 * (QORDER-1)]; /* 4 */
+  int16_t tempin_ch1[FRAMESAMPLES/2 + QLOOKAHEAD];
+  int16_t tempin_ch2[FRAMESAMPLES/2 + QLOOKAHEAD];
+  int32_t tmpState_ch1[2 * (QORDER-1)]; /* 4 */
+  int32_t tmpState_ch2[2 * (QORDER-1)]; /* 4 */
 
   /* High pass filter */
   WebRtcIsacfix_HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix);
@@ -213,13 +213,13 @@
 
   /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
   for (k=0; k<FRAMESAMPLES/2 + QLOOKAHEAD; k++) {
-    WebRtc_Word32 tmp1, tmp2, tmp3;
-    tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
-    tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
-    tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
-    LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
-    tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
-    HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
+    int32_t tmp1, tmp2, tmp3;
+    tmp1 = (int32_t)tempin_ch1[k]; // Q0 -> Q0
+    tmp2 = (int32_t)tempin_ch2[k]; // Q0 -> Q0
+    tmp3 = (int32_t)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
+    LP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
+    tmp3 = (int32_t)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
+    HP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
   }
 
 }/*end of WebRtcIsacfix_SplitAndFilter */
@@ -228,9 +228,9 @@
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 /* Without lookahead */
-void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16 *pin,
-                                   WebRtc_Word16 *LP16,
-                                   WebRtc_Word16 *HP16,
+void WebRtcIsacfix_SplitAndFilter2(int16_t *pin,
+                                   int16_t *LP16,
+                                   int16_t *HP16,
                                    PreFiltBankstr *prefiltdata)
 {
   /* Function WebRtcIsacfix_SplitAndFilter2 */
@@ -239,8 +239,8 @@
 
   int k;
 
-  WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2];
-  WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2];
+  int16_t tempin_ch1[FRAMESAMPLES/2];
+  int16_t tempin_ch2[FRAMESAMPLES/2];
 
 
   /* High pass filter */
@@ -272,13 +272,13 @@
 
   /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
   for (k=0; k<FRAMESAMPLES/2; k++) {
-    WebRtc_Word32 tmp1, tmp2, tmp3;
-    tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
-    tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
-    tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
-    LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
-    tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
-    HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
+    int32_t tmp1, tmp2, tmp3;
+    tmp1 = (int32_t)tempin_ch1[k]; // Q0 -> Q0
+    tmp2 = (int32_t)tempin_ch2[k]; // Q0 -> Q0
+    tmp3 = (int32_t)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
+    LP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
+    tmp3 = (int32_t)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
+    HP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
   }
 
 }/*end of WebRtcIsacfix_SplitAndFilter */
@@ -308,13 +308,13 @@
    high-pass signals.
    postfiltdata: the input data structure containing the filterbank
    states is updated for the next decoding iteration */
-void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16 *tempin_ch1,
-                                     WebRtc_Word16 *tempin_ch2,
-                                     WebRtc_Word16 *out16,
+void WebRtcIsacfix_FilterAndCombine1(int16_t *tempin_ch1,
+                                     int16_t *tempin_ch2,
+                                     int16_t *out16,
                                      PostFiltBankstr *postfiltdata)
 {
   int k;
-  WebRtc_Word16 in[FRAMESAMPLES];
+  int16_t in[FRAMESAMPLES];
 
   /* all-pass filter the new upper and lower channel signal.
      For upper channel, use the all-pass filter factors that were used as a
@@ -368,14 +368,14 @@
    high-pass signals.
    postfiltdata: the input data structure containing the filterbank
    states is updated for the next decoding iteration */
-void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16 *tempin_ch1,
-                                     WebRtc_Word16 *tempin_ch2,
-                                     WebRtc_Word16 *out16,
+void WebRtcIsacfix_FilterAndCombine2(int16_t *tempin_ch1,
+                                     int16_t *tempin_ch2,
+                                     int16_t *out16,
                                      PostFiltBankstr *postfiltdata,
-                                     WebRtc_Word16 len)
+                                     int16_t len)
 {
   int k;
-  WebRtc_Word16 in[FRAMESAMPLES];
+  int16_t in[FRAMESAMPLES];
 
   /* all-pass filter the new upper and lower channel signal.
      For upper channel, use the all-pass filter factors that were used as a
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters.c b/modules/audio_coding/codecs/isac/fix/source/filters.c
index a5ebd39..eb0e87a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filters.c
@@ -14,11 +14,11 @@
 
 // Autocorrelation function in fixed point.
 // NOTE! Different from SPLIB-version in how it scales the signal.
-int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
-                            const WebRtc_Word16* __restrict x,
-                            WebRtc_Word16 N,
-                            WebRtc_Word16 order,
-                            WebRtc_Word16* __restrict scale) {
+int WebRtcIsacfix_AutocorrC(int32_t* __restrict r,
+                            const int16_t* __restrict x,
+                            int16_t N,
+                            int16_t order,
+                            int16_t* __restrict scale) {
   int i = 0;
   int j = 0;
   int16_t scaling = 0;
@@ -59,17 +59,17 @@
   return(order + 1);
 }
 
-static const WebRtc_Word32 kApUpperQ15[ALLPASSSECTIONS] = { 1137, 12537 };
-static const WebRtc_Word32 kApLowerQ15[ALLPASSSECTIONS] = { 5059, 24379 };
+static const int32_t kApUpperQ15[ALLPASSSECTIONS] = { 1137, 12537 };
+static const int32_t kApLowerQ15[ALLPASSSECTIONS] = { 5059, 24379 };
 
 
-static void AllpassFilterForDec32(WebRtc_Word16         *InOut16, //Q0
-                                  const WebRtc_Word32   *APSectionFactors, //Q15
-                                  WebRtc_Word16         lengthInOut,
-                                  WebRtc_Word32          *FilterState) //Q16
+static void AllpassFilterForDec32(int16_t         *InOut16, //Q0
+                                  const int32_t   *APSectionFactors, //Q15
+                                  int16_t         lengthInOut,
+                                  int32_t          *FilterState) //Q16
 {
   int n, j;
-  WebRtc_Word32 a, b;
+  int32_t a, b;
 
   for (j=0; j<ALLPASSSECTIONS; j++) {
     for (n=0;n<lengthInOut;n+=2){
@@ -77,12 +77,12 @@
       a = WEBRTC_SPL_LSHIFT_W32(a, 1); // Q15 -> Q16
       b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16
       a = WEBRTC_SPL_MUL_16_32_RSFT16(
-          (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16),
+          (int16_t) WEBRTC_SPL_RSHIFT_W32(b, 16),
           -APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
       FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(
           WEBRTC_SPL_LSHIFT_W32(a,1),
-          WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n], 16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
-      InOut16[n] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
+          WEBRTC_SPL_LSHIFT_W32((uint32_t)InOut16[n], 16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
+      InOut16[n] = (int16_t) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
     }
   }
 }
@@ -90,20 +90,20 @@
 
 
 
-void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
-                                     WebRtc_Word32 *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
-                                     WebRtc_Word16 N,                /* number of input samples */
-                                     WebRtc_Word16 *out)             /* array of size N/2 */
+void WebRtcIsacfix_DecimateAllpass32(const int16_t *in,
+                                     int32_t *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
+                                     int16_t N,                /* number of input samples */
+                                     int16_t *out)             /* array of size N/2 */
 {
   int n;
-  WebRtc_Word16 data_vec[PITCH_FRAME_LEN];
+  int16_t data_vec[PITCH_FRAME_LEN];
 
   /* copy input */
-  memcpy(data_vec+1, in, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (N-1)));
+  memcpy(data_vec+1, in, WEBRTC_SPL_MUL_16_16(sizeof(int16_t), (N-1)));
 
 
-  data_vec[0] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)],16);   //the z^(-1) state
-  state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)] = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)in[N-1],16);
+  data_vec[0] = (int16_t) WEBRTC_SPL_RSHIFT_W32(state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)],16);   //the z^(-1) state
+  state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)] = WEBRTC_SPL_LSHIFT_W32((uint32_t)in[N-1],16);
 
 
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_neon.S b/modules/audio_coding/codecs/isac/fix/source/filters_neon.S
index a970333..3c5ac64 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters_neon.S
+++ b/modules/audio_coding/codecs/isac/fix/source/filters_neon.S
@@ -15,11 +15,11 @@
 .align  2
 
 @ int WebRtcIsacfix_AutocorrNeon(
-@     WebRtc_Word32* __restrict r,
-@     const WebRtc_Word16* __restrict x,
-@     WebRtc_Word16 N,
-@     WebRtc_Word16 order,
-@     WebRtc_Word16* __restrict scale);
+@     int32_t* __restrict r,
+@     const int16_t* __restrict x,
+@     int16_t N,
+@     int16_t order,
+@     int16_t* __restrict scale);
 
 DEFINE_FUNCTION WebRtcIsacfix_AutocorrNeon
   push       {r3 - r12}
diff --git a/modules/audio_coding/codecs/isac/fix/source/initialize.c b/modules/audio_coding/codecs/isac/fix/source/initialize.c
index 4d11af5..dbcd49c 100644
--- a/modules/audio_coding/codecs/isac/fix/source/initialize.c
+++ b/modules/audio_coding/codecs/isac/fix/source/initialize.c
@@ -25,18 +25,18 @@
   int k;
 
   for (k = 0; k < WINLEN; k++) {
-    maskdata->DataBufferLoQ0[k] = (WebRtc_Word16) 0;
-    maskdata->DataBufferHiQ0[k] = (WebRtc_Word16) 0;
+    maskdata->DataBufferLoQ0[k] = (int16_t) 0;
+    maskdata->DataBufferHiQ0[k] = (int16_t) 0;
   }
   for (k = 0; k < ORDERLO+1; k++) {
-    maskdata->CorrBufLoQQ[k] = (WebRtc_Word32) 0;
+    maskdata->CorrBufLoQQ[k] = (int32_t) 0;
     maskdata->CorrBufLoQdom[k] = 0;
 
     maskdata->PreStateLoGQ15[k] = 0;
 
   }
   for (k = 0; k < ORDERHI+1; k++) {
-    maskdata->CorrBufHiQQ[k] = (WebRtc_Word32) 0;
+    maskdata->CorrBufHiQQ[k] = (int32_t) 0;
     maskdata->CorrBufHiQdom[k] = 0;
     maskdata->PreStateHiGQ15[k] = 0;
   }
diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index f9c481d..945475f 100644
--- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -42,8 +42,8 @@
  *
  */
 
-WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes) {
-  *sizeinbytes=sizeof(ISACFIX_SubStruct)*2/sizeof(WebRtc_Word16);
+int16_t WebRtcIsacfix_AssignSize(int *sizeinbytes) {
+  *sizeinbytes=sizeof(ISACFIX_SubStruct)*2/sizeof(int16_t);
   return(0);
 }
 
@@ -56,7 +56,7 @@
  * If successful, Return 0, else Return -1
  */
 
-WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) {
+int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) {
   if (ISACFIX_inst_Addr!=NULL) {
     *inst = (ISACFIX_MainStruct*)ISACFIX_inst_Addr;
     (*(ISACFIX_SubStruct**)inst)->errorcode = 0;
@@ -84,7 +84,7 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst)
+int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst)
 {
   ISACFIX_SubStruct *tempo;
   tempo = malloc(1 * sizeof(ISACFIX_SubStruct));
@@ -113,7 +113,7 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -147,7 +147,7 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst)
 {
   free(ISAC_main_inst);
   return(0);
@@ -165,7 +165,7 @@
  *                             -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -217,11 +217,11 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
-                                        WebRtc_Word16  CodingMode)
+int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
+                                  int16_t  CodingMode)
 {
   int k;
-  WebRtc_Word16 statusInit;
+  int16_t statusInit;
   ISACFIX_SubStruct *ISAC_inst;
 
   statusInit = 0;
@@ -321,12 +321,12 @@
  *                          : -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                                   const WebRtc_Word16    *speechIn,
-                                   WebRtc_Word16          *encoded)
+int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                             const int16_t    *speechIn,
+                             int16_t          *encoded)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  WebRtc_Word16 stream_len;
+  int16_t stream_len;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
@@ -341,20 +341,20 @@
     return (-1);
   }
 
-  stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechIn,
-                                    &ISAC_inst->ISACenc_obj,
-                                    &ISAC_inst->bwestimator_obj,
-                                    ISAC_inst->CodingMode);
+  stream_len = WebRtcIsacfix_EncodeImpl((int16_t*)speechIn,
+                                        &ISAC_inst->ISACenc_obj,
+                                        &ISAC_inst->bwestimator_obj,
+                                        ISAC_inst->CodingMode);
   if (stream_len<0) {
     ISAC_inst->errorcode = - stream_len;
     return -1;
   }
 
 
-  /* convert from bytes to WebRtc_Word16 */
+  /* convert from bytes to int16_t */
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
-    encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
+    encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
                                   | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
   }
 
@@ -396,15 +396,15 @@
  *                          : -1 - Error
  */
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                      const WebRtc_Word16    *speechIn,
-                                      WebRtc_Word16          *encoded)
+int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+                               const int16_t    *speechIn,
+                               int16_t          *encoded)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  WebRtc_Word16 stream_len;
-  WebRtc_Word16 speechInWB[FRAMESAMPLES_10ms];
-  WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES_10ms/2];
-  WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES_10ms/2];
+  int16_t stream_len;
+  int16_t speechInWB[FRAMESAMPLES_10ms];
+  int16_t Vector_Word16_1[FRAMESAMPLES_10ms/2];
+  int16_t Vector_Word16_2[FRAMESAMPLES_10ms/2];
 
   int k;
 
@@ -431,20 +431,20 @@
 
 
   /* Encode WB signal */
-  stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechInWB,
-                                    &ISAC_inst->ISACenc_obj,
-                                    &ISAC_inst->bwestimator_obj,
-                                    ISAC_inst->CodingMode);
+  stream_len = WebRtcIsacfix_EncodeImpl((int16_t*)speechInWB,
+                                        &ISAC_inst->ISACenc_obj,
+                                        &ISAC_inst->bwestimator_obj,
+                                        ISAC_inst->CodingMode);
   if (stream_len<0) {
     ISAC_inst->errorcode = - stream_len;
     return -1;
   }
 
 
-  /* convert from bytes to WebRtc_Word16 */
+  /* convert from bytes to int16_t */
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
-    encoded[k] = (WebRtc_Word16)(((WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8)
+    encoded[k] = (int16_t)(((uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8)
                                  | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
   }
 
@@ -478,13 +478,13 @@
  *                          : -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
-                                            WebRtc_Word16      bweIndex,
-                                            float              scale,
-                                            WebRtc_Word16        *encoded)
+int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
+                                      int16_t      bweIndex,
+                                      float              scale,
+                                      int16_t        *encoded)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  WebRtc_Word16 stream_len;
+  int16_t stream_len;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
@@ -509,7 +509,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
-    encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
+    encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
                                   | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
   }
 
@@ -536,7 +536,7 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -579,22 +579,22 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
-                                     const WebRtc_UWord16   *encoded,
-                                     WebRtc_Word32          packet_size,
-                                     WebRtc_UWord16         rtp_seq_number,
-                                     WebRtc_UWord32         arr_ts)
+int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
+                                        const uint16_t   *encoded,
+                                        int32_t          packet_size,
+                                        uint16_t         rtp_seq_number,
+                                        uint32_t         arr_ts)
 {
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
-  WebRtc_UWord16 partOfStream[5];
+  uint16_t partOfStream[5];
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   /* Set stream pointer to point at partOfStream */
-  streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+  streamdata.stream = (uint16_t *)partOfStream;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -623,7 +623,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<5; k++) {
-    streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
 #else
   memcpy(streamdata.stream, encoded, 5);
@@ -666,23 +666,23 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
-                                       const WebRtc_UWord16   *encoded,
-                                       WebRtc_Word32          packet_size,
-                                       WebRtc_UWord16         rtp_seq_number,
-                                       WebRtc_UWord32         send_ts,
-                                       WebRtc_UWord32         arr_ts)
+int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
+                                       const uint16_t   *encoded,
+                                       int32_t          packet_size,
+                                       uint16_t         rtp_seq_number,
+                                       uint32_t         send_ts,
+                                       uint32_t         arr_ts)
 {
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
-  WebRtc_UWord16 partOfStream[5];
+  uint16_t partOfStream[5];
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   /* Set stream pointer to point at partOfStream */
-  streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+  streamdata.stream = (uint16_t *)partOfStream;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -711,7 +711,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<5; k++) {
-    streamdata.stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    streamdata.stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
 #else
   memcpy(streamdata.stream, encoded, 5);
@@ -755,20 +755,20 @@
  */
 
 
-WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                                     const WebRtc_UWord16   *encoded,
-                                     WebRtc_Word16          len,
-                                     WebRtc_Word16          *decoded,
-                                     WebRtc_Word16     *speechType)
+int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+                             const uint16_t   *encoded,
+                             int16_t          len,
+                             int16_t          *decoded,
+                             int16_t     *speechType)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
-  WebRtc_Word16     number_of_samples;
+  int16_t     number_of_samples;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 declen = 0;
+  int16_t declen = 0;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -790,15 +790,15 @@
     return -1;
   }
 
-  (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
+  (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded;
 
-  /* convert bitstream from WebRtc_Word16 to bytes */
+  /* convert bitstream from int16_t to bytes */
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<(len>>1); k++) {
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
   if (len & 0x0001)
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
+    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] & 0xFF)<<8);
 #endif
 
   /* added for NetEq purposes (VAD/DTX related) */
@@ -809,7 +809,7 @@
   if (declen < 0) {
     /* Some error inside the decoder */
     ISAC_inst->errorcode = -declen;
-    memset(decoded, 0, sizeof(WebRtc_Word16) * MAX_FRAMESAMPLES);
+    memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES);
     return -1;
   }
 
@@ -818,13 +818,13 @@
   if (declen & 0x0001) {
     if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
-      memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+      memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   } else {
     if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
-      memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+      memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   }
@@ -858,21 +858,21 @@
  */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                        const WebRtc_UWord16   *encoded,
-                                        WebRtc_Word16          len,
-                                        WebRtc_Word16          *decoded,
-                                        WebRtc_Word16    *speechType)
+int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+                               const uint16_t   *encoded,
+                               int16_t          len,
+                               int16_t          *decoded,
+                               int16_t    *speechType)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* twice the number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
-  WebRtc_Word16     number_of_samples;
+  int16_t     number_of_samples;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 declen = 0;
-  WebRtc_Word16 dummy[FRAMESAMPLES/2];
+  int16_t declen = 0;
+  int16_t dummy[FRAMESAMPLES/2];
 
 
   /* typecast pointer to real structure */
@@ -891,15 +891,15 @@
     return -1;
   }
 
-  (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
+  (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded;
 
-  /* convert bitstream from WebRtc_Word16 to bytes */
+  /* convert bitstream from int16_t to bytes */
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<(len>>1); k++) {
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
   if (len & 0x0001)
-    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
+    (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] & 0xFF)<<8);
 #endif
 
   /* added for NetEq purposes (VAD/DTX related) */
@@ -910,7 +910,7 @@
   if (declen < 0) {
     /* Some error inside the decoder */
     ISAC_inst->errorcode = -declen;
-    memset(decoded, 0, sizeof(WebRtc_Word16) * FRAMESAMPLES);
+    memset(decoded, 0, sizeof(int16_t) * FRAMESAMPLES);
     return -1;
   }
 
@@ -919,13 +919,13 @@
   if (declen & 0x0001) {
     if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
-      memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+      memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   } else {
     if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
-      memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+      memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   }
@@ -963,14 +963,14 @@
  */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                         WebRtc_Word16          *decoded,
-                                         WebRtc_Word16 noOfLostFrames )
+int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
+                                  int16_t          *decoded,
+                                  int16_t noOfLostFrames )
 {
-  WebRtc_Word16 no_of_samples, declen, k, ok;
-  WebRtc_Word16 outframeNB[FRAMESAMPLES];
-  WebRtc_Word16 outframeWB[FRAMESAMPLES];
-  WebRtc_Word16 dummy[FRAMESAMPLES/2];
+  int16_t no_of_samples, declen, k, ok;
+  int16_t outframeNB[FRAMESAMPLES];
+  int16_t outframeWB[FRAMESAMPLES];
+  int16_t dummy[FRAMESAMPLES/2];
 
 
   ISACFIX_SubStruct *ISAC_inst;
@@ -1029,13 +1029,13 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
-                                      WebRtc_Word16          *decoded,
-                                      WebRtc_Word16 noOfLostFrames)
+int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
+                                int16_t          *decoded,
+                                int16_t noOfLostFrames)
 {
 
-  WebRtc_Word16 no_of_samples, declen, k, ok;
-  WebRtc_Word16 outframe16[MAX_FRAMESAMPLES];
+  int16_t no_of_samples, declen, k, ok;
+  int16_t outframe16[MAX_FRAMESAMPLES];
 
   ISACFIX_SubStruct *ISAC_inst;
   /* typecast pointer to real structure */
@@ -1081,9 +1081,9 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                                    WebRtc_Word16          rate,
-                                    WebRtc_Word16          framesize)
+int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
+                              int16_t          rate,
+                              int16_t          framesize)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* typecast pointer to real structure */
@@ -1140,10 +1140,10 @@
  *         -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
-                                        WebRtc_Word16 rateBPS,
-                                        WebRtc_Word16 frameSizeMs,
-                                        WebRtc_Word16 enforceFrameSize)
+int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
+                                 int16_t rateBPS,
+                                 int16_t frameSizeMs,
+                                 int16_t enforceFrameSize)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* Typecast pointer to real structure */
@@ -1168,7 +1168,7 @@
   /* Set initial rate, if value between 10000 and 32000,                */
   /* if rateBPS is 0, keep the default initial bottleneck value (15000) */
   if ((rateBPS >= 10000) && (rateBPS <= 32000)) {
-    ISAC_inst->bwestimator_obj.sendBwAvg = (((WebRtc_UWord32)rateBPS) << 7);
+    ISAC_inst->bwestimator_obj.sendBwAvg = (((uint32_t)rateBPS) << 7);
   } else if (rateBPS != 0) {
     ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK;
     return -1;
@@ -1203,8 +1203,8 @@
  *
  */
 
-WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
-                                       WebRtc_Word16*     rateIndex)
+int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
+                                         int16_t*     rateIndex)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -1230,10 +1230,10 @@
  *
  */
 
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
-                                   WebRtc_Word16     rateIndex)
+int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
+                                     int16_t     rateIndex)
 {
-  WebRtc_Word16 err = 0;
+  int16_t err = 0;
   ISACFIX_SubStruct *ISAC_inst;
 
   /* typecast pointer to real structure */
@@ -1262,18 +1262,18 @@
  *
  */
 
-WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
-                                        WebRtc_Word16* frameLength)
+int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
+                                   int16_t* frameLength)
 {
   Bitstr_dec streamdata;
-  WebRtc_UWord16 partOfStream[5];
+  uint16_t partOfStream[5];
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   /* Set stream pointer to point at partOfStream */
-  streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+  streamdata.stream = (uint16_t *)partOfStream;
 
   streamdata.W_upper = 0xFFFFFFFF;
   streamdata.streamval = 0;
@@ -1282,7 +1282,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<5; k++) {
-    streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
 #else
   memcpy(streamdata.stream, encoded, 5);
@@ -1311,18 +1311,18 @@
  *
  */
 
-WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
-                                   WebRtc_Word16* rateIndex)
+int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
+                                  int16_t* rateIndex)
 {
   Bitstr_dec streamdata;
-  WebRtc_UWord16 partOfStream[5];
+  uint16_t partOfStream[5];
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   /* Set stream pointer to point at partOfStream */
-  streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+  streamdata.stream = (uint16_t *)partOfStream;
 
   streamdata.W_upper = 0xFFFFFFFF;
   streamdata.streamval = 0;
@@ -1331,7 +1331,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k=0; k<5; k++) {
-    streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+    streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
 #else
   memcpy(streamdata.stream, encoded, 5);
@@ -1367,7 +1367,7 @@
  * Return value             : Error code
  */
 
-WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* typecast pointer to real structure */
@@ -1389,12 +1389,12 @@
  * Return value             : bitrate
  */
 
-WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst)
+int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
   BwEstimatorstr * bw = (BwEstimatorstr*)&(ISAC_inst->bwestimator_obj);
 
-  return (WebRtc_Word32) WebRtcIsacfix_GetUplinkBandwidth(bw);
+  return (int32_t) WebRtcIsacfix_GetUplinkBandwidth(bw);
 }
 
 /****************************************************************************
@@ -1408,7 +1408,7 @@
  * Return value             :  frame lenght in samples
  */
 
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst)
+int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
   return ISAC_inst->ISACenc_obj.new_framelength;
@@ -1433,8 +1433,8 @@
  *                           -1 if error happens
  */
 
-WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
-                                              WebRtc_Word16 maxPayloadBytes)
+int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
+                                        int16_t maxPayloadBytes)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
@@ -1492,11 +1492,11 @@
  *                           -1 if error happens
  */
 
-WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
-                                       WebRtc_Word32 maxRate)
+int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
+                                 int32_t maxRate)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  WebRtc_Word16 maxRateInBytes;
+  int16_t maxRateInBytes;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -1512,7 +1512,7 @@
        maximum rate. Multiply with 30/1000 to get number of bits per 30 msec,
        divide by 8 to get number of bytes per 30 msec:
        maxRateInBytes = floor((maxRate * 30/1000) / 8); */
-    maxRateInBytes = (WebRtc_Word16)( WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_MUL(maxRate, 3), 800) );
+    maxRateInBytes = (int16_t)( WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_MUL(maxRate, 3), 800) );
 
     /* Store the value for usage in the WebRtcIsacfix_SetMaxPayloadSize-function */
     ISAC_inst->ISACenc_obj.maxRateInBytes = maxRateInBytes;
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice.c b/modules/audio_coding/codecs/isac/fix/source/lattice.c
index bbc5993..6b3a6bf 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice.c
@@ -19,17 +19,17 @@
 #include "settings.h"
 
 #define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32)                  \
-  ((WebRtc_Word32)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
+  ((int32_t)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
 /* This macro is FORBIDDEN to use elsewhere than in a function in this file and
    its corresponding neon version. It might give unpredictable results, since a
-   general WebRtc_Word32*WebRtc_Word32 multiplication results in a 64 bit value.
+   general int32_t*int32_t multiplication results in a 64 bit value.
    The result is then shifted just 16 steps to the right, giving need for 48
-   bits, i.e. in the generel case, it will NOT fit in a WebRtc_Word32. In the
-   cases used in here, the WebRtc_Word32 will be enough, since (for a good
+   bits, i.e. in the generel case, it will NOT fit in a int32_t. In the
+   cases used in here, the int32_t will be enough, since (for a good
    reason) the involved multiplicands aren't big enough to overflow a
-   WebRtc_Word32 after shifting right 16 bits. I have compared the result of a
+   int32_t after shifting right 16 bits. I have compared the result of a
    multiplication between t32 and tmp32, done in two ways:
-   1) Using (WebRtc_Word32) (((float)(tmp32))*((float)(tmp32b))/65536.0);
+   1) Using (int32_t) (((float)(tmp32))*((float)(tmp32b))/65536.0);
    2) Using LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
    By running 25 files, I haven't found any bigger diff than 64 - this was in the
    case when  method 1) gave 650235648 and 2) gave 650235712.
@@ -86,43 +86,43 @@
 
 /* filter the signal using normalized lattice filter */
 /* MA filter */
-void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
-                                       WebRtc_Word32 *stateGQ15,
-                                       WebRtc_Word16 *lat_inQ0,
-                                       WebRtc_Word16 *filt_coefQ15,
-                                       WebRtc_Word32 *gain_lo_hiQ17,
-                                       WebRtc_Word16 lo_hi,
-                                       WebRtc_Word16 *lat_outQ9)
+void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
+                                       int32_t *stateGQ15,
+                                       int16_t *lat_inQ0,
+                                       int16_t *filt_coefQ15,
+                                       int32_t *gain_lo_hiQ17,
+                                       int16_t lo_hi,
+                                       int16_t *lat_outQ9)
 {
-  WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
-  WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
+  int16_t sthQ15[MAX_AR_MODEL_ORDER];
+  int16_t cthQ15[MAX_AR_MODEL_ORDER];
 
   int u, i, k, n;
-  WebRtc_Word16 temp2,temp3;
-  WebRtc_Word16 ord_1 = orderCoef+1;
-  WebRtc_Word32 inv_cthQ16[MAX_AR_MODEL_ORDER];
+  int16_t temp2,temp3;
+  int16_t ord_1 = orderCoef+1;
+  int32_t inv_cthQ16[MAX_AR_MODEL_ORDER];
 
-  WebRtc_Word32 gain32, fQtmp;
-  WebRtc_Word16 gain16;
-  WebRtc_Word16 gain_sh;
+  int32_t gain32, fQtmp;
+  int16_t gain16;
+  int16_t gain_sh;
 
-  WebRtc_Word32 tmp32, tmp32b;
-  WebRtc_Word32 fQ15vec[HALF_SUBFRAMELEN];
-  WebRtc_Word32 gQ15[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
-  WebRtc_Word16 sh;
-  WebRtc_Word16 t16a;
-  WebRtc_Word16 t16b;
+  int32_t tmp32, tmp32b;
+  int32_t fQ15vec[HALF_SUBFRAMELEN];
+  int32_t gQ15[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
+  int16_t sh;
+  int16_t t16a;
+  int16_t t16b;
 
   for (u=0;u<SUBFRAMES;u++)
   {
     int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
 
     /* set the Direct Form coefficients */
-    temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
-    temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u)+lo_hi;
+    temp2 = (int16_t)WEBRTC_SPL_MUL_16_16(u, orderCoef);
+    temp3 = (int16_t)WEBRTC_SPL_MUL_16_16(2, u)+lo_hi;
 
     /* compute lattice filter coefficients */
-    memcpy(sthQ15, &filt_coefQ15[temp2], orderCoef * sizeof(WebRtc_Word16));
+    memcpy(sthQ15, &filt_coefQ15[temp2], orderCoef * sizeof(int16_t));
 
     WebRtcSpl_SqrtOfOneMinusXSquared(sthQ15, orderCoef, cthQ15);
 
@@ -134,9 +134,9 @@
     for (k=0;k<orderCoef;k++)
     {
       gain32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[k], gain32); //Q15*Q(17+gain_sh)>>15 = Q(17+gain_sh)
-      inv_cthQ16[k] = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, cthQ15[k]); // 1/cth[k] in Q31/Q15 = Q16
+      inv_cthQ16[k] = WebRtcSpl_DivW32W16((int32_t)2147483647, cthQ15[k]); // 1/cth[k] in Q31/Q15 = Q16
     }
-    gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(gain32, 16); //Q(1+gain_sh)
+    gain16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(gain32, 16); //Q(1+gain_sh)
 
     /* normalized lattice filter */
     /*****************************/
@@ -144,8 +144,8 @@
     /* initial conditions */
     for (i=0;i<HALF_SUBFRAMELEN;i++)
     {
-      fQ15vec[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
-      gQ15[0][i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
+      fQ15vec[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)lat_inQ0[i + temp1], 15); //Q15
+      gQ15[0][i] = WEBRTC_SPL_LSHIFT_W32((int32_t)lat_inQ0[i + temp1], 15); //Q15
     }
 
 
@@ -158,8 +158,8 @@
       tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], stateGQ15[i-1]);//Q15*Q15>>15 = Q15
       tmp32b= fQtmp + tmp32; //Q15+Q15=Q15
       tmp32 = inv_cthQ16[i-1]; //Q16
-      t16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
-      t16b = (WebRtc_Word16) (tmp32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)t16a), 16));
+      t16a = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+      t16b = (int16_t) (tmp32-WEBRTC_SPL_LSHIFT_W32(((int32_t)t16a), 16));
       if (t16b<0) t16a++;
       tmp32 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
       fQtmp = tmp32; // Q15
@@ -189,7 +189,7 @@
       //gain32 = WEBRTC_SPL_RSHIFT_W32(gain32, gain_sh); // Q(17+gain_sh) -> Q17
       tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(gain16, fQ15vec[n]); //Q(1+gain_sh)*Q15>>16 = Q(gain_sh)
       sh = 9-gain_sh; //number of needed shifts to reach Q9
-      t16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh);
+      t16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32, sh);
       lat_outQ9[n + temp1] = t16a;
     }
 
@@ -210,39 +210,39 @@
 
 /* ----------------AR filter-------------------------*/
 /* filter the signal using normalized lattice filter */
-void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
-                                       WebRtc_Word16 *stateGQ0,
-                                       WebRtc_Word32 *lat_inQ25,
-                                       WebRtc_Word16 *filt_coefQ15,
-                                       WebRtc_Word32 *gain_lo_hiQ17,
-                                       WebRtc_Word16 lo_hi,
-                                       WebRtc_Word16 *lat_outQ0)
+void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
+                                       int16_t *stateGQ0,
+                                       int32_t *lat_inQ25,
+                                       int16_t *filt_coefQ15,
+                                       int32_t *gain_lo_hiQ17,
+                                       int16_t lo_hi,
+                                       int16_t *lat_outQ0)
 {
   int ii,n,k,i,u;
-  WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
-  WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
-  WebRtc_Word32 tmp32;
+  int16_t sthQ15[MAX_AR_MODEL_ORDER];
+  int16_t cthQ15[MAX_AR_MODEL_ORDER];
+  int32_t tmp32;
 
 
-  WebRtc_Word16 tmpAR;
-  WebRtc_Word16 ARfQ0vec[HALF_SUBFRAMELEN];
-  WebRtc_Word16 ARgQ0vec[MAX_AR_MODEL_ORDER+1];
+  int16_t tmpAR;
+  int16_t ARfQ0vec[HALF_SUBFRAMELEN];
+  int16_t ARgQ0vec[MAX_AR_MODEL_ORDER+1];
 
-  WebRtc_Word32 inv_gain32;
-  WebRtc_Word16 inv_gain16;
-  WebRtc_Word16 den16;
-  WebRtc_Word16 sh;
+  int32_t inv_gain32;
+  int16_t inv_gain16;
+  int16_t den16;
+  int16_t sh;
 
-  WebRtc_Word16 temp2,temp3;
-  WebRtc_Word16 ord_1 = orderCoef+1;
+  int16_t temp2,temp3;
+  int16_t ord_1 = orderCoef+1;
 
   for (u=0;u<SUBFRAMES;u++)
   {
     int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
 
     //set the denominator and numerator of the Direct Form
-    temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
-    temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u) + lo_hi;
+    temp2 = (int16_t)WEBRTC_SPL_MUL_16_16(u, orderCoef);
+    temp3 = (int16_t)WEBRTC_SPL_MUL_16_16(2, u) + lo_hi;
 
     for (ii=0; ii<orderCoef; ii++) {
       sthQ15[ii] = filt_coefQ15[temp2+ii];
@@ -263,11 +263,11 @@
     }
 
     sh = WebRtcSpl_NormW32(tmp32); // tmp32 is the gain
-    den16 = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh-16); //Q(27+sh-16) = Q(sh+11) (all 16 bits are value bits)
-    inv_gain32 = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, den16); // 1/gain in Q31/Q(sh+11) = Q(20-sh)
+    den16 = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32, sh-16); //Q(27+sh-16) = Q(sh+11) (all 16 bits are value bits)
+    inv_gain32 = WebRtcSpl_DivW32W16((int32_t)2147483647, den16); // 1/gain in Q31/Q(sh+11) = Q(20-sh)
 
     //initial conditions
-    inv_gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inv_gain32, 2); // 1/gain in Q(20-sh-2) = Q(18-sh)
+    inv_gain16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(inv_gain32, 2); // 1/gain in Q(20-sh-2) = Q(18-sh)
 
     for (i=0;i<HALF_SUBFRAMELEN;i++)
     {
@@ -276,16 +276,16 @@
       tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh)
       tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0
 
-      ARfQ0vec[i] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+      ARfQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
     }
 
     for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
     {
       tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(cthQ15[i],ARfQ0vec[0])) - (WEBRTC_SPL_MUL_16_16(sthQ15[i],stateGQ0[i])) + 16384), 15);
-      tmpAR = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+      tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
 
       tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sthQ15[i],ARfQ0vec[0])) + (WEBRTC_SPL_MUL_16_16(cthQ15[i], stateGQ0[i])) + 16384), 15);
-      ARgQ0vec[i+1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+      ARgQ0vec[i+1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
       ARfQ0vec[0] = tmpAR;
     }
     ARgQ0vec[0] = ARfQ0vec[0];
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c b/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
index 80ccf39..d5b2d0e 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
@@ -40,8 +40,8 @@
               - (WEBRTC_SPL_MUL_16_16(sth_Q15[k], ar_g_Q0[k])) + 16384), 15);
       tmp32_2 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sth_Q15[k], tmpAR))
                 + (WEBRTC_SPL_MUL_16_16(cth_Q15[k], ar_g_Q0[k])) + 16384), 15);
-      tmpAR   = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32);
-      ar_g_Q0[k + 1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32_2);
+      tmpAR   = (int16_t)WebRtcSpl_SatW32ToW16(tmp32);
+      ar_g_Q0[k + 1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2);
     }
     ar_f_Q0[n + 1] = tmpAR;
     ar_g_Q0[0] = tmpAR;
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
index 4be438e..0dc8174 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
@@ -24,36 +24,36 @@
 
 /* The conversion is implemented by the step-down algorithm */
 void WebRtcSpl_AToK_JSK(
-    WebRtc_Word16 *a16, /* Q11 */
-    WebRtc_Word16 useOrder,
-    WebRtc_Word16 *k16  /* Q15 */
+    int16_t *a16, /* Q11 */
+    int16_t useOrder,
+    int16_t *k16  /* Q15 */
                         )
 {
   int m, k;
-  WebRtc_Word32 tmp32[MAX_AR_MODEL_ORDER];
-  WebRtc_Word32 tmp32b;
-  WebRtc_Word32 tmp_inv_denum32;
-  WebRtc_Word16 tmp_inv_denum16;
+  int32_t tmp32[MAX_AR_MODEL_ORDER];
+  int32_t tmp32b;
+  int32_t tmp_inv_denum32;
+  int16_t tmp_inv_denum16;
 
   k16[useOrder-1]= WEBRTC_SPL_LSHIFT_W16(a16[useOrder], 4); //Q11<<4 => Q15
 
   for (m=useOrder-1; m>0; m--) {
-    tmp_inv_denum32 = ((WebRtc_Word32) 1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]); // (1 - k^2) in Q30
-    tmp_inv_denum16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp_inv_denum32, 15); // (1 - k^2) in Q15
+    tmp_inv_denum32 = ((int32_t) 1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]); // (1 - k^2) in Q30
+    tmp_inv_denum16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp_inv_denum32, 15); // (1 - k^2) in Q15
 
     for (k=1; k<=m; k++) {
-      tmp32b = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16) -
+      tmp32b = WEBRTC_SPL_LSHIFT_W32((int32_t)a16[k], 16) -
           WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
 
       tmp32[k] = WebRtcSpl_DivW32W16(tmp32b, tmp_inv_denum16); //Q27/Q15 = Q12
     }
 
     for (k=1; k<m; k++) {
-      a16[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q12>>1 => Q11
+      a16[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q12>>1 => Q11
     }
 
     tmp32[m] = WEBRTC_SPL_SAT(4092, tmp32[m], -4092);
-    k16[m-1] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmp32[m], 3); //Q12<<3 => Q15
+    k16[m-1] = (int16_t) WEBRTC_SPL_LSHIFT_W32(tmp32[m], 3); //Q12<<3 => Q15
   }
 
   return;
@@ -63,25 +63,25 @@
 
 
 
-WebRtc_Word16 WebRtcSpl_LevinsonW32_JSK(
-    WebRtc_Word32 *R,  /* (i) Autocorrelation of length >= order+1 */
-    WebRtc_Word16 *A,  /* (o) A[0..order] LPC coefficients (Q11) */
-    WebRtc_Word16 *K,  /* (o) K[0...order-1] Reflection coefficients (Q15) */
-    WebRtc_Word16 order /* (i) filter order */
+int16_t WebRtcSpl_LevinsonW32_JSK(
+    int32_t *R,  /* (i) Autocorrelation of length >= order+1 */
+    int16_t *A,  /* (o) A[0..order] LPC coefficients (Q11) */
+    int16_t *K,  /* (o) K[0...order-1] Reflection coefficients (Q15) */
+    int16_t order /* (i) filter order */
                                         ) {
-  WebRtc_Word16 i, j;
-  WebRtc_Word16 R_hi[LEVINSON_MAX_ORDER+1], R_low[LEVINSON_MAX_ORDER+1];
+  int16_t i, j;
+  int16_t R_hi[LEVINSON_MAX_ORDER+1], R_low[LEVINSON_MAX_ORDER+1];
   /* Aurocorr coefficients in high precision */
-  WebRtc_Word16 A_hi[LEVINSON_MAX_ORDER+1], A_low[LEVINSON_MAX_ORDER+1];
+  int16_t A_hi[LEVINSON_MAX_ORDER+1], A_low[LEVINSON_MAX_ORDER+1];
   /* LPC coefficients in high precicion */
-  WebRtc_Word16 A_upd_hi[LEVINSON_MAX_ORDER+1], A_upd_low[LEVINSON_MAX_ORDER+1];
+  int16_t A_upd_hi[LEVINSON_MAX_ORDER+1], A_upd_low[LEVINSON_MAX_ORDER+1];
   /* LPC coefficients for next iteration */
-  WebRtc_Word16 K_hi, K_low;      /* reflection coefficient in high precision */
-  WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp; /* Prediction gain Alpha in high precision
+  int16_t K_hi, K_low;      /* reflection coefficient in high precision */
+  int16_t Alpha_hi, Alpha_low, Alpha_exp; /* Prediction gain Alpha in high precision
                                                    and with scale factor */
-  WebRtc_Word16 tmp_hi, tmp_low;
-  WebRtc_Word32 temp1W32, temp2W32, temp3W32;
-  WebRtc_Word16 norm;
+  int16_t tmp_hi, tmp_low;
+  int32_t temp1W32, temp2W32, temp3W32;
+  int16_t norm;
 
   /* Normalize the autocorrelation R[0]...R[order+1] */
 
@@ -90,14 +90,14 @@
   for (i=order;i>=0;i--) {
     temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
     /* Put R in hi and low format */
-    R_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
+    R_hi[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    R_low[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)R_hi[i], 16)), 1);
   }
 
   /* K = A[1] = -R[1] / R[0] */
 
-  temp2W32  = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16) +
-      WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1);     /* R[1] in Q31      */
+  temp2W32  = WEBRTC_SPL_LSHIFT_W32((int32_t)R_hi[1],16) +
+      WEBRTC_SPL_LSHIFT_W32((int32_t)R_low[1],1);     /* R[1] in Q31      */
   temp3W32  = WEBRTC_SPL_ABS_W32(temp2W32);      /* abs R[1]         */
   temp1W32  = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); /* abs(R[1])/R[0] in Q31 */
   /* Put back the sign on R[1] */
@@ -106,8 +106,8 @@
   }
 
   /* Put K in hi and low format */
-  K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-  K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+  K_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+  K_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)K_hi, 16)), 1);
 
   /* Store first reflection coefficient */
   K[0] = K_hi;
@@ -115,8 +115,8 @@
   temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4);    /* A[1] in Q27      */
 
   /* Put A[1] in hi and low format */
-  A_hi[1] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-  A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
+  A_hi[1] = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+  A_low[1] = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)A_hi[1], 16)), 1);
 
   /*  Alpha = R[0] * (1-K^2) */
 
@@ -124,11 +124,11 @@
                                       WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1); /* temp1W32 = k^2 in Q31 */
 
   temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32);    /* Guard against <0 */
-  temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32;    /* temp1W32 = (1 - K[0]*K[0]) in Q31 */
+  temp1W32 = (int32_t)0x7fffffffL - temp1W32;    /* temp1W32 = (1 - K[0]*K[0]) in Q31 */
 
   /* Store temp1W32 = 1 - K[0]*K[0] on hi and low format */
-  tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-  tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+  tmp_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+  tmp_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)tmp_hi, 16)), 1);
 
   /* Calculate Alpha in Q31 */
   temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi) +
@@ -139,8 +139,8 @@
 
   Alpha_exp = WebRtcSpl_NormW32(temp1W32);
   temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
-  Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-  Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+  Alpha_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+  Alpha_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)Alpha_hi, 16)), 1);
 
   /* Perform the iterative calculations in the
      Levinson Durbin algorithm */
@@ -166,8 +166,8 @@
     }
 
     temp1W32  = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
-    temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16) +
-                 WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
+    temp1W32 += (WEBRTC_SPL_LSHIFT_W32((int32_t)R_hi[i], 16) +
+                 WEBRTC_SPL_LSHIFT_W32((int32_t)R_low[i], 1));
 
     /* K = -temp1W32 / Alpha */
     temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32);      /* abs(temp1W32) */
@@ -185,16 +185,16 @@
     } else {
       if (temp3W32 > 0)
       {
-        temp3W32 = (WebRtc_Word32)0x7fffffffL;
+        temp3W32 = (int32_t)0x7fffffffL;
       } else
       {
-        temp3W32 = (WebRtc_Word32)0x80000000L;
+        temp3W32 = (int32_t)0x80000000L;
       }
     }
 
     /* Put K on hi and low format */
-    K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-    K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+    K_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+    K_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)K_hi, 16)), 1);
 
     /* Store Reflection coefficient in Q15 */
     K[i-1] = K_hi;
@@ -203,7 +203,7 @@
        user decide what to do in that case
     */
 
-    if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32740) {
+    if ((int32_t)WEBRTC_SPL_ABS_W16(K_hi) > (int32_t)32740) {
       return(-i); /* Unstable filter */
     }
 
@@ -215,23 +215,23 @@
 
     for(j=1; j<i; j++)
     {
-      temp1W32  = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16) +
-          WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);    /* temp1W32 = A[j] in Q27 */
+      temp1W32  = WEBRTC_SPL_LSHIFT_W32((int32_t)A_hi[j],16) +
+          WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[j],1);    /* temp1W32 = A[j] in Q27 */
 
       temp1W32 += WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j]) +
                                            WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]), 15) +
                                            WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]), 15) ), 1); /* temp1W32 += K*A[i-j] in Q27 */
 
       /* Put Anew in hi and low format */
-      A_upd_hi[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-      A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
+      A_upd_hi[j] = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+      A_upd_low[j] = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)A_upd_hi[j], 16)), 1);
     }
 
     temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);     /* temp3W32 = K in Q27 (Convert from Q31 to Q27) */
 
     /* Store Anew in hi and low format */
-    A_upd_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-    A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
+    A_upd_hi[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+    A_upd_low[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)A_upd_hi[i], 16)), 1);
 
     /*  Alpha = Alpha * (1-K^2) */
 
@@ -239,11 +239,11 @@
                                         WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1);  /* K*K in Q31 */
 
     temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32);      /* Guard against <0 */
-    temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32;      /* 1 - K*K  in Q31 */
+    temp1W32 = (int32_t)0x7fffffffL - temp1W32;      /* 1 - K*K  in Q31 */
 
     /* Convert 1- K^2 in hi and low format */
-    tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+    tmp_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    tmp_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)tmp_hi, 16)), 1);
 
     /* Calculate Alpha = Alpha * (1-K^2) in Q31 */
     temp1W32 = WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi) +
@@ -255,8 +255,8 @@
     norm = WebRtcSpl_NormW32(temp1W32);
     temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
 
-    Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+    Alpha_hi = (int16_t) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    Alpha_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((int32_t)Alpha_hi, 16)), 1);
 
     /* Update the total nomalization of Alpha */
     Alpha_exp = Alpha_exp + norm;
@@ -279,10 +279,10 @@
 
   for(i=1; i<=order; i++) {
     /* temp1W32 in Q27 */
-    temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16) +
-        WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
+    temp1W32 = WEBRTC_SPL_LSHIFT_W32((int32_t)A_hi[i], 16) +
+        WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[i], 1);
     /* Round and store upper word */
-    A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32+(WebRtc_Word32)32768, 16);
+    A[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(temp1W32+(int32_t)32768, 16);
   }
   return(1); /* Stable filters */
 }
@@ -297,7 +297,7 @@
  *  for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end
  * All values are multiplyed with 2^21 in fixed point code.
  */
-static const WebRtc_Word16 kWindowAutocorr[WINLEN] = {
+static const int16_t kWindowAutocorr[WINLEN] = {
   0,     0,     0,     0,     0,     1,     1,     2,     2,     3,     5,     6,
   8,    10,    12,    14,    17,    20,    24,    28,    33,    38,    43,    49,
   56,    63,    71,    79,    88,    98,   108,   119,   131,   143,   157,   171,
@@ -337,20 +337,20 @@
    round(kPolyVecLo*32768)
    round(kPolyVecHi*32768)
 */
-static const WebRtc_Word16 kPolyVecLo[12] = {
+static const int16_t kPolyVecLo[12] = {
   29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425, 10283, 9255
 };
-static const WebRtc_Word16 kPolyVecHi[6] = {
+static const int16_t kPolyVecHi[6] = {
   26214, 20972, 16777, 13422, 10737, 8590
 };
 
-static __inline WebRtc_Word32 log2_Q8_LPC( WebRtc_UWord32 x ) {
+static __inline int32_t log2_Q8_LPC( uint32_t x ) {
 
-  WebRtc_Word32 zeros, lg2;
-  WebRtc_Word16 frac;
+  int32_t zeros, lg2;
+  int16_t frac;
 
   zeros=WebRtcSpl_NormU32(x);
-  frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
+  frac=(int16_t)WEBRTC_SPL_RSHIFT_W32(((uint32_t)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
 
   /* log2(x) */
 
@@ -359,22 +359,22 @@
 
 }
 
-static const WebRtc_Word16 kMulPitchGain = -25; /* 200/256 in Q5 */
-static const WebRtc_Word16 kChngFactor = 3523; /* log10(2)*10/4*0.4/1.4=log10(2)/1.4= 0.2150 in Q14 */
-static const WebRtc_Word16 kExp2 = 11819; /* 1/log(2) */
+static const int16_t kMulPitchGain = -25; /* 200/256 in Q5 */
+static const int16_t kChngFactor = 3523; /* log10(2)*10/4*0.4/1.4=log10(2)/1.4= 0.2150 in Q14 */
+static const int16_t kExp2 = 11819; /* 1/log(2) */
 const int kShiftLowerBand = 11;  /* Shift value for lower band in Q domain. */
 const int kShiftHigherBand = 12;  /* Shift value for higher band in Q domain. */
 
-void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input, const WebRtc_Word16 *pitchGains_Q12,
-                           WebRtc_UWord32 *oldEnergy, WebRtc_Word16 *varscale)
+void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12,
+                           uint32_t *oldEnergy, int16_t *varscale)
 {
   int k;
-  WebRtc_UWord32 nrgQ[4];
-  WebRtc_Word16 nrgQlog[4];
-  WebRtc_Word16 tmp16, chng1, chng2, chng3, chng4, tmp, chngQ, oldNrgQlog, pgQ, pg3;
-  WebRtc_Word32 expPg32;
-  WebRtc_Word16 expPg, divVal;
-  WebRtc_Word16 tmp16_1, tmp16_2;
+  uint32_t nrgQ[4];
+  int16_t nrgQlog[4];
+  int16_t tmp16, chng1, chng2, chng3, chng4, tmp, chngQ, oldNrgQlog, pgQ, pg3;
+  int32_t expPg32;
+  int16_t expPg, divVal;
+  int16_t tmp16_1, tmp16_2;
 
   /* Calculate energies of first and second frame halfs */
   nrgQ[0]=0;
@@ -395,9 +395,9 @@
   }
 
   for ( k=0; k<4; k++) {
-    nrgQlog[k] = (WebRtc_Word16)log2_Q8_LPC(nrgQ[k]); /* log2(nrgQ) */
+    nrgQlog[k] = (int16_t)log2_Q8_LPC(nrgQ[k]); /* log2(nrgQ) */
   }
-  oldNrgQlog = (WebRtc_Word16)log2_Q8_LPC(*oldEnergy);
+  oldNrgQlog = (int16_t)log2_Q8_LPC(*oldEnergy);
 
   /* Calculate average level change */
   chng1 = WEBRTC_SPL_ABS_W16(nrgQlog[3]-nrgQlog[2]);
@@ -405,7 +405,7 @@
   chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]);
   chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog);
   tmp = chng1+chng2+chng3+chng4;
-  chngQ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
+  chngQ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
   chngQ += 2926; /* + 1.0/1.4 in Q12 */
 
   /* Find average pitch gain */
@@ -415,34 +415,34 @@
     pgQ += pitchGains_Q12[k];
   }
 
-  pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
-  pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18  */
-  pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10  kMulPitchGain = -25 = -200 in Q-3. */
+  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
+  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18  */
+  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10  kMulPitchGain = -25 = -200 in Q-3. */
 
-  tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
+  tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
   if (tmp16<0) {
     tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
-    tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
+    tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((uint16_t)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
     if (tmp16_1<0)
-      expPg=(WebRtc_Word16) -WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+      expPg=(int16_t) -WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
     else
-      expPg=(WebRtc_Word16) -WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+      expPg=(int16_t) -WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
   } else
-    expPg = (WebRtc_Word16) -16384; /* 1 in Q14, since 2^0=1 */
+    expPg = (int16_t) -16384; /* 1 in Q14, since 2^0=1 */
 
-  expPg32 = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W16((WebRtc_Word32)expPg, 8); /* Q22 */
+  expPg32 = (int32_t)WEBRTC_SPL_LSHIFT_W16((int32_t)expPg, 8); /* Q22 */
   divVal = WebRtcSpl_DivW32W16ResW16(expPg32, chngQ); /* Q22/Q12=Q10 */
 
-  tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,divVal,13);/* Q13*Q10>>13 => Q10*/
+  tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,divVal,13);/* Q13*Q10>>13 => Q10*/
   if (tmp16<0) {
     tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
-    tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
+    tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((uint16_t)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
     if (tmp16_1<0)
-      expPg=(WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+      expPg=(int16_t) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
     else
-      expPg=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+      expPg=(int16_t) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
   } else
-    expPg = (WebRtc_Word16) 16384; /* 1 in Q14, since 2^0=1 */
+    expPg = (int16_t) 16384; /* 1 in Q14, since 2^0=1 */
 
   *varscale = expPg-1;
   *oldEnergy = nrgQ[3];
@@ -450,16 +450,16 @@
 
 
 
-static __inline WebRtc_Word16  exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
+static __inline int16_t  exp2_Q10_T(int16_t x) { // Both in and out in Q10
 
-  WebRtc_Word16 tmp16_1, tmp16_2;
+  int16_t tmp16_1, tmp16_2;
 
-  tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
-  tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+  tmp16_2=(int16_t)(0x0400|(x&0x03FF));
+  tmp16_1=-(int16_t)WEBRTC_SPL_RSHIFT_W16(x,10);
   if(tmp16_1>0)
-    return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+    return (int16_t) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
   else
-    return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+    return (int16_t) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
 
 }
 
@@ -538,63 +538,63 @@
   return residual_energy;
 }
 
-void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
-                              WebRtc_Word16 *inHiQ0,
+void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
+                              int16_t *inHiQ0,
                               MaskFiltstr_enc *maskdata,
-                              WebRtc_Word16 snrQ10,
-                              const WebRtc_Word16 *pitchGains_Q12,
-                              WebRtc_Word32 *gain_lo_hiQ17,
-                              WebRtc_Word16 *lo_coeffQ15,
-                              WebRtc_Word16 *hi_coeffQ15)
+                              int16_t snrQ10,
+                              const int16_t *pitchGains_Q12,
+                              int32_t *gain_lo_hiQ17,
+                              int16_t *lo_coeffQ15,
+                              int16_t *hi_coeffQ15)
 {
   int k, n, ii;
   int pos1, pos2;
   int sh_lo, sh_hi, sh, ssh, shMem;
-  WebRtc_Word16 varscaleQ14;
+  int16_t varscaleQ14;
 
-  WebRtc_Word16 tmpQQlo, tmpQQhi;
-  WebRtc_Word32 tmp32;
-  WebRtc_Word16 tmp16,tmp16b;
+  int16_t tmpQQlo, tmpQQhi;
+  int32_t tmp32;
+  int16_t tmp16,tmp16b;
 
-  WebRtc_Word16 polyHI[ORDERHI+1];
-  WebRtc_Word16 rcQ15_lo[ORDERLO], rcQ15_hi[ORDERHI];
+  int16_t polyHI[ORDERHI+1];
+  int16_t rcQ15_lo[ORDERLO], rcQ15_hi[ORDERHI];
 
 
-  WebRtc_Word16 DataLoQ6[WINLEN], DataHiQ6[WINLEN];
-  WebRtc_Word32 corrloQQ[ORDERLO+2];
-  WebRtc_Word32 corrhiQQ[ORDERHI+1];
-  WebRtc_Word32 corrlo2QQ[ORDERLO+1];
-  WebRtc_Word16 scale;
-  WebRtc_Word16 QdomLO, QdomHI, newQdomHI, newQdomLO;
+  int16_t DataLoQ6[WINLEN], DataHiQ6[WINLEN];
+  int32_t corrloQQ[ORDERLO+2];
+  int32_t corrhiQQ[ORDERHI+1];
+  int32_t corrlo2QQ[ORDERLO+1];
+  int16_t scale;
+  int16_t QdomLO, QdomHI, newQdomHI, newQdomLO;
 
-  WebRtc_Word32 res_nrgQQ;
-  WebRtc_Word32 sqrt_nrg;
+  int32_t res_nrgQQ;
+  int32_t sqrt_nrg;
 
   /* less-noise-at-low-frequencies factor */
-  WebRtc_Word16 aaQ14;
+  int16_t aaQ14;
 
   /* Multiplication with 1/sqrt(12) ~= 0.28901734104046 can be done by convertion to
      Q15, i.e. round(0.28901734104046*32768) = 9471, and use 9471/32768.0 ~= 0.289032
   */
-  WebRtc_Word16 snrq;
+  int16_t snrq;
   int shft;
 
-  WebRtc_Word16 tmp16a;
-  WebRtc_Word32 tmp32a, tmp32b, tmp32c;
+  int16_t tmp16a;
+  int32_t tmp32a, tmp32b, tmp32c;
 
-  WebRtc_Word16 a_LOQ11[ORDERLO+1];
-  WebRtc_Word16 k_vecloQ15[ORDERLO];
-  WebRtc_Word16 a_HIQ12[ORDERHI+1];
-  WebRtc_Word16 k_vechiQ15[ORDERHI];
+  int16_t a_LOQ11[ORDERLO+1];
+  int16_t k_vecloQ15[ORDERLO];
+  int16_t a_HIQ12[ORDERHI+1];
+  int16_t k_vechiQ15[ORDERHI];
 
-  WebRtc_Word16 stab;
+  int16_t stab;
 
   snrq=snrQ10;
 
   /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/
-  tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
+  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
   tmp16b = exp2_Q10_T(tmp16); // Q10
-  snrq = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
+  snrq = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
 
   /* change quallevel depending on pitch gains and level fluctuations */
   WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14);
@@ -604,17 +604,17 @@
      With 0.35 in Q16 (0.35 ~= 22938/65536.0 = 0.3500061) and varscaleQ14 in Q14,
      we get Q16*Q14>>16 = Q14
   */
-  aaQ14 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+  aaQ14 = (int16_t) WEBRTC_SPL_RSHIFT_W32(
       (WEBRTC_SPL_MUL_16_16(22938, (8192 + WEBRTC_SPL_RSHIFT_W32(varscaleQ14, 1)))
-       + ((WebRtc_Word32)32768)), 16);
+       + ((int32_t)32768)), 16);
 
   /* Calculate tmp = (1.0 + aa*aa); in Q12 */
-  tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
+  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
   tmpQQlo = 4096 + WEBRTC_SPL_RSHIFT_W16(tmp16, 1); // Q12 + Q13>>1 = Q12
 
   /* Calculate tmp = (1.0+aa) * (1.0+aa); */
   tmp16 = 8192 + WEBRTC_SPL_RSHIFT_W16(aaQ14, 1); // 1+a in Q13
-  tmpQQhi = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
+  tmpQQhi = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
 
   /* replace data in buffer by new look-ahead data */
   for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) {
@@ -627,18 +627,18 @@
     for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
       maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2];
       maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2];
-      DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
           maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
           maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
     }
-    pos2 = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(k, UPDATE)/2);
+    pos2 = (int16_t)(WEBRTC_SPL_MUL_16_16(k, UPDATE)/2);
     for (n = 0; n < UPDATE/2; n++, pos1++) {
       maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2];
       maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++];
-      DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
           maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
           maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
     }
 
@@ -700,9 +700,9 @@
     /* Calculate corrlo2[0] += 9.5367431640625e-7; and
        corrhi[0]  += 9.5367431640625e-7, where the constant is 1/2^20 */
 
-    tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomLO-20);
+    tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t) 1, QdomLO-20);
     corrlo2QQ[0] += tmp32;
-    tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomHI-20);
+    tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t) 1, QdomHI-20);
     corrhiQQ[0]  += tmp32;
 
     /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) before the following
@@ -721,10 +721,10 @@
     newQdomLO = QdomLO;
 
     for (n = 0; n <= ORDERLO; n++) {
-      WebRtc_Word32 tmp, tmpB, tmpCorr;
-      WebRtc_Word16 alpha=328; //0.01 in Q15
-      WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
-      WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q15
+      int32_t tmp, tmpB, tmpCorr;
+      int16_t alpha=328; //0.01 in Q15
+      int16_t beta=324; //(1-0.01)*0.01=0.0099 in Q15
+      int16_t gamma=32440; //(1-0.01)=0.99 in Q15
 
       if (maskdata->CorrBufLoQQ[n] != 0) {
         shMem=WebRtcSpl_NormW32(maskdata->CorrBufLoQQ[n]);
@@ -768,10 +768,10 @@
     newQdomHI = QdomHI;
 
     for (n = 0; n <= ORDERHI; n++) {
-      WebRtc_Word32 tmp, tmpB, tmpCorr;
-      WebRtc_Word16 alpha=328; //0.01 in Q15
-      WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
-      WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q1
+      int32_t tmp, tmpB, tmpCorr;
+      int16_t alpha=328; //0.01 in Q15
+      int16_t beta=324; //(1-0.01)*0.01=0.0099 in Q15
+      int16_t gamma=32440; //(1-0.01)=0.99 in Q1
       if (maskdata->CorrBufHiQQ[n] != 0) {
         shMem=WebRtcSpl_NormW32(maskdata->CorrBufHiQQ[n]);
         sh = QdomHI - maskdata->CorrBufHiQdom[n];
@@ -834,13 +834,13 @@
 
     /* bandwidth expansion */
     for (n = 1; n <= ORDERLO; n++) {
-      a_LOQ11[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]);
+      a_LOQ11[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]);
     }
 
 
     polyHI[0] = a_HIQ12[0];
     for (n = 1; n <= ORDERHI; n++) {
-      a_HIQ12[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]);
+      a_HIQ12[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]);
       polyHI[n] = a_HIQ12[n];
     }
 
@@ -876,7 +876,7 @@
 
 
       //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
-      tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1);  // H_T_HQ19=65536 (16-17=-1)   ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1);  // sqrt_nrg is in Qssh
+      tmp32a=WEBRTC_SPL_RSHIFT_W32((int32_t) varscaleQ14,1);  // H_T_HQ19=65536 (16-17=-1)   ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1);  // sqrt_nrg is in Qssh
       ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1);  // sqrt_nrg is in Qssh
       sh = ssh - 14;
       tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
@@ -885,7 +885,7 @@
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
-      tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft)  (denominator)
+      tmp16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft)  (denominator)
 
       tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
       sh = ssh-shft-7;
@@ -893,13 +893,13 @@
     }
     else
     {
-      *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17);  // Gains in Q17
+      *gain_lo_hiQ17 = 100; //(int32_t)WEBRTC_SPL_LSHIFT_W32( (int32_t)1, 17);  // Gains in Q17
     }
     gain_lo_hiQ17++;
 
     /* copy coefficients to output array */
     for (n = 0; n < ORDERLO; n++) {
-      *lo_coeffQ15 = (WebRtc_Word16) (rcQ15_lo[n]);
+      *lo_coeffQ15 = (int16_t) (rcQ15_lo[n]);
       lo_coeffQ15++;
     }
     /* residual energy */
@@ -925,7 +925,7 @@
       /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
 
       //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
-      tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1);  // H_T_HQ19=65536 (16-17=-1)
+      tmp32a=WEBRTC_SPL_RSHIFT_W32((int32_t) varscaleQ14,1);  // H_T_HQ19=65536 (16-17=-1)
 
       ssh= WEBRTC_SPL_RSHIFT_W32(sh_hi, 1);  // sqrt_nrg is in Qssh
       sh = ssh - 14;
@@ -935,7 +935,7 @@
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
-      tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft)  (denominator)
+      tmp16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft)  (denominator)
 
       tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
       sh = ssh-shft-7;
@@ -943,7 +943,7 @@
     }
     else
     {
-      *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17);  // Gains in Q17
+      *gain_lo_hiQ17 = 100; //(int32_t)WEBRTC_SPL_LSHIFT_W32( (int32_t)1, 17);  // Gains in Q17
     }
     gain_lo_hiQ17++;
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
index e06a207..72e0cfc 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
@@ -24,19 +24,19 @@
 
 #include "structs.h"
 
-void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input,
-                           const WebRtc_Word16 *pitchGains_Q12,
-                           WebRtc_UWord32 *oldEnergy,
-                           WebRtc_Word16 *varscale);
+void WebRtcIsacfix_GetVars(const int16_t *input,
+                           const int16_t *pitchGains_Q12,
+                           uint32_t *oldEnergy,
+                           int16_t *varscale);
 
-void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
-                              WebRtc_Word16 *inHiQ0,
+void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
+                              int16_t *inHiQ0,
                               MaskFiltstr_enc *maskdata,
-                              WebRtc_Word16 snrQ10,
-                              const WebRtc_Word16 *pitchGains_Q12,
-                              WebRtc_Word32 *gain_lo_hiQ17,
-                              WebRtc_Word16 *lo_coeffQ15,
-                              WebRtc_Word16 *hi_coeffQ15);
+                              int16_t snrQ10,
+                              const int16_t *pitchGains_Q12,
+                              int32_t *gain_lo_hiQ17,
+                              int16_t *lo_coeffQ15,
+                              int16_t *hi_coeffQ15);
 
 typedef int32_t (*CalculateResidualEnergy)(int lpc_order,
                                            int32_t q_val_corr,
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
index 90cc9af..bc0f129 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
@@ -20,11 +20,11 @@
 #include "lpc_tables.h"
 
 /* indices of KLT coefficients used */
-const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12] = {
+const uint16_t WebRtcIsacfix_kSelIndGain[12] = {
   0,  1,  2,  3,  4,  5,  6,  7,  8,  9,
   10,  11};
 
-const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108] = {
+const uint16_t WebRtcIsacfix_kSelIndShape[108] = {
   0,  1,  2,  3,  4,  5,  6,  7,  8,  9,
   10,  11,  12,  13,  14,  15,  16,  17,  18,  19,
   20,  21,  22,  23,  24,  25,  26,  27,  28,  29,
@@ -39,26 +39,26 @@
 };
 
 /* cdf array for model indicator */
-const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[4] = {
+const uint16_t WebRtcIsacfix_kModelCdf[4] = {
   0,  15434,  37548,  65535
 };
 
 /* pointer to cdf array for model indicator */
-const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1] = {
+const uint16_t *WebRtcIsacfix_kModelCdfPtr[1] = {
   WebRtcIsacfix_kModelCdf
 };
 
 /* initial cdf index for decoder of model indicator */
-const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1] = {
+const uint16_t WebRtcIsacfix_kModelInitIndex[1] = {
   1
 };
 
 /* offset to go from rounded value to quantization index */
-const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12] ={
+const int16_t WebRtcIsacfix_kQuantMinGain[12] ={
   3,  6,  4,  6,  6,  9,  5,  16,  11,  34,  32,  47
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108] = {
+const int16_t WebRtcIsacfix_kQuantMinShape[108] = {
   0,  0,  0,  0,  0,  0,  0,  0,  0,  1,
   1,  1,  1,  1,  2,  2,  2,  3,  0,  0,
   0,  0,  1,  0,  0,  0,  0,  1,  1,  1,
@@ -73,11 +73,11 @@
 };
 
 /* maximum quantization index */
-const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12] = {
+const uint16_t WebRtcIsacfix_kMaxIndGain[12] = {
   6,  12,  8,  14,  10,  19,  12,  31,  22,  56,  52,  138
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108] = {
+const uint16_t WebRtcIsacfix_kMaxIndShape[108] = {
   0,  0,  0,  0,  0,  0,  0,  0,  0,  1,
   2,  2,  2,  2,  4,  4,  5,  6,  0,  0,
   0,  0,  1,  0,  0,  0,  0,  1,  2,  2,
@@ -92,13 +92,13 @@
 };
 
 /* index offset */
-const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[3][12] = {
+const uint16_t WebRtcIsacfix_kOffsetGain[3][12] = {
   { 0,  7,  20,  29,  44,  55,  75,  88,  120,  143,  200,  253},
   { 0,  7,  19,  27,  42,  53,  73,  86,  117,  140,  197,  249},
   { 0,  7,  20,  28,  44,  55,  75,  89,  121,  145,  202,  257}
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[3][108] = {
+const uint16_t WebRtcIsacfix_kOffsetShape[3][108] = {
   {
     0,  1,  2,  3,  4,  5,  6,  7,  8,  9,
     11,  14,  17,  20,  23,  28,  33,  39,  46,  47,
@@ -141,13 +141,13 @@
 };
 
 /* initial cdf index for KLT coefficients */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[3][12] = {
+const uint16_t WebRtcIsacfix_kInitIndexGain[3][12] = {
   { 3,  6,  4,  7,  5,  10,  6,  16,  11,  28,  26,  69},
   { 3,  6,  4,  7,  5,  10,  6,  15,  11,  28,  26,  69},
   { 3,  6,  4,  8,  5,  10,  7,  16,  12,  28,  27,  70}
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[3][108] = {
+const uint16_t WebRtcIsacfix_kInitIndexShape[3][108] = {
   {
     0,  0,  0,  0,  0,  0,  0,  0,  0,  1,
     1,  1,  1,  1,  2,  2,  3,  3,  0,  0,
@@ -190,11 +190,11 @@
 };
 
 /* offsets for quantizer representation levels*/
-const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3] = {
+const uint16_t WebRtcIsacfix_kOfLevelsGain[3] = {
   0,  392,  779
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3] = {
+const uint16_t WebRtcIsacfix_kOfLevelsShape[3] = {
   0,  578,  1152
 };
 
@@ -202,7 +202,7 @@
 
 
 
-const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176] = {
+const int32_t WebRtcIsacfix_kLevelsGainQ17[1176] = {
   -364547,-231664,-102123,-573,104807,238257,368823,-758583,-640135,-510291
   ,-377662,-252785,-113177,2627,112906,248601,389461,522691,644517,763974
   ,-538963,-368179,-245823,-106095,-890,104299,241111,350730,493190,-800763
@@ -325,7 +325,7 @@
 
 
 
-const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735] = {
+const int16_t WebRtcIsacfix_kLevelsShapeQ10[1735] = {
   0,     0,    -1,     0,     0,     1,     0,     1,     0,  -821
   ,     1,  -763,    -1,   656,  -620,     0,   633,  -636,     4,   615
   ,  -630,     1,   649, -1773,  -670,     5,   678,  1810, -1876,  -676
@@ -503,7 +503,7 @@
 };
 
 /* cdf tables for quantizer indices */
-const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212] = {
+const uint16_t WebRtcIsacfix_kCdfGain[1212] = {
   0,  13,  301,  3730,  61784,  65167,  65489,  65535,  0,  17,
   142,  314,  929,  2466,  7678,  56450,  63463,  64740,  65204,  65426,
   65527,  65535,  0,  8,  100,  724,  6301,  60105,  65125,  65510,
@@ -628,7 +628,7 @@
   65533,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059] = {
+const uint16_t WebRtcIsacfix_kCdfShape[2059] = {
   0,  65535,  0,  65535,  0,  65535,  0,  65535,  0,  65535,
   0,  65535,  0,  65535,  0,  65535,  0,  65535,  0,  4,
   65535,  0,  8,  65514,  65535,  0,  29,  65481,  65535,  0,
@@ -838,7 +838,7 @@
 };
 
 /* pointers to cdf tables for quantizer indices */
-const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[3][12] = {
+const uint16_t *WebRtcIsacfix_kCdfGainPtr[3][12] = {
   { WebRtcIsacfix_kCdfGain +0 +0,   WebRtcIsacfix_kCdfGain +0 +8,   WebRtcIsacfix_kCdfGain +0 +22,
     WebRtcIsacfix_kCdfGain +0 +32,  WebRtcIsacfix_kCdfGain +0 +48,  WebRtcIsacfix_kCdfGain +0 +60,
     WebRtcIsacfix_kCdfGain +0 +81,  WebRtcIsacfix_kCdfGain +0 +95,  WebRtcIsacfix_kCdfGain +0 +128,
@@ -856,7 +856,7 @@
   }
 };
 
-const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[3][108] = {
+const uint16_t *WebRtcIsacfix_kCdfShapePtr[3][108] = {
   { WebRtcIsacfix_kCdfShape +0 +0,   WebRtcIsacfix_kCdfShape +0 +2,   WebRtcIsacfix_kCdfShape +0 +4,
     WebRtcIsacfix_kCdfShape +0 +6,   WebRtcIsacfix_kCdfShape +0 +8,   WebRtcIsacfix_kCdfShape +0 +10,
     WebRtcIsacfix_kCdfShape +0 +12,  WebRtcIsacfix_kCdfShape +0 +14,  WebRtcIsacfix_kCdfShape +0 +16,
@@ -972,7 +972,7 @@
 
 /* code length for all coefficients using different models */
 
-const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392] = {
+const int16_t WebRtcIsacfix_kCodeLenGainQ11[392] = {
   25189, 16036,  8717,   358,  8757, 15706, 21456, 24397, 18502, 17559
   , 13794, 11088,  7480,   873,  6603, 11636, 14627, 16805, 19132, 26624
   , 26624, 19408, 13751,  7280,   583,  7591, 15178, 23773, 28672, 25189
@@ -1015,7 +1015,7 @@
   , 30720, 30720
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577] = {
+const int16_t WebRtcIsacfix_kCodeLenShapeQ11[577] = {
   0,     0,     0,     0,     0,     0,     0,     0,     0, 28672
   ,     0, 26624,     1, 23773, 22819,     4, 20982, 18598,    10, 19282
   , 16587,    22, 16442, 26624, 13126,    60, 14245, 26624, 26624, 12736
@@ -1077,7 +1077,7 @@
 };
 
 /* left KLT transforms */
-const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[3][4] = {
+const int16_t WebRtcIsacfix_kT1GainQ15[3][4] = {
   { -26130, 19773, 19773, 26130 },
   { -26664, 19046, 19046, 26664 },
   { -23538, 22797, 22797, 23538 }
@@ -1085,7 +1085,7 @@
 
 
 
-const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[3][324] = {
+const int16_t WebRtcIsacfix_kT1ShapeQ15[3][324] = {
   { 52,16,168,7,439,-138,-89,306,671,882,
     157,1301,291,1598,-3571,-1943,-1119,32404,96,-12,
     379,-64,-307,345,-836,539,1045,2541,-2865,-992,
@@ -1191,7 +1191,7 @@
 };
 
 /* right KLT transforms */
-const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[3][36] = {
+const int16_t WebRtcIsacfix_kT2GainQ15[3][36] = {
   {   4775, -14892,  20313, -17104,  10533,  -3613,  -6782,  16044,  -8889,
       -11019,  21330, -10720,  13193, -15678, -11101,  14461,  12250, -13096,
       -16951,   2167,  16066,  15569,   -702, -16754, -19195, -12823,  -4321,
@@ -1209,7 +1209,7 @@
   }
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[3][36] = {
+const int16_t WebRtcIsacfix_kT2ShapeQ15[3][36] = {
   {   4400, -11512,  17205, -19470,  14770,  -5345,   9784, -19222,  11228,
       6842, -18371,   9909,  14191, -13496, -11563,  14015,  11827, -14839,
       -15439,    948,  17802,  14827,  -2053, -17132,  18723,  14516,   4135,
@@ -1228,7 +1228,7 @@
 };
 
 /* means of log gains and LAR coefficients*/
-const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[3][12] = {
+const int16_t WebRtcIsacfix_kMeansGainQ8[3][12] = {
   { -1758, -1370, -1758, -1373, -1757, -1375,
     -1758, -1374, -1758, -1373, -1755, -1370
   },
@@ -1241,7 +1241,7 @@
 };
 
 
-const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108] = {
+const int32_t WebRtcIsacfix_kMeansShapeQ17[3][108] = {
   { -119581, 34418, -44193, 11112, -4428, 18906, 9222, 8068, 1953, 5425,
     1871, 1689, 109933, 33751, 10471, -2566, 1090, 2320, -119219, 33728,
     -43759, 11450, -4870, 19117, 9174, 8037, 1972, 5331, 1872, 1843,
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
index 4f2e0e7..587bcd4 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
@@ -22,77 +22,77 @@
 
 
 /* indices of KLT coefficients used */
-extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12];
+extern const uint16_t WebRtcIsacfix_kSelIndGain[12];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108];
+extern const uint16_t WebRtcIsacfix_kSelIndShape[108];
 
 /* cdf array for model indicator */
-extern const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS+1];
+extern const uint16_t WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS+1];
 
 /* pointer to cdf array for model indicator */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1];
+extern const uint16_t *WebRtcIsacfix_kModelCdfPtr[1];
 
 /* initial cdf index for decoder of model indicator */
-extern const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1];
+extern const uint16_t WebRtcIsacfix_kModelInitIndex[1];
 
 /* offset to go from rounded value to quantization index */
-extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12];
+extern const int16_t WebRtcIsacfix_kQuantMinGain[12];
 
-extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108];
+extern const int16_t WebRtcIsacfix_kQuantMinShape[108];
 
 /* maximum quantization index */
-extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12];
+extern const uint16_t WebRtcIsacfix_kMaxIndGain[12];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108];
+extern const uint16_t WebRtcIsacfix_kMaxIndShape[108];
 
 /* index offset */
-extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[KLT_NUM_MODELS][12];
+extern const uint16_t WebRtcIsacfix_kOffsetGain[KLT_NUM_MODELS][12];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[KLT_NUM_MODELS][108];
+extern const uint16_t WebRtcIsacfix_kOffsetShape[KLT_NUM_MODELS][108];
 
 /* initial cdf index for KLT coefficients */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[KLT_NUM_MODELS][12];
+extern const uint16_t WebRtcIsacfix_kInitIndexGain[KLT_NUM_MODELS][12];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[KLT_NUM_MODELS][108];
+extern const uint16_t WebRtcIsacfix_kInitIndexShape[KLT_NUM_MODELS][108];
 
 /* offsets for quantizer representation levels */
-extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3];
+extern const uint16_t WebRtcIsacfix_kOfLevelsGain[3];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3];
+extern const uint16_t WebRtcIsacfix_kOfLevelsShape[3];
 
 /* quantizer representation levels */
-extern const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176];
+extern const int32_t WebRtcIsacfix_kLevelsGainQ17[1176];
 
-extern const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735];
+extern const int16_t WebRtcIsacfix_kLevelsShapeQ10[1735];
 
 /* cdf tables for quantizer indices */
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212];
+extern const uint16_t WebRtcIsacfix_kCdfGain[1212];
 
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059];
+extern const uint16_t WebRtcIsacfix_kCdfShape[2059];
 
 /* pointers to cdf tables for quantizer indices */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
+extern const uint16_t *WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
 
-extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
+extern const uint16_t *WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
 
 /* code length for all coefficients using different models */
-extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392];
+extern const int16_t WebRtcIsacfix_kCodeLenGainQ11[392];
 
-extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577];
+extern const int16_t WebRtcIsacfix_kCodeLenShapeQ11[577];
 
 /* left KLT transforms */
-extern const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4];
+extern const int16_t WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4];
 
-extern const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324];
+extern const int16_t WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324];
 
 /* right KLT transforms */
-extern const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36];
+extern const int16_t WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36];
 
-extern const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36];
+extern const int16_t WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36];
 
 /* means of log gains and LAR coefficients */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[KLT_NUM_MODELS][12];
+extern const int16_t WebRtcIsacfix_kMeansGainQ8[KLT_NUM_MODELS][12];
 
-extern const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108];
+extern const int32_t WebRtcIsacfix_kMeansShapeQ17[3][108];
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
index 6af02d8..a1dced9 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
@@ -24,24 +24,24 @@
 #include "system_wrappers/interface/compile_assert.h"
 
 /* log2[0.2, 0.5, 0.98] in Q8 */
-static const WebRtc_Word16 kLogLagWinQ8[3] = {
+static const int16_t kLogLagWinQ8[3] = {
   -594, -256, -7
 };
 
 /* [1 -0.75 0.25] in Q12 */
-static const WebRtc_Word16 kACoefQ12[3] = {
+static const int16_t kACoefQ12[3] = {
   4096, -3072, 1024
 };
 
 
 
-static __inline WebRtc_Word32 Log2Q8( WebRtc_UWord32 x ) {
+static __inline int32_t Log2Q8( uint32_t x ) {
 
-  WebRtc_Word32 zeros, lg2;
-  WebRtc_Word16 frac;
+  int32_t zeros, lg2;
+  int16_t frac;
 
   zeros=WebRtcSpl_NormU32(x);
-  frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)(WEBRTC_SPL_LSHIFT_W32(x, zeros))&0x7FFFFFFF), 23);
+  frac=(int16_t)WEBRTC_SPL_RSHIFT_W32(((uint32_t)(WEBRTC_SPL_LSHIFT_W32(x, zeros))&0x7FFFFFFF), 23);
   /* log2(magn(i)) */
 
   lg2= (WEBRTC_SPL_LSHIFT_W32((31-zeros), 8)+frac);
@@ -49,27 +49,27 @@
 
 }
 
-static __inline WebRtc_Word16 Exp2Q10(WebRtc_Word16 x) { // Both in and out in Q10
+static __inline int16_t Exp2Q10(int16_t x) { // Both in and out in Q10
 
-  WebRtc_Word16 tmp16_1, tmp16_2;
+  int16_t tmp16_1, tmp16_2;
 
-  tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
-  tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+  tmp16_2=(int16_t)(0x0400|(x&0x03FF));
+  tmp16_1=-(int16_t)WEBRTC_SPL_RSHIFT_W16(x,10);
   if(tmp16_1>0)
-    return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+    return (int16_t) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
   else
-    return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+    return (int16_t) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
 
 }
 
 
 
 /* 1D parabolic interpolation . All input and output values are in Q8 */
-static __inline void Intrp1DQ8(WebRtc_Word32 *x, WebRtc_Word32 *fx, WebRtc_Word32 *y, WebRtc_Word32 *fy) {
+static __inline void Intrp1DQ8(int32_t *x, int32_t *fx, int32_t *y, int32_t *fy) {
 
-  WebRtc_Word16 sign1=1, sign2=1;
-  WebRtc_Word32 r32, q32, t32, nom32, den32;
-  WebRtc_Word16 t16, tmp16, tmp16_1;
+  int16_t sign1=1, sign2=1;
+  int32_t r32, q32, t32, nom32, den32;
+  int16_t t16, tmp16, tmp16_1;
 
   if ((fx[0]>0) && (fx[2]>0)) {
     r32=fx[1]-fx[2];
@@ -85,7 +85,7 @@
     /* (Signs are removed because WebRtcSpl_DivResultInQ31 can't handle negative numbers) */
     t32=WebRtcSpl_DivResultInQ31(WEBRTC_SPL_MUL_32_16(nom32, sign1),WEBRTC_SPL_MUL_32_16(den32, sign2)); /* t in Q31, without signs */
 
-    t16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(t32, 23);  /* Q8 */
+    t16=(int16_t)WEBRTC_SPL_RSHIFT_W32(t32, 23);  /* Q8 */
     t16=t16*sign1*sign2;        /* t in Q8 with signs */
 
     *y = x[0]+t16;          /* Q8 */
@@ -95,9 +95,9 @@
     /* fy = 0.5 * t * (t-1) * fx[0] + (1-t*t) * fx[1] + 0.5 * t * (t+1) * fx[2]; */
 
     /* Part I: 0.5 * t * (t-1) * fx[0] */
-    tmp16_1=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16,t16); /* Q8*Q8=Q16 */
+    tmp16_1=(int16_t)WEBRTC_SPL_MUL_16_16(t16,t16); /* Q8*Q8=Q16 */
     tmp16_1 = WEBRTC_SPL_RSHIFT_W16(tmp16_1,2);  /* Q16>>2 = Q14 */
-    t16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16, 64);           /* Q8<<6 = Q14  */
+    t16 = (int16_t)WEBRTC_SPL_MUL_16_16(t16, 64);           /* Q8<<6 = Q14  */
     tmp16 = tmp16_1-t16;
     *fy = WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[0]); /* (Q14 * Q8 >>15)/2 = Q8 */
 
@@ -115,10 +115,10 @@
 }
 
 
-static void FindFour32(WebRtc_Word32 *in, WebRtc_Word16 length, WebRtc_Word16 *bestind)
+static void FindFour32(int32_t *in, int16_t length, int16_t *bestind)
 {
-  WebRtc_Word32 best[4]= {-100, -100, -100, -100};
-  WebRtc_Word16 k;
+  int32_t best[4]= {-100, -100, -100, -100};
+  int16_t k;
 
   for (k=0; k<length; k++) {
     if (in[k] > best[3]) {
@@ -159,35 +159,35 @@
 
 
 
-static void PCorr2Q32(const WebRtc_Word16 *in, WebRtc_Word32 *logcorQ8)
+static void PCorr2Q32(const int16_t *in, int32_t *logcorQ8)
 {
-  WebRtc_Word16 scaling,n,k;
-  WebRtc_Word32 ysum32,csum32, lys, lcs;
-  WebRtc_Word32 oneQ8;
+  int16_t scaling,n,k;
+  int32_t ysum32,csum32, lys, lcs;
+  int32_t oneQ8;
 
 
-  const WebRtc_Word16 *x, *inptr;
+  const int16_t *x, *inptr;
 
-  oneQ8 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, 8);  // 1.00 in Q8
+  oneQ8 = WEBRTC_SPL_LSHIFT_W32((int32_t)1, 8);  // 1.00 in Q8
 
   x = in + PITCH_MAX_LAG/2 + 2;
-  scaling = WebRtcSpl_GetScalingSquare ((WebRtc_Word16 *) in, PITCH_CORR_LEN2, PITCH_CORR_LEN2);
+  scaling = WebRtcSpl_GetScalingSquare ((int16_t *) in, PITCH_CORR_LEN2, PITCH_CORR_LEN2);
   ysum32 = 1;
   csum32 = 0;
   x = in + PITCH_MAX_LAG/2 + 2;
   for (n = 0; n < PITCH_CORR_LEN2; n++) {
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[n],(WebRtc_Word16) in[n], scaling);  // Q0
-    csum32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) x[n],(WebRtc_Word16) in[n], scaling); // Q0
+    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[n],(int16_t) in[n], scaling);  // Q0
+    csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t) x[n],(int16_t) in[n], scaling); // Q0
   }
 
   logcorQ8 += PITCH_LAG_SPAN2 - 1;
 
-  lys=Log2Q8((WebRtc_UWord32) ysum32); // Q8
+  lys=Log2Q8((uint32_t) ysum32); // Q8
   lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
 
   if (csum32>0) {
 
-    lcs=Log2Q8((WebRtc_UWord32) csum32);   // 2log(csum) in Q8
+    lcs=Log2Q8((uint32_t) csum32);   // 2log(csum) in Q8
 
     if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2 in Q8
       *logcorQ8 = lcs - lys;  // log2(csum/sqrt(ysum))
@@ -202,8 +202,8 @@
 
   for (k = 1; k < PITCH_LAG_SPAN2; k++) {
     inptr = &in[k];
-    ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[k-1],(WebRtc_Word16) in[k-1], scaling);
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1],(WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1], scaling);
+    ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[k-1],(int16_t) in[k-1], scaling);
+    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[PITCH_CORR_LEN2 + k - 1],(int16_t) in[PITCH_CORR_LEN2 + k - 1], scaling);
 
 #ifdef WEBRTC_ARCH_ARM_NEON
     {
@@ -244,12 +244,12 @@
 
     logcorQ8--;
 
-    lys=Log2Q8((WebRtc_UWord32)ysum32); // Q8
+    lys=Log2Q8((uint32_t)ysum32); // Q8
     lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
 
     if (csum32>0) {
 
-      lcs=Log2Q8((WebRtc_UWord32) csum32);   // 2log(csum) in Q8
+      lcs=Log2Q8((uint32_t) csum32);   // 2log(csum) in Q8
 
       if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2
         *logcorQ8 = lcs - lys;  // log2(csum/sqrt(ysum))
@@ -265,34 +265,34 @@
 
 
 
-void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in, /* Q0 */
+void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
                                 PitchAnalysisStruct *State,
-                                WebRtc_Word16 *lagsQ7                   /* Q7 */
+                                int16_t *lagsQ7                   /* Q7 */
                                 )
 {
-  WebRtc_Word16 buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
-  WebRtc_Word32 *crrvecQ8_1,*crrvecQ8_2;
-  WebRtc_Word32 cv1q[PITCH_LAG_SPAN2+2],cv2q[PITCH_LAG_SPAN2+2], peakvq[PITCH_LAG_SPAN2+2];
+  int16_t buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+  int32_t *crrvecQ8_1,*crrvecQ8_2;
+  int32_t cv1q[PITCH_LAG_SPAN2+2],cv2q[PITCH_LAG_SPAN2+2], peakvq[PITCH_LAG_SPAN2+2];
   int k;
-  WebRtc_Word16 peaks_indq;
-  WebRtc_Word16 peakiq[PITCH_LAG_SPAN2];
-  WebRtc_Word32 corr;
-  WebRtc_Word32 corr32, corr_max32, corr_max_o32;
-  WebRtc_Word16 npkq;
-  WebRtc_Word16 best4q[4]={0,0,0,0};
-  WebRtc_Word32 xq[3],yq[1],fyq[1];
-  WebRtc_Word32 *fxq;
-  WebRtc_Word32 best_lag1q, best_lag2q;
-  WebRtc_Word32 tmp32a,tmp32b,lag32,ratq;
-  WebRtc_Word16 start;
-  WebRtc_Word16 oldgQ12, tmp16a, tmp16b, gain_bias16,tmp16c, tmp16d, bias16;
-  WebRtc_Word32 tmp32c,tmp32d, tmp32e;
-  WebRtc_Word16 old_lagQ;
-  WebRtc_Word32 old_lagQ8;
-  WebRtc_Word32 lagsQ8[4];
+  int16_t peaks_indq;
+  int16_t peakiq[PITCH_LAG_SPAN2];
+  int32_t corr;
+  int32_t corr32, corr_max32, corr_max_o32;
+  int16_t npkq;
+  int16_t best4q[4]={0,0,0,0};
+  int32_t xq[3],yq[1],fyq[1];
+  int32_t *fxq;
+  int32_t best_lag1q, best_lag2q;
+  int32_t tmp32a,tmp32b,lag32,ratq;
+  int16_t start;
+  int16_t oldgQ12, tmp16a, tmp16b, gain_bias16,tmp16c, tmp16d, bias16;
+  int32_t tmp32c,tmp32d, tmp32e;
+  int16_t old_lagQ;
+  int32_t old_lagQ8;
+  int32_t lagsQ8[4];
 
   old_lagQ = State->PFstr_wght.oldlagQ7; // Q7
-  old_lagQ8= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)old_lagQ,1); //Q8
+  old_lagQ8= WEBRTC_SPL_LSHIFT_W32((int32_t)old_lagQ,1); //Q8
 
   oldgQ12= State->PFstr_wght.oldgainQ12;
 
@@ -301,7 +301,7 @@
 
 
   /* copy old values from state buffer */
-  memcpy(buf_dec16, State->dec_buffer16, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2)));
+  memcpy(buf_dec16, State->dec_buffer16, WEBRTC_SPL_MUL_16_16(sizeof(int16_t), (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2)));
 
   /* decimation; put result after the old values */
   WebRtcIsacfix_DecimateAllpass32(in, State->decimator_state32, PITCH_FRAME_LEN,
@@ -309,7 +309,7 @@
 
   /* low-pass filtering */
   start= PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2;
-  WebRtcSpl_FilterARFastQ12(&buf_dec16[start],&buf_dec16[start],(WebRtc_Word16*)kACoefQ12,3, PITCH_FRAME_LEN/2);
+  WebRtcSpl_FilterARFastQ12(&buf_dec16[start],&buf_dec16[start],(int16_t*)kACoefQ12,3, PITCH_FRAME_LEN/2);
 
   /* copy end part back into state buffer */
   for (k = 0; k < (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2); k++)
@@ -322,25 +322,25 @@
 
 
   /* bias towards pitch lag of previous frame */
-  tmp32a = Log2Q8((WebRtc_UWord32) old_lagQ8) - 2304; // log2(0.5*oldlag) in Q8
+  tmp32a = Log2Q8((uint32_t) old_lagQ8) - 2304; // log2(0.5*oldlag) in Q8
   tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0;
-  gain_bias16 = (WebRtc_Word16) tmp32b;  //Q12
+  gain_bias16 = (int16_t) tmp32b;  //Q12
   if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12
 
 
   for (k = 0; k < PITCH_LAG_SPAN2; k++)
   {
     if (crrvecQ8_1[k]>0) {
-      tmp32b = Log2Q8((WebRtc_UWord32) (k + (PITCH_MIN_LAG/2-2)));
-      tmp16a = (WebRtc_Word16) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
+      tmp32b = Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2)));
+      tmp16a = (int16_t) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
       tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10
-      tmp16b = (WebRtc_Word16) tmp32c; // Q10 & <8
+      tmp16b = (int16_t) tmp32c; // Q10 & <8
       tmp32d = WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 177 , 8); // mult with ln2 in Q8
-      tmp16c = (WebRtc_Word16) tmp32d; // Q10 & <4
-      tmp16d = Exp2Q10((WebRtc_Word16) -tmp16c); //Q10
+      tmp16c = (int16_t) tmp32d; // Q10 & <4
+      tmp16d = Exp2Q10((int16_t) -tmp16c); //Q10
       tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10  & * 0.5
-      bias16 = (WebRtc_Word16) (1024 + tmp32c); // Q10
-      tmp32b = Log2Q8((WebRtc_UWord32) bias16) - 2560; // Q10 in -> Q8 out with 10*2^8 offset
+      bias16 = (int16_t) (1024 + tmp32c); // Q10
+      tmp32b = Log2Q8((uint32_t) bias16) - 2560; // Q10 in -> Q8 out with 10*2^8 offset
       crrvecQ8_1[k] += tmp32b ; // -10*2^8 offset
     }
   }
@@ -402,7 +402,7 @@
   corr_max32=0;
   best_lag1q =0;
   if (peaks_indq > 0) {
-    FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
+    FindFour32(peakvq, (int16_t) peaks_indq, best4q);
     npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
 
     for (k=0;k<npkq;k++) {
@@ -413,10 +413,10 @@
       xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8);
       Intrp1DQ8(xq, fxq, yq, fyq);
 
-      tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
+      tmp32a= Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -42, 8);
+      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -42, 8);
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
@@ -440,12 +440,12 @@
   for (k = 1; k <= PITCH_LAG_SPAN2; k++)
   {
     tmp32a = WEBRTC_SPL_LSHIFT_W32(k, 7); // 0.5*k Q8
-    tmp32b = (WebRtc_Word32) (WEBRTC_SPL_LSHIFT_W32(tmp32a, 1)) - ratq; // Q8
-    tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32b, (WebRtc_Word16) tmp32b, 8); // Q8
+    tmp32b = (int32_t) (WEBRTC_SPL_LSHIFT_W32(tmp32a, 1)) - ratq; // Q8
+    tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32b, (int16_t) tmp32b, 8); // Q8
 
-    tmp32b = (WebRtc_Word32) tmp32c + (WebRtc_Word32)  WEBRTC_SPL_RSHIFT_W32(ratq, 1); // (k-r)^2 + 0.5 * r  Q8
-    tmp32c = Log2Q8((WebRtc_UWord32) tmp32a) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
-    tmp32d = Log2Q8((WebRtc_UWord32) tmp32b) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
+    tmp32b = (int32_t) tmp32c + (int32_t)  WEBRTC_SPL_RSHIFT_W32(ratq, 1); // (k-r)^2 + 0.5 * r  Q8
+    tmp32c = Log2Q8((uint32_t) tmp32a) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
+    tmp32d = Log2Q8((uint32_t) tmp32b) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
     tmp32e =  tmp32c -tmp32d;
 
     cv2q[k] += WEBRTC_SPL_RSHIFT_W32(tmp32e, 1);
@@ -474,7 +474,7 @@
   best_lag2q =0;
   if (peaks_indq > 0) {
 
-    FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
+    FindFour32(peakvq, (int16_t) peaks_indq, best4q);
     npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
     for (k=0;k<npkq;k++) {
 
@@ -487,8 +487,8 @@
 
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */
-      tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -82, 8);
+      tmp32a= Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
+      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -82, 8);
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
@@ -506,24 +506,24 @@
     lagsQ8[3] = lagsQ8[0];
   }
 
-  lagsQ7[0]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[0], 1);
-  lagsQ7[1]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[1], 1);
-  lagsQ7[2]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[2], 1);
-  lagsQ7[3]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[3], 1);
+  lagsQ7[0]=(int16_t) WEBRTC_SPL_RSHIFT_W32(lagsQ8[0], 1);
+  lagsQ7[1]=(int16_t) WEBRTC_SPL_RSHIFT_W32(lagsQ8[1], 1);
+  lagsQ7[2]=(int16_t) WEBRTC_SPL_RSHIFT_W32(lagsQ8[2], 1);
+  lagsQ7[3]=(int16_t) WEBRTC_SPL_RSHIFT_W32(lagsQ8[3], 1);
 
 
 }
 
 
 
-void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *inn,               /* PITCH_FRAME_LEN samples */
-                                 WebRtc_Word16 *outQ0,                  /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+void WebRtcIsacfix_PitchAnalysis(const int16_t *inn,               /* PITCH_FRAME_LEN samples */
+                                 int16_t *outQ0,                  /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
                                  PitchAnalysisStruct *State,
-                                 WebRtc_Word16 *PitchLags_Q7,
-                                 WebRtc_Word16 *PitchGains_Q12)
+                                 int16_t *PitchLags_Q7,
+                                 int16_t *PitchGains_Q12)
 {
-  WebRtc_Word16 inbufQ0[PITCH_FRAME_LEN + QLOOKAHEAD];
-  WebRtc_Word16 k;
+  int16_t inbufQ0[PITCH_FRAME_LEN + QLOOKAHEAD];
+  int16_t k;
 
   /* inital pitch estimate */
   WebRtcIsacfix_InitialPitch(inn, State,  PitchLags_Q7);
@@ -537,7 +537,7 @@
     inbufQ0[k] = State->inbuf[k];
   }
   for (k = 0; k < PITCH_FRAME_LEN; k++) {
-    inbufQ0[k+QLOOKAHEAD] = (WebRtc_Word16) inn[k];
+    inbufQ0[k+QLOOKAHEAD] = (int16_t) inn[k];
   }
 
   /* lookahead pitch filtering for masking analysis */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
index 6225256..93c81c8 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
@@ -20,42 +20,42 @@
 
 #include "structs.h"
 
-void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *in,               /* PITCH_FRAME_LEN samples */
-                                 WebRtc_Word16 *outQ0,                  /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+void WebRtcIsacfix_PitchAnalysis(const int16_t *in,               /* PITCH_FRAME_LEN samples */
+                                 int16_t *outQ0,                  /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
                                  PitchAnalysisStruct *State,
-                                 WebRtc_Word16 *lagsQ7,
-                                 WebRtc_Word16 *PitchGains_Q12);
+                                 int16_t *lagsQ7,
+                                 int16_t *PitchGains_Q12);
 
-void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in,
+void WebRtcIsacfix_InitialPitch(const int16_t *in,
                                 PitchAnalysisStruct *State,
-                                WebRtc_Word16 *qlags);
+                                int16_t *qlags);
 
-void WebRtcIsacfix_PitchFilter(WebRtc_Word16 *indatFix,
-                               WebRtc_Word16 *outdatQQ,
+void WebRtcIsacfix_PitchFilter(int16_t *indatFix,
+                               int16_t *outdatQQ,
                                PitchFiltstr *pfp,
-                               WebRtc_Word16 *lagsQ7,
-                               WebRtc_Word16 *gainsQ12,
-                               WebRtc_Word16 type);
+                               int16_t *lagsQ7,
+                               int16_t *gainsQ12,
+                               int16_t type);
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
-                                   WebRtc_Word16 gain,
+                                   int16_t gain,
                                    int index,
-                                   WebRtc_Word16 sign,
-                                   WebRtc_Word16* inputState,
-                                   WebRtc_Word16* outputBuff2,
-                                   const WebRtc_Word16* coefficient,
-                                   WebRtc_Word16* inputBuf,
-                                   WebRtc_Word16* outputBuf,
+                                   int16_t sign,
+                                   int16_t* inputState,
+                                   int16_t* outputBuff2,
+                                   const int16_t* coefficient,
+                                   int16_t* inputBuf,
+                                   int16_t* outputBuf,
                                    int* index2);
 
-void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16 *indatQ0,
+void WebRtcIsacfix_PitchFilterGains(const int16_t *indatQ0,
                                     PitchFiltstr *pfp,
-                                    WebRtc_Word16 *lagsQ7,
-                                    WebRtc_Word16 *gainsQ12);
+                                    int16_t *lagsQ7,
+                                    int16_t *gainsQ12);
 
-void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
-                                     WebRtc_Word32 *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
-                                     WebRtc_Word16 N,                   /* number of input samples */
-                                     WebRtc_Word16 *out);             /* array of size N/2 */
+void WebRtcIsacfix_DecimateAllpass32(const int16_t *in,
+                                     int32_t *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
+                                     int16_t N,                   /* number of input samples */
+                                     int16_t *out);             /* array of size N/2 */
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
index f30293e..89d884a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
@@ -25,16 +25,16 @@
 static const int kSegments = 5;
 
 // A division factor of 1/5 in Q15.
-static const WebRtc_Word16 kDivFactor = 6553;
+static const int16_t kDivFactor = 6553;
 
 // Filter coefficicients in Q15.
-static const WebRtc_Word16 kDampFilter[PITCH_DAMPORDER] = {
+static const int16_t kDampFilter[PITCH_DAMPORDER] = {
   -2294, 8192, 20972, 8192, -2294
 };
 
 // Interpolation coefficients; generated by design_pitch_filter.m.
 // Coefficients are stored in Q14.
-static const WebRtc_Word16 kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+static const int16_t kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
   {-367, 1090, -2706,  9945, 10596, -3318,  1626, -781,  287},
   {-325,  953, -2292,  7301, 12963, -3320,  1570, -743,  271},
   {-240,  693, -1622,  4634, 14809, -2782,  1262, -587,  212},
@@ -48,44 +48,44 @@
 // Function prototype for pitch filtering.
 // TODO(Turaj): Add descriptions of input and output parameters.
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
-                                   WebRtc_Word16 gain,
+                                   int16_t gain,
                                    int index,
-                                   WebRtc_Word16 sign,
-                                   WebRtc_Word16* inputState,
-                                   WebRtc_Word16* outputBuf2,
-                                   const WebRtc_Word16* coefficient,
-                                   WebRtc_Word16* inputBuf,
-                                   WebRtc_Word16* outputBuf,
+                                   int16_t sign,
+                                   int16_t* inputState,
+                                   int16_t* outputBuf2,
+                                   const int16_t* coefficient,
+                                   int16_t* inputBuf,
+                                   int16_t* outputBuf,
                                    int* index2);
 
-static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal,
-                                         WebRtc_Word16 qDomain) {
-  WebRtc_Word32 intgr;
-  WebRtc_Word32 roundVal;
+static __inline int32_t CalcLrIntQ(int32_t fixVal,
+                                   int16_t qDomain) {
+  int32_t intgr;
+  int32_t roundVal;
 
-  roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,  qDomain - 1);
+  roundVal = WEBRTC_SPL_LSHIFT_W32((int32_t)1,  qDomain - 1);
   intgr = WEBRTC_SPL_RSHIFT_W32(fixVal + roundVal, qDomain);
 
   return intgr;
 }
 
-void WebRtcIsacfix_PitchFilter(WebRtc_Word16* indatQQ, // Q10 if type is 1 or 4,
+void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
                                                        // Q0 if type is 2.
-                               WebRtc_Word16* outdatQQ,
+                               int16_t* outdatQQ,
                                PitchFiltstr* pfp,
-                               WebRtc_Word16* lagsQ7,
-                               WebRtc_Word16* gainsQ12,
-                               WebRtc_Word16 type) {
+                               int16_t* lagsQ7,
+                               int16_t* gainsQ12,
+                               int16_t type) {
   int    k, ind, cnt;
-  WebRtc_Word16 sign = 1;
-  WebRtc_Word16 inystateQQ[PITCH_DAMPORDER];
-  WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE + QLOOKAHEAD];
-  const WebRtc_Word16 Gain = 21299;     // 1.3 in Q14
-  WebRtc_Word16 oldLagQ7;
-  WebRtc_Word16 oldGainQ12, lagdeltaQ7, curLagQ7, gaindeltaQ12, curGainQ12;
+  int16_t sign = 1;
+  int16_t inystateQQ[PITCH_DAMPORDER];
+  int16_t ubufQQ[PITCH_INTBUFFSIZE + QLOOKAHEAD];
+  const int16_t Gain = 21299;     // 1.3 in Q14
+  int16_t oldLagQ7;
+  int16_t oldGainQ12, lagdeltaQ7, curLagQ7, gaindeltaQ12, curGainQ12;
   int indW32 = 0, frcQQ = 0;
-  WebRtc_Word32 tmpW32;
-  const WebRtc_Word16* fracoeffQQ = NULL;
+  int32_t tmpW32;
+  const int16_t* fracoeffQQ = NULL;
 
   // Assumptions in ARM assembly for WebRtcIsacfix_PitchFilterCoreARM().
   COMPILE_ASSERT(PITCH_FRACORDER == 9);
@@ -104,7 +104,7 @@
 
     // Make output more periodic.
     for (k = 0; k < PITCH_SUBFRAMES; k++) {
-      gainsQ12[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+      gainsQ12[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
           gainsQ12[k], Gain, 14);
     }
   }
@@ -121,11 +121,11 @@
   for (k = 0; k < PITCH_SUBFRAMES; k++) {
     // Calculate interpolation steps.
     lagdeltaQ7 = lagsQ7[k] - oldLagQ7;
-    lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+    lagdeltaQ7 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
                   lagdeltaQ7, kDivFactor, 15);
     curLagQ7 = oldLagQ7;
     gaindeltaQ12 = gainsQ12[k] - oldGainQ12;
-    gaindeltaQ12 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+    gaindeltaQ12 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
                     gaindeltaQ12, kDivFactor, 15);
 
     curGainQ12 = oldGainQ12;
@@ -172,18 +172,18 @@
 }
 
 
-void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16* indatQ0,
+void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
                                     PitchFiltstr* pfp,
-                                    WebRtc_Word16* lagsQ7,
-                                    WebRtc_Word16* gainsQ12) {
+                                    int16_t* lagsQ7,
+                                    int16_t* gainsQ12) {
   int  k, n, m, ind, pos, pos3QQ;
 
-  WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE];
-  WebRtc_Word16 oldLagQ7, lagdeltaQ7, curLagQ7;
-  const WebRtc_Word16* fracoeffQQ = NULL;
-  WebRtc_Word16 scale;
-  WebRtc_Word16 cnt = 0, frcQQ, indW16 = 0, tmpW16;
-  WebRtc_Word32 tmpW32, tmp2W32, csum1QQ, esumxQQ;
+  int16_t ubufQQ[PITCH_INTBUFFSIZE];
+  int16_t oldLagQ7, lagdeltaQ7, curLagQ7;
+  const int16_t* fracoeffQQ = NULL;
+  int16_t scale;
+  int16_t cnt = 0, frcQQ, indW16 = 0, tmpW16;
+  int32_t tmpW32, tmp2W32, csum1QQ, esumxQQ;
 
   // Set up buffer and states.
   memcpy(ubufQQ, pfp->ubufQQ, sizeof(pfp->ubufQQ));
@@ -202,7 +202,7 @@
 
     // Calculate interpolation steps.
     lagdeltaQ7 = lagsQ7[k] - oldLagQ7;
-    lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+    lagdeltaQ7 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
                    lagdeltaQ7, kDivFactor, 15);
     curLagQ7 = oldLagQ7;
     oldLagQ7 = lagsQ7[k];
@@ -215,7 +215,7 @@
     for (cnt = 0; cnt < kSegments; cnt++) {
       // Update parameters for each segment.
       curLagQ7 += lagdeltaQ7;
-      indW16 = (WebRtc_Word16)CalcLrIntQ(curLagQ7, 7);
+      indW16 = (int16_t)CalcLrIntQ(curLagQ7, 7);
       tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16, 7);
       tmpW16 -= curLagQ7;
       frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16, 4);
@@ -241,7 +241,7 @@
 
         tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32);
         tmpW32 += 8192;
-        tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
+        tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
         tmpW32 = WEBRTC_SPL_MUL_16_16(tmpW16, tmpW16);
 
         if ((tmp2W32 > 1073700000) || (csum1QQ > 1073700000) ||
@@ -269,7 +269,7 @@
     } else {
       tmpW32 = 4096;
     }
-    gainsQ12[k] = (WebRtc_Word16)WEBRTC_SPL_SAT(PITCH_MAX_GAIN_Q12, tmpW32, 0);
+    gainsQ12[k] = (int16_t)WEBRTC_SPL_SAT(PITCH_MAX_GAIN_Q12, tmpW32, 0);
   }
 
   // Export buffer and states.
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
index ffd0e63..57796b0 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
@@ -20,14 +20,14 @@
 .align  2
 
 @ void WebRtcIsacfix_PitchFilterCore(int loopNumber,
-@                                    WebRtc_Word16 gain,
+@                                    int16_t gain,
 @                                    int index,
-@                                    WebRtc_Word16 sign,
-@                                    WebRtc_Word16* inputState,
-@                                    WebRtc_Word16* outputBuf2,
-@                                    const WebRtc_Word16* coefficient,
-@                                    WebRtc_Word16* inputBuf,
-@                                    WebRtc_Word16* outputBuf,
+@                                    int16_t sign,
+@                                    int16_t* inputState,
+@                                    int16_t* outputBuf2,
+@                                    const int16_t* coefficient,
+@                                    int16_t* inputBuf,
+@                                    int16_t* outputBuf,
 @                                    int* index2) {
 DEFINE_FUNCTION WebRtcIsacfix_PitchFilterCore
   push {r4-r11}
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
index 29b4b6a..d3c90b3 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
@@ -12,26 +12,26 @@
 #include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 
 /* Filter coefficicients in Q15. */
-static const WebRtc_Word16 kDampFilter[PITCH_DAMPORDER] = {
+static const int16_t kDampFilter[PITCH_DAMPORDER] = {
   -2294, 8192, 20972, 8192, -2294
 };
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
-                                   WebRtc_Word16 gain,
+                                   int16_t gain,
                                    int index,
-                                   WebRtc_Word16 sign,
-                                   WebRtc_Word16* inputState,
-                                   WebRtc_Word16* outputBuf2,
-                                   const WebRtc_Word16* coefficient,
-                                   WebRtc_Word16* inputBuf,
-                                   WebRtc_Word16* outputBuf,
+                                   int16_t sign,
+                                   int16_t* inputState,
+                                   int16_t* outputBuf2,
+                                   const int16_t* coefficient,
+                                   int16_t* inputBuf,
+                                   int16_t* outputBuf,
                                    int* index2) {
   int i = 0, j = 0;  /* Loop counters. */
-  WebRtc_Word16* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)];
-  WebRtc_Word16 tmpW16 = 0;
+  int16_t* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)];
+  int16_t tmpW16 = 0;
 
   for (i = 0; i < loopNumber; i++) {
-    WebRtc_Word32 tmpW32 = 0;
+    int32_t tmpW32 = 0;
 
     /* Filter to get fractional pitch. */
     for (j = 0; j < PITCH_FRACORDER; j++) {
@@ -41,12 +41,12 @@
     /* Saturate to avoid overflow in tmpW16. */
     tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104);
     tmpW32 += 8192;
-    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
+    tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
 
     /* Shift low pass filter state. */
     memmove(&inputState[1], &inputState[0],
-            (PITCH_DAMPORDER - 1) * sizeof(WebRtc_Word16));
-    inputState[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+            (PITCH_DAMPORDER - 1) * sizeof(int16_t));
+    inputState[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
                       gain, tmpW16, 12);
 
     /* Low pass filter. */
@@ -60,7 +60,7 @@
     /* Saturate to avoid overflow in tmpW16. */
     tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208);
     tmpW32 += 16384;
-    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 15);
+    tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 15);
 
     /* Subtract from input and update buffer. */
     tmpW32 = inputBuf[*index2] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
index 50ea658..bfd83b7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
@@ -21,7 +21,7 @@
 /********************* Pitch Filter Gain Coefficient Tables ************************/
 
 /* cdf for quantized pitch filter gains */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255] = {
+const uint16_t WebRtcIsacfix_kPitchGainCdf[255] = {
   0,  2,  4,  6,  64,  901,  903,  905,  16954,  16956,
   16961,  17360,  17362,  17364,  17366,  17368,  17370,  17372,  17374,  17411,
   17514,  17516,  17583,  18790,  18796,  18802,  20760,  20777,  20782,  21722,
@@ -51,25 +51,25 @@
 };
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3] = {
+const int16_t WebRtcIsacfix_kLowerlimiGain[3] = {
   -7, -2, -1
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3] = {
+const int16_t WebRtcIsacfix_kUpperlimitGain[3] = {
   0,  3,  1
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2] = {
+const uint16_t WebRtcIsacfix_kMultsGain[2] = {
   18,  3
 };
 
 /* size of cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1] = {
+const uint16_t WebRtcIsacfix_kCdfTableSizeGain[1] = {
   256
 };
 
 /* mean values of pitch filter gains in FIXED point Q12 */
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144] = {
+const int16_t WebRtcIsacfix_kPitchGain1[144] = {
   843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839,
   1843, 1843, 1843, 1843, 1843,   1843, 1843, 814, 846, 1092, 1013,
   1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843,   1843,
@@ -86,7 +86,7 @@
   1265
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144] = {
+const int16_t WebRtcIsacfix_kPitchGain2[144] = {
   1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784,
   1606, 1843, 1843, 1711, 1843,   1843, 1814, 1389, 1275, 1040, 1564,
   1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720,   1475,
@@ -103,7 +103,7 @@
   87
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144] = {
+const int16_t WebRtcIsacfix_kPitchGain3[144] = {
   1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639,
   1393, 1760, 1525, 1285, 1656,   1419, 1176, 1835, 1718, 1475, 1841,
   1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299,   1040,
@@ -121,7 +121,7 @@
 };
 
 
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144] = {
+const int16_t WebRtcIsacfix_kPitchGain4[144] = {
   1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434,
   1656, 843, 1092, 1336, 504,    757, 1007, 1843, 1843, 1843, 1838,
   1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821,    1092,
@@ -141,7 +141,7 @@
 
 
 /* transform matrix in Q12*/
-const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4] = {
+const int16_t WebRtcIsacfix_kTransform[4][4] = {
   { -2048, -2048, -2048, -2048 },
   {  2748,   916,  -916, -2748 },
   {  2048, -2048, -2048,  2048 },
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
index 788e553..c4e0be5 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
@@ -23,23 +23,23 @@
 
 /********************* Pitch Filter Gain Coefficient Tables ************************/
 /* cdf for quantized pitch filter gains */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255];
+extern const uint16_t WebRtcIsacfix_kPitchGainCdf[255];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3];
-extern const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2];
+extern const int16_t WebRtcIsacfix_kLowerlimiGain[3];
+extern const int16_t WebRtcIsacfix_kUpperlimitGain[3];
+extern const uint16_t WebRtcIsacfix_kMultsGain[2];
 
 /* mean values of pitch filter gains in Q12*/
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144];
+extern const int16_t WebRtcIsacfix_kPitchGain1[144];
+extern const int16_t WebRtcIsacfix_kPitchGain2[144];
+extern const int16_t WebRtcIsacfix_kPitchGain3[144];
+extern const int16_t WebRtcIsacfix_kPitchGain4[144];
 
 /* size of cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1];
+extern const uint16_t WebRtcIsacfix_kCdfTableSizeGain[1];
 
 /* transform matrix */
-extern const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4];
+extern const int16_t WebRtcIsacfix_kTransform[4][4];
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
index 81700e4..4566b6eb 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
@@ -24,7 +24,7 @@
 /* tables for use with small pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf1Lo[127] = {
   0,  134,  336,  549,  778,  998,  1264,  1512,  1777,  2070,
   2423,  2794,  3051,  3361,  3708,  3979,  4315,  4610,  4933,  5269,
   5575,  5896,  6155,  6480,  6816,  7129,  7477,  7764,  8061,  8358,
@@ -40,20 +40,20 @@
   65152,  65535,  65535,  65535,  65535,  65535,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf2Lo[20] = {
   0,  429,  3558,  5861,  8558,  11639,  15210,  19502,  24773,  31983,
   42602,  48567,  52601,  55676,  58160,  60172,  61889,  63235,  65383,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf3Lo[2] = {
   0,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf4Lo[10] = {
   0,  2966,  6368,  11182,  19431,  37793,  48532,  55353,  60626,  65535
 };
 
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4] = {
+const uint16_t *WebRtcIsacfix_kPitchLagPtrLo[4] = {
   WebRtcIsacfix_kPitchLagCdf1Lo,
   WebRtcIsacfix_kPitchLagCdf2Lo,
   WebRtcIsacfix_kPitchLagCdf3Lo,
@@ -61,32 +61,32 @@
 };
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1] = {
+const uint16_t WebRtcIsacfix_kPitchLagSizeLo[1] = {
   128
 };
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4] = {
+const int16_t WebRtcIsacfix_kLowerLimitLo[4] = {
   -140, -9,  0, -4
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4] = {
+const int16_t WebRtcIsacfix_kUpperLimitLo[4] = {
   -20,  9,  0,  4
 };
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3] = {
+const uint16_t WebRtcIsacfix_kInitIndLo[3] = {
   10,  1,  5
 };
 
 /* mean values of pitch filter lags in Q10 */
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19] = {
+const int16_t WebRtcIsacfix_kMeanLag2Lo[19] = {
   -17627, -16207, -14409, -12319, -10253, -8200, -6054, -3986, -1948, -19,
   1937, 3974, 6064, 8155, 10229, 12270, 14296, 16127, 17520
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9] = {
+const int16_t WebRtcIsacfix_kMeanLag4Lo[9] = {
   -7949, -6063, -4036, -1941, 38, 1977, 4060, 6059
 };
 
@@ -95,7 +95,7 @@
 /* tables for use with medium pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf1Mid[255] = {
   0,  28,  61,  88,  121,  149,  233,  331,  475,  559,
   624,  661,  689,  712,  745,  791,  815,  843,  866,  922,
   959,  1024,  1061,  1117,  1178,  1238,  1280,  1350,  1453,  1513,
@@ -124,23 +124,23 @@
   65535,  65535,  65535,  65535,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf2Mid[36] = {
   0,  71,  335,  581,  836,  1039,  1323,  1795,  2258,  2608,
   3005,  3591,  4243,  5344,  7163,  10583,  16848,  28078,  49448,  57007,
   60357,  61850,  62837,  63437,  63872,  64188,  64377,  64614,  64774,  64949,
   65039,  65115,  65223,  65360,  65474,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf3Mid[2] = {
   0,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf4Mid[20] = {
   0,  28,  246,  459,  667,  1045,  1523,  2337,  4337,  11347,
   44231,  56709,  60781,  62243,  63161,  63969,  64608,  65062,  65502,  65535
 };
 
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4] = {
+const uint16_t *WebRtcIsacfix_kPitchLagPtrMid[4] = {
   WebRtcIsacfix_kPitchLagCdf1Mid,
   WebRtcIsacfix_kPitchLagCdf2Mid,
   WebRtcIsacfix_kPitchLagCdf3Mid,
@@ -148,27 +148,27 @@
 };
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1] = {
+const uint16_t WebRtcIsacfix_kPitchLagSizeMid[1] = {
   256
 };
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4] = {
+const int16_t WebRtcIsacfix_kLowerLimitMid[4] = {
   -280, -17,  0, -9
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4] = {
+const int16_t WebRtcIsacfix_kUpperLimitMid[4] = {
   -40,  17,  0,  9
 };
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3] = {
+const uint16_t WebRtcIsacfix_kInitIndMid[3] = {
   18,  1,  10
 };
 
 /* mean values of pitch filter lags in Q10 */
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35] = {
+const int16_t WebRtcIsacfix_kMeanLag2Mid[35] = {
   -17297, -16250, -15416, -14343, -13341, -12363, -11270,
   -10355, -9122, -8217, -7172, -6083, -5102, -4004, -3060,
   -1982, -952, -18, 935, 1976, 3040, 4032,
@@ -177,7 +177,7 @@
 };
 
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19] = {
+const int16_t WebRtcIsacfix_kMeanLag4Mid[19] = {
   -8811, -8081, -7203, -6003, -5057, -4025, -2983, -1964,
   -891, 29, 921, 1920, 2988, 4064, 5187, 6079, 7173, 8074, 8849
 };
@@ -186,7 +186,7 @@
 /* tables for use with large pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf1Hi[511] = {
   0,  7,  18,  33,  69,  105,  156,  228,  315,  612,
   680,  691,  709,  724,  735,  738,  742,  746,  749,  753,
   756,  760,  764,  774,  782,  785,  789,  796,  800,  803,
@@ -241,7 +241,7 @@
   65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf2Hi[68] = {
   0,  7,  11,  22,  37,  52,  56,  59,  81,  85,
   89,  96,  115,  130,  137,  152,  170,  181,  193,  200,
   207,  233,  237,  259,  289,  318,  363,  433,  592,  992,
@@ -251,18 +251,18 @@
   65483,  65491,  65498,  65505,  65516,  65520,  65528,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf3Hi[2] = {
   0,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35] = {
+const uint16_t WebRtcIsacfix_kPitchLagCdf4Hi[35] = {
   0,  7,  19,  30,  41,  48,  63,  74,  82,  96,
   122,  152,  215,  330,  701,  2611,  10931,  48106,  61177,  64341,
   65112,  65238,  65309,  65338,  65364,  65379,  65401,  65427,  65453,
   65465,  65476,  65490,  65509,  65528,  65535
 };
 
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4] = {
+const uint16_t *WebRtcIsacfix_kPitchLagPtrHi[4] = {
   WebRtcIsacfix_kPitchLagCdf1Hi,
   WebRtcIsacfix_kPitchLagCdf2Hi,
   WebRtcIsacfix_kPitchLagCdf3Hi,
@@ -270,27 +270,27 @@
 };
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1] = {
+const uint16_t WebRtcIsacfix_kPitchLagSizeHi[1] = {
   512
 };
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4] = {
+const int16_t WebRtcIsacfix_kLowerLimitHi[4] = {
   -552, -34,  0, -16
 };
 
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4] = {
+const int16_t WebRtcIsacfix_kUpperLimitHi[4] = {
   -80,  32,  0,  17
 };
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3] = {
+const uint16_t WebRtcIsacfix_kInitIndHi[3] = {
   34,  1,  18
 };
 
 /* mean values of pitch filter lags */
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67] = {
+const int16_t WebRtcIsacfix_kMeanLag2Hi[67] = {
   -17482, -16896, -16220, -15929, -15329, -14848, -14336, -13807, -13312, -12800, -12218, -11720,
   -11307, -10649, -10396, -9742, -9148, -8668, -8297, -7718, -7155, -6656, -6231, -5600, -5129,
   -4610, -4110, -3521, -3040, -2525, -2016, -1506, -995, -477, -5, 469, 991, 1510, 2025, 2526, 3079,
@@ -299,7 +299,7 @@
 };
 
 
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34] = {
+const int16_t WebRtcIsacfix_kMeanLag4Hi[34] = {
   -8175, -7659, -7205, -6684, -6215, -5651, -5180, -4566, -4087, -3536, -3096,
   -2532, -1990, -1482, -959, -440, 11, 451, 954, 1492, 2020, 2562, 3059,
   3577, 4113, 4618, 5134, 5724, 6060, 6758, 7015, 7716, 8066, 8741
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
index 9517c29..bb8b39a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
@@ -27,77 +27,77 @@
 /* tables for use with small pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Lo[127];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Lo[20];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Lo[2];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Lo[10];
 
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4];
+extern const uint16_t *WebRtcIsacfix_kPitchLagPtrLo[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1];
+extern const uint16_t WebRtcIsacfix_kPitchLagSizeLo[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4];
+extern const int16_t WebRtcIsacfix_kLowerLimitLo[4];
+extern const int16_t WebRtcIsacfix_kUpperLimitLo[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3];
+extern const uint16_t WebRtcIsacfix_kInitIndLo[3];
 
 /* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9];
+extern const int16_t WebRtcIsacfix_kMeanLag2Lo[19];
+extern const int16_t WebRtcIsacfix_kMeanLag4Lo[9];
 
 
 
 /* tables for use with medium pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Mid[255];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Mid[36];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Mid[2];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Mid[20];
 
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4];
+extern const uint16_t *WebRtcIsacfix_kPitchLagPtrMid[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1];
+extern const uint16_t WebRtcIsacfix_kPitchLagSizeMid[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4];
+extern const int16_t WebRtcIsacfix_kLowerLimitMid[4];
+extern const int16_t WebRtcIsacfix_kUpperLimitMid[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3];
+extern const uint16_t WebRtcIsacfix_kInitIndMid[3];
 
 /* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19];
+extern const int16_t WebRtcIsacfix_kMeanLag2Mid[35];
+extern const int16_t WebRtcIsacfix_kMeanLag4Mid[19];
 
 
 /* tables for use with large pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Hi[511];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Hi[68];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Hi[2];
+extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Hi[35];
 
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4];
+extern const uint16_t *WebRtcIsacfix_kPitchLagPtrHi[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1];
+extern const uint16_t WebRtcIsacfix_kPitchLagSizeHi[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4];
+extern const int16_t WebRtcIsacfix_kLowerLimitHi[4];
+extern const int16_t WebRtcIsacfix_kUpperLimitHi[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3];
+extern const uint16_t WebRtcIsacfix_kInitIndHi[3];
 
 /* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34];
+extern const int16_t WebRtcIsacfix_kMeanLag2Hi[67];
+extern const int16_t WebRtcIsacfix_kMeanLag4Hi[34];
 
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/settings.h b/modules/audio_coding/codecs/isac/fix/source/settings.h
index da88ba2..2149480 100644
--- a/modules/audio_coding/codecs/isac/fix/source/settings.h
+++ b/modules/audio_coding/codecs/isac/fix/source/settings.h
@@ -22,11 +22,11 @@
 /* sampling frequency (Hz) */
 #define FS                                      16000
 /* 1.5 times Sampling frequency */
-#define FS_1_HALF        (WebRtc_UWord32) 24000
+#define FS_1_HALF        (uint32_t) 24000
 /* Three times Sampling frequency */
-#define FS3          (WebRtc_UWord32) 48000
+#define FS3          (uint32_t) 48000
 /* Eight times Sampling frequency */
-#define FS8          (WebRtc_UWord32) 128000
+#define FS8          (uint32_t) 128000
 
 /* number of samples per frame (either 480 (30ms) or 960 (60ms)) */
 #define INITIAL_FRAMESAMPLES     960
diff --git a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
index 81b932f..cf2dea7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
+++ b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
@@ -22,84 +22,84 @@
 /********************* AR Coefficient Tables ************************/
 
 /* cdf for quantized reflection coefficient 1 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc1Cdf[12] = {
   0,  2,  4,  129,  7707,  57485,  65495,  65527,  65529,  65531,
   65533,  65535
 };
 
 /* cdf for quantized reflection coefficient 2 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc2Cdf[12] = {
   0,  2,  4,  7,  531,  25298,  64525,  65526,  65529,  65531,
   65533,  65535
 };
 
 /* cdf for quantized reflection coefficient 3 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc3Cdf[12] = {
   0,  2,  4,  6,  620,  22898,  64843,  65527,  65529,  65531,
   65533,  65535
 };
 
 /* cdf for quantized reflection coefficient 4 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc4Cdf[12] = {
   0,  2,  4,  6,  35,  10034,  60733,  65506,  65529,  65531,
   65533,  65535
 };
 
 /* cdf for quantized reflection coefficient 5 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc5Cdf[12] = {
   0,  2,  4,  6,  36,  7567,  56727,  65385,  65529,  65531,
   65533,  65535
 };
 
 /* cdf for quantized reflection coefficient 6 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12] = {
+const uint16_t WebRtcIsacfix_kRc6Cdf[12] = {
   0,  2,  4,  6,  14,  6579,  57360,  65409,  65529,  65531,
   65533,  65535
 };
 
 /* representation levels for quantized reflection coefficient 1 */
-const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11] = {
+const int16_t WebRtcIsacfix_kRc1Levels[11] = {
   -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 2 */
-const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11] = {
+const int16_t WebRtcIsacfix_kRc2Levels[11] = {
   -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 3 */
-const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11] = {
+const int16_t WebRtcIsacfix_kRc3Levels[11] = {
   -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 4 */
-const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11] = {
+const int16_t WebRtcIsacfix_kRc4Levels[11] = {
   -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 5 */
-const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11] = {
+const int16_t WebRtcIsacfix_kRc5Levels[11] = {
   -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 6 */
-const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11] = {
+const int16_t WebRtcIsacfix_kRc6Levels[11] = {
   -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
 };
 
 /* quantization boundary levels for reflection coefficients */
-const WebRtc_Word16 WebRtcIsacfix_kRcBound[12] = {
+const int16_t WebRtcIsacfix_kRcBound[12] = {
   -32768, -31441, -27566, -21458, -13612, -4663,
   4663, 13612, 21458, 27566, 31441, 32767
 };
 
 /* initial index for AR reflection coefficient quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[6] = {
+const uint16_t WebRtcIsacfix_kRcInitInd[6] = {
   5,  5,  5,  5,  5,  5
 };
 
 /* pointers to AR cdf tables */
-const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER] = {
+const uint16_t *WebRtcIsacfix_kRcCdfPtr[AR_ORDER] = {
   WebRtcIsacfix_kRc1Cdf,
   WebRtcIsacfix_kRc2Cdf,
   WebRtcIsacfix_kRc3Cdf,
@@ -109,7 +109,7 @@
 };
 
 /* pointers to AR representation levels tables */
-const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = {
+const int16_t *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = {
   WebRtcIsacfix_kRc1Levels,
   WebRtcIsacfix_kRc2Levels,
   WebRtcIsacfix_kRc3Levels,
@@ -122,30 +122,30 @@
 /******************** GAIN Coefficient Tables ***********************/
 
 /* cdf for Gain coefficient */
-const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19] = {
+const uint16_t WebRtcIsacfix_kGainCdf[19] = {
   0,  2,  4,  6,  8,  10,  12,  14,  16,  1172,
   11119,  29411,  51699,  64445,  65527,  65529,  65531,  65533,  65535
 };
 
 /* representation levels for quantized squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18] = {
+const int32_t WebRtcIsacfix_kGain2Lev[18] = {
   128, 128, 128, 128, 128, 215, 364, 709, 1268,
   1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000
 };
 
 /* quantization boundary levels for squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19] = {
+const int32_t WebRtcIsacfix_kGain2Bound[19] = {
   0, 21, 35, 59, 99, 166, 280, 475, 815, 1414,
   2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF
 };
 
 /* pointers to Gain cdf table */
-const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1] = {
+const uint16_t *WebRtcIsacfix_kGainPtr[1] = {
   WebRtcIsacfix_kGainCdf
 };
 
 /* gain initial index for gain quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1] = {
+const uint16_t WebRtcIsacfix_kGainInitInd[1] = {
   11
 };
 
@@ -153,7 +153,7 @@
 /************************* Cosine Tables ****************************/
 
 /* cosine table */
-const WebRtc_Word16 WebRtcIsacfix_kCos[6][60] = {
+const int16_t WebRtcIsacfix_kCos[6][60] = {
   { 512,   512,   511,   510,   508,   507,   505,   502,   499,   496,
         493,   489,   485,   480,   476,   470,   465,   459,   453,   447,
  440,   433,   426,   418,   410,   402,   394,   385,   376,   367,
diff --git a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
index b506d0e..115509a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
@@ -25,72 +25,72 @@
 
 /********************* AR Coefficient Tables ************************/
 /* cdf for quantized reflection coefficient 1 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc1Cdf[12];
 
 /* cdf for quantized reflection coefficient 2 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc2Cdf[12];
 
 /* cdf for quantized reflection coefficient 3 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc3Cdf[12];
 
 /* cdf for quantized reflection coefficient 4 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc4Cdf[12];
 
 /* cdf for quantized reflection coefficient 5 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc5Cdf[12];
 
 /* cdf for quantized reflection coefficient 6 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12];
+extern const uint16_t WebRtcIsacfix_kRc6Cdf[12];
 
 /* representation levels for quantized reflection coefficient 1 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11];
+extern const int16_t WebRtcIsacfix_kRc1Levels[11];
 
 /* representation levels for quantized reflection coefficient 2 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11];
+extern const int16_t WebRtcIsacfix_kRc2Levels[11];
 
 /* representation levels for quantized reflection coefficient 3 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11];
+extern const int16_t WebRtcIsacfix_kRc3Levels[11];
 
 /* representation levels for quantized reflection coefficient 4 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11];
+extern const int16_t WebRtcIsacfix_kRc4Levels[11];
 
 /* representation levels for quantized reflection coefficient 5 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11];
+extern const int16_t WebRtcIsacfix_kRc5Levels[11];
 
 /* representation levels for quantized reflection coefficient 6 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11];
+extern const int16_t WebRtcIsacfix_kRc6Levels[11];
 
 /* quantization boundary levels for reflection coefficients */
-extern const WebRtc_Word16 WebRtcIsacfix_kRcBound[12];
+extern const int16_t WebRtcIsacfix_kRcBound[12];
 
 /* initial indices for AR reflection coefficient quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[AR_ORDER];
+extern const uint16_t WebRtcIsacfix_kRcInitInd[AR_ORDER];
 
 /* pointers to AR cdf tables */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
+extern const uint16_t *WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
 
 /* pointers to AR representation levels tables */
-extern const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER];
+extern const int16_t *WebRtcIsacfix_kRcLevPtr[AR_ORDER];
 
 
 /******************** GAIN Coefficient Tables ***********************/
 /* cdf for Gain coefficient */
-extern const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19];
+extern const uint16_t WebRtcIsacfix_kGainCdf[19];
 
 /* representation levels for quantized Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18];
+extern const int32_t WebRtcIsacfix_kGain2Lev[18];
 
 /* squared quantization boundary levels for Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19];
+extern const int32_t WebRtcIsacfix_kGain2Bound[19];
 
 /* pointer to Gain cdf table */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1];
+extern const uint16_t *WebRtcIsacfix_kGainPtr[1];
 
 /* Gain initial index for gain quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1];
+extern const uint16_t WebRtcIsacfix_kGainInitInd[1];
 
 /************************* Cosine Tables ****************************/
 /* Cosine table */
-extern const WebRtc_Word16 WebRtcIsacfix_kCos[6][60];
+extern const int16_t WebRtcIsacfix_kCos[6][60];
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/structs.h b/modules/audio_coding/codecs/isac/fix/source/structs.h
index c038a43..4d04356 100644
--- a/modules/audio_coding/codecs/isac/fix/source/structs.h
+++ b/modules/audio_coding/codecs/isac/fix/source/structs.h
@@ -26,11 +26,11 @@
 /* Bitstream struct for decoder */
 typedef struct Bitstreamstruct_dec {
 
-  WebRtc_UWord16  *stream;          /* Pointer to bytestream to decode */
-  WebRtc_UWord32  W_upper;          /* Upper boundary of interval W */
-  WebRtc_UWord32  streamval;
-  WebRtc_UWord16  stream_index;     /* Index to the current position in bytestream */
-  WebRtc_Word16   full;             /* 0 - first byte in memory filled, second empty*/
+  uint16_t  *stream;          /* Pointer to bytestream to decode */
+  uint32_t  W_upper;          /* Upper boundary of interval W */
+  uint32_t  streamval;
+  uint16_t  stream_index;     /* Index to the current position in bytestream */
+  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
   /* 1 - both bytes are empty (we just filled the previous memory */
 
 } Bitstr_dec;
@@ -38,11 +38,11 @@
 /* Bitstream struct for encoder */
 typedef struct Bitstreamstruct_enc {
 
-  WebRtc_UWord16  stream[STREAM_MAXW16_60MS];   /* Vector for adding encoded bytestream */
-  WebRtc_UWord32  W_upper;          /* Upper boundary of interval W */
-  WebRtc_UWord32  streamval;
-  WebRtc_UWord16  stream_index;     /* Index to the current position in bytestream */
-  WebRtc_Word16   full;             /* 0 - first byte in memory filled, second empty*/
+  uint16_t  stream[STREAM_MAXW16_60MS];   /* Vector for adding encoded bytestream */
+  uint32_t  W_upper;          /* Upper boundary of interval W */
+  uint32_t  streamval;
+  uint16_t  stream_index;     /* Index to the current position in bytestream */
+  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
   /* 1 - both bytes are empty (we just filled the previous memory */
 
 } Bitstr_enc;
@@ -50,19 +50,19 @@
 
 typedef struct {
 
-  WebRtc_Word16 DataBufferLoQ0[WINLEN];
-  WebRtc_Word16 DataBufferHiQ0[WINLEN];
+  int16_t DataBufferLoQ0[WINLEN];
+  int16_t DataBufferHiQ0[WINLEN];
 
-  WebRtc_Word32 CorrBufLoQQ[ORDERLO+1];
-  WebRtc_Word32 CorrBufHiQQ[ORDERHI+1];
+  int32_t CorrBufLoQQ[ORDERLO+1];
+  int32_t CorrBufHiQQ[ORDERHI+1];
 
-  WebRtc_Word16 CorrBufLoQdom[ORDERLO+1];
-  WebRtc_Word16 CorrBufHiQdom[ORDERHI+1];
+  int16_t CorrBufLoQdom[ORDERLO+1];
+  int16_t CorrBufHiQdom[ORDERHI+1];
 
-  WebRtc_Word32 PreStateLoGQ15[ORDERLO+1];
-  WebRtc_Word32 PreStateHiGQ15[ORDERHI+1];
+  int32_t PreStateLoGQ15[ORDERLO+1];
+  int32_t PreStateHiGQ15[ORDERHI+1];
 
-  WebRtc_UWord32 OldEnergy;
+  uint32_t OldEnergy;
 
 } MaskFiltstr_enc;
 
@@ -70,10 +70,10 @@
 
 typedef struct {
 
-  WebRtc_Word16 PostStateLoGQ0[ORDERLO+1];
-  WebRtc_Word16 PostStateHiGQ0[ORDERHI+1];
+  int16_t PostStateLoGQ0[ORDERLO+1];
+  int16_t PostStateHiGQ0[ORDERHI+1];
 
-  WebRtc_UWord32 OldEnergy;
+  uint32_t OldEnergy;
 
 } MaskFiltstr_dec;
 
@@ -88,13 +88,13 @@
 
   //state vectors for each of the two analysis filters
 
-  WebRtc_Word32 INSTAT1_fix[2*(QORDER-1)];
-  WebRtc_Word32 INSTAT2_fix[2*(QORDER-1)];
-  WebRtc_Word16 INLABUF1_fix[QLOOKAHEAD];
-  WebRtc_Word16 INLABUF2_fix[QLOOKAHEAD];
+  int32_t INSTAT1_fix[2*(QORDER-1)];
+  int32_t INSTAT2_fix[2*(QORDER-1)];
+  int16_t INLABUF1_fix[QLOOKAHEAD];
+  int16_t INLABUF2_fix[QLOOKAHEAD];
 
   /* High pass filter */
-  WebRtc_Word32 HPstates_fix[HPORDER];
+  int32_t HPstates_fix[HPORDER];
 
 } PreFiltBankstr;
 
@@ -102,13 +102,13 @@
 typedef struct {
 
   //state vectors for each of the two analysis filters
-  WebRtc_Word32 STATE_0_LOWER_fix[2*POSTQORDER];
-  WebRtc_Word32 STATE_0_UPPER_fix[2*POSTQORDER];
+  int32_t STATE_0_LOWER_fix[2*POSTQORDER];
+  int32_t STATE_0_UPPER_fix[2*POSTQORDER];
 
   /* High pass filter */
 
-  WebRtc_Word32 HPstates1_fix[HPORDER];
-  WebRtc_Word32 HPstates2_fix[HPORDER];
+  int32_t HPstates1_fix[HPORDER];
+  int32_t HPstates2_fix[HPORDER];
 
 } PostFiltBankstr;
 
@@ -116,14 +116,14 @@
 
 
   /* data buffer for pitch filter */
-  WebRtc_Word16 ubufQQ[PITCH_BUFFSIZE];
+  int16_t ubufQQ[PITCH_BUFFSIZE];
 
   /* low pass state vector */
-  WebRtc_Word16 ystateQQ[PITCH_DAMPORDER];
+  int16_t ystateQQ[PITCH_DAMPORDER];
 
   /* old lag and gain */
-  WebRtc_Word16 oldlagQ7;
-  WebRtc_Word16 oldgainQ12;
+  int16_t oldlagQ7;
+  int16_t oldgainQ12;
 
 } PitchFiltstr;
 
@@ -132,9 +132,9 @@
 typedef struct {
 
   //for inital estimator
-  WebRtc_Word16   dec_buffer16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2];
-  WebRtc_Word32   decimator_state32[2*ALLPASSSECTIONS+1];
-  WebRtc_Word16   inbuf[QLOOKAHEAD];
+  int16_t   dec_buffer16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2];
+  int32_t   decimator_state32[2*ALLPASSSECTIONS+1];
+  int16_t   inbuf[QLOOKAHEAD];
 
   PitchFiltstr  PFstr_wght;
   PitchFiltstr  PFstr;
@@ -147,41 +147,41 @@
   /* Parameters used in PLC to avoid re-computation       */
 
   /* --- residual signals --- */
-  WebRtc_Word16 prevPitchInvIn[FRAMESAMPLES/2];
-  WebRtc_Word16 prevPitchInvOut[PITCH_MAX_LAG + 10];            // [FRAMESAMPLES/2]; save 90
-  WebRtc_Word32 prevHP[PITCH_MAX_LAG + 10];                     // [FRAMESAMPLES/2]; save 90
+  int16_t prevPitchInvIn[FRAMESAMPLES/2];
+  int16_t prevPitchInvOut[PITCH_MAX_LAG + 10];            // [FRAMESAMPLES/2]; save 90
+  int32_t prevHP[PITCH_MAX_LAG + 10];                     // [FRAMESAMPLES/2]; save 90
 
 
-  WebRtc_Word16 decayCoeffPriodic; /* how much to supress a sample */
-  WebRtc_Word16 decayCoeffNoise;
-  WebRtc_Word16 used;       /* if PLC is used */
+  int16_t decayCoeffPriodic; /* how much to supress a sample */
+  int16_t decayCoeffNoise;
+  int16_t used;       /* if PLC is used */
 
 
-  WebRtc_Word16 *lastPitchLP;                                  // [FRAMESAMPLES/2]; saved 240;
+  int16_t *lastPitchLP;                                  // [FRAMESAMPLES/2]; saved 240;
 
 
   /* --- LPC side info --- */
-  WebRtc_Word16 lofilt_coefQ15[ ORDERLO ];
-  WebRtc_Word16 hifilt_coefQ15[ ORDERHI ];
-  WebRtc_Word32 gain_lo_hiQ17[2];
+  int16_t lofilt_coefQ15[ ORDERLO ];
+  int16_t hifilt_coefQ15[ ORDERHI ];
+  int32_t gain_lo_hiQ17[2];
 
   /* --- LTP side info --- */
-  WebRtc_Word16 AvgPitchGain_Q12;
-  WebRtc_Word16 lastPitchGain_Q12;
-  WebRtc_Word16 lastPitchLag_Q7;
+  int16_t AvgPitchGain_Q12;
+  int16_t lastPitchGain_Q12;
+  int16_t lastPitchLag_Q7;
 
   /* --- Add-overlap in recovery packet --- */
-  WebRtc_Word16 overlapLP[ RECOVERY_OVERLAP ];                 // [FRAMESAMPLES/2]; saved 160
+  int16_t overlapLP[ RECOVERY_OVERLAP ];                 // [FRAMESAMPLES/2]; saved 160
 
-  WebRtc_Word16 pitchCycles;
-  WebRtc_Word16 A;
-  WebRtc_Word16 B;
-  WebRtc_Word16 pitchIndex;
-  WebRtc_Word16 stretchLag;
-  WebRtc_Word16 *prevPitchLP;                                  // [ FRAMESAMPLES/2 ]; saved 240
-  WebRtc_Word16 seed;
+  int16_t pitchCycles;
+  int16_t A;
+  int16_t B;
+  int16_t pitchIndex;
+  int16_t stretchLag;
+  int16_t *prevPitchLP;                                  // [ FRAMESAMPLES/2 ]; saved 240
+  int16_t seed;
 
-  WebRtc_Word16 std;
+  int16_t std;
 } PLCstr;
 
 
@@ -189,60 +189,60 @@
 /* Have instance of struct together with other iSAC structs */
 typedef struct {
 
-  WebRtc_Word16   prevFrameSizeMs;      /* Previous frame size (in ms) */
-  WebRtc_UWord16  prevRtpNumber;      /* Previous RTP timestamp from received packet */
+  int16_t   prevFrameSizeMs;      /* Previous frame size (in ms) */
+  uint16_t  prevRtpNumber;      /* Previous RTP timestamp from received packet */
   /* (in samples relative beginning)  */
-  WebRtc_UWord32  prevSendTime;   /* Send time for previous packet, from RTP header */
-  WebRtc_UWord32  prevArrivalTime;      /* Arrival time for previous packet (in ms using timeGetTime()) */
-  WebRtc_UWord16  prevRtpRate;          /* rate of previous packet, derived from RTP timestamps (in bits/s) */
-  WebRtc_UWord32  lastUpdate;           /* Time since the last update of the Bottle Neck estimate (in samples) */
-  WebRtc_UWord32  lastReduction;        /* Time sinse the last reduction (in samples) */
-  WebRtc_Word32   countUpdates;         /* How many times the estimate was update in the beginning */
+  uint32_t  prevSendTime;   /* Send time for previous packet, from RTP header */
+  uint32_t  prevArrivalTime;      /* Arrival time for previous packet (in ms using timeGetTime()) */
+  uint16_t  prevRtpRate;          /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+  uint32_t  lastUpdate;           /* Time since the last update of the Bottle Neck estimate (in samples) */
+  uint32_t  lastReduction;        /* Time sinse the last reduction (in samples) */
+  int32_t   countUpdates;         /* How many times the estimate was update in the beginning */
 
   /* The estimated bottle neck rate from there to here (in bits/s)                */
-  WebRtc_UWord32  recBw;
-  WebRtc_UWord32  recBwInv;
-  WebRtc_UWord32  recBwAvg;
-  WebRtc_UWord32  recBwAvgQ;
+  uint32_t  recBw;
+  uint32_t  recBwInv;
+  uint32_t  recBwAvg;
+  uint32_t  recBwAvgQ;
 
-  WebRtc_UWord32  minBwInv;
-  WebRtc_UWord32  maxBwInv;
+  uint32_t  minBwInv;
+  uint32_t  maxBwInv;
 
   /* The estimated mean absolute jitter value, as seen on this side (in ms)       */
-  WebRtc_Word32   recJitter;
-  WebRtc_Word32   recJitterShortTerm;
-  WebRtc_Word32   recJitterShortTermAbs;
-  WebRtc_Word32   recMaxDelay;
-  WebRtc_Word32   recMaxDelayAvgQ;
+  int32_t   recJitter;
+  int32_t   recJitterShortTerm;
+  int32_t   recJitterShortTermAbs;
+  int32_t   recMaxDelay;
+  int32_t   recMaxDelayAvgQ;
 
 
-  WebRtc_Word16   recHeaderRate;         /* (assumed) bitrate for headers (bps) */
+  int16_t   recHeaderRate;         /* (assumed) bitrate for headers (bps) */
 
-  WebRtc_UWord32  sendBwAvg;           /* The estimated bottle neck rate from here to there (in bits/s) */
-  WebRtc_Word32   sendMaxDelayAvg;    /* The estimated mean absolute jitter value, as seen on the other siee (in ms)  */
+  uint32_t  sendBwAvg;           /* The estimated bottle neck rate from here to there (in bits/s) */
+  int32_t   sendMaxDelayAvg;    /* The estimated mean absolute jitter value, as seen on the other siee (in ms)  */
 
 
-  WebRtc_Word16   countRecPkts;          /* number of packets received since last update */
-  WebRtc_Word16   highSpeedRec;        /* flag for marking that a high speed network has been detected downstream */
+  int16_t   countRecPkts;          /* number of packets received since last update */
+  int16_t   highSpeedRec;        /* flag for marking that a high speed network has been detected downstream */
 
   /* number of consecutive pkts sent during which the bwe estimate has
      remained at a value greater than the downstream threshold for determining highspeed network */
-  WebRtc_Word16   countHighSpeedRec;
+  int16_t   countHighSpeedRec;
 
   /* flag indicating bwe should not adjust down immediately for very late pckts */
-  WebRtc_Word16   inWaitPeriod;
+  int16_t   inWaitPeriod;
 
   /* variable holding the time of the start of a window of time when
      bwe should not adjust down immediately for very late pckts */
-  WebRtc_UWord32  startWaitPeriod;
+  uint32_t  startWaitPeriod;
 
   /* number of consecutive pkts sent during which the bwe estimate has
      remained at a value greater than the upstream threshold for determining highspeed network */
-  WebRtc_Word16   countHighSpeedSent;
+  int16_t   countHighSpeedSent;
 
   /* flag indicated the desired number of packets over threshold rate have been sent and
      bwe will assume the connection is over broadband network */
-  WebRtc_Word16   highSpeedSend;
+  int16_t   highSpeedSend;
 
 
 
@@ -253,15 +253,15 @@
 typedef struct {
 
   /* boolean, flags if previous packet exceeded B.N. */
-  WebRtc_Word16    PrevExceed;
+  int16_t    PrevExceed;
   /* ms */
-  WebRtc_Word16    ExceedAgo;
+  int16_t    ExceedAgo;
   /* packets left to send in current burst */
-  WebRtc_Word16    BurstCounter;
+  int16_t    BurstCounter;
   /* packets */
-  WebRtc_Word16    InitCounter;
+  int16_t    InitCounter;
   /* ms remaining in buffer when next packet will be sent */
-  WebRtc_Word16    StillBuffered;
+  int16_t    StillBuffered;
 
 } RateModel;
 
@@ -276,24 +276,24 @@
   int     startIdx;
 
   /* Frame length in samples */
-  WebRtc_Word16         framelength;
+  int16_t         framelength;
 
   /* Pitch Gain */
-  WebRtc_Word16   pitchGain_index[2];
+  int16_t   pitchGain_index[2];
 
   /* Pitch Lag */
-  WebRtc_Word32   meanGain[2];
-  WebRtc_Word16   pitchIndex[PITCH_SUBFRAMES*2];
+  int32_t   meanGain[2];
+  int16_t   pitchIndex[PITCH_SUBFRAMES*2];
 
   /* LPC */
-  WebRtc_Word32         LPCcoeffs_g[12*2]; /* KLT_ORDER_GAIN = 12 */
-  WebRtc_Word16   LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
-  WebRtc_Word16   LPCindex_g[12*2];  /* KLT_ORDER_GAIN = 12 */
+  int32_t         LPCcoeffs_g[12*2]; /* KLT_ORDER_GAIN = 12 */
+  int16_t   LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
+  int16_t   LPCindex_g[12*2];  /* KLT_ORDER_GAIN = 12 */
 
   /* Encode Spec */
-  WebRtc_Word16   fre[FRAMESAMPLES];
-  WebRtc_Word16   fim[FRAMESAMPLES];
-  WebRtc_Word16   AvgPitchGain[2];
+  int16_t   fre[FRAMESAMPLES];
+  int16_t   fim[FRAMESAMPLES];
+  int16_t   AvgPitchGain[2];
 
   /* Used in adaptive mode only */
   int     minBytes;
@@ -309,29 +309,29 @@
   PitchAnalysisStruct pitchanalysisstr_obj;
   RateModel           rate_data_obj;
 
-  WebRtc_Word16         buffer_index;
-  WebRtc_Word16         current_framesamples;
+  int16_t         buffer_index;
+  int16_t         current_framesamples;
 
-  WebRtc_Word16      data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES
+  int16_t      data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES
 
-  WebRtc_Word16         frame_nb;
-  WebRtc_Word16         BottleNeck;
-  WebRtc_Word16         MaxDelay;
-  WebRtc_Word16         new_framelength;
-  WebRtc_Word16         s2nr;
-  WebRtc_UWord16        MaxBits;
+  int16_t         frame_nb;
+  int16_t         BottleNeck;
+  int16_t         MaxDelay;
+  int16_t         new_framelength;
+  int16_t         s2nr;
+  uint16_t        MaxBits;
 
-  WebRtc_Word16         bitstr_seed;
+  int16_t         bitstr_seed;
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
   PostFiltBankstr     interpolatorstr_obj;
 #endif
 
   ISAC_SaveEncData_t *SaveEnc_ptr;
-  WebRtc_Word16         payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
-  WebRtc_Word16         payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
-  WebRtc_Word16         maxPayloadBytes;     /* Maximum allowed number of bits for both 30 and 60 msec packet */
-  WebRtc_Word16         maxRateInBytes;      /* Maximum allowed rate in bytes per 30 msec packet */
-  WebRtc_Word16         enforceFrameSize;    /* If set iSAC will never change packet size */
+  int16_t         payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t         payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t         maxPayloadBytes;     /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  int16_t         maxRateInBytes;      /* Maximum allowed rate in bytes per 30 msec packet */
+  int16_t         enforceFrameSize;    /* If set iSAC will never change packet size */
 
 } ISACFIX_EncInst_t;
 
@@ -357,23 +357,23 @@
   ISACFIX_EncInst_t ISACenc_obj;
   ISACFIX_DecInst_t ISACdec_obj;
   BwEstimatorstr     bwestimator_obj;
-  WebRtc_Word16         CodingMode;       /* 0 = adaptive; 1 = instantaneous */
-  WebRtc_Word16   errorcode;
-  WebRtc_Word16   initflag;  /* 0 = nothing initiated; 1 = encoder or decoder */
+  int16_t         CodingMode;       /* 0 = adaptive; 1 = instantaneous */
+  int16_t   errorcode;
+  int16_t   initflag;  /* 0 = nothing initiated; 1 = encoder or decoder */
   /* not initiated; 2 = all initiated */
 } ISACFIX_SubStruct;
 
 
 typedef struct {
-  WebRtc_Word32   lpcGains[12];     /* 6 lower-band & 6 upper-band we may need to double it for 60*/
+  int32_t   lpcGains[12];     /* 6 lower-band & 6 upper-band we may need to double it for 60*/
   /* */
-  WebRtc_UWord32  W_upper;          /* Upper boundary of interval W */
-  WebRtc_UWord32  streamval;
-  WebRtc_UWord16  stream_index;     /* Index to the current position in bytestream */
-  WebRtc_Word16   full;             /* 0 - first byte in memory filled, second empty*/
+  uint32_t  W_upper;          /* Upper boundary of interval W */
+  uint32_t  streamval;
+  uint16_t  stream_index;     /* Index to the current position in bytestream */
+  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
   /* 1 - both bytes are empty (we just filled the previous memory */
-  WebRtc_UWord16  beforeLastWord;
-  WebRtc_UWord16  lastWord;
+  uint16_t  beforeLastWord;
+  uint16_t  lastWord;
 } transcode_obj;
 
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform.c b/modules/audio_coding/codecs/isac/fix/source/transform.c
index c7a3e80..67e513c 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform.c
+++ b/modules/audio_coding/codecs/isac/fix/source/transform.c
@@ -22,14 +22,14 @@
 #if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON)
 /* Tables are defined in ARM assembly files. */
 /* Cosine table 1 in Q14 */
-extern const WebRtc_Word16 WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2];
+extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2];
 /* Sine table 1 in Q14 */
-extern const WebRtc_Word16 WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2];
+extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2];
 /* Sine table 2 in Q14 */
-extern const WebRtc_Word16 WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4];
+extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4];
 #else
 /* Cosine table 1 in Q14 */
-static const WebRtc_Word16 WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = {
+static const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = {
   16384,  16383,  16378,  16371,  16362,  16349,  16333,  16315,  16294,  16270,
   16244,  16214,  16182,  16147,  16110,  16069,  16026,  15980,  15931,  15880,
   15826,  15769,  15709,  15647,  15582,  15515,  15444,  15371,  15296,  15218,
@@ -58,7 +58,7 @@
 };
 
 /* Sine table 1 in Q14 */
-static const WebRtc_Word16 WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2] = {
+static const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2] = {
   0,   214,   429,   643,   857,  1072,  1285,  1499,  1713,  1926,
   2139,  2351,  2563,  2775,  2986,  3196,  3406,  3616,  3825,  4033,
   4240,  4447,  4653,  4859,  5063,  5266,  5469,  5671,  5872,  6071,
@@ -87,7 +87,7 @@
 
 
 /* Sine table 2 in Q14 */
-static const WebRtc_Word16 WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = {
+static const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = {
   16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257,
   16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853,
   15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178,
@@ -103,18 +103,18 @@
 };
 #endif  // WEBRTC_DETECT_ARM_NEON || WEBRTC_ARCH_ARM_NEON
 
-void WebRtcIsacfix_Time2SpecC(WebRtc_Word16 *inre1Q9,
-                             WebRtc_Word16 *inre2Q9,
-                             WebRtc_Word16 *outreQ7,
-                             WebRtc_Word16 *outimQ7)
+void WebRtcIsacfix_Time2SpecC(int16_t *inre1Q9,
+                              int16_t *inre2Q9,
+                              int16_t *outreQ7,
+                              int16_t *outimQ7)
 {
 
   int k;
-  WebRtc_Word32 tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2];
-  WebRtc_Word16 tmp1rQ14, tmp1iQ14;
-  WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
-  WebRtc_Word32 v1Q16, v2Q16;
-  WebRtc_Word16 factQ19, sh;
+  int32_t tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2];
+  int16_t tmp1rQ14, tmp1iQ14;
+  int32_t xrQ16, xiQ16, yrQ16, yiQ16;
+  int32_t v1Q16, v2Q16;
+  int16_t factQ19, sh;
 
   /* Multiply with complex exponentials and combine into one complex vector */
   factQ19 = 16921; // 0.5/sqrt(240) in Q19 is round(.5/sqrt(240)*(2^19)) = 16921
@@ -141,14 +141,14 @@
   //"Fastest" vectors
   if (sh>=0) {
     for (k=0; k<FRAMESAMPLES/2; k++) {
-      inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpreQ16[k], sh); //Q(16+sh)
-      inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpimQ16[k], sh); //Q(16+sh)
+      inre1Q9[k] = (int16_t) WEBRTC_SPL_LSHIFT_W32(tmpreQ16[k], sh); //Q(16+sh)
+      inre2Q9[k] = (int16_t) WEBRTC_SPL_LSHIFT_W32(tmpimQ16[k], sh); //Q(16+sh)
     }
   } else {
-    WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
+    int32_t round = WEBRTC_SPL_LSHIFT_W32((int32_t)1, -sh-1);
     for (k=0; k<FRAMESAMPLES/2; k++) {
-      inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpreQ16[k]+round, -sh); //Q(16+sh)
-      inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpimQ16[k]+round, -sh); //Q(16+sh)
+      inre1Q9[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmpreQ16[k]+round, -sh); //Q(16+sh)
+      inre2Q9[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmpimQ16[k]+round, -sh); //Q(16+sh)
     }
   }
 
@@ -158,13 +158,13 @@
   //"Fastest" vectors
   if (sh>=0) {
     for (k=0; k<FRAMESAMPLES/2; k++) {
-      tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre1Q9[k], sh); //Q(16+sh) -> Q16
-      tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre2Q9[k], sh); //Q(16+sh) -> Q16
+      tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32((int32_t)inre1Q9[k], sh); //Q(16+sh) -> Q16
+      tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32((int32_t)inre2Q9[k], sh); //Q(16+sh) -> Q16
     }
   } else {
     for (k=0; k<FRAMESAMPLES/2; k++) {
-      tmpreQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre1Q9[k], -sh); //Q(16+sh) -> Q16
-      tmpimQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre2Q9[k], -sh); //Q(16+sh) -> Q16
+      tmpreQ16[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)inre1Q9[k], -sh); //Q(16+sh) -> Q16
+      tmpimQ16[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)inre2Q9[k], -sh); //Q(16+sh) -> Q16
     }
   }
 
@@ -179,36 +179,36 @@
     tmp1iQ14 = WebRtcIsacfix_kSinTab2[k];
     v1Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xiQ16);
     v2Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xiQ16);
-    outreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v1Q16, 9);
-    outimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v2Q16, 9);
+    outreQ7[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(v1Q16, 9);
+    outimQ7[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(v2Q16, 9);
     v1Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yiQ16);
     v2Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yiQ16);
-    outreQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v1Q16, 9); //CalcLrIntQ(v1Q16, 9);
-    outimQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v2Q16, 9); //CalcLrIntQ(v2Q16, 9);
+    outreQ7[FRAMESAMPLES/2 - 1 - k] = (int16_t)WEBRTC_SPL_RSHIFT_W32(v1Q16, 9); //CalcLrIntQ(v1Q16, 9);
+    outimQ7[FRAMESAMPLES/2 - 1 - k] = (int16_t)WEBRTC_SPL_RSHIFT_W32(v2Q16, 9); //CalcLrIntQ(v2Q16, 9);
 
   }
 }
 
 
-void WebRtcIsacfix_Spec2TimeC(WebRtc_Word16 *inreQ7, WebRtc_Word16 *inimQ7, WebRtc_Word32 *outre1Q16, WebRtc_Word32 *outre2Q16)
+void WebRtcIsacfix_Spec2TimeC(int16_t *inreQ7, int16_t *inimQ7, int32_t *outre1Q16, int32_t *outre2Q16)
 {
 
   int k;
-  WebRtc_Word16 tmp1rQ14, tmp1iQ14;
-  WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
-  WebRtc_Word32 tmpInRe, tmpInIm, tmpInRe2, tmpInIm2;
-  WebRtc_Word16 factQ11;
-  WebRtc_Word16 sh;
+  int16_t tmp1rQ14, tmp1iQ14;
+  int32_t xrQ16, xiQ16, yrQ16, yiQ16;
+  int32_t tmpInRe, tmpInIm, tmpInRe2, tmpInIm2;
+  int16_t factQ11;
+  int16_t sh;
 
   for (k = 0; k < FRAMESAMPLES/4; k++) {
     /* Move zero in time to beginning of frames */
     tmp1rQ14 = -WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 - 1 - k];
     tmp1iQ14 = WebRtcIsacfix_kSinTab2[k];
 
-    tmpInRe = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[k], 9);  // Q7 -> Q16
-    tmpInIm = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[k], 9);  // Q7 -> Q16
-    tmpInRe2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[FRAMESAMPLES/2 - 1 - k], 9);  // Q7 -> Q16
-    tmpInIm2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[FRAMESAMPLES/2 - 1 - k], 9);  // Q7 -> Q16
+    tmpInRe = WEBRTC_SPL_LSHIFT_W32((int32_t) inreQ7[k], 9);  // Q7 -> Q16
+    tmpInIm = WEBRTC_SPL_LSHIFT_W32((int32_t) inimQ7[k], 9);  // Q7 -> Q16
+    tmpInRe2 = WEBRTC_SPL_LSHIFT_W32((int32_t) inreQ7[FRAMESAMPLES/2 - 1 - k], 9);  // Q7 -> Q16
+    tmpInIm2 = WEBRTC_SPL_LSHIFT_W32((int32_t) inimQ7[FRAMESAMPLES/2 - 1 - k], 9);  // Q7 -> Q16
 
     xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm);
     xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe);
@@ -236,14 +236,14 @@
   //"Fastest" vectors
   if (sh>=0) {
     for (k=0; k<240; k++) {
-      inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre1Q16[k], sh); //Q(16+sh)
-      inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre2Q16[k], sh); //Q(16+sh)
+      inreQ7[k] = (int16_t) WEBRTC_SPL_LSHIFT_W32(outre1Q16[k], sh); //Q(16+sh)
+      inimQ7[k] = (int16_t) WEBRTC_SPL_LSHIFT_W32(outre2Q16[k], sh); //Q(16+sh)
     }
   } else {
-    WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
+    int32_t round = WEBRTC_SPL_LSHIFT_W32((int32_t)1, -sh-1);
     for (k=0; k<240; k++) {
-      inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre1Q16[k]+round, -sh); //Q(16+sh)
-      inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre2Q16[k]+round, -sh); //Q(16+sh)
+      inreQ7[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(outre1Q16[k]+round, -sh); //Q(16+sh)
+      inimQ7[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(outre2Q16[k]+round, -sh); //Q(16+sh)
     }
   }
 
@@ -252,13 +252,13 @@
   //"Fastest" vectors
   if (sh>=0) {
     for (k=0; k<240; k++) {
-      outre1Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inreQ7[k], sh); //Q(16+sh) -> Q16
-      outre2Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inimQ7[k], sh); //Q(16+sh) -> Q16
+      outre1Q16[k] = WEBRTC_SPL_RSHIFT_W32((int32_t)inreQ7[k], sh); //Q(16+sh) -> Q16
+      outre2Q16[k] = WEBRTC_SPL_RSHIFT_W32((int32_t)inimQ7[k], sh); //Q(16+sh) -> Q16
     }
   } else {
     for (k=0; k<240; k++) {
-      outre1Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inreQ7[k], -sh); //Q(16+sh) -> Q16
-      outre2Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inimQ7[k], -sh); //Q(16+sh) -> Q16
+      outre1Q16[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)inreQ7[k], -sh); //Q(16+sh) -> Q16
+      outre2Q16[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)inimQ7[k], -sh); //Q(16+sh) -> Q16
     }
   }
 
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform.h b/modules/audio_coding/codecs/isac/fix/source/transform.h
index d9bd462..564385e 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform.h
+++ b/modules/audio_coding/codecs/isac/fix/source/transform.h
@@ -19,16 +19,16 @@
 #include "webrtc/typedefs.h"
 
 /* Cosine table 1 in Q14 */
-extern const WebRtc_Word16 kCosTab1[FRAMESAMPLES/2];
+extern const int16_t kCosTab1[FRAMESAMPLES/2];
 
 /* Sine table 1 in Q14 */
-extern const WebRtc_Word16 kSinTab1[FRAMESAMPLES/2];
+extern const int16_t kSinTab1[FRAMESAMPLES/2];
 
 /* Cosine table 2 in Q14 */
-extern const WebRtc_Word16 kCosTab2[FRAMESAMPLES/4];
+extern const int16_t kCosTab2[FRAMESAMPLES/4];
 
 /* Sine table 2 in Q14 */
-extern const WebRtc_Word16 kSinTab2[FRAMESAMPLES/4];
+extern const int16_t kSinTab2[FRAMESAMPLES/4];
 
 #ifdef __cplusplus
 } /* extern "C" */
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_neon.S b/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
index 135f130..46682ac 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
@@ -20,10 +20,10 @@
 GLOBAL_LABEL WebRtcIsacfix_kCosTab1
 GLOBAL_LABEL WebRtcIsacfix_kSinTab2
 
-@ void WebRtcIsacfix_Time2SpecNeon(WebRtc_Word16* inre1Q9,
-@                                  WebRtc_Word16* inre2Q9,
-@                                  WebRtc_Word16* outreQ7,
-@                                  WebRtc_Word16* outimQ7);
+@ void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9,
+@                                  int16_t* inre2Q9,
+@                                  int16_t* outreQ7,
+@                                  int16_t* outimQ7);
 
 DEFINE_FUNCTION WebRtcIsacfix_Time2SpecNeon
 .align  2
@@ -317,10 +317,10 @@
   .short 3406, 3196, 2986, 2775, 2563, 2351, 2139, 1926
   .short 1713, 1499, 1285, 1072, 857, 643, 429, 214
 
-@ void WebRtcIsacfix_Spec2TimeNeon(WebRtc_Word16 *inreQ7,
-@                                  WebRtc_Word16 *inimQ7,
-@                                  WebRtc_Word32 *outre1Q16,
-@                                  WebRtc_Word32 *outre2Q16);
+@ void WebRtcIsacfix_Spec2TimeNeon(int16_t *inreQ7,
+@                                  int16_t *inimQ7,
+@                                  int32_t *outre1Q16,
+@                                  int32_t *outre2Q16);
 
 DEFINE_FUNCTION WebRtcIsacfix_Spec2TimeNeon
 .align  2
diff --git a/modules/audio_coding/codecs/isac/fix/test/Isac_test.cc b/modules/audio_coding/codecs/isac/fix/test/Isac_test.cc
index 2791db4..61d79e4 100644
--- a/modules/audio_coding/codecs/isac/fix/test/Isac_test.cc
+++ b/modules/audio_coding/codecs/isac/fix/test/Isac_test.cc
@@ -25,9 +25,9 @@
 
 
 typedef struct {
-	WebRtc_UWord32 arrival_time;            /* samples */
-	WebRtc_UWord32 sample_count;            /* samples */
-	WebRtc_UWord16 rtp_number;
+	uint32_t arrival_time;            /* samples */
+	uint32_t sample_count;            /* samples */
+	uint16_t rtp_number;
 } BottleNeckModel;
 
 void get_arrival_time(int current_framesamples,   /* samples */
@@ -60,18 +60,18 @@
 
 	/* Parameters */
 	FILE *pInFile, *pOutFile, *pChcFile; 
-	WebRtc_Word8 inFile[40];
-	WebRtc_Word8 outFile[40];
-	WebRtc_Word8 chcFile[40];
-	WebRtc_Word8 codec[10];
-	WebRtc_Word16 bitrt, spType, size;
-	WebRtc_UWord16 frameLen;
-	WebRtc_Word16 sigOut[1000], sigIn[1000]; 
-	WebRtc_UWord16 bitStream[500]; /* double to 32 kbps for 60 ms */
+	int8_t inFile[40];
+	int8_t outFile[40];
+	int8_t chcFile[40];
+	int8_t codec[10];
+	int16_t bitrt, spType, size;
+	uint16_t frameLen;
+	int16_t sigOut[1000], sigIn[1000]; 
+	uint16_t bitStream[500]; /* double to 32 kbps for 60 ms */
 
-	WebRtc_Word16 chc, ok;
+	int16_t chc, ok;
 	int noOfCalls, cdlen;
-	WebRtc_Word16 noOfLostFrames;
+	int16_t noOfLostFrames;
 	int err, errtype;
 
 	BottleNeckModel       BN_data;
@@ -170,12 +170,12 @@
 				exit(EXIT_FAILURE);
 			}
 		/* loop over frame */
-		while (fread(sigIn,sizeof(WebRtc_Word16),frameLen,pInFile) == frameLen) {
+		while (fread(sigIn,sizeof(int16_t),frameLen,pInFile) == frameLen) {
 			
 			noOfCalls=0;
 			cdlen=0;
 			while (cdlen<=0) {
-				cdlen=WebRtcIsacfix_Encode(ISACfix_inst,&sigIn[noOfCalls*160],(WebRtc_Word16*)bitStream);
+				cdlen=WebRtcIsacfix_Encode(ISACfix_inst,&sigIn[noOfCalls*160],(int16_t*)bitStream);
 				if(cdlen==-1){
 					errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
 					printf("\n\nError in encoder: %d.\n\n", errtype);
@@ -186,7 +186,7 @@
 	
 	
 			if(_stricmp("none", chcFile)){
-				if (fread(&chc,sizeof(WebRtc_Word16),1,pChcFile)!=1) /* packet may be lost */
+				if (fread(&chc,sizeof(int16_t),1,pChcFile)!=1) /* packet may be lost */
 					break;
 			} else {
 				chc = 1; /* packets never lost */
diff --git a/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index ee70eb0..c3c6f13 100644
--- a/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -27,11 +27,11 @@
 #define FS           16000 /* sampling frequency (Hz) */
 
 /* Function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
+int readframe(int16_t *data, FILE *inp, int length) {
 
   short k, rlen, status = 0;
 
-  rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
+  rlen = fread(data, sizeof(int16_t), length, inp);
   if (rlen < length) {
     for (k = rlen; k < length; k++)
       data[k] = 0;
@@ -43,10 +43,10 @@
 
 /* Struct for bottleneck model */
 typedef struct {
-  WebRtc_UWord32 send_time;            /* samples */
-  WebRtc_UWord32 arrival_time;         /* samples */
-  WebRtc_UWord32 sample_count;         /* samples */
-  WebRtc_UWord16 rtp_number;
+  uint32_t send_time;            /* samples */
+  uint32_t arrival_time;         /* samples */
+  uint32_t sample_count;         /* samples */
+  uint16_t rtp_number;
 } BottleNeckModel;
 
 void get_arrival_time(int current_framesamples,   /* samples */
@@ -99,25 +99,25 @@
   int endfile;
 
   int i, errtype, h = 0, k, packetLossPercent = 0;
-  WebRtc_Word16 CodingMode;
-  WebRtc_Word16 bottleneck;
-  WebRtc_Word16 framesize = 30;           /* ms */
+  int16_t CodingMode;
+  int16_t bottleneck;
+  int16_t framesize = 30;           /* ms */
   int cur_framesmpls, err = 0, lostPackets = 0;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
-  WebRtc_Word16 stream_len = 0;
-  WebRtc_Word16 framecnt, declen = 0;
-  WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
-  WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
-  WebRtc_UWord16 streamdata[500];
-  WebRtc_Word16 speechType[1];
-  WebRtc_Word16 prevFrameSize = 1;
-  WebRtc_Word16 rateBPS = 0;
-  WebRtc_Word16 fixedFL = 0;
-  WebRtc_Word16 payloadSize = 0;
-  WebRtc_Word32 payloadRate = 0;
+  int16_t stream_len = 0;
+  int16_t framecnt, declen = 0;
+  int16_t shortdata[FRAMESAMPLES_10ms];
+  int16_t decoded[MAX_FRAMESAMPLES];
+  uint16_t streamdata[500];
+  int16_t speechType[1];
+  int16_t prevFrameSize = 1;
+  int16_t rateBPS = 0;
+  int16_t fixedFL = 0;
+  int16_t payloadSize = 0;
+  int32_t payloadRate = 0;
   int setControlBWE = 0;
   int readLoss;
   FILE  *plFile = NULL;
@@ -127,7 +127,7 @@
 
   int totalbits =0;
   int totalsmpls =0;
-  WebRtc_Word16 testNum, testCE;
+  int16_t testNum, testCE;
 
   FILE *fp_gns = NULL;
   int gns = 0;
@@ -135,7 +135,7 @@
   char gns_file[100];
 
   int nbTest = 0;
-  WebRtc_Word16 lostFrame;
+  int16_t lostFrame;
   float scale = (float)0.7;
   /* only one structure used for ISAC encoder */
   ISACFIX_MainStruct *ISAC_main_inst = NULL;
@@ -388,7 +388,7 @@
           exit(0);
         }
       }
-      bottleneck = (WebRtc_Word16)aux_var;
+      bottleneck = (int16_t)aux_var;
       /* Bottleneck is a cosine function
        * Matlab code for writing the bottleneck file:
        * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
@@ -565,18 +565,18 @@
           /* Encode */
           stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
                                             shortdata,
-                                            (WebRtc_Word16*)streamdata);
+                                            (int16_t*)streamdata);
 
           /* If packet is ready, and CE testing, call the different API
              functions from the internal API. */
           if (stream_len>0) {
             if (testCE == 1) {
-              err = WebRtcIsacfix_ReadBwIndex((WebRtc_Word16*)streamdata, &bwe);
+              err = WebRtcIsacfix_ReadBwIndex((int16_t*)streamdata, &bwe);
               stream_len = WebRtcIsacfix_GetNewBitStream(
                   ISAC_main_inst,
                   bwe,
                   scale,
-                  (WebRtc_Word16*)streamdata);
+                  (int16_t*)streamdata);
             } else if (testCE == 2) {
               /* transcode function not supported */
             } else if (testCE == 3) {
@@ -637,7 +637,7 @@
             exit(0);
           }
         }
-        bottleneck = (WebRtc_Word16)aux_var;
+        bottleneck = (int16_t)aux_var;
         if (CodingMode == 1) {
           WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
         }
@@ -712,7 +712,7 @@
       }
 
       if( readLoss == 1 ) {
-        if( fread( &lostFrame, sizeof(WebRtc_Word16), 1, plFile ) != 1 ) {
+        if( fread( &lostFrame, sizeof(int16_t), 1, plFile ) != 1 ) {
           rewind( plFile );
         }
         lostFrame = !lostFrame;
@@ -740,7 +740,7 @@
         if (nbTest !=2 ) {
           short FL;
           /* Call getFramelen, only used here for function test */
-          err = WebRtcIsacfix_ReadFrameLen((WebRtc_Word16*)streamdata, &FL);
+          err = WebRtcIsacfix_ReadFrameLen((int16_t*)streamdata, &FL);
           declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
                                          decoded, speechType );
           /* Error check */
@@ -768,7 +768,7 @@
       }
 
       /* Write decoded speech frame to file */
-      if (fwrite(decoded, sizeof(WebRtc_Word16),
+      if (fwrite(decoded, sizeof(int16_t),
                  declen, outp) != (size_t)declen) {
         return -1;
       }
diff --git a/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 57c30ca..d5682b2 100644
--- a/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -42,11 +42,11 @@
 
 
 /* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
+int readframe(int16_t *data, FILE *inp, int length) {
 	
 	short k, rlen, status = 0;
 		
-	rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
+	rlen = fread(data, sizeof(int16_t), length, inp);
 	if (rlen < length) {
 		for (k = rlen; k < length; k++)
 			data[k] = 0;
@@ -57,10 +57,10 @@
 }
 
 typedef struct {
-	WebRtc_UWord32 send_time;            /* samples */
-	WebRtc_UWord32 arrival_time;            /* samples */
-	WebRtc_UWord32 sample_count;            /* samples */
-	WebRtc_UWord16 rtp_number;
+	uint32_t send_time;            /* samples */
+	uint32_t arrival_time;            /* samples */
+	uint32_t sample_count;            /* samples */
+	uint16_t rtp_number;
 } BottleNeckModel;
 
 void get_arrival_time(int current_framesamples,   /* samples */
@@ -96,11 +96,11 @@
 
 
 	int i,j,errtype, plc=0;
-	WebRtc_Word16 CodingMode;
-	WebRtc_Word16 bottleneck;
+	int16_t CodingMode;
+	int16_t bottleneck;
 
-	WebRtc_Word16 framesize = 30;           /* ms */
-    //WebRtc_Word16 framesize = 60; /* To invoke cisco complexity case at frame 2252 */
+	int16_t framesize = 30;           /* ms */
+    //int16_t framesize = 60; /* To invoke cisco complexity case at frame 2252 */
 	
 	int cur_framesmpls, err;
 	
@@ -109,15 +109,15 @@
 	double runtime;
 	double length_file;
 	
-	WebRtc_Word16 stream_len = 0;
-	WebRtc_Word16 declen;
+	int16_t stream_len = 0;
+	int16_t declen;
 	
-	WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
-	WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
-	WebRtc_UWord16 streamdata[600];
-	WebRtc_Word16	speechType[1];
+	int16_t shortdata[FRAMESAMPLES_10ms];
+	int16_t decoded[MAX_FRAMESAMPLES];
+	uint16_t streamdata[600];
+	int16_t	speechType[1];
 	
-//	WebRtc_Word16	*iSACstruct;
+//	int16_t	*iSACstruct;
 
 	char version_number[20];
 	int mode=-1, tmp, nbTest=0; /*,sss;*/
@@ -641,7 +641,7 @@
 		}
 
 		/* Write decoded speech frame to file */
-		fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
+		fwrite(decoded, sizeof(int16_t), declen, outp);
 		}
 
 		fprintf(stderr,"  \rframe = %d", framecnt);
diff --git a/modules/audio_coding/codecs/isac/main/interface/isac.h b/modules/audio_coding/codecs/isac/main/interface/isac.h
index 54c630a..f937b34 100644
--- a/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -39,7 +39,7 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_AssignSize(
+  int16_t WebRtcIsac_AssignSize(
       int* sizeinbytes);
 
 
@@ -58,7 +58,7 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_Assign(
+  int16_t WebRtcIsac_Assign(
       ISACStruct** ISAC_main_inst,
       void*        ISAC_inst_Addr);
 
@@ -76,7 +76,7 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_Create(
+  int16_t WebRtcIsac_Create(
       ISACStruct** ISAC_main_inst);
 
 
@@ -92,7 +92,7 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_Free(
+  int16_t WebRtcIsac_Free(
       ISACStruct* ISAC_main_inst);
 
 
@@ -115,9 +115,9 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_EncoderInit(
+  int16_t WebRtcIsac_EncoderInit(
       ISACStruct* ISAC_main_inst,
-      WebRtc_Word16 CodingMode);
+      int16_t CodingMode);
 
 
   /******************************************************************************
@@ -144,10 +144,10 @@
    *                            : -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_Encode(
+  int16_t WebRtcIsac_Encode(
       ISACStruct*        ISAC_main_inst,
-      const WebRtc_Word16* speechIn,
-      WebRtc_Word16*       encoded);
+      const int16_t* speechIn,
+      int16_t*       encoded);
 
 
   /******************************************************************************
@@ -163,7 +163,7 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_DecoderInit(
+  int16_t WebRtcIsac_DecoderInit(
       ISACStruct* ISAC_main_inst);
 
 
@@ -185,13 +185,13 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(
+  int16_t WebRtcIsac_UpdateBwEstimate(
       ISACStruct*         ISAC_main_inst,
-      const WebRtc_UWord16* encoded,
-      WebRtc_Word32         packet_size,
-      WebRtc_UWord16        rtp_seq_number,
-      WebRtc_UWord32        send_ts,
-      WebRtc_UWord32        arr_ts);
+      const uint16_t* encoded,
+      int32_t         packet_size,
+      uint16_t        rtp_seq_number,
+      uint32_t        send_ts,
+      uint32_t        arr_ts);
 
 
   /******************************************************************************
@@ -214,12 +214,12 @@
    *                              -1 - Error.
    */
 
-  WebRtc_Word16 WebRtcIsac_Decode(
+  int16_t WebRtcIsac_Decode(
       ISACStruct*           ISAC_main_inst,
-      const WebRtc_UWord16* encoded,
-      WebRtc_Word16         len,
-      WebRtc_Word16*        decoded,
-      WebRtc_Word16*        speechType);
+      const uint16_t* encoded,
+      int16_t         len,
+      int16_t*        decoded,
+      int16_t*        speechType);
 
 
   /******************************************************************************
@@ -241,10 +241,10 @@
    *                              -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_DecodePlc(
+  int16_t WebRtcIsac_DecodePlc(
       ISACStruct*  ISAC_main_inst,
-      WebRtc_Word16* decoded,
-      WebRtc_Word16  noOfLostFrames);
+      int16_t* decoded,
+      int16_t  noOfLostFrames);
 
 
   /******************************************************************************
@@ -266,10 +266,10 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_Control(
+  int16_t WebRtcIsac_Control(
       ISACStruct*   ISAC_main_inst,
-      WebRtc_Word32 rate,
-      WebRtc_Word16 framesize);
+      int32_t rate,
+      int16_t framesize);
 
 
   /******************************************************************************
@@ -297,11 +297,11 @@
    *                             -1 - Error
    */
 
-  WebRtc_Word16 WebRtcIsac_ControlBwe(
+  int16_t WebRtcIsac_ControlBwe(
       ISACStruct* ISAC_main_inst,
-      WebRtc_Word32 rateBPS,
-      WebRtc_Word16 frameSizeMs,
-      WebRtc_Word16 enforceFrameSize);
+      int32_t rateBPS,
+      int16_t frameSizeMs,
+      int16_t enforceFrameSize);
 
 
   /******************************************************************************
@@ -317,10 +317,10 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsac_ReadFrameLen(
+  int16_t WebRtcIsac_ReadFrameLen(
       ISACStruct*          ISAC_main_inst,
-      const WebRtc_Word16* encoded,
-      WebRtc_Word16*       frameLength);
+      const int16_t* encoded,
+      int16_t*       frameLength);
 
 
   /******************************************************************************
@@ -351,7 +351,7 @@
    * Return value               : Error code
    */
 
-  WebRtc_Word16 WebRtcIsac_GetErrorCode(
+  int16_t WebRtcIsac_GetErrorCode(
       ISACStruct* ISAC_main_inst);
 
 
@@ -379,9 +379,9 @@
    *                               0 bit-rates computed correctly.
    */
 
-  WebRtc_Word16 WebRtcIsac_GetUplinkBw(
+  int16_t WebRtcIsac_GetUplinkBw(
       ISACStruct*    ISAC_main_inst,
-      WebRtc_Word32* bottleneck);
+      int32_t* bottleneck);
 
 
   /******************************************************************************
@@ -419,9 +419,9 @@
    *                             -1 if error happens
    */
 
-  WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(
+  int16_t WebRtcIsac_SetMaxPayloadSize(
       ISACStruct* ISAC_main_inst,
-      WebRtc_Word16 maxPayloadBytes);
+      int16_t maxPayloadBytes);
 
 
   /******************************************************************************
@@ -463,9 +463,9 @@
    *                             -1 if error happens
    */
 
-  WebRtc_Word16 WebRtcIsac_SetMaxRate(
+  int16_t WebRtcIsac_SetMaxRate(
       ISACStruct* ISAC_main_inst,
-      WebRtc_Word32 maxRate);
+      int32_t maxRate);
 
 
   /******************************************************************************
@@ -479,7 +479,7 @@
    *
    */
 
-  WebRtc_UWord16 WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
+  uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
 
 
   /******************************************************************************
@@ -492,7 +492,7 @@
    *
    */
 
-  WebRtc_UWord16 WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst);
+  uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst);
 
 
   /******************************************************************************
@@ -509,8 +509,8 @@
    *                             -1 if failed.
    */
 
-  WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
-                                          WebRtc_UWord16 samp_rate_hz);
+  int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
+                                          uint16_t samp_rate_hz);
 
 
   /******************************************************************************
@@ -529,8 +529,8 @@
    *                             -1 if failed.
    */
 
-  WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
-                                          WebRtc_UWord16 sample_rate_hz);
+  int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
+                                          uint16_t sample_rate_hz);
 
 
 
@@ -569,13 +569,13 @@
    *                                 the struct since it is only allowed to read
    *                                 the struct.
    */
-  WebRtc_Word16 WebRtcIsac_GetNewBitStream(
+  int16_t WebRtcIsac_GetNewBitStream(
       ISACStruct*    ISAC_main_inst,
-      WebRtc_Word16  bweIndex,
-      WebRtc_Word16  jitterInfo,
-      WebRtc_Word32  rate,
-      WebRtc_Word16* encoded,
-      WebRtc_Word16  isRCU);
+      int16_t  bweIndex,
+      int16_t  jitterInfo,
+      int32_t  rate,
+      int16_t* encoded,
+      int16_t  isRCU);
 
 
 
@@ -593,10 +593,10 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(
+  int16_t WebRtcIsac_GetDownLinkBwIndex(
       ISACStruct*  ISAC_main_inst,
-      WebRtc_Word16* bweIndex,
-      WebRtc_Word16* jitterInfo);
+      int16_t* bweIndex,
+      int16_t* jitterInfo);
 
 
   /****************************************************************************
@@ -611,9 +611,9 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(
+  int16_t WebRtcIsac_UpdateUplinkBw(
       ISACStruct* ISAC_main_inst,
-      WebRtc_Word16 bweIndex);
+      int16_t bweIndex);
 
 
   /****************************************************************************
@@ -630,9 +630,9 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsac_ReadBwIndex(
-      const WebRtc_Word16* encoded,
-      WebRtc_Word16*       bweIndex);
+  int16_t WebRtcIsac_ReadBwIndex(
+      const int16_t* encoded,
+      int16_t*       bweIndex);
 
 
 
@@ -652,7 +652,7 @@
    *
    */
 
-  WebRtc_Word16 WebRtcIsac_GetNewFrameLen(
+  int16_t WebRtcIsac_GetNewFrameLen(
       ISACStruct* ISAC_main_inst);
 
 
@@ -677,9 +677,9 @@
    *
    *
    */
-  WebRtc_Word16 WebRtcIsac_GetRedPayload(
+  int16_t WebRtcIsac_GetRedPayload(
       ISACStruct*    ISAC_main_inst,
-      WebRtc_Word16* encoded);
+      int16_t* encoded);
 
 
   /****************************************************************************
@@ -701,12 +701,12 @@
    * Return value              : >0 - number of samples in decoded vector
    *                             -1 - Error
    */
-  WebRtc_Word16 WebRtcIsac_DecodeRcu(
+  int16_t WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
-      const WebRtc_UWord16* encoded,
-      WebRtc_Word16         len,
-      WebRtc_Word16*        decoded,
-      WebRtc_Word16*        speechType);
+      const uint16_t* encoded,
+      int16_t         len,
+      int16_t*        decoded,
+      int16_t*        speechType);
 
 
 #if defined(__cplusplus)
diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines.c b/modules/audio_coding/codecs/isac/main/source/arith_routines.c
index 31c441a..5c901bb 100644
--- a/modules/audio_coding/codecs/isac/main/source/arith_routines.c
+++ b/modules/audio_coding/codecs/isac/main/source/arith_routines.c
@@ -18,7 +18,7 @@
  */
 int WebRtcIsac_EncTerminate(Bitstr *streamdata) /* in-/output struct containing bitstream */
 {
-  WebRtc_UWord8 *stream_ptr;
+  uint8_t *stream_ptr;
 
 
   /* point to the right place in the stream buffer */
@@ -37,7 +37,7 @@
       stream_ptr = streamdata->stream + streamdata->stream_index;
     }
     /* write remaining data to bitstream */
-    *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+    *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24);
   }
   else
   {
@@ -51,8 +51,8 @@
       stream_ptr = streamdata->stream + streamdata->stream_index;
     }
     /* write remaining data to bitstream */
-    *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
-    *stream_ptr++ = (WebRtc_UWord8) ((streamdata->streamval >> 16) & 0x00FF);
+    *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24);
+    *stream_ptr++ = (uint8_t) ((streamdata->streamval >> 16) & 0x00FF);
   }
 
   /* calculate stream length */
diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines.h b/modules/audio_coding/codecs/isac/main/source/arith_routines.h
index 8e5f496..43ba40e 100644
--- a/modules/audio_coding/codecs/isac/main/source/arith_routines.h
+++ b/modules/audio_coding/codecs/isac/main/source/arith_routines.h
@@ -23,41 +23,41 @@
 
 int WebRtcIsac_EncLogisticMulti2(
     Bitstr *streamdata,              /* in-/output struct containing bitstream */
-    WebRtc_Word16 *dataQ7,           /* input: data vector */
-    const WebRtc_UWord16 *env,       /* input: side info vector defining the width of the pdf */
+    int16_t *dataQ7,           /* input: data vector */
+    const uint16_t *env,       /* input: side info vector defining the width of the pdf */
     const int N,                     /* input: data vector length */
-    const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
+    const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
 
 /* returns the number of bytes in the stream */
 int WebRtcIsac_EncTerminate(Bitstr *streamdata); /* in-/output struct containing bitstream */
 
 /* returns the number of bytes in the stream so far */
 int WebRtcIsac_DecLogisticMulti2(
-    WebRtc_Word16 *data,             /* output: data vector */
+    int16_t *data,             /* output: data vector */
     Bitstr *streamdata,              /* in-/output struct containing bitstream */
-    const WebRtc_UWord16 *env,       /* input: side info vector defining the width of the pdf */
-    const WebRtc_Word16 *dither,     /* input: dither vector */
+    const uint16_t *env,       /* input: side info vector defining the width of the pdf */
+    const int16_t *dither,     /* input: dither vector */
     const int N,                     /* input: data vector length */
-    const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
+    const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
 
 void WebRtcIsac_EncHistMulti(
     Bitstr *streamdata,         /* in-/output struct containing bitstream */
     const int *data,            /* input: data vector */
-    const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+    const uint16_t **cdf, /* input: array of cdf arrays */
     const int N);               /* input: data vector length */
 
 int WebRtcIsac_DecHistBisectMulti(
     int *data,                      /* output: data vector */
     Bitstr *streamdata,             /* in-/output struct containing bitstream */
-    const WebRtc_UWord16 **cdf,     /* input: array of cdf arrays */
-    const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
+    const uint16_t **cdf,     /* input: array of cdf arrays */
+    const uint16_t *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
     const int N);                   /* input: data vector length */
 
 int WebRtcIsac_DecHistOneStepMulti(
     int *data,                       /* output: data vector */
     Bitstr *streamdata,              /* in-/output struct containing bitstream */
-    const WebRtc_UWord16 **cdf,      /* input: array of cdf arrays */
-    const WebRtc_UWord16 *init_index,/* input: vector of initial cdf table search entries */
+    const uint16_t **cdf,      /* input: array of cdf arrays */
+    const uint16_t *init_index,/* input: vector of initial cdf table search entries */
     const int N);                    /* input: data vector length */
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c b/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
index f4a13d6..63e4928 100644
--- a/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
+++ b/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
@@ -17,14 +17,14 @@
  */
 void WebRtcIsac_EncHistMulti(Bitstr *streamdata, /* in-/output struct containing bitstream */
                              const int *data,  /* input: data vector */
-                             const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+                             const uint16_t **cdf, /* input: array of cdf arrays */
                              const int N)   /* input: data vector length */
 {
-  WebRtc_UWord32 W_lower, W_upper;
-  WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
-  WebRtc_UWord8 *stream_ptr;
-  WebRtc_UWord8 *stream_ptr_carry;
-  WebRtc_UWord32 cdf_lo, cdf_hi;
+  uint32_t W_lower, W_upper;
+  uint32_t W_upper_LSB, W_upper_MSB;
+  uint8_t *stream_ptr;
+  uint8_t *stream_ptr_carry;
+  uint32_t cdf_lo, cdf_hi;
   int k;
 
 
@@ -35,8 +35,8 @@
   for (k=N; k>0; k--)
   {
     /* fetch cdf_lower and cdf_upper from cdf tables */
-    cdf_lo = (WebRtc_UWord32) *(*cdf + *data);
-    cdf_hi = (WebRtc_UWord32) *(*cdf++ + *data++ + 1);
+    cdf_lo = (uint32_t) *(*cdf + *data);
+    cdf_hi = (uint32_t) *(*cdf++ + *data++ + 1);
 
     /* update interval */
     W_upper_LSB = W_upper & 0x0000FFFF;
@@ -64,7 +64,7 @@
     while ( !(W_upper & 0xFF000000) )      /* W_upper < 2^24 */
     {
       W_upper <<= 8;
-      *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+      *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24);
       streamdata->streamval <<= 8;
     }
   }
@@ -84,16 +84,16 @@
  */
 int WebRtcIsac_DecHistBisectMulti(int *data,     /* output: data vector */
                                   Bitstr *streamdata,   /* in-/output struct containing bitstream */
-                                  const WebRtc_UWord16 **cdf,  /* input: array of cdf arrays */
-                                  const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
+                                  const uint16_t **cdf,  /* input: array of cdf arrays */
+                                  const uint16_t *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
                                   const int N)    /* input: data vector length */
 {
-  WebRtc_UWord32    W_lower, W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord32    W_upper_LSB, W_upper_MSB;
-  WebRtc_UWord32    streamval;
-  const   WebRtc_UWord8 *stream_ptr;
-  const   WebRtc_UWord16 *cdf_ptr;
+  uint32_t    W_lower, W_upper;
+  uint32_t    W_tmp;
+  uint32_t    W_upper_LSB, W_upper_MSB;
+  uint32_t    streamval;
+  const   uint8_t *stream_ptr;
+  const   uint16_t *cdf_ptr;
   int     size_tmp;
   int     k;
 
@@ -192,16 +192,16 @@
  */
 int WebRtcIsac_DecHistOneStepMulti(int *data,        /* output: data vector */
                                    Bitstr *streamdata,      /* in-/output struct containing bitstream */
-                                   const WebRtc_UWord16 **cdf,   /* input: array of cdf arrays */
-                                   const WebRtc_UWord16 *init_index, /* input: vector of initial cdf table search entries */
+                                   const uint16_t **cdf,   /* input: array of cdf arrays */
+                                   const uint16_t *init_index, /* input: vector of initial cdf table search entries */
                                    const int N)     /* input: data vector length */
 {
-  WebRtc_UWord32    W_lower, W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord32    W_upper_LSB, W_upper_MSB;
-  WebRtc_UWord32    streamval;
-  const   WebRtc_UWord8 *stream_ptr;
-  const   WebRtc_UWord16 *cdf_ptr;
+  uint32_t    W_lower, W_upper;
+  uint32_t    W_tmp;
+  uint32_t    W_upper_LSB, W_upper_MSB;
+  uint32_t    streamval;
+  const   uint8_t *stream_ptr;
+  const   uint16_t *cdf_ptr;
   int     k;
 
 
diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c b/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
index 422855a..eeed7ae 100644
--- a/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
+++ b/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
@@ -21,7 +21,7 @@
 
 
 
-static const WebRtc_Word32 kHistEdgesQ15[51] = {
+static const int32_t kHistEdgesQ15[51] = {
   -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
   -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644,
   -65536, -52429, -39322, -26215, -13108,  0,  13107,  26214,  39321,  52428,
@@ -49,10 +49,10 @@
 
 
 /* function to be converted to fixed point */
-static __inline WebRtc_UWord32 piecewise(WebRtc_Word32 xinQ15) {
+static __inline uint32_t piecewise(int32_t xinQ15) {
 
-  WebRtc_Word32 ind, qtmp1, qtmp2, qtmp3;
-  WebRtc_UWord32 tmpUW32;
+  int32_t ind, qtmp1, qtmp2, qtmp3;
+  uint32_t tmpUW32;
 
 
   qtmp2 = xinQ15;
@@ -79,17 +79,17 @@
 
 int WebRtcIsac_EncLogisticMulti2(
     Bitstr *streamdata,      /* in-/output struct containing bitstream */
-    WebRtc_Word16 *dataQ7,    /* input: data vector */
-    const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
+    int16_t *dataQ7,    /* input: data vector */
+    const uint16_t *envQ8, /* input: side info vector defining the width of the pdf */
     const int N,       /* input: data vector length / 2 */
-    const WebRtc_Word16 isSWB12kHz)
+    const int16_t isSWB12kHz)
 {
-  WebRtc_UWord32 W_lower, W_upper;
-  WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
-  WebRtc_UWord8 *stream_ptr;
-  WebRtc_UWord8 *maxStreamPtr;
-  WebRtc_UWord8 *stream_ptr_carry;
-  WebRtc_UWord32 cdf_lo, cdf_hi;
+  uint32_t W_lower, W_upper;
+  uint32_t W_upper_LSB, W_upper_MSB;
+  uint8_t *stream_ptr;
+  uint8_t *maxStreamPtr;
+  uint8_t *stream_ptr_carry;
+  uint32_t cdf_lo, cdf_hi;
   int k;
 
   /* point to beginning of stream buffer */
@@ -149,7 +149,7 @@
     while ( !(W_upper & 0xFF000000) )      /* W_upper < 2^24 */
     {
       W_upper <<= 8;
-      *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+      *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24);
 
       if(stream_ptr > maxStreamPtr)
       {
@@ -169,20 +169,20 @@
 
 
 int WebRtcIsac_DecLogisticMulti2(
-    WebRtc_Word16 *dataQ7,       /* output: data vector */
+    int16_t *dataQ7,       /* output: data vector */
     Bitstr *streamdata,      /* in-/output struct containing bitstream */
-    const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
-    const WebRtc_Word16 *ditherQ7,/* input: dither vector */
+    const uint16_t *envQ8, /* input: side info vector defining the width of the pdf */
+    const int16_t *ditherQ7,/* input: dither vector */
     const int N,         /* input: data vector length */
-    const WebRtc_Word16 isSWB12kHz)
+    const int16_t isSWB12kHz)
 {
-  WebRtc_UWord32    W_lower, W_upper;
-  WebRtc_UWord32    W_tmp;
-  WebRtc_UWord32    W_upper_LSB, W_upper_MSB;
-  WebRtc_UWord32    streamval;
-  const WebRtc_UWord8 *stream_ptr;
-  WebRtc_UWord32    cdf_tmp;
-  WebRtc_Word16     candQ7;
+  uint32_t    W_lower, W_upper;
+  uint32_t    W_tmp;
+  uint32_t    W_upper_LSB, W_upper_MSB;
+  uint32_t    streamval;
+  const uint8_t *stream_ptr;
+  uint32_t    cdf_tmp;
+  int16_t     candQ7;
   int             k;
 
   stream_ptr = streamdata->stream + streamdata->stream_index;
diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index d0a50c5..c4ceb59 100644
--- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@@ -41,7 +41,7 @@
 
 
 
-WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
+int32_t WebRtcIsac_InitBandwidthEstimator(
     BwEstimatorstr*              bwest_str,
     enum IsacSamplingRate encoderSampRate,
     enum IsacSamplingRate decoderSampRate)
@@ -67,7 +67,7 @@
         bwest_str->prev_frame_length = INIT_FRAME_LEN_WB;
         bwest_str->rec_bw_inv        = 1.0f /
             (INIT_BN_EST_WB + INIT_HDR_RATE_WB);
-        bwest_str->rec_bw            = (WebRtc_Word32)INIT_BN_EST_WB;
+        bwest_str->rec_bw            = (int32_t)INIT_BN_EST_WB;
         bwest_str->rec_bw_avg_Q      = INIT_BN_EST_WB;
         bwest_str->rec_bw_avg        = INIT_BN_EST_WB + INIT_HDR_RATE_WB;
         bwest_str->rec_header_rate   = INIT_HDR_RATE_WB;
@@ -78,7 +78,7 @@
         bwest_str->prev_frame_length = INIT_FRAME_LEN_SWB;
         bwest_str->rec_bw_inv        = 1.0f /
             (INIT_BN_EST_SWB + INIT_HDR_RATE_SWB);
-        bwest_str->rec_bw            = (WebRtc_Word32)INIT_BN_EST_SWB;
+        bwest_str->rec_bw            = (int32_t)INIT_BN_EST_SWB;
         bwest_str->rec_bw_avg_Q      = INIT_BN_EST_SWB;
         bwest_str->rec_bw_avg        = INIT_BN_EST_SWB + INIT_HDR_RATE_SWB;
         bwest_str->rec_header_rate   = INIT_HDR_RATE_SWB;
@@ -131,14 +131,14 @@
 /* pksize        - size of packet in bytes, from NetEq                                               */
 /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
 /* returns 0 if everything went fine, -1 otherwise                                                   */
-WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
+int16_t WebRtcIsac_UpdateBandwidthEstimator(
     BwEstimatorstr *bwest_str,
-    const WebRtc_UWord16 rtp_number,
-    const WebRtc_Word32  frame_length,
-    const WebRtc_UWord32 send_ts,
-    const WebRtc_UWord32 arr_ts,
-    const WebRtc_Word32  pksize
-    /*,    const WebRtc_UWord16 Index*/)
+    const uint16_t rtp_number,
+    const int32_t  frame_length,
+    const uint32_t send_ts,
+    const uint32_t arr_ts,
+    const int32_t  pksize
+    /*,    const uint16_t Index*/)
 {
   float weight = 0.0f;
   float curr_bw_inv = 0.0f;
@@ -207,7 +207,7 @@
       // that strict -DH
     {
       /* if not been updated for a long time, reduce the BN estimate */
-      if((WebRtc_UWord32)(arr_ts - bwest_str->last_update_ts) *
+      if((uint32_t)(arr_ts - bwest_str->last_update_ts) *
          1000.0f / FS > 3000)
       {
         //how many frames should have been received since the last
@@ -222,7 +222,7 @@
            0.9)
         {
           float inv_bitrate = (float) pow( 0.99995,
-                                           (double)((WebRtc_UWord32)(arr_ts -
+                                           (double)((uint32_t)(arr_ts -
                                                                      bwest_str->last_reduction_ts)*1000.0f/FS) );
 
           if ( inv_bitrate )
@@ -303,7 +303,7 @@
       float averageLatencyMs = latencyMs / bwest_str->numConsecLatePkts;
       delay_correction_factor = frame_length / (frame_length + averageLatencyMs);
       immediate_set = 1;
-      bwest_str->inWaitLatePkts = (WebRtc_Word16)((bwest_str->consecLatency/(FS/1000)) / 30);// + 150;
+      bwest_str->inWaitLatePkts = (int16_t)((bwest_str->consecLatency/(FS/1000)) / 30);// + 150;
       bwest_str->start_wait_period = arr_ts;
     }
     ///////////////////////////////////////////////
@@ -466,17 +466,17 @@
   bwest_str->prev_rec_send_ts = send_ts;
 
   /* Replace bwest_str->rec_bw by the new value (atomic operation) */
-  bwest_str->rec_bw = (WebRtc_Word32)(1.0f / bwest_str->rec_bw_inv -
+  bwest_str->rec_bw = (int32_t)(1.0f / bwest_str->rec_bw_inv -
                                       bwest_str->rec_header_rate);
 
   if (immediate_set)
   {
-    bwest_str->rec_bw = (WebRtc_Word32) (delay_correction_factor *
+    bwest_str->rec_bw = (int32_t) (delay_correction_factor *
                                          (float) bwest_str->rec_bw);
 
-    if (bwest_str->rec_bw < (WebRtc_Word32) MIN_ISAC_BW)
+    if (bwest_str->rec_bw < (int32_t) MIN_ISAC_BW)
     {
-      bwest_str->rec_bw = (WebRtc_Word32) MIN_ISAC_BW;
+      bwest_str->rec_bw = (int32_t) MIN_ISAC_BW;
     }
 
     bwest_str->rec_bw_avg = bwest_str->rec_bw +
@@ -503,9 +503,9 @@
 /* This function updates the send bottle neck rate                                                   */
 /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
 /* returns 0 if everything went fine, -1 otherwise                                                   */
-WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
+int16_t WebRtcIsac_UpdateUplinkBwImpl(
     BwEstimatorstr*           bwest_str,
-    WebRtc_Word16               index,
+    int16_t               index,
     enum IsacSamplingRate encoderSamplingFreq)
 {
   if((index < 0) || (index > 23))
@@ -560,9 +560,9 @@
 
 // called when there is upper-band bit-stream to update jitter
 // statistics.
-WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
+int16_t WebRtcIsac_UpdateUplinkJitter(
     BwEstimatorstr*              bwest_str,
-    WebRtc_Word32                  index)
+    int32_t                  index)
 {
   if((index < 0) || (index > 23))
   {
@@ -589,25 +589,25 @@
 
 // Returns the bandwidth/jitter estimation code (integer 0...23)
 // to put in the sending iSAC payload
-WebRtc_UWord16
+uint16_t
 WebRtcIsac_GetDownlinkBwJitIndexImpl(
     BwEstimatorstr*           bwest_str,
-    WebRtc_Word16*              bottleneckIndex,
-    WebRtc_Word16*              jitterInfo,
+    int16_t*              bottleneckIndex,
+    int16_t*              jitterInfo,
     enum IsacSamplingRate decoderSamplingFreq)
 {
   float MaxDelay;
-  //WebRtc_UWord16 MaxDelayBit;
+  //uint16_t MaxDelayBit;
 
   float rate;
   float r;
   float e1, e2;
   const float weight = 0.1f;
   const float* ptrQuantizationTable;
-  WebRtc_Word16 addJitterInfo;
-  WebRtc_Word16 minInd;
-  WebRtc_Word16 maxInd;
-  WebRtc_Word16 midInd;
+  int16_t addJitterInfo;
+  int16_t minInd;
+  int16_t maxInd;
+  int16_t midInd;
 
   /* Get Max Delay Bit */
   /* get unquantized max delay */
@@ -691,9 +691,9 @@
 
 
 /* get the bottle neck rate from far side to here, as estimated on this side */
-WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
+int32_t WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
 {
-  WebRtc_Word32  rec_bw;
+  int32_t  rec_bw;
   float   jitter_sign;
   float   bw_adjust;
 
@@ -705,7 +705,7 @@
   bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign);
 
   /* adjust Rate if jitter sign is mostly constant */
-  rec_bw = (WebRtc_Word32)(bwest_str->rec_bw * bw_adjust);
+  rec_bw = (int32_t)(bwest_str->rec_bw * bw_adjust);
 
   /* limit range of bottle neck rate */
   if (rec_bw < MIN_ISAC_BW)
@@ -720,12 +720,12 @@
 }
 
 /* Returns the max delay (in ms) */
-WebRtc_Word32
+int32_t
 WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
 {
-  WebRtc_Word32 rec_max_delay;
+  int32_t rec_max_delay;
 
-  rec_max_delay = (WebRtc_Word32)(bwest_str->rec_max_delay);
+  rec_max_delay = (int32_t)(bwest_str->rec_max_delay);
 
   /* limit range of jitter estimate */
   if (rec_max_delay < MIN_ISAC_MD)
@@ -743,7 +743,7 @@
 void
 WebRtcIsac_GetUplinkBandwidth(
     const BwEstimatorstr* bwest_str,
-    WebRtc_Word32*          bitRate)
+    int32_t*          bitRate)
 {
   /* limit range of bottle neck rate */
   if (bwest_str->send_bw_avg < MIN_ISAC_BW)
@@ -756,18 +756,18 @@
   }
   else
   {
-    *bitRate = (WebRtc_Word32)(bwest_str->send_bw_avg);
+    *bitRate = (int32_t)(bwest_str->send_bw_avg);
   }
   return;
 }
 
 /* Returns the max delay value from the other side in ms */
-WebRtc_Word32
+int32_t
 WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str)
 {
-  WebRtc_Word32 send_max_delay;
+  int32_t send_max_delay;
 
-  send_max_delay = (WebRtc_Word32)(bwest_str->send_max_delay_avg);
+  send_max_delay = (int32_t)(bwest_str->send_max_delay_avg);
 
   /* limit range of jitter estimate */
   if (send_max_delay < MIN_ISAC_MD)
@@ -793,7 +793,7 @@
     const double       BottleNeck,    /* bottle neck rate; excl headers (bps) */
     const double       DelayBuildUp,  /* max delay from bottleneck buffering (ms) */
     enum ISACBandwidth bandwidth
-    /*,WebRtc_Word16        frequentLargePackets*/)
+    /*,int16_t        frequentLargePackets*/)
 {
   double MinRate = 0.0;
   int    MinBytes;
diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
index 5604d7b..edabdff 100644
--- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
+++ b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
@@ -75,7 +75,7 @@
   /* This function initializes the struct                    */
   /* to be called before using the struct for anything else  */
   /* returns 0 if everything went fine, -1 otherwise         */
-  WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
+  int32_t WebRtcIsac_InitBandwidthEstimator(
       BwEstimatorstr*           bwest_str,
       enum IsacSamplingRate encoderSampRate,
       enum IsacSamplingRate decoderSampRate);
@@ -89,42 +89,42 @@
   /* pksize        - size of packet in bytes, from NetEq                                               */
   /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
   /* returns 0 if everything went fine, -1 otherwise                                                   */
-  WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
+  int16_t WebRtcIsac_UpdateBandwidthEstimator(
       BwEstimatorstr*    bwest_str,
-      const WebRtc_UWord16 rtp_number,
-      const WebRtc_Word32  frame_length,
-      const WebRtc_UWord32 send_ts,
-      const WebRtc_UWord32 arr_ts,
-      const WebRtc_Word32  pksize);
+      const uint16_t rtp_number,
+      const int32_t  frame_length,
+      const uint32_t send_ts,
+      const uint32_t arr_ts,
+      const int32_t  pksize);
 
   /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
-  WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
+  int16_t WebRtcIsac_UpdateUplinkBwImpl(
       BwEstimatorstr*           bwest_str,
-      WebRtc_Word16               Index,
+      int16_t               Index,
       enum IsacSamplingRate encoderSamplingFreq);
 
   /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
-  WebRtc_UWord16 WebRtcIsac_GetDownlinkBwJitIndexImpl(
+  uint16_t WebRtcIsac_GetDownlinkBwJitIndexImpl(
       BwEstimatorstr*           bwest_str,
-      WebRtc_Word16*              bottleneckIndex,
-      WebRtc_Word16*              jitterInfo,
+      int16_t*              bottleneckIndex,
+      int16_t*              jitterInfo,
       enum IsacSamplingRate decoderSamplingFreq);
 
   /* Returns the bandwidth estimation (in bps) */
-  WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth(
+  int32_t WebRtcIsac_GetDownlinkBandwidth(
       const BwEstimatorstr *bwest_str);
 
   /* Returns the max delay (in ms) */
-  WebRtc_Word32 WebRtcIsac_GetDownlinkMaxDelay(
+  int32_t WebRtcIsac_GetDownlinkMaxDelay(
       const BwEstimatorstr *bwest_str);
 
   /* Returns the bandwidth that iSAC should send with in bps */
   void WebRtcIsac_GetUplinkBandwidth(
       const BwEstimatorstr* bwest_str,
-      WebRtc_Word32*          bitRate);
+      int32_t*          bitRate);
 
   /* Returns the max delay value from the other side in ms */
-  WebRtc_Word32 WebRtcIsac_GetUplinkMaxDelay(
+  int32_t WebRtcIsac_GetUplinkMaxDelay(
       const BwEstimatorstr *bwest_str);
 
 
@@ -139,7 +139,7 @@
       const double       BottleNeck,    /* bottle neck rate; excl headers (bps) */
       const double       DelayBuildUp,  /* max delay from bottleneck buffering (ms) */
       enum ISACBandwidth bandwidth
-      /*,WebRtc_Word16        frequentLargePackets*/);
+      /*,int16_t        frequentLargePackets*/);
 
   /*
    * update long-term average bitrate and amount of data in buffer
@@ -165,9 +165,9 @@
       int    new_framelength);
 
 
-  WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
+  int16_t WebRtcIsac_UpdateUplinkJitter(
       BwEstimatorstr*              bwest_str,
-      WebRtc_Word32                  index);
+      int32_t                  index);
 
 #if defined(__cplusplus)
 }
diff --git a/modules/audio_coding/codecs/isac/main/source/codec.h b/modules/audio_coding/codecs/isac/main/source/codec.h
index 0b4d862..845f357 100644
--- a/modules/audio_coding/codecs/isac/main/source/codec.h
+++ b/modules/audio_coding/codecs/isac/main/source/codec.h
@@ -25,21 +25,21 @@
 void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
 
 int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
-                                 WebRtc_Word32 packet_size,
-                                 WebRtc_UWord16 rtp_seq_number,
-                                 WebRtc_UWord32 send_ts, WebRtc_UWord32 arr_ts,
+                                 int32_t packet_size,
+                                 uint16_t rtp_seq_number,
+                                 uint32_t send_ts, uint32_t arr_ts,
                                  enum IsacSamplingRate encoderSampRate,
                                  enum IsacSamplingRate decoderSampRate);
 
 int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
-                        WebRtc_Word16* current_framesamples,
-                        WebRtc_Word16 isRCUPayload);
+                        int16_t* current_framesamples,
+                        int16_t isRCUPayload);
 
 int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
-                           WebRtc_Word16* current_framesamples);
+                           int16_t* current_framesamples);
 
 int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
-                        WebRtc_Word16 codingMode, WebRtc_Word16
+                        int16_t codingMode, int16_t
                         bottleneckIndex);
 
 int WebRtcIsac_EncodeStoredDataLb(const ISAC_SaveEncData_t* ISACSavedEnc_obj,
@@ -48,9 +48,9 @@
 
 int WebRtcIsac_EncodeStoredDataUb(
     const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
-    WebRtc_Word32 jitterInfo, float scale, enum ISACBandwidth bandwidth);
+    int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
 
-WebRtc_Word16 WebRtcIsac_GetRedPayloadUb(
+int16_t WebRtcIsac_GetRedPayloadUb(
     const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
     enum ISACBandwidth bandwidth);
 
@@ -72,10 +72,10 @@
  *                                -1 if failed to allocate rates.
  */
 
-WebRtc_Word16 WebRtcIsac_RateAllocation(WebRtc_Word32 inRateBitPerSec,
-                                        double* rateLBBitPerSec,
-                                        double* rateUBBitPerSec,
-                                        enum ISACBandwidth* bandwidthKHz);
+int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
+                                  double* rateLBBitPerSec,
+                                  double* rateUBBitPerSec,
+                                  enum ISACBandwidth* bandwidthKHz);
 
 
 /******************************************************************************
@@ -94,7 +94,7 @@
  *                             <0 if an error occurred.
  */
 int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdec_obj,
-                          WebRtc_Word16 isRCUPayload);
+                          int16_t isRCUPayload);
 
 
 /******************************************************************************
@@ -113,7 +113,7 @@
  *                             <0 if an error occurred.
  */
 int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdec_obj,
-                          WebRtc_Word16 isRCUPayload);
+                          int16_t isRCUPayload);
 
 
 /******************************************************************************
@@ -132,7 +132,7 @@
  *                             <0 if an error occurred.
  */
 int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACenc_obj,
-                          WebRtc_Word32 jitterInfo);
+                          int32_t jitterInfo);
 
 
 /******************************************************************************
@@ -151,7 +151,7 @@
  *                             <0 if an error occurred.
  */
 int WebRtcIsac_EncodeUb12(float* in, ISACUBEncStruct* ISACenc_obj,
-                          WebRtc_Word32 jitterInfo);
+                          int32_t jitterInfo);
 
 /************************** initialization functions *************************/
 
@@ -170,8 +170,8 @@
 
 void WebRtcIsac_InitTransform();
 
-void WebRtcIsac_Time2Spec(double* inre1, double* inre2, WebRtc_Word16* outre,
-                          WebRtc_Word16* outim, FFTstr* fftstr_obj);
+void WebRtcIsac_Time2Spec(double* inre1, double* inre2, int16_t* outre,
+                          int16_t* outim, FFTstr* fftstr_obj);
 
 void WebRtcIsac_Spec2time(double* inre, double* inim, double* outre1,
                           double* outre2, FFTstr* fftstr_obj);
diff --git a/modules/audio_coding/codecs/isac/main/source/crc.c b/modules/audio_coding/codecs/isac/main/source/crc.c
index 098e4b7..1d36ff0 100644
--- a/modules/audio_coding/codecs/isac/main/source/crc.c
+++ b/modules/audio_coding/codecs/isac/main/source/crc.c
@@ -15,7 +15,7 @@
 #define POLYNOMIAL 0x04c11db7L
 
 
-static const WebRtc_UWord32 kCrcTable[256] = {
+static const uint32_t kCrcTable[256] = {
   0,          0x4c11db7,  0x9823b6e,  0xd4326d9,  0x130476dc, 0x17c56b6b,
   0x1a864db2, 0x1e475005, 0x2608edb8, 0x22c9f00f, 0x2f8ad6d6, 0x2b4bcb61,
   0x350c9b64, 0x31cd86d3, 0x3c8ea00a, 0x384fbdbd, 0x4c11db70, 0x48d0c6c7,
@@ -80,12 +80,12 @@
  *                             -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsac_GetCrc(const WebRtc_Word16* bitstream,
-                                WebRtc_Word16        len_bitstream_in_bytes,
-                                WebRtc_UWord32*      crc)
+int16_t WebRtcIsac_GetCrc(const int16_t* bitstream,
+                          int16_t        len_bitstream_in_bytes,
+                          uint32_t*      crc)
 {
-  WebRtc_UWord8* bitstream_ptr_uw8;
-  WebRtc_UWord32 crc_state;
+  uint8_t* bitstream_ptr_uw8;
+  uint32_t crc_state;
   int byte_cntr;
   int crc_tbl_indx;
 
@@ -94,7 +94,7 @@
     return -1;
   }
   /* cast to UWord8 pointer */
-  bitstream_ptr_uw8 = (WebRtc_UWord8 *)bitstream;
+  bitstream_ptr_uw8 = (uint8_t *)bitstream;
 
   /* initialize */
   crc_state = 0xFFFFFFFF;
diff --git a/modules/audio_coding/codecs/isac/main/source/crc.h b/modules/audio_coding/codecs/isac/main/source/crc.h
index 0151278..dba8749 100644
--- a/modules/audio_coding/codecs/isac/main/source/crc.h
+++ b/modules/audio_coding/codecs/isac/main/source/crc.h
@@ -36,10 +36,10 @@
  *                   -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsac_GetCrc(
-    const WebRtc_Word16* encoded,
-    WebRtc_Word16        no_of_word8s,
-    WebRtc_UWord32*      crc);
+int16_t WebRtcIsac_GetCrc(
+    const int16_t* encoded,
+    int16_t        no_of_word8s,
+    uint32_t*      crc);
 
 
 
diff --git a/modules/audio_coding/codecs/isac/main/source/decode.c b/modules/audio_coding/codecs/isac/main/source/decode.c
index 2b06cd0..e23765b 100644
--- a/modules/audio_coding/codecs/isac/main/source/decode.c
+++ b/modules/audio_coding/codecs/isac/main/source/decode.c
@@ -36,11 +36,11 @@
  * returns the total number of bytes in the stream
  */
 int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
-                        WebRtc_Word16* current_framesamples,
-                        WebRtc_Word16 isRCUPayload) {
+                        int16_t* current_framesamples,
+                        int16_t isRCUPayload) {
   int k;
   int len, err;
-  WebRtc_Word16 bandwidthInd;
+  int16_t bandwidthInd;
 
   float LP_dec_float[FRAMESAMPLES_HALF];
   float HP_dec_float[FRAMESAMPLES_HALF];
@@ -58,8 +58,8 @@
   double PitchLags[4];
   double PitchGains[4];
   double AvgPitchGain;
-  WebRtc_Word16 PitchGains_Q12[4];
-  WebRtc_Word16 AvgPitchGain_Q12;
+  int16_t PitchGains_Q12[4];
+  int16_t AvgPitchGain_Q12;
 
   float gain;
 
@@ -182,7 +182,7 @@
  * frequency, but split to 12 sub-frames, i.e. twice as lower-band.
  */
 int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
-                          WebRtc_Word16 isRCUPayload) {
+                          int16_t isRCUPayload) {
   int len, err;
 
   double halfFrameFirst[FRAMESAMPLES_HALF];
@@ -193,7 +193,7 @@
 
   double real_f[FRAMESAMPLES_HALF];
   double imag_f[FRAMESAMPLES_HALF];
-  const WebRtc_Word16 kAveragePitchGain = 0; /* No pitch-gain for upper-band. */
+  const int16_t kAveragePitchGain = 0; /* No pitch-gain for upper-band. */
   len = 0;
 
   /* Decode & de-quantize filter coefficients. */
@@ -246,7 +246,7 @@
  * are combined, to reconstruct the upperband 8-16 kHz.
  */
 int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
-                      WebRtc_Word16 isRCUPayload) {
+                      int16_t isRCUPayload) {
   int len, err;
 
   float LP_dec_float[FRAMESAMPLES_HALF];
@@ -259,7 +259,7 @@
 
   double real_f[FRAMESAMPLES_HALF];
   double imag_f[FRAMESAMPLES_HALF];
-  const WebRtc_Word16 kAveragePitchGain = 0; /* No pitch-gain for upper-band. */
+  const int16_t kAveragePitchGain = 0; /* No pitch-gain for upper-band. */
   len = 0;
 
   /* Decode & dequantize filter coefficients. */
diff --git a/modules/audio_coding/codecs/isac/main/source/decode_bwe.c b/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
index a92b9b9..5abe204 100644
--- a/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
+++ b/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
@@ -18,19 +18,19 @@
 WebRtcIsac_EstimateBandwidth(
     BwEstimatorstr*           bwest_str,
     Bitstr*                   streamdata,
-    WebRtc_Word32               packet_size,
-    WebRtc_UWord16              rtp_seq_number,
-    WebRtc_UWord32              send_ts,
-    WebRtc_UWord32              arr_ts,
+    int32_t               packet_size,
+    uint16_t              rtp_seq_number,
+    uint32_t              send_ts,
+    uint32_t              arr_ts,
     enum IsacSamplingRate encoderSampRate,
     enum IsacSamplingRate decoderSampRate)
 {
-  WebRtc_Word16  index;
-  WebRtc_Word16  frame_samples;
-  WebRtc_UWord32 sendTimestampIn16kHz;
-  WebRtc_UWord32 arrivalTimestampIn16kHz;
-  WebRtc_UWord32 diffSendTime;
-  WebRtc_UWord32 diffArrivalTime;
+  int16_t  index;
+  int16_t  frame_samples;
+  uint32_t sendTimestampIn16kHz;
+  uint32_t arrivalTimestampIn16kHz;
+  uint32_t diffSendTime;
+  uint32_t diffArrivalTime;
   int err;
 
   /* decode framelength and BW estimation */
@@ -55,26 +55,26 @@
   // We like BWE to work at 16 kHz sampling rate,
   // therefore, we have to change the timestamps accordingly.
   // translate the send timestamp if required
-  diffSendTime = (WebRtc_UWord32)((WebRtc_UWord32)send_ts -
-                                  (WebRtc_UWord32)bwest_str->senderTimestamp);
+  diffSendTime = (uint32_t)((uint32_t)send_ts -
+                                  (uint32_t)bwest_str->senderTimestamp);
   bwest_str->senderTimestamp = send_ts;
 
-  diffArrivalTime = (WebRtc_UWord32)((WebRtc_UWord32)arr_ts -
-                                     (WebRtc_UWord32)bwest_str->receiverTimestamp);
+  diffArrivalTime = (uint32_t)((uint32_t)arr_ts -
+                                     (uint32_t)bwest_str->receiverTimestamp);
   bwest_str->receiverTimestamp = arr_ts;
 
   if(decoderSampRate == kIsacSuperWideband)
   {
-    diffArrivalTime = (WebRtc_UWord32)diffArrivalTime >> 1;
-    diffSendTime = (WebRtc_UWord32)diffSendTime >> 1;
+    diffArrivalTime = (uint32_t)diffArrivalTime >> 1;
+    diffSendTime = (uint32_t)diffSendTime >> 1;
   }
 
   // arrival timestamp in 16 kHz
-  arrivalTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
-                                             bwest_str->prev_rec_arr_ts + (WebRtc_UWord32)diffArrivalTime);
+  arrivalTimestampIn16kHz = (uint32_t)((uint32_t)
+                                             bwest_str->prev_rec_arr_ts + (uint32_t)diffArrivalTime);
   // send timestamp in 16 kHz
-  sendTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
-                                          bwest_str->prev_rec_send_ts + (WebRtc_UWord32)diffSendTime);
+  sendTimestampIn16kHz = (uint32_t)((uint32_t)
+                                          bwest_str->prev_rec_send_ts + (uint32_t)diffSendTime);
 
   err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number,
                                             (frame_samples * 1000) / FS, sendTimestampIn16kHz,
diff --git a/modules/audio_coding/codecs/isac/main/source/encode.c b/modules/audio_coding/codecs/isac/main/source/encode.c
index f6bdc17..5d8fa38 100644
--- a/modules/audio_coding/codecs/isac/main/source/encode.c
+++ b/modules/audio_coding/codecs/isac/main/source/encode.c
@@ -70,15 +70,15 @@
  */
 
 /*     38  39.17  40.33   41.5  42.67  43.83     45 */
-static const WebRtc_Word16 kLowerBandBitRate12[7] = {
+static const int16_t kLowerBandBitRate12[7] = {
     29000, 30000, 30000, 31000, 31000, 32000, 32000 };
-static const WebRtc_Word16 kUpperBandBitRate12[7] = {
+static const int16_t kUpperBandBitRate12[7] = {
     25000, 25000, 27000, 27000, 29000, 29000, 32000 };
 
 /*    50     51.2  52.4   53.6   54.8    56 */
-static const WebRtc_Word16 kLowerBandBitRate16[6] = {
+static const int16_t kLowerBandBitRate16[6] = {
     31000, 31000, 32000, 32000, 32000, 32000 };
-static const WebRtc_Word16 kUpperBandBitRate16[6] = {
+static const int16_t kUpperBandBitRate16[6] = {
     28000, 29000, 29000, 30000, 31000, 32000 };
 
 /******************************************************************************
@@ -99,18 +99,18 @@
  *                                -1 if failed to allocate rates.
  */
 
-WebRtc_Word16 WebRtcIsac_RateAllocation(WebRtc_Word32 inRateBitPerSec,
+int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
                                         double* rateLBBitPerSec,
                                         double* rateUBBitPerSec,
                                         enum ISACBandwidth* bandwidthKHz) {
-  WebRtc_Word16 idx;
+  int16_t idx;
   double idxD;
   double idxErr;
   if (inRateBitPerSec < 38000) {
     /* If the given overall bottleneck is less than 38000 then
      * then codec has to operate in wideband mode, i.e. 8 kHz
      * bandwidth. */
-    *rateLBBitPerSec = (WebRtc_Word16)((inRateBitPerSec > 32000) ?
+    *rateLBBitPerSec = (int16_t)((inRateBitPerSec > 32000) ?
         32000 : inRateBitPerSec);
     *rateUBBitPerSec = 0;
     *bandwidthKHz = isac8kHz;
@@ -123,15 +123,15 @@
      * step is (45000 - 38000)/6.0 we use the inverse of it. */
     const double stepSizeInv = 8.5714286e-4;
     idxD = (inRateBitPerSec - 38000) * stepSizeInv;
-    idx = (idxD >= 6) ? 6 : ((WebRtc_Word16)idxD);
+    idx = (idxD >= 6) ? 6 : ((int16_t)idxD);
     idxErr = idxD - idx;
     *rateLBBitPerSec = kLowerBandBitRate12[idx];
     *rateUBBitPerSec = kUpperBandBitRate12[idx];
 
     if (idx < 6) {
-      *rateLBBitPerSec += (WebRtc_Word16)(
+      *rateLBBitPerSec += (int16_t)(
           idxErr * (kLowerBandBitRate12[idx + 1] - kLowerBandBitRate12[idx]));
-      *rateUBBitPerSec += (WebRtc_Word16)(
+      *rateUBBitPerSec += (int16_t)(
           idxErr * (kUpperBandBitRate12[idx + 1] - kUpperBandBitRate12[idx]));
     }
     *bandwidthKHz = isac12kHz;
@@ -144,17 +144,17 @@
      * step is (56000 - 50000)/5 we use the inverse of it. */
     const double stepSizeInv = 8.3333333e-4;
     idxD = (inRateBitPerSec - 50000) * stepSizeInv;
-    idx = (idxD >= 5) ? 5 : ((WebRtc_Word16)idxD);
+    idx = (idxD >= 5) ? 5 : ((int16_t)idxD);
     idxErr = idxD - idx;
     *rateLBBitPerSec = kLowerBandBitRate16[idx];
     *rateUBBitPerSec  = kUpperBandBitRate16[idx];
 
     if (idx < 5) {
-      *rateLBBitPerSec += (WebRtc_Word16)(idxErr *
+      *rateLBBitPerSec += (int16_t)(idxErr *
           (kLowerBandBitRate16[idx + 1] -
               kLowerBandBitRate16[idx]));
 
-      *rateUBBitPerSec += (WebRtc_Word16)(idxErr *
+      *rateUBBitPerSec += (int16_t)(idxErr *
           (kUpperBandBitRate16[idx + 1] -
               kUpperBandBitRate16[idx]));
     }
@@ -178,8 +178,8 @@
 }
 
 int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
-                        WebRtc_Word16 codingMode,
-                        WebRtc_Word16 bottleneckIndex) {
+                        int16_t codingMode,
+                        int16_t bottleneckIndex) {
   int stream_length = 0;
   int err;
   int k;
@@ -197,20 +197,20 @@
 
   double HPw[FRAMESAMPLES_HALF];
   double LPw_pf[FRAMESAMPLES_HALF];
-  WebRtc_Word16 fre[FRAMESAMPLES_HALF];   /* Q7 */
-  WebRtc_Word16 fim[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fre[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fim[FRAMESAMPLES_HALF];   /* Q7 */
 
   double PitchLags[4];
   double PitchGains[4];
-  WebRtc_Word16 PitchGains_Q12[4];
-  WebRtc_Word16 AvgPitchGain_Q12;
+  int16_t PitchGains_Q12[4];
+  int16_t AvgPitchGain_Q12;
 
   int frame_mode; /* 0 for 30ms, 1 for 60ms */
   int status = 0;
   int my_index;
   transcode_obj transcodingParam;
   double bytesLeftSpecCoding;
-  WebRtc_UWord16 payloadLimitBytes;
+  uint16_t payloadLimitBytes;
 
   /* Copy new frame-length and bottleneck rate only for the first 10 ms data */
   if (ISACencLB_obj->buffer_index == 0) {
@@ -292,7 +292,7 @@
 
   /* Convert PitchGain to Fixed point. */
   for (k = 0; k < PITCH_SUBFRAMES; k++) {
-    PitchGains_Q12[k] = (WebRtc_Word16)(PitchGains[k] * 4096.0);
+    PitchGains_Q12[k] = (int16_t)(PitchGains[k] * 4096.0);
   }
 
   /* Set where to store data in multiple packets memory. */
@@ -458,8 +458,8 @@
 
     /* Scale DFT coefficients. */
     for (k = 0; k < FRAMESAMPLES_HALF; k++) {
-      fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale);
-      fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale);
+      fre[k] = (int16_t)(fre[k] * transcodeScale);
+      fim[k] = (int16_t)(fim[k] * transcodeScale);
     }
 
     /* Save data for multiple packets memory. */
@@ -531,17 +531,17 @@
 
 
 static int LimitPayloadUb(ISACUBEncStruct* ISACencUB_obj,
-                          WebRtc_UWord16 payloadLimitBytes,
+                          uint16_t payloadLimitBytes,
                           double bytesLeftSpecCoding,
                           transcode_obj* transcodingParam,
-                          WebRtc_Word16* fre, WebRtc_Word16* fim,
+                          int16_t* fre, int16_t* fim,
                           double* lpcGains, enum ISACBand band, int status) {
 
   int iterCntr = 0;
   int k;
   double bytesSpecCoderUsed;
   double transcodeScale;
-  const WebRtc_Word16 kAveragePitchGain = 0.0;
+  const int16_t kAveragePitchGain = 0.0;
 
   do {
     if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
@@ -580,8 +580,8 @@
 
     /* Scale DFT coefficients. */
     for (k = 0; k < FRAMESAMPLES_HALF; k++) {
-      fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale + 0.5);
-      fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale + 0.5);
+      fre[k] = (int16_t)(fre[k] * transcodeScale + 0.5);
+      fim[k] = (int16_t)(fim[k] * transcodeScale + 0.5);
     }
     /* Store FFT coefficients for multiple encoding. */
     memcpy(ISACencUB_obj->SaveEnc_obj.realFFT, fre,
@@ -642,7 +642,7 @@
 }
 
 int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACencUB_obj,
-                          WebRtc_Word32 jitterInfo) {
+                          int32_t jitterInfo) {
   int err;
   int k;
 
@@ -651,8 +651,8 @@
                             (1 + UB_LPC_ORDER)];
 
   double LP_lookahead[FRAMESAMPLES];
-  WebRtc_Word16 fre[FRAMESAMPLES_HALF];   /* Q7 */
-  WebRtc_Word16 fim[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fre[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fim[FRAMESAMPLES_HALF];   /* Q7 */
 
   int status = 0;
 
@@ -660,9 +660,9 @@
   double corr[SUBFRAMES << 1][UB_LPC_ORDER + 1];
   double lpcGains[SUBFRAMES << 1];
   transcode_obj transcodingParam;
-  WebRtc_UWord16 payloadLimitBytes;
+  uint16_t payloadLimitBytes;
   double s2nr;
-  const WebRtc_Word16 kAveragePitchGain = 0.0;
+  const int16_t kAveragePitchGain = 0.0;
   int bytesLeftSpecCoding;
 
   /* Buffer speech samples (by 10ms packet) until the frame-length is   */
@@ -827,7 +827,7 @@
 
 
 int WebRtcIsac_EncodeUb12(float* in, ISACUBEncStruct* ISACencUB_obj,
-                          WebRtc_Word32 jitterInfo) {
+                          int32_t jitterInfo) {
   int err;
   int k;
 
@@ -842,8 +842,8 @@
   double LPw[FRAMESAMPLES_HALF];
 
   double HPw[FRAMESAMPLES_HALF];
-  WebRtc_Word16 fre[FRAMESAMPLES_HALF];   /* Q7 */
-  WebRtc_Word16 fim[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fre[FRAMESAMPLES_HALF];   /* Q7 */
+  int16_t fim[FRAMESAMPLES_HALF];   /* Q7 */
 
   int status = 0;
 
@@ -852,9 +852,9 @@
   double corr[UB_LPC_GAIN_DIM][UB_LPC_ORDER + 1];
   double lpcGains[SUBFRAMES];
   transcode_obj transcodingParam;
-  WebRtc_UWord16 payloadLimitBytes;
+  uint16_t payloadLimitBytes;
   double s2nr;
-  const WebRtc_Word16 kAveragePitchGain = 0.0;
+  const int16_t kAveragePitchGain = 0.0;
   double bytesLeftSpecCoding;
 
   /* Buffer speech samples (by 10ms packet) until the framelength is  */
@@ -1011,13 +1011,13 @@
   int status;
   int BWno = BWnumber;
 
-  const WebRtc_UWord16* WebRtcIsac_kQPitchGainCdf_ptr[1];
-  const WebRtc_UWord16** cdf;
+  const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1];
+  const uint16_t** cdf;
 
   double tmpLPCcoeffs_lo[(ORDERLO + 1)*SUBFRAMES * 2];
   double tmpLPCcoeffs_hi[(ORDERHI + 1)*SUBFRAMES * 2];
   int tmpLPCindex_g[12 * 2];
-  WebRtc_Word16 tmp_fre[FRAMESAMPLES], tmp_fim[FRAMESAMPLES];
+  int16_t tmp_fre[FRAMESAMPLES], tmp_fim[FRAMESAMPLES];
   const int kModel = 0;
 
   /* Sanity Check - possible values for BWnumber is 0 - 23. */
@@ -1053,8 +1053,8 @@
     for (ii = 0;
         ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx));
         ii++) {
-      tmp_fre[ii] = (WebRtc_Word16)((scale) * (float)ISACSavedEnc_obj->fre[ii]);
-      tmp_fim[ii] = (WebRtc_Word16)((scale) * (float)ISACSavedEnc_obj->fim[ii]);
+      tmp_fre[ii] = (int16_t)((scale) * (float)ISACSavedEnc_obj->fre[ii]);
+      tmp_fim[ii] = (int16_t)((scale) * (float)ISACSavedEnc_obj->fim[ii]);
     }
   } else {
     for (ii = 0;
@@ -1134,17 +1134,17 @@
 int WebRtcIsac_EncodeStoredDataUb(
     const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
     Bitstr* bitStream,
-    WebRtc_Word32 jitterInfo,
+    int32_t jitterInfo,
     float scale,
     enum ISACBandwidth bandwidth) {
   int n;
   int err;
   double lpcGain[SUBFRAMES];
-  WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
-  WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
-  const WebRtc_UWord16** shape_cdf;
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
+  const uint16_t** shape_cdf;
   int shape_len;
-  const WebRtc_Word16 kAveragePitchGain = 0.0;
+  const int16_t kAveragePitchGain = 0.0;
   enum ISACBand band;
   /* Reset bitstream. */
   WebRtcIsac_ResetBitstream(bitStream);
@@ -1201,9 +1201,9 @@
     }
 
     for (n = 0; n < FRAMESAMPLES_HALF; n++) {
-      realFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->realFFT[n] +
+      realFFT[n] = (int16_t)(scale * (float)ISACSavedEnc_obj->realFFT[n] +
           0.5f);
-      imagFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->imagFFT[n] +
+      imagFFT[n] = (int16_t)(scale * (float)ISACSavedEnc_obj->imagFFT[n] +
           0.5f);
     }
     /* Store FFT coefficients. */
@@ -1219,24 +1219,24 @@
   return WebRtcIsac_EncTerminate(bitStream);
 }
 
-WebRtc_Word16 WebRtcIsac_GetRedPayloadUb(
+int16_t WebRtcIsac_GetRedPayloadUb(
     const ISACUBSaveEncDataStruct* ISACSavedEncObj,
     Bitstr*                        bitStreamObj,
     enum ISACBandwidth             bandwidth) {
   int n;
-  WebRtc_Word16 status;
-  WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
-  WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+  int16_t status;
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
   enum ISACBand band;
-  const WebRtc_Word16 kAveragePitchGain = 0.0;
+  const int16_t kAveragePitchGain = 0.0;
   /* Store bit-stream object. */
   memcpy(bitStreamObj, &ISACSavedEncObj->bitStreamObj, sizeof(Bitstr));
 
   /* Scale FFT coefficients. */
   for (n = 0; n < FRAMESAMPLES_HALF; n++) {
-    realFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->realFFT[n] *
+    realFFT[n] = (int16_t)((float)ISACSavedEncObj->realFFT[n] *
         RCU_TRANSCODING_SCALE_UB + 0.5);
-    imagFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->imagFFT[n] *
+    imagFFT[n] = (int16_t)((float)ISACSavedEncObj->imagFFT[n] *
         RCU_TRANSCODING_SCALE_UB + 0.5);
   }
 
diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
index 2bf4c36..67e77b5 100644
--- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
+++ b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
@@ -44,14 +44,14 @@
  *
  *
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_RemoveLarMean(
     double* lar,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 coeffCntr;
-  WebRtc_Word16 vecCntr;
-  WebRtc_Word16 numVec;
+  int16_t coeffCntr;
+  int16_t vecCntr;
+  int16_t numVec;
   const double* meanLAR;
   switch(bandwidth)
   {
@@ -98,18 +98,18 @@
  * Output:
  *      -out                : decorrelated LAR vectors.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_DecorrelateIntraVec(
     const double* data,
     double*       out,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
   const double* ptrData;
   const double* ptrRow;
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
-  WebRtc_Word16 larVecCntr;
-  WebRtc_Word16 numVec;
+  int16_t rowCntr;
+  int16_t colCntr;
+  int16_t larVecCntr;
+  int16_t numVec;
   const double* decorrMat;
   switch(bandwidth)
   {
@@ -172,17 +172,17 @@
  * Output:
  *      -out                : decorrelated LAR vectors.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_DecorrelateInterVec(
     const double* data,
     double* out,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 coeffCntr;
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
+  int16_t coeffCntr;
+  int16_t rowCntr;
+  int16_t colCntr;
   const double* decorrMat;
-  WebRtc_Word16 interVecDim;
+  int16_t interVecDim;
 
   switch(bandwidth)
   {
@@ -245,14 +245,14 @@
 WebRtcIsac_QuantizeUncorrLar(
     double* data,
     int* recIdx,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 cntr;
-  WebRtc_Word32 idx;
-  WebRtc_Word16 interVecDim;
+  int16_t cntr;
+  int32_t idx;
+  int16_t interVecDim;
   const double* leftRecPoint;
   double quantizationStepSize;
-  const WebRtc_Word16* numQuantCell;
+  const int16_t* numQuantCell;
   switch(bandwidth)
   {
     case isac12kHz:
@@ -280,7 +280,7 @@
   //
   for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++)
   {
-    idx = (WebRtc_Word32)floor((*data - leftRecPoint[cntr]) /
+    idx = (int32_t)floor((*data - leftRecPoint[cntr]) /
                                quantizationStepSize + 0.5);
     if(idx < 0)
     {
@@ -311,14 +311,14 @@
  * Output:
  *      -out                : pointer to quantized values.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_DequantizeLpcParam(
     const int* idx,
     double*    out,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 cntr;
-  WebRtc_Word16 interVecDim;
+  int16_t cntr;
+  int16_t interVecDim;
   const double* leftRecPoint;
   double quantizationStepSize;
 
@@ -367,16 +367,16 @@
  * Output:
  *      -out                : correlated parametrs.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_CorrelateIntraVec(
     const double* data,
     double*       out,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 vecCntr;
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
-  WebRtc_Word16 numVec;
+  int16_t vecCntr;
+  int16_t rowCntr;
+  int16_t colCntr;
+  int16_t numVec;
   const double* ptrData;
   const double* intraVecDecorrMat;
 
@@ -430,16 +430,16 @@
  * Output:
  *      -out                : correlated parametrs.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_CorrelateInterVec(
     const double* data,
     double*       out,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 coeffCntr;
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
-  WebRtc_Word16 interVecDim;
+  int16_t coeffCntr;
+  int16_t rowCntr;
+  int16_t colCntr;
+  int16_t interVecDim;
   double myVec[UB16_LPC_VEC_PER_FRAME];
   const double* interVecDecorrMat;
 
@@ -495,14 +495,14 @@
  * Output:
  *      -data               : pointer to LARs.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_AddLarMean(
     double* data,
-    WebRtc_Word16 bandwidth)
+    int16_t bandwidth)
 {
-  WebRtc_Word16 coeffCntr;
-  WebRtc_Word16 vecCntr;
-  WebRtc_Word16 numVec;
+  int16_t coeffCntr;
+  int16_t vecCntr;
+  int16_t numVec;
   const double* meanLAR;
 
   switch(bandwidth)
@@ -544,11 +544,11 @@
  * Output:
  *      -lpcGain            : mean-removed in log domain.
  */
-WebRtc_Word16
+int16_t
 WebRtcIsac_ToLogDomainRemoveMean(
     double* data)
 {
-  WebRtc_Word16 coeffCntr;
+  int16_t coeffCntr;
   for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
   {
     data[coeffCntr] = log(data[coeffCntr]) - WebRtcIsac_kMeanLpcGain;
@@ -569,12 +569,12 @@
  * Output:
  *      -out                : decorrelated parameters.
  */
-WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
+int16_t WebRtcIsac_DecorrelateLPGain(
     const double* data,
     double* out)
 {
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
+  int16_t rowCntr;
+  int16_t colCntr;
 
   for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++)
   {
@@ -604,7 +604,7 @@
     double* data,
     int*    idx)
 {
-  WebRtc_Word16 coeffCntr;
+  int16_t coeffCntr;
   for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
   {
     *idx = (int)floor((*data - WebRtcIsac_kLeftRecPointLpcGain[coeffCntr]) /
@@ -638,11 +638,11 @@
  * Output:
  *      -lpcGains           : quantized values of the given parametes.
  */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
+int16_t WebRtcIsac_DequantizeLpcGain(
     const int* idx,
     double*    out)
 {
-  WebRtc_Word16 coeffCntr;
+  int16_t coeffCntr;
   for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
   {
     *out = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx *
@@ -664,12 +664,12 @@
  * Output:
  *      -out                : correlated parameters.
  */
-WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
+int16_t WebRtcIsac_CorrelateLpcGain(
     const double* data,
     double* out)
 {
-  WebRtc_Word16 rowCntr;
-  WebRtc_Word16 colCntr;
+  int16_t rowCntr;
+  int16_t colCntr;
 
   for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++)
   {
@@ -696,10 +696,10 @@
  * Output:
  *      -lpcGain            : LPC gain in normal domain.
  */
-WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
+int16_t WebRtcIsac_AddMeanToLinearDomain(
     double* lpcGains)
 {
-  WebRtc_Word16 coeffCntr;
+  int16_t coeffCntr;
   for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
   {
     lpcGains[coeffCntr] = exp(lpcGains[coeffCntr] + WebRtcIsac_kMeanLpcGain);
diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
index e7f1a76..eab98c1 100644
--- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
+++ b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
@@ -40,9 +40,9 @@
  *
  *
  */
-WebRtc_Word16 WebRtcIsac_RemoveLarMean(
+int16_t WebRtcIsac_RemoveLarMean(
     double*     lar,
-    WebRtc_Word16 bandwidth);
+    int16_t bandwidth);
 
 /******************************************************************************
  * WebRtcIsac_DecorrelateIntraVec()
@@ -60,10 +60,10 @@
  * Output:
  *      -out                : decorrelated LAR vectors.
  */
-WebRtc_Word16 WebRtcIsac_DecorrelateIntraVec(
+int16_t WebRtcIsac_DecorrelateIntraVec(
     const double* inLAR,
     double*       out,
-    WebRtc_Word16   bandwidth);
+    int16_t   bandwidth);
 
 
 /******************************************************************************
@@ -83,10 +83,10 @@
  * Output:
  *      -out                : decorrelated LAR vectors.
  */
-WebRtc_Word16 WebRtcIsac_DecorrelateInterVec(
+int16_t WebRtcIsac_DecorrelateInterVec(
     const double* data,
     double*       out,
-    WebRtc_Word16   bandwidth);
+    int16_t   bandwidth);
 
 
 /******************************************************************************
@@ -106,7 +106,7 @@
 double WebRtcIsac_QuantizeUncorrLar(
     double*     data,
     int*        idx,
-    WebRtc_Word16 bandwidth);
+    int16_t bandwidth);
 
 
 /******************************************************************************
@@ -122,10 +122,10 @@
  * Output:
  *      -out                : correlated parametrs.
  */
-WebRtc_Word16 WebRtcIsac_CorrelateIntraVec(
+int16_t WebRtcIsac_CorrelateIntraVec(
     const double* data,
     double*       out,
-    WebRtc_Word16   bandwidth);
+    int16_t   bandwidth);
 
 
 /******************************************************************************
@@ -141,10 +141,10 @@
  * Output:
  *      -out                : correlated parametrs.
  */
-WebRtc_Word16 WebRtcIsac_CorrelateInterVec(
+int16_t WebRtcIsac_CorrelateInterVec(
     const double* data,
     double*       out,
-    WebRtc_Word16   bandwidth);
+    int16_t   bandwidth);
 
 
 /******************************************************************************
@@ -160,9 +160,9 @@
  * Output:
  *      -data               : pointer to LARs.
  */
-WebRtc_Word16 WebRtcIsac_AddLarMean(
+int16_t WebRtcIsac_AddLarMean(
     double*     data,
-    WebRtc_Word16 bandwidth);
+    int16_t bandwidth);
 
 
 /******************************************************************************
@@ -178,10 +178,10 @@
  * Output:
  *      -out                : pointer to quantized values.
  */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcParam(
+int16_t WebRtcIsac_DequantizeLpcParam(
     const int*  idx,
     double*     out,
-    WebRtc_Word16 bandwidth);
+    int16_t bandwidth);
 
 
 /******************************************************************************
@@ -195,7 +195,7 @@
  * Output:
  *      -lpcGain            : mean-removed in log domain.
  */
-WebRtc_Word16 WebRtcIsac_ToLogDomainRemoveMean(
+int16_t WebRtcIsac_ToLogDomainRemoveMean(
     double* lpGains);
 
 
@@ -211,7 +211,7 @@
  * Output:
  *      -out                : decorrelated parameters.
  */
-WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
+int16_t WebRtcIsac_DecorrelateLPGain(
     const double* data,
     double*       out);
 
@@ -244,7 +244,7 @@
  * Output:
  *      -lpcGains           : quantized values of the given parametes.
  */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
+int16_t WebRtcIsac_DequantizeLpcGain(
     const int* idx,
     double*    lpGains);
 
@@ -260,7 +260,7 @@
  * Output:
  *      -out                : correlated parameters.
  */
-WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
+int16_t WebRtcIsac_CorrelateLpcGain(
     const double* data,
     double*       out);
 
@@ -276,7 +276,7 @@
  * Output:
  *      -lpcGain            : LPC gain in normal domain.
  */
-WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
+int16_t WebRtcIsac_AddMeanToLinearDomain(
     double* lpcGains);
 
 
diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
index 0ef1b38..66bf06d 100644
--- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
+++ b/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
@@ -34,32 +34,32 @@
 #include <math.h>
 #include <string.h>
 
-static const WebRtc_UWord16 kLpcVecPerSegmentUb12 = 5;
-static const WebRtc_UWord16 kLpcVecPerSegmentUb16 = 4;
+static const uint16_t kLpcVecPerSegmentUb12 = 5;
+static const uint16_t kLpcVecPerSegmentUb16 = 4;
 
 /* CDF array for encoder bandwidth (12 vs 16 kHz) indicator. */
-static const WebRtc_UWord16 kOneBitEqualProbCdf[3] = {
+static const uint16_t kOneBitEqualProbCdf[3] = {
     0, 32768, 65535 };
 
 /* Pointer to cdf array for encoder bandwidth (12 vs 16 kHz) indicator. */
-static const WebRtc_UWord16* kOneBitEqualProbCdf_ptr[1] = {
+static const uint16_t* kOneBitEqualProbCdf_ptr[1] = {
     kOneBitEqualProbCdf };
 
 /*
  * Initial cdf index for decoder of encoded bandwidth
  * (12 vs 16 kHz) indicator.
  */
-static const WebRtc_UWord16 kOneBitEqualProbInitIndex[1] = { 1 };
+static const uint16_t kOneBitEqualProbInitIndex[1] = { 1 };
 
 
 static const int kIsSWB12 = 1;
 
 /* compute correlation from power spectrum */
-static void FindCorrelation(WebRtc_Word32* PSpecQ12, WebRtc_Word32* CorrQ7) {
-  WebRtc_Word32 summ[FRAMESAMPLES / 8];
-  WebRtc_Word32 diff[FRAMESAMPLES / 8];
-  const WebRtc_Word16* CS_ptrQ9;
-  WebRtc_Word32 sum;
+static void FindCorrelation(int32_t* PSpecQ12, int32_t* CorrQ7) {
+  int32_t summ[FRAMESAMPLES / 8];
+  int32_t diff[FRAMESAMPLES / 8];
+  const int16_t* CS_ptrQ9;
+  int32_t sum;
   int k, n;
 
   for (k = 0; k < FRAMESAMPLES / 8; k++) {
@@ -92,15 +92,15 @@
 
 /* compute inverse AR power spectrum */
 /* Changed to the function used in iSAC FIX for compatibility reasons */
-static void FindInvArSpec(const WebRtc_Word16* ARCoefQ12,
-                          const WebRtc_Word32 gainQ10,
-                          WebRtc_Word32* CurveQ16) {
-  WebRtc_Word32 CorrQ11[AR_ORDER + 1];
-  WebRtc_Word32 sum, tmpGain;
-  WebRtc_Word32 diffQ16[FRAMESAMPLES / 8];
-  const WebRtc_Word16* CS_ptrQ9;
+static void FindInvArSpec(const int16_t* ARCoefQ12,
+                          const int32_t gainQ10,
+                          int32_t* CurveQ16) {
+  int32_t CorrQ11[AR_ORDER + 1];
+  int32_t sum, tmpGain;
+  int32_t diffQ16[FRAMESAMPLES / 8];
+  const int16_t* CS_ptrQ9;
   int k, n;
-  WebRtc_Word16 round, shftVal = 0, sh;
+  int16_t round, shftVal = 0, sh;
 
   sum = 0;
   for (n = 0; n < AR_ORDER + 1; n++) {
@@ -174,10 +174,10 @@
 }
 
 /* Generate array of dither samples in Q7. */
-static void GenerateDitherQ7Lb(WebRtc_Word16* bufQ7, WebRtc_UWord32 seed,
-                               int length, WebRtc_Word16 AvgPitchGain_Q12) {
+static void GenerateDitherQ7Lb(int16_t* bufQ7, uint32_t seed,
+                               int length, int16_t AvgPitchGain_Q12) {
   int   k, shft;
-  WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14;
+  int16_t dither1_Q7, dither2_Q7, dither_gain_Q14;
 
   /* This threshold should be equal to that in decode_spec(). */
   if (AvgPitchGain_Q12 < 614) {
@@ -187,13 +187,13 @@
 
       /* Fixed-point dither sample between -64 and 64 (Q7). */
       /* dither = seed * 128 / 4294967295 */
-      dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216) >> 25);
+      dither1_Q7 = (int16_t)(((int)seed + 16777216) >> 25);
 
       /* New random unsigned int. */
       seed = (seed * 196314165) + 907633515;
 
       /* Fixed-point dither sample between -64 and 64. */
-      dither2_Q7 = (WebRtc_Word16)(((int)seed + 16777216) >> 25);
+      dither2_Q7 = (int16_t)(((int)seed + 16777216) >> 25);
 
       shft = (seed >> 25) & 15;
       if (shft < 5) {
@@ -211,7 +211,7 @@
       }
     }
   } else {
-    dither_gain_Q14 = (WebRtc_Word16)(22528 - 10 * AvgPitchGain_Q12);
+    dither_gain_Q14 = (int16_t)(22528 - 10 * AvgPitchGain_Q12);
 
     /* Dither on half of the coefficients. */
     for (k = 0; k < length - 1; k += 2) {
@@ -219,7 +219,7 @@
       seed = (seed * 196314165) + 907633515;
 
       /* Fixed-point dither sample between -64 and 64. */
-      dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216) >> 25);
+      dither1_Q7 = (int16_t)(((int)seed + 16777216) >> 25);
 
       /* Dither sample is placed in either even or odd index. */
       shft = (seed >> 25) & 1;     /* Either 0 or 1 */
@@ -249,8 +249,8 @@
  *      -bufQ7              : pointer to a buffer where dithers are written to.
  */
 static void GenerateDitherQ7LbUB(
-    WebRtc_Word16* bufQ7,
-    WebRtc_UWord32 seed,
+    int16_t* bufQ7,
+    uint32_t seed,
     int length) {
   int k;
   for (k = 0; k < length; k++) {
@@ -259,10 +259,10 @@
 
     /* Fixed-point dither sample between -64 and 64 (Q7). */
     /* bufQ7 = seed * 128 / 4294967295 */
-    bufQ7[k] = (WebRtc_Word16)(((int)seed + 16777216) >> 25);
+    bufQ7[k] = (int16_t)(((int)seed + 16777216) >> 25);
 
     /* Scale by 0.35. */
-    bufQ7[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(bufQ7[k], 2048, 13);
+    bufQ7[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(bufQ7[k], 2048, 13);
   }
 }
 
@@ -270,18 +270,18 @@
  * Function to decode the complex spectrum from the bit stream
  * returns the total number of bytes in the stream.
  */
-int WebRtcIsac_DecodeSpec(Bitstr* streamdata, WebRtc_Word16 AvgPitchGain_Q12,
+int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
                           enum ISACBand band, double* fr, double* fi) {
-  WebRtc_Word16  DitherQ7[FRAMESAMPLES];
-  WebRtc_Word16  data[FRAMESAMPLES];
-  WebRtc_Word32  invARSpec2_Q16[FRAMESAMPLES_QUARTER];
-  WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
-  WebRtc_Word16  ARCoefQ12[AR_ORDER + 1];
-  WebRtc_Word16  RCQ15[AR_ORDER];
-  WebRtc_Word16  gainQ10;
-  WebRtc_Word32  gain2_Q10, res;
-  WebRtc_Word32  in_sqrt;
-  WebRtc_Word32  newRes;
+  int16_t  DitherQ7[FRAMESAMPLES];
+  int16_t  data[FRAMESAMPLES];
+  int32_t  invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+  uint16_t invARSpecQ8[FRAMESAMPLES_QUARTER];
+  int16_t  ARCoefQ12[AR_ORDER + 1];
+  int16_t  RCQ15[AR_ORDER];
+  int16_t  gainQ10;
+  int32_t  gain2_Q10, res;
+  int32_t  in_sqrt;
+  int32_t  newRes;
   int k, len, i;
   int is_12khz = !kIsSWB12;
   int num_dft_coeff = FRAMESAMPLES;
@@ -326,7 +326,7 @@
       newRes = (in_sqrt / res + res) >> 1;
     } while (newRes != res && i-- > 0);
 
-    invARSpecQ8[k] = (WebRtc_Word16)newRes;
+    invARSpecQ8[k] = (int16_t)newRes;
   }
 
   len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8, DitherQ7,
@@ -339,8 +339,8 @@
   switch (band) {
     case kIsacLowerBand: {
       /* Scale down spectral samples with low SNR. */
-      WebRtc_Word32 p1;
-      WebRtc_Word32 p2;
+      int32_t p1;
+      int32_t p2;
       if (AvgPitchGain_Q12 <= 614) {
         p1 = 30 << 10;
         p2 = 32768 + (33 << 16);
@@ -349,7 +349,7 @@
         p2 = 32768 + (40 << 16);
       }
       for (k = 0; k < FRAMESAMPLES; k += 4) {
-        gainQ10 = WebRtcSpl_DivW32W16ResW16(p1, (WebRtc_Word16)(
+        gainQ10 = WebRtcSpl_DivW32W16ResW16(p1, (int16_t)(
             (invARSpec2_Q16[k >> 2] + p2) >> 16));
         *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
         *fi++ = (double)((data[k + 1] * gainQ10 + 512) >> 10) / 128.0;
@@ -391,26 +391,26 @@
 }
 
 
-int WebRtcIsac_EncodeSpec(const WebRtc_Word16* fr, const WebRtc_Word16* fi,
-                          WebRtc_Word16 AvgPitchGain_Q12, enum ISACBand band,
+int WebRtcIsac_EncodeSpec(const int16_t* fr, const int16_t* fi,
+                          int16_t AvgPitchGain_Q12, enum ISACBand band,
                           Bitstr* streamdata) {
-  WebRtc_Word16 ditherQ7[FRAMESAMPLES];
-  WebRtc_Word16 dataQ7[FRAMESAMPLES];
-  WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
-  WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
-  WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
-  WebRtc_Word32 CorrQ7[AR_ORDER + 1];
-  WebRtc_Word32 CorrQ7_norm[AR_ORDER + 1];
-  WebRtc_Word16 RCQ15[AR_ORDER];
-  WebRtc_Word16 ARCoefQ12[AR_ORDER + 1];
-  WebRtc_Word32 gain2_Q10;
-  WebRtc_Word16 val;
-  WebRtc_Word32 nrg, res;
-  WebRtc_UWord32 sum;
-  WebRtc_Word32 in_sqrt;
-  WebRtc_Word32 newRes;
-  WebRtc_Word16 err;
-  WebRtc_UWord32 nrg_u32;
+  int16_t ditherQ7[FRAMESAMPLES];
+  int16_t dataQ7[FRAMESAMPLES];
+  int32_t PSpec[FRAMESAMPLES_QUARTER];
+  int32_t invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+  uint16_t invARSpecQ8[FRAMESAMPLES_QUARTER];
+  int32_t CorrQ7[AR_ORDER + 1];
+  int32_t CorrQ7_norm[AR_ORDER + 1];
+  int16_t RCQ15[AR_ORDER];
+  int16_t ARCoefQ12[AR_ORDER + 1];
+  int32_t gain2_Q10;
+  int16_t val;
+  int32_t nrg, res;
+  uint32_t sum;
+  int32_t in_sqrt;
+  int32_t newRes;
+  int16_t err;
+  uint32_t nrg_u32;
   int shift_var;
   int k, n, j, i;
   int is_12khz = !kIsSWB12;
@@ -542,7 +542,7 @@
     }
   }
 
-  nrg_u32 = (WebRtc_UWord32)nrg;
+  nrg_u32 = (uint32_t)nrg;
   if (shift_var > 0) {
     nrg_u32 = nrg_u32 >> shift_var;
   } else {
@@ -551,7 +551,7 @@
   if (nrg_u32 > 0x7FFFFFFF) {
     nrg = 0x7FFFFFFF;
   }  else {
-    nrg = (WebRtc_Word32)nrg_u32;
+    nrg = (int32_t)nrg_u32;
   }
   /* Also shifts 31 bits to the left! */
   gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg);
@@ -579,7 +579,7 @@
       newRes = (in_sqrt / res + res) >> 1;
     } while (newRes != res && i-- > 0);
 
-    invARSpecQ8[k] = (WebRtc_Word16)newRes;
+    invARSpecQ8[k] = (int16_t)newRes;
   }
   /* arithmetic coding of spectrum */
   err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
@@ -682,13 +682,13 @@
 }
 
 
-WebRtc_Word16 WebRtcIsac_Poly2LarUB(double* lpcVecs, WebRtc_Word16 bandwidth) {
+int16_t WebRtcIsac_Poly2LarUB(double* lpcVecs, int16_t bandwidth) {
   double      poly[MAX_ORDER];
   double      rc[MAX_ORDER];
   double*     ptrIO;
-  WebRtc_Word16 vecCntr;
-  WebRtc_Word16 vecSize;
-  WebRtc_Word16 numVec;
+  int16_t vecCntr;
+  int16_t vecSize;
+  int16_t numVec;
 
   vecSize = UB_LPC_ORDER;
   switch (bandwidth) {
@@ -791,16 +791,16 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata,
-                                             double* percepFilterParams,
-                                             WebRtc_Word16 bandwidth) {
+int16_t WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata,
+                                       double* percepFilterParams,
+                                       int16_t bandwidth) {
   double lpcCoeff[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
   int err;
   int interpolCntr;
   int subframeCntr;
-  WebRtc_Word16 numSegments;
-  WebRtc_Word16 numVecPerSegment;
-  WebRtc_Word16 numGains;
+  int16_t numSegments;
+  int16_t numVecPerSegment;
+  int16_t numGains;
 
   double percepFilterGains[SUBFRAMES << 1];
   double* ptrOutParam = percepFilterParams;
@@ -1181,9 +1181,9 @@
 }
 
 
-WebRtc_Word16 WebRtcIsac_EncodeLpcUB(double* lpcVecs, Bitstr* streamdata,
-                                     double* interpolLPCCoeff,
-                                     WebRtc_Word16 bandwidth,
+int16_t WebRtcIsac_EncodeLpcUB(double* lpcVecs, Bitstr* streamdata,
+                               double* interpolLPCCoeff,
+                               int16_t bandwidth,
                                      ISACUBSaveEncDataStruct* encData) {
   double    U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
   int     idx[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
@@ -1402,7 +1402,7 @@
 
 
 
-WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata) {
+int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata) {
   double U[UB_LPC_GAIN_DIM];
   int idx[UB_LPC_GAIN_DIM];
   int err;
@@ -1422,7 +1422,7 @@
 
 
 /* decode & dequantize RC */
-int WebRtcIsac_DecodeRc(Bitstr* streamdata, WebRtc_Word16* RCQ15) {
+int WebRtcIsac_DecodeRc(Bitstr* streamdata, int16_t* RCQ15) {
   int k, err;
   int index[AR_ORDER];
 
@@ -1442,7 +1442,7 @@
 
 
 /* quantize & code RC */
-void WebRtcIsac_EncodeRc(WebRtc_Word16* RCQ15, Bitstr* streamdata) {
+void WebRtcIsac_EncodeRc(int16_t* RCQ15, Bitstr* streamdata) {
   int k;
   int index[AR_ORDER];
 
@@ -1466,7 +1466,7 @@
 
 
 /* decode & dequantize squared Gain */
-int WebRtcIsac_DecodeGain2(Bitstr* streamdata, WebRtc_Word32* gainQ10) {
+int WebRtcIsac_DecodeGain2(Bitstr* streamdata, int32_t* gainQ10) {
   int index, err;
 
   /* entropy decoding of quantization index */
@@ -1483,7 +1483,7 @@
 
 
 /* quantize & code squared Gain */
-int WebRtcIsac_EncodeGain2(WebRtc_Word32* gainQ10, Bitstr* streamdata) {
+int WebRtcIsac_EncodeGain2(int32_t* gainQ10, Bitstr* streamdata) {
   int index;
 
   /* find quantization index */
@@ -1508,9 +1508,9 @@
 
 /* decode & dequantize Pitch Gains */
 int WebRtcIsac_DecodePitchGain(Bitstr* streamdata,
-                               WebRtc_Word16* PitchGains_Q12) {
+                               int16_t* PitchGains_Q12) {
   int index_comb, err;
-  const WebRtc_UWord16* WebRtcIsac_kQPitchGainCdf_ptr[1];
+  const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1];
 
   /* Entropy decoding of quantization indices */
   *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
@@ -1531,7 +1531,7 @@
 
 
 /* Quantize & code Pitch Gains. */
-void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12,
+void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12,
                                 Bitstr* streamdata,
                                 ISAC_SaveEncData_t* encData) {
   int k, j;
@@ -1539,7 +1539,7 @@
   double S[PITCH_SUBFRAMES];
   int index[3];
   int index_comb;
-  const WebRtc_UWord16* WebRtcIsac_kQPitchGainCdf_ptr[1];
+  const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1];
   double PitchGains[PITCH_SUBFRAMES] = {0, 0, 0, 0};
 
   /* Take the asin. */
@@ -1589,7 +1589,7 @@
 
 /* Pitch LAG */
 /* Decode & de-quantize Pitch Lags. */
-int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, WebRtc_Word16* PitchGain_Q12,
+int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, int16_t* PitchGain_Q12,
                               double* PitchLags) {
   int k, err;
   double StepSize;
@@ -1597,10 +1597,10 @@
   int index[PITCH_SUBFRAMES];
   double mean_gain;
   const double* mean_val2, *mean_val3, *mean_val4;
-  const WebRtc_Word16* lower_limit;
-  const WebRtc_UWord16* init_index;
-  const WebRtc_UWord16* cdf_size;
-  const WebRtc_UWord16** cdf;
+  const int16_t* lower_limit;
+  const uint16_t* init_index;
+  const uint16_t* cdf_size;
+  const uint16_t** cdf;
   double PitchGain[4] = {0, 0, 0, 0};
 
   /* compute mean pitch gain */
@@ -1676,7 +1676,7 @@
 
 
 /* Quantize & code pitch lags. */
-void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12,
+void WebRtcIsac_EncodePitchLag(double* PitchLags, int16_t* PitchGain_Q12,
                                Bitstr* streamdata,
                                ISAC_SaveEncData_t* encData) {
   int k, j;
@@ -1685,8 +1685,8 @@
   int index[PITCH_SUBFRAMES];
   double mean_gain;
   const double* mean_val2, *mean_val3, *mean_val4;
-  const WebRtc_Word16* lower_limit, *upper_limit;
-  const WebRtc_UWord16** cdf;
+  const int16_t* lower_limit, *upper_limit;
+  const uint16_t** cdf;
   double PitchGain[4] = {0, 0, 0, 0};
 
   /* compute mean pitch gain */
@@ -1777,18 +1777,18 @@
 
 
 /* cdf array for frame length indicator */
-const WebRtc_UWord16 WebRtcIsac_kFrameLengthCdf[4] = {
+const uint16_t WebRtcIsac_kFrameLengthCdf[4] = {
     0, 21845, 43690, 65535 };
 
 /* pointer to cdf array for frame length indicator */
-const WebRtc_UWord16* WebRtcIsac_kFrameLengthCdf_ptr[1] = {
+const uint16_t* WebRtcIsac_kFrameLengthCdf_ptr[1] = {
     WebRtcIsac_kFrameLengthCdf };
 
 /* initial cdf index for decoder of frame length indicator */
-const WebRtc_UWord16 WebRtcIsac_kFrameLengthInitIndex[1] = { 1 };
+const uint16_t WebRtcIsac_kFrameLengthInitIndex[1] = { 1 };
 
 
-int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, WebRtc_Word16* framesamples) {
+int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framesamples) {
   int frame_mode, err;
   err = 0;
   /* entropy decoding of frame length [1:30ms,2:60ms] */
@@ -1811,7 +1811,7 @@
   return err;
 }
 
-int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr* streamdata) {
+int WebRtcIsac_EncodeFrameLen(int16_t framesamples, Bitstr* streamdata) {
   int frame_mode, status;
 
   status = 0;
@@ -1837,19 +1837,19 @@
 }
 
 /* cdf array for estimated bandwidth */
-static const WebRtc_UWord16 kBwCdf[25] = {
+static const uint16_t kBwCdf[25] = {
     0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
     32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
     62804, 65535 };
 
 /* pointer to cdf array for estimated bandwidth */
-static const WebRtc_UWord16* kBwCdfPtr[1] = { kBwCdf };
+static const uint16_t* kBwCdfPtr[1] = { kBwCdf };
 
 /* initial cdf index for decoder of estimated bandwidth*/
-static const WebRtc_UWord16 kBwInitIndex[1] = { 7 };
+static const uint16_t kBwInitIndex[1] = { 7 };
 
 
-int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, WebRtc_Word16* BWno) {
+int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, int16_t* BWno) {
   int BWno32, err;
 
   /* entropy decoding of sender's BW estimation [0..23] */
@@ -1858,7 +1858,7 @@
   if (err < 0) {
     return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
   }
-  *BWno = (WebRtc_Word16)BWno32;
+  *BWno = (int16_t)BWno32;
   return err;
 }
 
@@ -1950,7 +1950,7 @@
 /* Decode & de-quantize LPC Coefficients. */
 int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs,
                                double* percepFilterGains,
-                               WebRtc_Word16 bandwidth) {
+                               int16_t bandwidth) {
   int  index_s[KLT_ORDER_SHAPE];
 
   double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
@@ -1993,8 +1993,8 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
-                                         Bitstr* streamData) {
+int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
+                                   Bitstr* streamData) {
   int bandwidthMode;
   switch (bandwidth) {
     case isac12kHz: {
@@ -2013,8 +2013,8 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
-                                         enum ISACBandwidth* bandwidth) {
+int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
+                                   enum ISACBandwidth* bandwidth) {
   int bandwidthMode;
   if (WebRtcIsac_DecHistOneStepMulti(&bandwidthMode, streamData,
                                      kOneBitEqualProbCdf_ptr,
@@ -2036,8 +2036,8 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(WebRtc_Word32 jitterIndex,
-                                          Bitstr* streamData) {
+int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
+                                    Bitstr* streamData) {
   /* This is to avoid LINUX warning until we change 'int' to 'Word32'. */
   int intVar;
 
@@ -2051,8 +2051,8 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
-                                          WebRtc_Word32* jitterInfo) {
+int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
+                                    int32_t* jitterInfo) {
   int intVar;
   /* Use the same CDF table as for bandwidth
    * both take two values with equal probability. */
@@ -2061,6 +2061,6 @@
                                      kOneBitEqualProbInitIndex, 1) < 0) {
     return -ISAC_RANGE_ERROR_DECODE_BANDWITH;
   }
-  *jitterInfo = (WebRtc_Word16)(intVar);
+  *jitterInfo = (int16_t)(intVar);
   return 0;
 }
diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h b/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
index 1a46953..1a86ce2 100644
--- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
+++ b/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
@@ -46,7 +46,7 @@
  * Return value             : < 0 if an error occures
  *                              0 if succeeded.
  */
-int WebRtcIsac_DecodeSpec(Bitstr* streamdata, WebRtc_Word16 AvgPitchGain_Q12,
+int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
                           enum ISACBand band, double* fr, double* fi);
 
 /******************************************************************************
@@ -72,15 +72,15 @@
  * Return value             : < 0 if an error occures
  *                              0 if succeeded.
  */
-int WebRtcIsac_EncodeSpec(const WebRtc_Word16* fr, const WebRtc_Word16* fi,
-                          WebRtc_Word16 AvgPitchGain_Q12, enum ISACBand band,
+int WebRtcIsac_EncodeSpec(const int16_t* fr, const int16_t* fi,
+                          int16_t AvgPitchGain_Q12, enum ISACBand band,
                           Bitstr* streamdata);
 
 /* decode & dequantize LPC Coef */
 int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef);
 int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs,
                                double* percepFilterGains,
-                               WebRtc_Word16 bandwidth);
+                               int16_t bandwidth);
 
 int WebRtcIsac_DecodeLpc(Bitstr* streamdata, double* LPCCoef_lo,
                          double* LPCCoef_hi);
@@ -126,10 +126,10 @@
  * Return value             : 0 if encoding is successful,
  *                           <0 if failed to encode.
  */
-WebRtc_Word16 WebRtcIsac_EncodeLpcUB(double* lpcCoeff, Bitstr* streamdata,
-                                     double* interpolLPCCoeff,
-                                     WebRtc_Word16 bandwidth,
-                                     ISACUBSaveEncDataStruct* encData);
+int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff, Bitstr* streamdata,
+                               double* interpolLPCCoeff,
+                               int16_t bandwidth,
+                               ISACUBSaveEncDataStruct* encData);
 
 /******************************************************************************
  * WebRtcIsac_DecodeInterpolLpcUb()
@@ -159,37 +159,37 @@
  * Return value             : 0 if encoding is successful,
  *                           <0 if failed to encode.
  */
-WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata,
-                                             double* percepFilterParam,
-                                             WebRtc_Word16 bandwidth);
+int16_t WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata,
+                                       double* percepFilterParam,
+                                       int16_t bandwidth);
 
 /* Decode & dequantize RC */
-int WebRtcIsac_DecodeRc(Bitstr* streamdata, WebRtc_Word16* RCQ15);
+int WebRtcIsac_DecodeRc(Bitstr* streamdata, int16_t* RCQ15);
 
 /* Quantize & code RC */
-void WebRtcIsac_EncodeRc(WebRtc_Word16* RCQ15, Bitstr* streamdata);
+void WebRtcIsac_EncodeRc(int16_t* RCQ15, Bitstr* streamdata);
 
 /* Decode & dequantize squared Gain */
-int WebRtcIsac_DecodeGain2(Bitstr* streamdata, WebRtc_Word32* Gain2);
+int WebRtcIsac_DecodeGain2(Bitstr* streamdata, int32_t* Gain2);
 
 /* Quantize & code squared Gain (input is squared gain) */
-int WebRtcIsac_EncodeGain2(WebRtc_Word32* gain2, Bitstr* streamdata);
+int WebRtcIsac_EncodeGain2(int32_t* gain2, Bitstr* streamdata);
 
-void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12,
+void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12,
                                 Bitstr* streamdata,
                                 ISAC_SaveEncData_t* encData);
 
-void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12,
+void WebRtcIsac_EncodePitchLag(double* PitchLags, int16_t* PitchGain_Q12,
                                Bitstr* streamdata, ISAC_SaveEncData_t* encData);
 
 int WebRtcIsac_DecodePitchGain(Bitstr* streamdata,
-                               WebRtc_Word16* PitchGain_Q12);
-int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, WebRtc_Word16* PitchGain_Q12,
+                               int16_t* PitchGain_Q12);
+int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, int16_t* PitchGain_Q12,
                               double* PitchLag);
 
-int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, WebRtc_Word16* framelength);
-int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framelength, Bitstr* streamdata);
-int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, WebRtc_Word16* BWno);
+int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength);
+int WebRtcIsac_EncodeFrameLen(int16_t framelength, Bitstr* streamdata);
+int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, int16_t* BWno);
 void WebRtcIsac_EncodeReceiveBw(int* BWno, Bitstr* streamdata);
 
 /* Step-down */
@@ -253,7 +253,7 @@
  * Return value             : 0 if succeeded.
  *                           <0 if failed.
  */
-WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
+int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
 
 
 /******************************************************************************
@@ -272,8 +272,8 @@
  * Return value             : 0 if succeeded.
  *                           <0 if failed.
  */
-WebRtc_Word16 WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
-                                         Bitstr* streamData);
+int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
+                                   Bitstr* streamData);
 
 
 /******************************************************************************
@@ -293,8 +293,8 @@
  * Return value             : 0 if succeeded.
  *                           <0 if failed.
  */
-WebRtc_Word16 WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
-                                         enum ISACBandwidth* bandwidth);
+int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
+                                   enum ISACBandwidth* bandwidth);
 
 
 /******************************************************************************
@@ -314,8 +314,8 @@
  * Return value             : 0 if succeeded.
  *                           <0 if failed.
  */
-WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(WebRtc_Word32 jitterIndex,
-                                          Bitstr* streamData);
+int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
+                                    Bitstr* streamData);
 
 
 /******************************************************************************
@@ -335,7 +335,7 @@
  * Return value             : 0 if succeeded.
  *                           <0 if failed.
  */
-WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
-                                          WebRtc_Word32* jitterInfo);
+int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
+                                    int32_t* jitterInfo);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c
index 428fda8..ed794a5 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -55,10 +55,10 @@
  *
  */
 static void UpdatePayloadSizeLimit(ISACMainStruct* instISAC) {
-  WebRtc_Word16 lim30MsPayloadBytes = WEBRTC_SPL_MIN(
+  int16_t lim30MsPayloadBytes = WEBRTC_SPL_MIN(
                           (instISAC->maxPayloadSizeBytes),
                           (instISAC->maxRateBytesPer30Ms));
-  WebRtc_Word16 lim60MsPayloadBytes = WEBRTC_SPL_MIN(
+  int16_t lim60MsPayloadBytes = WEBRTC_SPL_MIN(
                           (instISAC->maxPayloadSizeBytes),
                           (instISAC->maxRateBytesPer30Ms << 1));
 
@@ -112,7 +112,7 @@
   if ((instISAC->codingMode == 0) &&
       (instISAC->instLB.ISACencLB_obj.buffer_index == 0) &&
       (instISAC->instLB.ISACencLB_obj.frame_nb == 0)) {
-    WebRtc_Word32 bottleneck;
+    int32_t bottleneck;
     WebRtcIsac_GetUplinkBandwidth(&(instISAC->bwestimator_obj),
                                   &bottleneck);
 
@@ -190,8 +190,8 @@
  *
  */
 static void GetSendBandwidthInfo(ISACMainStruct* instISAC,
-                                 WebRtc_Word16* bandwidthIndex,
-                                 WebRtc_Word16* jitterInfo) {
+                                 int16_t* bandwidthIndex,
+                                 int16_t* jitterInfo) {
   if ((instISAC->instLB.ISACencLB_obj.buffer_index ==
       (FRAMESAMPLES_10ms << 1)) &&
       (instISAC->instLB.ISACencLB_obj.frame_nb == 0)) {
@@ -216,8 +216,8 @@
  * Return value               : 0 - Ok
  *                             -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_AssignSize(int* sizeInBytes) {
-  *sizeInBytes = sizeof(ISACMainStruct) * 2 / sizeof(WebRtc_Word16);
+int16_t WebRtcIsac_AssignSize(int* sizeInBytes) {
+  *sizeInBytes = sizeof(ISACMainStruct) * 2 / sizeof(int16_t);
   return 0;
 }
 
@@ -235,8 +235,8 @@
  * Return value               : 0 - Ok
  *                             -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
-                                void* instISAC_Addr) {
+int16_t WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
+                          void* instISAC_Addr) {
   if (instISAC_Addr != NULL) {
     ISACMainStruct* instISAC = (ISACMainStruct*)instISAC_Addr;
     instISAC->errorCode = 0;
@@ -269,7 +269,7 @@
  * Return value               : 0 - Ok
  *                             -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
+int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
   ISACMainStruct* instISAC;
 
   instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1);
@@ -300,7 +300,7 @@
  * Return value               : 0 - Ok
  *                             -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_Free(ISACStruct* ISAC_main_inst) {
+int16_t WebRtcIsac_Free(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   WEBRTC_SPL_FREE(instISAC);
   return 0;
@@ -329,10 +329,10 @@
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-static WebRtc_Word16 EncoderInitLb(ISACLBStruct* instLB,
-                                   WebRtc_Word16 codingMode,
-                                   enum IsacSamplingRate sampRate) {
-  WebRtc_Word16 statusInit = 0;
+static int16_t EncoderInitLb(ISACLBStruct* instLB,
+                             int16_t codingMode,
+                             enum IsacSamplingRate sampRate) {
+  int16_t statusInit = 0;
   int k;
 
   /* Init stream vector to zero */
@@ -371,9 +371,9 @@
   return statusInit;
 }
 
-static WebRtc_Word16 EncoderInitUb(ISACUBStruct* instUB,
-                                   WebRtc_Word16 bandwidth) {
-  WebRtc_Word16 statusInit = 0;
+static int16_t EncoderInitUb(ISACUBStruct* instUB,
+                             int16_t bandwidth) {
+  int16_t statusInit = 0;
   int k;
 
   /* Init stream vector to zero. */
@@ -406,10 +406,10 @@
 }
 
 
-WebRtc_Word16 WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
-                                     WebRtc_Word16 codingMode) {
+int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
+                               int16_t codingMode) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  WebRtc_Word16 status;
+  int16_t status;
 
   if ((codingMode != 0) && (codingMode != 1)) {
     instISAC->errorCode = ISAC_DISALLOWED_CODING_MODE;
@@ -449,9 +449,9 @@
   if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
     /* Initialize encoder filter-bank. */
     memset(instISAC->analysisFBState1, 0,
-           FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+           FB_STATE_SIZE_WORD32 * sizeof(int32_t));
     memset(instISAC->analysisFBState2, 0,
-           FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+           FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
     status = EncoderInitUb(&(instISAC->instUB),
                            instISAC->bandwidthKHz);
@@ -489,21 +489,21 @@
  *                                 samples.
  *                            : -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
-                                const WebRtc_Word16* speechIn,
-                                WebRtc_Word16* encoded) {
+int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
+                          const int16_t* speechIn,
+                          int16_t* encoded) {
   float inFrame[FRAMESAMPLES_10ms];
-  WebRtc_Word16 speechInLB[FRAMESAMPLES_10ms];
-  WebRtc_Word16 speechInUB[FRAMESAMPLES_10ms];
-  WebRtc_Word16 streamLenLB = 0;
-  WebRtc_Word16 streamLenUB = 0;
-  WebRtc_Word16 streamLen = 0;
-  WebRtc_Word16 k = 0;
-  WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
+  int16_t speechInLB[FRAMESAMPLES_10ms];
+  int16_t speechInUB[FRAMESAMPLES_10ms];
+  int16_t streamLenLB = 0;
+  int16_t streamLenUB = 0;
+  int16_t streamLen = 0;
+  int16_t k = 0;
+  uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
   int garbageLen = 0;
-  WebRtc_Word32 bottleneck = 0;
-  WebRtc_Word16 bottleneckIdx = 0;
-  WebRtc_Word16 jitterInfo = 0;
+  int32_t bottleneck = 0;
+  int16_t bottleneckIdx = 0;
+  int16_t jitterInfo = 0;
 
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   ISACLBStruct* instLB = &(instISAC->instLB);
@@ -641,7 +641,7 @@
     memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
     streamLen = streamLenLB;
     if (streamLenUB > 0) {
-      ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)(streamLenUB + 1 +
+      ptrEncodedUW8[streamLenLB] = (uint8_t)(streamLenUB + 1 +
                                                    LEN_CHECK_SUM_WORD8);
       memcpy(&ptrEncodedUW8[streamLenLB + 1],
              instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
@@ -664,7 +664,7 @@
   if (instISAC->codingMode == 0) {
     int minBytes;
     int limit;
-    WebRtc_UWord8* ptrGarbage;
+    uint8_t* ptrGarbage;
 
     instISAC->MaxDelay = (double)WebRtcIsac_GetUplinkMaxDelay(
                            &instISAC->bwestimator_obj);
@@ -706,20 +706,20 @@
     /* If bit-stream too short then add garbage at the end. */
     if (garbageLen > 0) {
       for (k = 0; k < garbageLen; k++) {
-        ptrGarbage[k] = (WebRtc_UWord8)(rand() & 0xFF);
+        ptrGarbage[k] = (uint8_t)(rand() & 0xFF);
       }
       /* For a correct length of the upper-band bit-stream together
        * with the garbage. Garbage is embeded in upper-band bit-stream.
        * That is the only way to preserve backward compatibility. */
       if ((instISAC->bandwidthKHz == isac8kHz) ||
           (streamLenUB == 0)) {
-        ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)garbageLen;
+        ptrEncodedUW8[streamLenLB] = (uint8_t)garbageLen;
       } else {
-        ptrEncodedUW8[streamLenLB] += (WebRtc_UWord8)garbageLen;
+        ptrEncodedUW8[streamLenLB] += (uint8_t)garbageLen;
         /* Write the length of the garbage at the end of the upper-band
          *  bit-stream, if exists. This helps for sanity check. */
         ptrEncodedUW8[streamLenLB + 1 + streamLenUB] =
-            (WebRtc_UWord8)garbageLen;
+            (uint8_t)garbageLen;
 
       }
       streamLen += garbageLen;
@@ -734,14 +734,14 @@
 
   /* Generate CRC if required. */
   if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) {
-    WebRtc_UWord32 crc;
+    uint32_t crc;
 
-    WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
+    WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
                       streamLenUB + garbageLen, &crc);
 #ifndef WEBRTC_BIG_ENDIAN
     for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
       ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
-        (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+        (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
     }
 #else
     memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
@@ -782,27 +782,27 @@
  *                                 the struct since it is only allowed to read
  *                                 the struct.
  */
-WebRtc_Word16 WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
-                                         WebRtc_Word16  bweIndex,
-                                         WebRtc_Word16  jitterInfo,
-                                         WebRtc_Word32  rate,
-                                         WebRtc_Word16* encoded,
-                                         WebRtc_Word16  isRCU) {
+int16_t WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
+                                   int16_t  bweIndex,
+                                   int16_t  jitterInfo,
+                                   int32_t  rate,
+                                   int16_t* encoded,
+                                   int16_t  isRCU) {
   Bitstr iSACBitStreamInst;   /* Local struct for bitstream handling */
-  WebRtc_Word16 streamLenLB;
-  WebRtc_Word16 streamLenUB;
-  WebRtc_Word16 totalStreamLen;
+  int16_t streamLenLB;
+  int16_t streamLenUB;
+  int16_t totalStreamLen;
   double gain2;
   double gain1;
   float scale;
   enum ISACBandwidth bandwidthKHz;
   double rateLB;
   double rateUB;
-  WebRtc_Word32 currentBN;
-  WebRtc_UWord8* encodedPtrUW8 = (WebRtc_UWord8*)encoded;
-  WebRtc_UWord32 crc;
+  int32_t currentBN;
+  uint8_t* encodedPtrUW8 = (uint8_t*)encoded;
+  uint32_t crc;
 #ifndef WEBRTC_BIG_ENDIAN
-  WebRtc_Word16  k;
+  int16_t  k;
 #endif
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
@@ -849,7 +849,7 @@
     return -1;
   }
 
-  /* Convert from bytes to WebRtc_Word16. */
+  /* Convert from bytes to int16_t. */
   memcpy(encoded, iSACBitStreamInst.stream, streamLenLB);
 
   if (bandwidthKHz == isac8kHz) {
@@ -890,12 +890,12 @@
   memcpy(&encodedPtrUW8[streamLenLB + 1], iSACBitStreamInst.stream,
          streamLenUB);
 
-  WebRtcIsac_GetCrc((WebRtc_Word16*)(&(encodedPtrUW8[streamLenLB + 1])),
+  WebRtcIsac_GetCrc((int16_t*)(&(encodedPtrUW8[streamLenLB + 1])),
                     streamLenUB, &crc);
 #ifndef WEBRTC_BIG_ENDIAN
   for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
     encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
-      (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+      (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
   }
 #else
   memcpy(&encodedPtrUW8[streamLenLB + streamLenUB + 1], &crc,
@@ -921,7 +921,7 @@
  *                            :  0 - Ok
  *                              -1 - Error
  */
-static WebRtc_Word16 DecoderInitLb(ISACLBStruct* instISAC) {
+static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
   int i;
   /* Initialize stream vector to zero. */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -935,7 +935,7 @@
   return 0;
 }
 
-static WebRtc_Word16 DecoderInitUb(ISACUBStruct* instISAC) {
+static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
   int i;
   /* Init stream vector to zero */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -948,7 +948,7 @@
   return (0);
 }
 
-WebRtc_Word16 WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
+int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   if (DecoderInitLb(&instISAC->instLB) < 0) {
@@ -956,9 +956,9 @@
   }
   if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
     memset(instISAC->synthesisFBState1, 0,
-           FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+           FB_STATE_SIZE_WORD32 * sizeof(int32_t));
     memset(instISAC->synthesisFBState2, 0,
-           FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+           FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
     if (DecoderInitUb(&(instISAC->instUB)) < 0) {
       return -1;
@@ -996,18 +996,18 @@
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
-                                          const WebRtc_UWord16* encoded,
-                                          WebRtc_Word32 packet_size,
-                                          WebRtc_UWord16 rtp_seq_number,
-                                          WebRtc_UWord32 send_ts,
-                                          WebRtc_UWord32 arr_ts) {
+int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
+                                    const uint16_t* encoded,
+                                    int32_t packet_size,
+                                    uint16_t rtp_seq_number,
+                                    uint32_t send_ts,
+                                    uint32_t arr_ts) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   Bitstr streamdata;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   /* Check if decoder initiated. */
   if ((instISAC->initFlag & BIT_MASK_DEC_INIT) != BIT_MASK_DEC_INIT) {
@@ -1025,7 +1025,7 @@
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
-    streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
+    streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
                                             ((k & 1) << 3)) & 0xFF);
   }
 #else
@@ -1044,30 +1044,30 @@
   return 0;
 }
 
-static WebRtc_Word16 Decode(ISACStruct* ISAC_main_inst,
-                            const WebRtc_UWord16* encoded,
-                            WebRtc_Word16 lenEncodedBytes,
-                            WebRtc_Word16* decoded,
-                            WebRtc_Word16* speechType,
-                            WebRtc_Word16 isRCUPayload) {
+static int16_t Decode(ISACStruct* ISAC_main_inst,
+                      const uint16_t* encoded,
+                      int16_t lenEncodedBytes,
+                      int16_t* decoded,
+                      int16_t* speechType,
+                      int16_t isRCUPayload) {
   /* Number of samples (480 or 960), output from decoder
      that were actually used in the encoder/decoder
      (determined on the fly). */
-  WebRtc_Word16 numSamplesLB;
-  WebRtc_Word16 numSamplesUB;
-  WebRtc_Word16 speechIdx;
+  int16_t numSamplesLB;
+  int16_t numSamplesUB;
+  int16_t speechIdx;
   float outFrame[MAX_FRAMESAMPLES];
-  WebRtc_Word16 outFrameLB[MAX_FRAMESAMPLES];
-  WebRtc_Word16 outFrameUB[MAX_FRAMESAMPLES];
-  WebRtc_Word16 numDecodedBytesLB;
-  WebRtc_Word16 numDecodedBytesUB;
-  WebRtc_Word16 lenEncodedLBBytes;
-  WebRtc_Word16 validChecksum = 1;
-  WebRtc_Word16 k;
-  WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
-  WebRtc_UWord16 numLayer;
-  WebRtc_Word16 totSizeBytes;
-  WebRtc_Word16 err;
+  int16_t outFrameLB[MAX_FRAMESAMPLES];
+  int16_t outFrameUB[MAX_FRAMESAMPLES];
+  int16_t numDecodedBytesLB;
+  int16_t numDecodedBytesUB;
+  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;
 
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   ISACUBDecStruct* decInstUB = &(instISAC->instUB.ISACdecUB_obj);
@@ -1131,12 +1131,12 @@
       } else if (outFrame[k] < -32768) {
         decoded[k] = -32768;
       } else {
-        decoded[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
+        decoded[k] = (int16_t)WebRtcIsac_lrint(outFrame[k]);
       }
     }
     numSamplesUB = 0;
   } else {
-    WebRtc_UWord32 crc;
+    uint32_t crc;
     /* We don't accept larger than 30ms (480 samples at lower-band)
      * frame-size. */
     for (k = 0; k < numSamplesLB; k++) {
@@ -1145,7 +1145,7 @@
       } else if (outFrame[k] < -32768) {
         outFrameLB[k] = -32768;
       } else {
-        outFrameLB[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
+        outFrameLB[k] = (int16_t)WebRtcIsac_lrint(outFrame[k]);
       }
     }
 
@@ -1153,13 +1153,13 @@
     if (numDecodedBytesLB == lenEncodedBytes) {
       /* Decoding was successful. No super-wideband bit-stream exists. */
       numSamplesUB = numSamplesLB;
-      memset(outFrameUB, 0, sizeof(WebRtc_Word16) *  numSamplesUB);
+      memset(outFrameUB, 0, sizeof(int16_t) *  numSamplesUB);
 
       /* Prepare for the potential increase of signal bandwidth. */
       instISAC->resetFlag_8kHz = 2;
     } else {
       /* This includes the checksum and the bytes that stores the length. */
-      WebRtc_Word16 lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
+      int16_t lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
 
       /* Is this garbage or valid super-wideband bit-stream?
        * Check if checksum is valid. */
@@ -1169,7 +1169,7 @@
         validChecksum = 0;
       } else {
         /* Run CRC to see if the checksum match. */
-        WebRtcIsac_GetCrc((WebRtc_Word16*)(
+        WebRtcIsac_GetCrc((int16_t*)(
                             &ptrEncodedUW8[numDecodedBytesLB + 1]),
                           lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
 
@@ -1185,11 +1185,11 @@
         /* This is a garbage, we have received a wideband
          * bit-stream with garbage. */
         numSamplesUB = numSamplesLB;
-        memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
+        memset(outFrameUB, 0, sizeof(int16_t) * numSamplesUB);
       } else {
         /* A valid super-wideband biststream exists. */
         enum ISACBandwidth bandwidthKHz;
-        WebRtc_Word32 maxDelayBit;
+        int32_t maxDelayBit;
 
         /* If we have super-wideband bit-stream, we cannot
          * have 60 ms frame-size. */
@@ -1298,7 +1298,7 @@
           } else if (outFrame[k] < -32768) {
             outFrameUB[k] = -32768;
           } else {
-            outFrameUB[k] = (WebRtc_Word16)WebRtcIsac_lrint(
+            outFrameUB[k] = (int16_t)WebRtcIsac_lrint(
                               outFrame[k]);
           }
         }
@@ -1344,12 +1344,12 @@
  *                              -1 - Error
  */
 
-WebRtc_Word16 WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
-                                const WebRtc_UWord16* encoded,
-                                WebRtc_Word16 lenEncodedBytes,
-                                WebRtc_Word16* decoded,
-                                WebRtc_Word16* speechType) {
-  WebRtc_Word16 isRCUPayload = 0;
+int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
+                          const uint16_t* encoded,
+                          int16_t lenEncodedBytes,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  int16_t isRCUPayload = 0;
   return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
                 speechType, isRCUPayload);
 }
@@ -1376,12 +1376,12 @@
 
 
 
-WebRtc_Word16 WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
-                                   const WebRtc_UWord16* encoded,
-                                   WebRtc_Word16 lenEncodedBytes,
-                                   WebRtc_Word16* decoded,
-                                   WebRtc_Word16* speechType) {
-  WebRtc_Word16 isRCUPayload = 1;
+int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
+                             const uint16_t* encoded,
+                             int16_t lenEncodedBytes,
+                             int16_t* decoded,
+                             int16_t* speechType) {
+  int16_t isRCUPayload = 1;
   return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
                 speechType, isRCUPayload);
 }
@@ -1404,10 +1404,10 @@
  * Return value               : >0 - number of samples in decoded PLC vector
  *                              -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
-                                   WebRtc_Word16* decoded,
-                                   WebRtc_Word16 noOfLostFrames) {
-  WebRtc_Word16 numSamples = 0;
+int16_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
+                             int16_t* decoded,
+                             int16_t noOfLostFrames) {
+  int16_t numSamples = 0;
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   /* Limit number of frames to two = 60 millisecond.
@@ -1429,7 +1429,7 @@
   }
 
   /* Set output samples to zero. */
-  memset(decoded, 0, numSamples * sizeof(WebRtc_Word16));
+  memset(decoded, 0, numSamples * sizeof(int16_t));
   return numSamples;
 }
 
@@ -1451,8 +1451,8 @@
  * Return value               : 0 - ok
  *                             -1 - Error
  */
-static WebRtc_Word16 ControlLb(ISACLBStruct* instISAC, double rate,
-                               WebRtc_Word16 frameSize) {
+static int16_t ControlLb(ISACLBStruct* instISAC, double rate,
+                         int16_t frameSize) {
   if ((rate >= 10000) && (rate <= 32000)) {
     instISAC->ISACencLB_obj.bottleneck = rate;
   } else {
@@ -1468,7 +1468,7 @@
   return 0;
 }
 
-static WebRtc_Word16 ControlUb(ISACUBStruct* instISAC, double rate) {
+static int16_t ControlUb(ISACUBStruct* instISAC, double rate) {
   if ((rate >= 10000) && (rate <= 32000)) {
     instISAC->ISACencUB_obj.bottleneck = rate;
   } else {
@@ -1477,11 +1477,11 @@
   return 0;
 }
 
-WebRtc_Word16 WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
-                                 WebRtc_Word32 bottleneckBPS,
-                                 WebRtc_Word16 frameSize) {
+int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
+                           int32_t bottleneckBPS,
+                           int16_t frameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  WebRtc_Word16 status;
+  int16_t status;
   double rateLB;
   double rateUB;
   enum ISACBandwidth bandwidthKHz;
@@ -1586,10 +1586,10 @@
  * Return value               : 0 - ok
  *                             -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
-                                    WebRtc_Word32 bottleneckBPS,
-                                    WebRtc_Word16 frameSizeMs,
-                                    WebRtc_Word16 enforceFrameSize) {
+int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
+                              int32_t bottleneckBPS,
+                              int16_t frameSizeMs,
+                              int16_t enforceFrameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum ISACBandwidth bandwidth;
 
@@ -1659,9 +1659,9 @@
  *        - bweIndex         : Bandwidth estimate to transmit to other side.
  *
  */
-WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst,
-                                            WebRtc_Word16* bweIndex,
-                                            WebRtc_Word16* jitterInfo) {
+int16_t WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst,
+                                      int16_t* bweIndex,
+                                      int16_t* jitterInfo) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   /* Check if encoder initialized. */
@@ -1692,10 +1692,10 @@
  * Return value               : 0 - ok
  *                             -1 - index out of range
  */
-WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst,
-                                        WebRtc_Word16 bweIndex) {
+int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst,
+                                  int16_t bweIndex) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  WebRtc_Word16 returnVal;
+  int16_t returnVal;
 
   /* Check if encoder initiated. */
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
@@ -1732,19 +1732,19 @@
  *        - bweIndex          : Bandwidth estimate in bit-stream
  *
  */
-WebRtc_Word16 WebRtcIsac_ReadBwIndex(const WebRtc_Word16* encoded,
-                                     WebRtc_Word16* bweIndex) {
+int16_t WebRtcIsac_ReadBwIndex(const int16_t* encoded,
+                               int16_t* bweIndex) {
   Bitstr streamdata;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
 
   WebRtcIsac_ResetBitstream(&(streamdata));
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
-    streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
+    streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
         ((k & 1) << 3)) & 0xFF);
   }
 #else
@@ -1780,21 +1780,21 @@
  *        - frameLength       : Length of frame in packet (in samples)
  *
  */
-WebRtc_Word16 WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
-                                      const WebRtc_Word16* encoded,
-                                      WebRtc_Word16* frameLength) {
+int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
+                                const int16_t* encoded,
+                                int16_t* frameLength) {
   Bitstr streamdata;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
 #endif
-  WebRtc_Word16 err;
+  int16_t err;
   ISACMainStruct* instISAC;
 
   WebRtcIsac_ResetBitstream(&(streamdata));
 
 #ifndef WEBRTC_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
-    streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
+    streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
                                             ((k & 1) << 3)) & 0xFF);
   }
 #else
@@ -1834,7 +1834,7 @@
  * Return Value                : frame lenght in samples
  *
  */
-WebRtc_Word16 WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) {
+int16_t WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   /* Return new frame length. */
@@ -1860,7 +1860,7 @@
  *
  * Return value               : Error code
  */
-WebRtc_Word16 WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst) {
+int16_t WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst) {
  return ((ISACMainStruct*)ISAC_main_inst)->errorCode;
 }
 
@@ -1886,13 +1886,13 @@
  * Return value               : -1 if error happens
  *                               0 bit-rates computed correctly.
  */
-WebRtc_Word16 WebRtcIsac_GetUplinkBw(ISACStruct*  ISAC_main_inst,
-                                     WebRtc_Word32* bottleneck) {
+int16_t WebRtcIsac_GetUplinkBw(ISACStruct*  ISAC_main_inst,
+                               int32_t* bottleneck) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   if (instISAC->codingMode == 0) {
     /* We are in adaptive mode then get the bottleneck from BWE. */
-    *bottleneck = (WebRtc_Word32)instISAC->bwestimator_obj.send_bw_avg;
+    *bottleneck = (int32_t)instISAC->bwestimator_obj.send_bw_avg;
   } else {
     *bottleneck = instISAC->bottleneck;
   }
@@ -1939,10 +1939,10 @@
  * Return value               : 0 if successful
  *                             -1 if error happens
  */
-WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst,
-                                           WebRtc_Word16 maxPayloadBytes) {
+int16_t WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst,
+                                     int16_t maxPayloadBytes) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  WebRtc_Word16 status = 0;
+  int16_t status = 0;
 
   /* Check if encoder initiated */
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
@@ -2022,11 +2022,11 @@
  * Return value               : 0 if successful
  *                             -1 if error happens
  */
-WebRtc_Word16 WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst,
-                                    WebRtc_Word32 maxRate) {
+int16_t WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst,
+                              int32_t maxRate) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  WebRtc_Word16 maxRateInBytesPer30Ms;
-  WebRtc_Word16 status = 0;
+  int16_t maxRateInBytesPer30Ms;
+  int16_t status = 0;
 
   /* check if encoder initiated */
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
@@ -2037,7 +2037,7 @@
      given maximum rate. Multiply with 30/1000 to get number of
      bits per 30 ms, divide by 8 to get number of bytes per 30 ms:
      maxRateInBytes = floor((maxRate * 30/1000) / 8); */
-  maxRateInBytesPer30Ms = (WebRtc_Word16)(maxRate * 3 / 800);
+  maxRateInBytesPer30Ms = (int16_t)(maxRate * 3 / 800);
 
   if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
     if (maxRate < 32000) {
@@ -2093,14 +2093,14 @@
  * Return value               : >0 - Length (in bytes) of coded data
  *                            : -1 - Error
  */
-WebRtc_Word16 WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
-                                       WebRtc_Word16* encoded) {
+int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
+                                 int16_t* encoded) {
   Bitstr iSACBitStreamInst;
-  WebRtc_Word16 streamLenLB;
-  WebRtc_Word16 streamLenUB;
-  WebRtc_Word16 streamLen;
-  WebRtc_Word16 totalLenUB;
-  WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
+  int16_t streamLenLB;
+  int16_t streamLenUB;
+  int16_t streamLen;
+  int16_t totalLenUB;
+  uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 #ifndef WEBRTC_BIG_ENDIAN
   int k;
@@ -2122,7 +2122,7 @@
     return -1;
   }
 
-  /* convert from bytes to WebRtc_Word16. */
+  /* convert from bytes to int16_t. */
   memcpy(ptrEncodedUW8, iSACBitStreamInst.stream, streamLenLB);
   streamLen = streamLenLB;
   if (instISAC->bandwidthKHz == isac8kHz) {
@@ -2150,18 +2150,18 @@
   /* Generate CRC if required. */
   if ((instISAC->bandwidthKHz != isac8kHz) &&
       (streamLenUB > 0)) {
-    WebRtc_UWord32 crc;
+    uint32_t crc;
     streamLen += totalLenUB;
-    ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)totalLenUB;
+    ptrEncodedUW8[streamLenLB] = (uint8_t)totalLenUB;
     memcpy(&ptrEncodedUW8[streamLenLB + 1], iSACBitStreamInst.stream,
            streamLenUB);
 
-    WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
+    WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
                       streamLenUB, &crc);
 #ifndef WEBRTC_BIG_ENDIAN
     for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
       ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
-        (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+        (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
     }
 #else
     memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
@@ -2209,8 +2209,8 @@
  * Return value               : 0 if successful
  *                             -1 if failed.
  */
-WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
-                                        WebRtc_UWord16 sample_rate_hz) {
+int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
+                                  uint16_t sample_rate_hz) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum IsacSamplingRate encoder_operational_rate;
 
@@ -2238,9 +2238,9 @@
     ISACLBStruct* instLB = &(instISAC->instLB);
     double bottleneckLB;
     double bottleneckUB;
-    WebRtc_Word32 bottleneck = instISAC->bottleneck;
-    WebRtc_Word16 codingMode = instISAC->codingMode;
-    WebRtc_Word16 frameSizeMs = instLB->ISACencLB_obj.new_framelength /
+    int32_t bottleneck = instISAC->bottleneck;
+    int16_t codingMode = instISAC->codingMode;
+    int16_t frameSizeMs = instLB->ISACencLB_obj.new_framelength /
         (FS / 1000);
 
     if ((encoder_operational_rate == kIsacWideband) &&
@@ -2269,9 +2269,9 @@
       EncoderInitUb(instUB, instISAC->bandwidthKHz);
 
       memset(instISAC->analysisFBState1, 0,
-             FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+             FB_STATE_SIZE_WORD32 * sizeof(int32_t));
       memset(instISAC->analysisFBState2, 0,
-             FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+             FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
       if (codingMode == 1) {
         instISAC->bottleneck = bottleneck;
@@ -2306,8 +2306,8 @@
  * Return value               : 0 if successful
  *                             -1 if failed.
  */
-WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
-                                        WebRtc_UWord16 sample_rate_hz) {
+int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
+                                  uint16_t sample_rate_hz) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum IsacSamplingRate decoder_operational_rate;
 
@@ -2326,9 +2326,9 @@
       /* Switching from wideband to super-wideband at the decoder
        * we need to reset the filter-bank and initialize upper-band decoder. */
       memset(instISAC->synthesisFBState1, 0,
-             FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+             FB_STATE_SIZE_WORD32 * sizeof(int32_t));
       memset(instISAC->synthesisFBState2, 0,
-             FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+             FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
       if (DecoderInitUb(&(instISAC->instUB)) < 0) {
         return -1;
@@ -2349,7 +2349,7 @@
  *                              is expected to be sampled in this rate.
  *
  */
-WebRtc_UWord16 WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
+uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   return instISAC->in_sample_rate_hz;
 }
@@ -2366,7 +2366,7 @@
  *                              sampled at this rate.
  *
  */
-WebRtc_UWord16 WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
+uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   return instISAC->decoderSamplingRateKHz == kIsacWideband ? 16000 : 32000;
 }
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
index 854b2d7..5198ebf 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
@@ -111,7 +111,7 @@
 
 
 //was static before, but didn't work with MEX file
-void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
+void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
                        double *oldEnergy, double *varscale)
 {
   double nrg[4], chng, pg;
@@ -206,7 +206,7 @@
 }
 
 void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
-                             double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
+                             double signal_noise_ratio, const int16_t *pitchGains_Q12,
                              double *lo_coeff, double *hi_coeff)
 {
   int k, n, j, pos1, pos2;
@@ -388,12 +388,12 @@
     double*      lpCoeff,
     double       corrMat[][UB_LPC_ORDER + 1],
     double*      varscale,
-    WebRtc_Word16  bandwidth)
+    int16_t  bandwidth)
 {
   int frameCntr, activeFrameCntr, n, pos1, pos2;
-  WebRtc_Word16 criterion1;
-  WebRtc_Word16 criterion2;
-  WebRtc_Word16 numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz));
+  int16_t criterion1;
+  int16_t criterion2;
+  int16_t numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz));
   double data[WINLEN];
   double corrSubFrame[UB_LPC_ORDER+2];
   double reflecCoeff[UB_LPC_ORDER];
@@ -492,8 +492,8 @@
     double        corrMat[][UB_LPC_ORDER + 1],
     const double* varscale)
 {
-  WebRtc_Word16 j, n;
-  WebRtc_Word16 subFrameCntr;
+  int16_t j, n;
+  int16_t subFrameCntr;
   double aPolynom[ORDERLO + 1];
   double res_nrg;
 
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
index 4eafeac..866c76d 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
@@ -23,11 +23,11 @@
 
 double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order);
 
-void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
+void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
                        double *oldEnergy, double *varscale);
 
 void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
-                             double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
+                             double signal_noise_ratio, const int16_t *pitchGains_Q12,
                              double *lo_coeff, double *hi_coeff);
 
 
@@ -45,6 +45,6 @@
     double*      lpCoeff,
     double       corr[][UB_LPC_ORDER + 1],
     double*      varscale,
-    WebRtc_Word16  bandwidth);
+    int16_t  bandwidth);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
index 25c69cb..8ce004b 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
@@ -36,7 +36,7 @@
 /*
 * Number of reconstruction points of quantizers for LPC Gains.
 */
-const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES] =
+const int16_t WebRtcIsac_kNumQCellLpcGain[SUBFRAMES] =
 {
     17,  20,  25,  45,  77, 170
 };
@@ -44,7 +44,7 @@
 * Starting index for entropy decoder to search for the right interval,
 * one entry per LAR coefficient
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES] =
+const uint16_t WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES] =
 {
      8,  10,  12,  22,  38,  85
 };
@@ -53,26 +53,26 @@
 * The following 6 vectors define CDF of 6 decorrelated LPC
 * gains.
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec0[18] =
 {
      0,    10,    27,    83,   234,   568,  1601,  4683, 16830, 57534, 63437,
  64767, 65229, 65408, 65483, 65514, 65527, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec1[21] =
 {
      0,    15,    33,    84,   185,   385,   807,  1619,  3529,  7850, 19488,
  51365, 62437, 64548, 65088, 65304, 65409, 65484, 65507, 65522, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec2[26] =
 {
      0,    15,    29,    54,    89,   145,   228,   380,   652,  1493,  4260,
  12359, 34133, 50749, 57224, 60814, 62927, 64078, 64742, 65103, 65311, 65418,
  65473, 65509, 65521, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec3[46] =
 {
      0,     8,    12,    16,    26,    42,    56,    76,   111,   164,   247,
    366,   508,   693,  1000,  1442,  2155,  3188,  4854,  7387, 11249, 17617,
@@ -81,7 +81,7 @@
  65523, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec4[78] =
 {
      0,    17,    29,    39,    51,    70,   104,   154,   234,   324,   443,
    590,   760,   971,  1202,  1494,  1845,  2274,  2797,  3366,  4088,  4905,
@@ -93,7 +93,7 @@
  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171] =
+const uint16_t WebRtcIsac_kLpcGainCdfVec5[171] =
 {
      0,    10,    12,    14,    16,    18,    23,    29,    35,    42,    51,
     58,    65,    72,    78,    87,    96,   103,   111,   122,   134,   150,
@@ -116,7 +116,7 @@
 /*
 * An array of pointers to CDFs of decorrelated LPC Gains
 */
-const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES] =
+const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES] =
 {
     WebRtcIsac_kLpcGainCdfVec0, WebRtcIsac_kLpcGainCdfVec1,
     WebRtcIsac_kLpcGainCdfVec2, WebRtcIsac_kLpcGainCdfVec3,
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
index 1eba97c..121d05e 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
@@ -26,23 +26,23 @@
 
 extern const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES];
 
-extern const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES];
+extern const int16_t WebRtcIsac_kNumQCellLpcGain[SUBFRAMES];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES];
+extern const uint16_t WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec0[18];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec1[21];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec2[26];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec3[46];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec4[78];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171];
+extern const uint16_t WebRtcIsac_kLpcGainCdfVec5[171];
 
-extern const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
+extern const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
 
 extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
 
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
index 695d583..2c5698f 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
@@ -72,7 +72,7 @@
 /*
 * Number of reconstruction points of quantizers for LAR coefficients.
 */
-const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
+const int16_t WebRtcIsac_kLpcShapeNumRecPointUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
 {
     13, 15, 19, 27, 19, 24, 32, 48
@@ -82,7 +82,7 @@
 * Starting index for entropy decoder to search for the right interval,
 * one entry per LAR coefficient
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
+const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
 {
      6,  7,  9, 13,  9, 12, 16, 24
@@ -92,52 +92,52 @@
 * The following 8 vectors define CDF of 8 decorrelated LAR
 * coefficients.
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14] =
 {
      0,    13,    95,   418,  1687,  6498, 21317, 44200, 59029, 63849, 65147,
  65449, 65525, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub12[16] =
 {
      0,    10,    59,   255,   858,  2667,  8200, 22609, 42988, 57202, 62947,
  64743, 65308, 65476, 65522, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub12[20] =
 {
      0,    18,    40,   118,   332,   857,  2017,  4822, 11321, 24330, 41279,
  54342, 60637, 63394, 64659, 65184, 65398, 65482, 65518, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub12[28] =
 {
      0,    21,    38,    90,   196,   398,   770,  1400,  2589,  4650,  8211,
  14933, 26044, 39592, 50814, 57452, 60971, 62884, 63995, 64621, 65019, 65273,
  65410, 65480, 65514, 65522, 65531, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub12[20] =
 {
      0,     7,    46,   141,   403,   969,  2132,  4649, 10633, 24902, 43254,
  54665, 59928, 62674, 64173, 64938, 65293, 65464, 65523, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub12[25] =
 {
      0,     7,    22,    72,   174,   411,   854,  1737,  3545,  6774, 13165,
  25221, 40980, 52821, 58714, 61706, 63472, 64437, 64989, 65287, 65430, 65503,
  65525, 65529, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33] =
 {
      0,    11,    21,    36,    65,   128,   228,   401,   707,  1241,  2126,
   3589,  6060, 10517, 18853, 31114, 42477, 49770, 54271, 57467, 59838, 61569,
  62831, 63772, 64433, 64833, 65123, 65306, 65419, 65466, 65499, 65519, 65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49] =
 {
      0,    14,    34,    67,   107,   167,   245,   326,   449,   645,   861,
   1155,  1508,  2003,  2669,  3544,  4592,  5961,  7583,  9887, 13256, 18765,
@@ -149,7 +149,7 @@
 /*
 * An array of pointers to CDFs of decorrelated LARs
 */
-const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
+const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
 {
     WebRtcIsac_kLpcShapeCdfVec0Ub12, WebRtcIsac_kLpcShapeCdfVec1Ub12,
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
index 1e93847..cef885a 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
@@ -37,29 +37,29 @@
 [UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME];
 
 
-extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
+extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
+extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub12[16];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub12[20];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub12[28];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub12[20];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub12[25];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
 
-extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
+extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12
 [UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
 
 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
index 89f4523..0f567ed 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
@@ -60,26 +60,26 @@
 * The following 16 vectors define CDF of 16 decorrelated LAR
 * coefficients.
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14] =
 {
      0,      2,     20,    159,   1034,   5688,  20892,  44653,
  59849,  64485,  65383,  65518,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub16[16] =
 {
      0,      1,      7,     43,    276,   1496,   6681,  21653,
  43891,  58859,  64022,  65248,  65489,  65529,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub16[18] =
 {
      0,      1,      9,     54,    238,    933,   3192,   9461,
  23226,  42146,  56138,  62413,  64623,  65300,  65473,  65521,
  65533,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub16[30] =
 {
      0,      2,      4,      8,     17,     36,     75,    155,
    329,    683,   1376,   2662,   5047,   9508,  17526,  29027,
@@ -87,27 +87,27 @@
  65273,  65429,  65497,  65526,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub16[16] =
 {
      0,      1,     10,     63,    361,   1785,   7407,  22242,
  43337,  58125,  63729,  65181,  65472,  65527,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub16[17] =
 {
      0,      1,      7,     29,    134,    599,   2443,   8590,
  22962,  42635,  56911,  63060,  64940,  65408,  65513,  65531,
  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub16[21] =
 {
      0,      1,      5,     16,     57,    191,    611,   1808,
   4847,  11755,  24612,  40910,  53789,  60698,  63729,  64924,
  65346,  65486,  65523,  65532,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub16[36] =
 {
      0,      1,      4,     12,     25,     55,    104,    184,
    314,    539,    926,   1550,   2479,   3861,   5892,   8845,
@@ -116,21 +116,21 @@
  65518,  65530,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec8Ub16[21] =
 {
      0,      1,      2,      7,     26,    103,    351,   1149,
   3583,  10204,  23846,  41711,  55361,  61917,  64382,  65186,
  65433,  65506,  65528,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub160[21] =
 {
      0,      6,     19,     63,    205,    638,   1799,   4784,
  11721,  24494,  40803,  53805,  60886,  63822,  64931,  65333,
  65472,  65517,  65530,  65533,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub161[28] =
 {
      0,      1,      3,     11,     31,     86,    221,    506,
   1101,   2296,   4486,   8477,  15356,  26079,  38941,  49952,
@@ -138,7 +138,7 @@
  65526,  65532,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub162[55] =
 {
      0,      3,     12,     23,     42,     65,     89,    115,
    150,    195,    248,    327,    430,    580,    784,   1099,
@@ -149,7 +149,7 @@
  65527,  65529,  65531,  65532,  65533,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub163[26] =
 {
      0,      2,      4,     10,     21,     48,    114,    280,
    701,   1765,   4555,  11270,  24267,  41213,  54285,  61003,
@@ -157,7 +157,7 @@
  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub164[28] =
 {
      0,      1,      3,      6,     15,     36,     82,    196,
    453,   1087,   2557,   5923,  13016,  25366,  40449,  52582,
@@ -165,7 +165,7 @@
  65529,  65533,  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34] =
 {
      0,      2,      4,      8,     18,     35,     73,    146,
    279,    524,    980,   1789,   3235,   5784,  10040,  16998,
@@ -174,7 +174,7 @@
  65534,  65535
 };
 
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71] =
+const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71] =
 {
      0,      1,      2,      6,     13,     26,     55,     92,
    141,    191,    242,    296,    355,    429,    522,    636,
@@ -190,7 +190,7 @@
 /*
 * An array of pointers to CDFs of decorrelated LARs
 */
-const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
+const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = {
      WebRtcIsac_kLpcShapeCdfVec01Ub16,
      WebRtcIsac_kLpcShapeCdfVec1Ub16,
@@ -224,7 +224,7 @@
 /*
 * Number of reconstruction points of quantizers for LAR coefficients.
 */
-const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
+const int16_t WebRtcIsac_kLpcShapeNumRecPointUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
 {
    13,    15,    17,    29,    15,    16,    20,    35,    20,
@@ -235,7 +235,7 @@
 * Starting index for entropy decoder to search for the right interval,
 * one entry per LAR coefficient
 */
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
+const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
 {
     6,     7,     8,    14,     7,     8,    10,    17,    10,
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
index 68d08b2..a4b3a59 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
@@ -30,48 +30,48 @@
 extern const double WebRtcIsac_kInterVecDecorrMatUb16
 [UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub16[16];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub16[18];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub16[30];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub16[16];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub16[17];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub16[21];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub16[36];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec8Ub16[21];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub160[21];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub161[28];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub162[55];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub163[26];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub164[28];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
+extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
 
-extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
+extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
 
 extern const double WebRtcIsac_kLpcShapeLeftRecPointUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
 
-extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
+extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
 
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
+extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb16
 [UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
 
 extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
index 985e43c..909809b 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
@@ -14,15 +14,15 @@
 #include "settings.h"
 
 /* cdf array for model indicator */
-const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[4] = {
+const uint16_t WebRtcIsac_kQKltModelCdf[4] = {
     0,  15434,  37548,  65535 };
 
 /* pointer to cdf array for model indicator */
-const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1] = {
+const uint16_t *WebRtcIsac_kQKltModelCdfPtr[1] = {
     WebRtcIsac_kQKltModelCdf };
 
 /* initial cdf index for decoder of model indicator */
-const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1] = { 1 };
+const uint16_t WebRtcIsac_kQKltModelInitIndex[1] = { 1 };
 
 /* offset to go from rounded value to quantization index */
 const short WebRtcIsac_kQKltQuantMinGain[12] = {
@@ -43,10 +43,10 @@
     5,  6,  7,  11,  9,  13,  12,  26 };
 
 /* maximum quantization index */
-const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12] = {
+const uint16_t WebRtcIsac_kQKltMaxIndGain[12] = {
     6,  12,  8,  14,  10,  19,  12,  31,  22,  56, 52,  138 };
 
-const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108] = {
+const uint16_t WebRtcIsac_kQKltMaxIndShape[108] = {
     0,  0,  0,  0,  0,  0,  0,  0,  0,  1,
     2,  2,  2,  2,  4,  4,  5,  6,  0,  0,
     0,  0,  1,  0,  0,  0,  0,  1,  2,  2,
@@ -60,10 +60,10 @@
     9,  10,  13,  19,  17,  23,  25,  49 };
 
 /* index offset */
-const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[12] = {
+const uint16_t WebRtcIsac_kQKltOffsetGain[12] = {
     0,  7,  20,  29,  44,  55,  75,  88,  120,  143,  200,  253 };
 
-const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[108] = {
+const uint16_t WebRtcIsac_kQKltOffsetShape[108] = {
     0,  1,  2,  3,  4,  5,  6,  7,  8,  9,
     11,  14,  17,  20,  23,  28,  33,  39,  46,  47,
     48,  49,  50,  52,  53,  54,  55,  56,  58,  61,
@@ -77,10 +77,10 @@
     405,  415,  426,  440,  460,  478,  502,  528 };
 
 /* initial cdf index for KLT coefficients */
-const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[12] = {
+const uint16_t WebRtcIsac_kQKltInitIndexGain[12] = {
     3,  6,  4,  7,  5,  10,  6,  16,  11,  28, 26,  69};
 
-const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[108] = {
+const uint16_t WebRtcIsac_kQKltInitIndexShape[108] = {
     0,  0,  0,  0,  0,  0,  0,  0,  0,  1,
     1,  1,  1,  1,  2,  2,  3,  3,  0,  0,
     0,  0,  1,  0,  0,  0,  0,  1,  1,  1,
@@ -296,7 +296,7 @@
 
 
 /* cdf tables for quantizer indices */
-const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[404] = {
+const uint16_t WebRtcIsac_kQKltCdfGain[404] = {
     0,  13,  301,  3730,  61784,  65167,  65489,  65535,  0,  17,
     142,  314,  929,  2466,  7678,  56450,  63463,  64740,  65204,  65426,
     65527,  65535,  0,  8,  100,  724,  6301,  60105,  65125,  65510,
@@ -340,7 +340,7 @@
     65514,  65516,  65518,  65522,  65531,  65533,  65535 };
 
 
-const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[686] = {
+const uint16_t WebRtcIsac_kQKltCdfShape[686] = {
     0,  65535,  0,  65535,  0,  65535,  0,  65535,  0,  65535,
     0,  65535,  0,  65535,  0,  65535,  0,  65535,  0,  4,
     65535,  0,  8,  65514,  65535,  0,  29,  65481,  65535,  0,
@@ -413,7 +413,7 @@
 
 
 /* pointers to cdf tables for quantizer indices */
-const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[12] = {
+const uint16_t *WebRtcIsac_kQKltCdfPtrGain[12] = {
     WebRtcIsac_kQKltCdfGain +0 +0, WebRtcIsac_kQKltCdfGain +0 +8,
     WebRtcIsac_kQKltCdfGain +0 +22, WebRtcIsac_kQKltCdfGain +0 +32,
     WebRtcIsac_kQKltCdfGain +0 +48, WebRtcIsac_kQKltCdfGain +0 +60,
@@ -421,7 +421,7 @@
     WebRtcIsac_kQKltCdfGain +0 +128, WebRtcIsac_kQKltCdfGain +0 +152,
     WebRtcIsac_kQKltCdfGain +0 +210, WebRtcIsac_kQKltCdfGain +0 +264 };
 
-const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[108] = {
+const uint16_t *WebRtcIsac_kQKltCdfPtrShape[108] = {
     WebRtcIsac_kQKltCdfShape +0 +0, WebRtcIsac_kQKltCdfShape +0 +2,
     WebRtcIsac_kQKltCdfShape +0 +4, WebRtcIsac_kQKltCdfShape +0 +6,
     WebRtcIsac_kQKltCdfShape +0 +8, WebRtcIsac_kQKltCdfShape +0 +10,
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
index c217804..51f6316 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
@@ -39,13 +39,13 @@
 #define KLT_ORDER_SHAPE    (LPC_SHAPE_ORDER * SUBFRAMES)
 
 /* cdf array for model indicator */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
+extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
 
 /* pointer to cdf array for model indicator */
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1];
+extern const uint16_t *WebRtcIsac_kQKltModelCdfPtr[1];
 
 /* initial cdf index for decoder of model indicator */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1];
+extern const uint16_t WebRtcIsac_kQKltModelInitIndex[1];
 
 /* offset to go from rounded value to quantization index */
 extern const short WebRtcIsac_kQKltQuantMinGain[12];
@@ -53,19 +53,19 @@
 extern const short WebRtcIsac_kQKltQuantMinShape[108];
 
 /* maximum quantization index */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12];
+extern const uint16_t WebRtcIsac_kQKltMaxIndGain[12];
 
-extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108];
+extern const uint16_t WebRtcIsac_kQKltMaxIndShape[108];
 
 /* index offset */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[12];
+extern const uint16_t WebRtcIsac_kQKltOffsetGain[12];
 
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[108];
+extern const uint16_t WebRtcIsac_kQKltOffsetShape[108];
 
 /* initial cdf index for KLT coefficients */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[12];
+extern const uint16_t WebRtcIsac_kQKltInitIndexGain[12];
 
-extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[108];
+extern const uint16_t WebRtcIsac_kQKltInitIndexShape[108];
 
 /* quantizer representation levels */
 extern const double WebRtcIsac_kQKltLevelsGain[392];
@@ -73,14 +73,14 @@
 extern const double WebRtcIsac_kQKltLevelsShape[578];
 
 /* cdf tables for quantizer indices */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[404];
+extern const uint16_t WebRtcIsac_kQKltCdfGain[404];
 
-extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[686];
+extern const uint16_t WebRtcIsac_kQKltCdfShape[686];
 
 /* pointers to cdf tables for quantizer indices */
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[12];
+extern const uint16_t *WebRtcIsac_kQKltCdfPtrGain[12];
 
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[108];
+extern const uint16_t *WebRtcIsac_kQKltCdfPtrShape[108];
 
 /* left KLT transforms */
 extern const double WebRtcIsac_kKltT1Gain[4];
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
index 5d998a2..947d3e7 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
@@ -15,7 +15,7 @@
 /* header file for coding tables for the pitch filter side-info in the entropy coder */
 /********************* Pitch Filter Gain Coefficient Tables ************************/
 /* cdf for quantized pitch filter gains */
-const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255] = {
+const uint16_t WebRtcIsac_kQPitchGainCdf[255] = {
   0,  2,  4,  6,  64,  901,  903,  905,  16954,  16956,
   16961,  17360,  17362,  17364,  17366,  17368,  17370,  17372,  17374,  17411,
   17514,  17516,  17583,  18790,  18796,  18802,  20760,  20777,  20782,  21722,
@@ -44,22 +44,22 @@
   65535,  65535,  65535,  65535,  65535};
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3] = {
+const int16_t WebRtcIsac_kIndexLowerLimitGain[3] = {
   -7, -2, -1};
 
-const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3] = {
+const int16_t WebRtcIsac_kIndexUpperLimitGain[3] = {
   0,  3,  1};
 
-const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2] = {
+const uint16_t WebRtcIsac_kIndexMultsGain[2] = {
   18,  3};
 
 /* size of cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1] = {
+const uint16_t WebRtcIsac_kQCdfTableSizeGain[1] = {
   256};
 
 ///////////////////////////FIXED POINT
 /* mean values of pitch filter gains in FIXED point */
-const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144] = {
+const int16_t WebRtcIsac_kQMeanGain1Q12[144] = {
    843,    1092,    1336,    1222,    1405,    1656,    1500,    1815,    1843,    1838,    1839,    1843,    1843,    1843,    1843,    1843,
   1843,    1843,     814,     846,    1092,    1013,    1174,    1383,    1391,    1511,    1584,    1734,    1753,    1843,    1843,    1843,
   1843,    1843,    1843,    1843,     524,     689,     777,     845,     947,    1069,    1090,    1263,    1380,    1447,    1559,    1676,
@@ -70,7 +70,7 @@
    112,     120,     190,     283,     442,     343,     526,     809,     684,     935,    1134,    1020,    1265,    1506,       0,       0,
      0,       0,       0,       0,       0,     111,     256,      87,     373,     597,     430,     684,     935,     770,    1020,    1265};
 
-const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144] = {
+const int16_t WebRtcIsac_kQMeanGain2Q12[144] = {
   1760,    1525,    1285,    1747,    1671,    1393,    1843,    1826,    1555,    1843,    1784,    1606,    1843,    1843,    1711,    1843,
   1843,    1814,    1389,    1275,    1040,    1564,    1414,    1252,    1610,    1495,    1343,    1753,    1592,    1405,    1804,    1720,
   1475,    1843,    1814,    1581,    1208,    1061,    856,    1349,    1148,    994,    1390,    1253,    1111,    1495,    1343,    1178,
@@ -81,7 +81,7 @@
   222,    38,        513,    271,    124,    624,    325,    157,    737,    484,    233,    849,    597,    343,    27,        0,
   0,    141,    0,    0,    256,    69,        0,    370,    87,        0,    484,    229,    0,    597,    343,    87};
 
-const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144] = {
+const int16_t WebRtcIsac_kQMeanGain3Q12[144] = {
   1843,    1843,    1711,    1843,    1818,    1606,    1843,    1827,    1511,    1814,    1639,    1393,    1760,    1525,    1285,    1656,
   1419,    1176,    1835,    1718,    1475,    1841,    1650,    1387,    1648,    1498,    1287,    1600,    1411,    1176,    1522,    1299,
   1040,    1419,    1176,    928,    1773,    1461,    1128,    1532,    1355,    1202,    1429,    1260,    1115,    1398,    1151,    1025,
@@ -93,7 +93,7 @@
   0,    370,    57,        0,    256,    43,        0,    141,    0,        0,    27,        0,    0,    0,    0,    0};
 
 
-const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144] = {
+const int16_t WebRtcIsac_kQMeanGain4Q12[144] = {
   1843,    1843,    1843,    1843,    1841,    1843,    1500,    1821,    1843,    1222,    1434,    1656,    843,    1092,    1336,    504,
   757,    1007,    1843,    1843,    1843,    1838,    1791,    1843,    1265,    1505,    1599,    965,    1219,    1425,    730,    821,
   1092,    249,    504,    757,    1783,    1819,    1843,    1351,    1567,    1727,    1096,    1268,    1409,    805,    961,    1131,
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
index f958f5d..e7518b1 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
@@ -23,23 +23,23 @@
 /* header file for coding tables for the pitch filter side-info in the entropy coder */
 /********************* Pitch Filter Gain Coefficient Tables ************************/
 /* cdf for quantized pitch filter gains */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255];
+extern const uint16_t WebRtcIsac_kQPitchGainCdf[255];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3];
+extern const int16_t WebRtcIsac_kIndexLowerLimitGain[3];
 
-extern const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3];
-extern const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2];
+extern const int16_t WebRtcIsac_kIndexUpperLimitGain[3];
+extern const uint16_t WebRtcIsac_kIndexMultsGain[2];
 
 /* mean values of pitch filter gains */
 //(Y)
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144];
+extern const int16_t WebRtcIsac_kQMeanGain1Q12[144];
+extern const int16_t WebRtcIsac_kQMeanGain2Q12[144];
+extern const int16_t WebRtcIsac_kQMeanGain3Q12[144];
+extern const int16_t WebRtcIsac_kQMeanGain4Q12[144];
 //(Y)
 
 /* size of cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1];
+extern const uint16_t WebRtcIsac_kQCdfTableSizeGain[1];
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
index 72a031e..f845a22 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
@@ -17,7 +17,7 @@
 /* tables for use with small pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf1Lo[127] = {
  0,  134,  336,  549,  778,  998,  1264,  1512,  1777,  2070,
  2423,  2794,  3051,  3361,  3708,  3979,  4315,  4610,  4933,  5269,
  5575,  5896,  6155,  6480,  6816,  7129,  7477,  7764,  8061,  8358,
@@ -32,30 +32,30 @@
  59288,  60179,  61076,  61806,  62474,  63129,  63656,  64160,  64533,  64856,
  65152,  65535,  65535,  65535,  65535,  65535,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20] = {
  0,  429,  3558,  5861,  8558,  11639,  15210,  19502,  24773,  31983,
  42602,  48567,  52601,  55676,  58160,  60172,  61889,  63235,  65383,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2] = {
  0,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10] = {
  0,  2966,  6368,  11182,  19431,  37793,  48532,  55353,  60626,  65535};
 
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4] = {WebRtcIsac_kQPitchLagCdf1Lo, WebRtcIsac_kQPitchLagCdf2Lo, WebRtcIsac_kQPitchLagCdf3Lo, WebRtcIsac_kQPitchLagCdf4Lo};
+const uint16_t *WebRtcIsac_kQPitchLagCdfPtrLo[4] = {WebRtcIsac_kQPitchLagCdf1Lo, WebRtcIsac_kQPitchLagCdf2Lo, WebRtcIsac_kQPitchLagCdf3Lo, WebRtcIsac_kQPitchLagCdf4Lo};
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1] = {128};
+const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1] = {128};
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4] = {
+const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4] = {
 -140, -9,  0, -4};
 
-const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4] = {
+const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4] = {
 -20,  9,  0,  4};
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3] = {
+const uint16_t WebRtcIsac_kQInitIndexLagLo[3] = {
  10,  1,  5};
 
 /* mean values of pitch filter lags */
@@ -75,7 +75,7 @@
 /* tables for use with medium pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf1Mid[255] = {
  0,  28,  61,  88,  121,  149,  233,  331,  475,  559,
  624,  661,  689,  712,  745,  791,  815,  843,  866,  922,
  959,  1024,  1061,  1117,  1178,  1238,  1280,  1350,  1453,  1513,
@@ -103,33 +103,33 @@
  65414,  65535,  65535,  65535,  65535,  65535,  65535,  65535,  65535,  65535,
  65535,  65535,  65535,  65535,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36] = {
  0,  71,  335,  581,  836,  1039,  1323,  1795,  2258,  2608,
  3005,  3591,  4243,  5344,  7163,  10583,  16848,  28078,  49448,  57007,
  60357,  61850,  62837,  63437,  63872,  64188,  64377,  64614,  64774,  64949,
  65039,  65115,  65223,  65360,  65474,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2] = {
  0,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20] = {
  0,  28,  246,  459,  667,  1045,  1523,  2337,  4337,  11347,
  44231,  56709,  60781,  62243,  63161,  63969,  64608,  65062,  65502,  65535};
 
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4] = {WebRtcIsac_kQPitchLagCdf1Mid, WebRtcIsac_kQPitchLagCdf2Mid, WebRtcIsac_kQPitchLagCdf3Mid, WebRtcIsac_kQPitchLagCdf4Mid};
+const uint16_t *WebRtcIsac_kQPitchLagCdfPtrMid[4] = {WebRtcIsac_kQPitchLagCdf1Mid, WebRtcIsac_kQPitchLagCdf2Mid, WebRtcIsac_kQPitchLagCdf3Mid, WebRtcIsac_kQPitchLagCdf4Mid};
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1] = {256};
+const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1] = {256};
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4] = {
+const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4] = {
 -280, -17,  0, -9};
 
-const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4] = {
+const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4] = {
 -40,  17,  0,  9};
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3] = {
+const uint16_t WebRtcIsac_kQInitIndexLagMid[3] = {
  18,  1,  10};
 
 /* mean values of pitch filter lags */
@@ -152,7 +152,7 @@
 /* tables for use with large pitch gain */
 
 /* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf1Hi[511] = {
  0,  7,  18,  33,  69,  105,  156,  228,  315,  612,
  680,  691,  709,  724,  735,  738,  742,  746,  749,  753,
  756,  760,  764,  774,  782,  785,  789,  796,  800,  803,
@@ -206,7 +206,7 @@
  65535,  65535,  65535,  65535,  65535,  65535,  65535,  65535,  65535,  65535,
  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68] = {
  0,  7,  11,  22,  37,  52,  56,  59,  81,  85,
  89,  96,  115,  130,  137,  152,  170,  181,  193,  200,
  207,  233,  237,  259,  289,  318,  363,  433,  592,  992,
@@ -215,29 +215,29 @@
  65413,  65420,  65428,  65435,  65439,  65450,  65454,  65468,  65472,  65476,
  65483,  65491,  65498,  65505,  65516,  65520,  65528,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2] = {
  0,  65535};
 
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35] = {
+const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35] = {
  0,  7,  19,  30,  41,  48,  63,  74,  82,  96,
  122,  152,  215,  330,  701,  2611,  10931,  48106,  61177,  64341,
  65112,  65238,  65309,  65338,  65364,  65379,  65401,  65427,  65453,  65465,
  65476,  65490,  65509,  65528,  65535};
 
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4] = {WebRtcIsac_kQPitchLagCdf1Hi, WebRtcIsac_kQPitchLagCdf2Hi, WebRtcIsac_kQPitchLagCdf3Hi, WebRtcIsac_kQPitchLagCdf4Hi};
+const uint16_t *WebRtcIsac_kQPitchLagCdfPtrHi[4] = {WebRtcIsac_kQPitchLagCdf1Hi, WebRtcIsac_kQPitchLagCdf2Hi, WebRtcIsac_kQPitchLagCdf3Hi, WebRtcIsac_kQPitchLagCdf4Hi};
 
 /* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1] = {512};
+const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1] = {512};
 
 /* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4] = {
+const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4] = {
 -552, -34,  0, -16};
 
-const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4] = {
+const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4] = {
 -80,  32,  0,  17};
 
 /* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3] = {
+const uint16_t WebRtcIsac_kQInitIndexLagHi[3] = {
  34,  1,  18};
 
 /* mean values of pitch filter lags */
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
index 67b02e5..7c72571 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
@@ -25,22 +25,22 @@
 /* tables for use with small pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf1Lo[127];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10];
 
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4];
+extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrLo[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1];
+extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4];
-extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4];
+extern const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4];
+extern const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3];
+extern const uint16_t WebRtcIsac_kQInitIndexLagLo[3];
 
 /* mean values of pitch filter lags */
 extern const double WebRtcIsac_kQMeanLag2Lo[19];
@@ -53,22 +53,22 @@
 /* tables for use with medium pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf1Mid[255];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20];
 
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4];
+extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrMid[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1];
+extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4];
-extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4];
+extern const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4];
+extern const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3];
+extern const uint16_t WebRtcIsac_kQInitIndexLagMid[3];
 
 /* mean values of pitch filter lags */
 extern const double WebRtcIsac_kQMeanLag2Mid[35];
@@ -81,22 +81,22 @@
 /* tables for use with large pitch gain */
 
 /* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf1Hi[511];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2];
+extern const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35];
 
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4];
+extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrHi[4];
 
 /* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1];
+extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1];
 
 /* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4];
-extern const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4];
+extern const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4];
+extern const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4];
 
 /* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3];
+extern const uint16_t WebRtcIsac_kQInitIndexLagHi[3];
 
 /* mean values of pitch filter lags */
 extern const double WebRtcIsac_kQMeanLag2Hi[67];
diff --git a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
index 92b9c4d..9eae055 100644
--- a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
@@ -13,82 +13,82 @@
 
 /********************* AR Coefficient Tables ************************/
 /* cdf for quantized reflection coefficient 1 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc1Cdf[12] = {
  0,  2,  4,  129,  7707,  57485,  65495,  65527,  65529,  65531,
  65533,  65535};
 
 /* cdf for quantized reflection coefficient 2 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc2Cdf[12] = {
  0,  2,  4,  7,  531,  25298,  64525,  65526,  65529,  65531,
  65533,  65535};
 
 /* cdf for quantized reflection coefficient 3 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc3Cdf[12] = {
  0,  2,  4,  6,  620,  22898,  64843,  65527,  65529,  65531,
  65533,  65535};
 
 /* cdf for quantized reflection coefficient 4 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc4Cdf[12] = {
  0,  2,  4,  6,  35,  10034,  60733,  65506,  65529,  65531,
  65533,  65535};
 
 /* cdf for quantized reflection coefficient 5 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc5Cdf[12] = {
  0,  2,  4,  6,  36,  7567,  56727,  65385,  65529,  65531,
  65533,  65535};
 
 /* cdf for quantized reflection coefficient 6 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12] = {
+const uint16_t WebRtcIsac_kQArRc6Cdf[12] = {
  0,  2,  4,  6,  14,  6579,  57360,  65409,  65529,  65531,
  65533,  65535};
 
 /* representation levels for quantized reflection coefficient 1 */
-const WebRtc_Word16 WebRtcIsac_kQArRc1Levels[11] = {
+const int16_t WebRtcIsac_kQArRc1Levels[11] = {
  -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 2 */
-const WebRtc_Word16 WebRtcIsac_kQArRc2Levels[11] = {
+const int16_t WebRtcIsac_kQArRc2Levels[11] = {
  -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 3 */
-const WebRtc_Word16 WebRtcIsac_kQArRc3Levels[11] = {
+const int16_t WebRtcIsac_kQArRc3Levels[11] = {
 -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 4 */
-const WebRtc_Word16 WebRtcIsac_kQArRc4Levels[11] = {
+const int16_t WebRtcIsac_kQArRc4Levels[11] = {
 -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 5 */
-const WebRtc_Word16 WebRtcIsac_kQArRc5Levels[11] = {
+const int16_t WebRtcIsac_kQArRc5Levels[11] = {
 -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
 };
 
 /* representation levels for quantized reflection coefficient 6 */
-const WebRtc_Word16 WebRtcIsac_kQArRc6Levels[11] = {
+const int16_t WebRtcIsac_kQArRc6Levels[11] = {
 -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
 };
 
 /* quantization boundary levels for reflection coefficients */
-const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12] = {
+const int16_t WebRtcIsac_kQArBoundaryLevels[12] = {
 -32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, 32767
 };
 
 /* initial index for AR reflection coefficient quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[6] = {
+const uint16_t WebRtcIsac_kQArRcInitIndex[6] = {
  5,  5,  5,  5,  5,  5};
 
 /* pointers to AR cdf tables */
-const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER] = {
+const uint16_t *WebRtcIsac_kQArRcCdfPtr[AR_ORDER] = {
   WebRtcIsac_kQArRc1Cdf, WebRtcIsac_kQArRc2Cdf, WebRtcIsac_kQArRc3Cdf,
   WebRtcIsac_kQArRc4Cdf, WebRtcIsac_kQArRc5Cdf, WebRtcIsac_kQArRc6Cdf
 };
 
 /* pointers to AR representation levels tables */
-const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER] = {
+const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER] = {
   WebRtcIsac_kQArRc1Levels, WebRtcIsac_kQArRc2Levels, WebRtcIsac_kQArRc3Levels,
   WebRtcIsac_kQArRc4Levels, WebRtcIsac_kQArRc5Levels, WebRtcIsac_kQArRc6Levels
 };
@@ -96,27 +96,27 @@
 
 /******************** GAIN Coefficient Tables ***********************/
 /* cdf for Gain coefficient */
-const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19] = {
+const uint16_t WebRtcIsac_kQGainCdf[19] = {
  0,  2,  4,  6,  8,  10,  12,  14,  16,  1172,
  11119,  29411,  51699,  64445,  65527,  65529,  65531,  65533,  65535};
 
 /* representation levels for quantized squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18] = {
+const int32_t WebRtcIsac_kQGain2Levels[18] = {
 // 17, 28, 46, 76, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
  128, 128, 128, 128, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
 /* quantization boundary levels for squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19] = {
+const int32_t WebRtcIsac_kQGain2BoundaryLevels[19] = {
 0, 21, 35, 59, 99, 166, 280, 475, 815, 1414, 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF};
 
 /* pointers to Gain cdf table */
-const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1] = {WebRtcIsac_kQGainCdf};
+const uint16_t *WebRtcIsac_kQGainCdf_ptr[1] = {WebRtcIsac_kQGainCdf};
 
 /* Gain initial index for gain quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1] = {11};
+const uint16_t WebRtcIsac_kQGainInitIndex[1] = {11};
 
 /************************* Cosine Tables ****************************/
 /* Cosine table */
-const WebRtc_Word16 WebRtcIsac_kCos[6][60] = {
+const int16_t WebRtcIsac_kCos[6][60] = {
 {512,  512,  511,  510,  508,  507,  505,  502,  499,  496,  493,  489,  485,  480,  476,  470,  465,  459,  453,  447,
 440,  433,  426,  418,  410,  402,  394,  385,  376,  367,  357,  348,  338,  327,  317,  306,  295,  284,  273,  262,
 250,  238,  226,  214,  202,  190,  177,  165,  152,  139,  126,  113,  100,  87,  73,  60,  47,  33,  20,  7},
diff --git a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
index 159245b..22fe6a2 100644
--- a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
@@ -23,54 +23,54 @@
 
 /********************* AR Coefficient Tables ************************/
 /* cdf for quantized reflection coefficient 1 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc1Cdf[12];
 
 /* cdf for quantized reflection coefficient 2 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc2Cdf[12];
 
 /* cdf for quantized reflection coefficient 3 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc3Cdf[12];
 
 /* cdf for quantized reflection coefficient 4 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc4Cdf[12];
 
 /* cdf for quantized reflection coefficient 5 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc5Cdf[12];
 
 /* cdf for quantized reflection coefficient 6 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12];
+extern const uint16_t WebRtcIsac_kQArRc6Cdf[12];
 
 /* quantization boundary levels for reflection coefficients */
-extern const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12];
+extern const int16_t WebRtcIsac_kQArBoundaryLevels[12];
 
 /* initial indices for AR reflection coefficient quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[AR_ORDER];
+extern const uint16_t WebRtcIsac_kQArRcInitIndex[AR_ORDER];
 
 /* pointers to AR cdf tables */
-extern const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
+extern const uint16_t *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
 
 /* pointers to AR representation levels tables */
-extern const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
+extern const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
 
 
 /******************** GAIN Coefficient Tables ***********************/
 /* cdf for Gain coefficient */
-extern const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19];
+extern const uint16_t WebRtcIsac_kQGainCdf[19];
 
 /* representation levels for quantized Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18];
+extern const int32_t WebRtcIsac_kQGain2Levels[18];
 
 /* squared quantization boundary levels for Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19];
+extern const int32_t WebRtcIsac_kQGain2BoundaryLevels[19];
 
 /* pointer to Gain cdf table */
-extern const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1];
+extern const uint16_t *WebRtcIsac_kQGainCdf_ptr[1];
 
 /* Gain initial index for gain quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1];
+extern const uint16_t WebRtcIsac_kQGainInitIndex[1];
 
 /************************* Cosine Tables ****************************/
 /* Cosine table */
-extern const WebRtc_Word16 WebRtcIsac_kCos[6][60];
+extern const int16_t WebRtcIsac_kCos[6][60];
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/structs.h b/modules/audio_coding/codecs/isac/main/source/structs.h
index 62dc204..1bd73e7 100644
--- a/modules/audio_coding/codecs/isac/main/source/structs.h
+++ b/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -25,10 +25,10 @@
 
 typedef struct Bitstreamstruct {
 
-  WebRtc_UWord8   stream[STREAM_SIZE_MAX];
-  WebRtc_UWord32  W_upper;
-  WebRtc_UWord32  streamval;
-  WebRtc_UWord32  stream_index;
+  uint8_t   stream[STREAM_SIZE_MAX];
+  uint32_t  W_upper;
+  uint32_t  streamval;
+  uint32_t  stream_index;
 
 } Bitstr;
 
@@ -149,32 +149,32 @@
 typedef struct {
 
   /* Previous frame length (in ms)                                    */
-  WebRtc_Word32    prev_frame_length;
+  int32_t    prev_frame_length;
 
   /* Previous RTP timestamp from received
      packet (in samples relative beginning)                           */
-  WebRtc_Word32    prev_rec_rtp_number;
+  int32_t    prev_rec_rtp_number;
 
   /* Send timestamp for previous packet (in ms using timeGetTime())   */
-  WebRtc_UWord32    prev_rec_send_ts;
+  uint32_t    prev_rec_send_ts;
 
   /* Arrival time for previous packet (in ms using timeGetTime())     */
-  WebRtc_UWord32    prev_rec_arr_ts;
+  uint32_t    prev_rec_arr_ts;
 
   /* rate of previous packet, derived from RTP timestamps (in bits/s) */
   float   prev_rec_rtp_rate;
 
   /* Time sinse the last update of the BN estimate (in ms)            */
-  WebRtc_UWord32    last_update_ts;
+  uint32_t    last_update_ts;
 
   /* Time sinse the last reduction (in ms)                            */
-  WebRtc_UWord32    last_reduction_ts;
+  uint32_t    last_reduction_ts;
 
   /* How many times the estimate was update in the beginning          */
-  WebRtc_Word32    count_tot_updates_rec;
+  int32_t    count_tot_updates_rec;
 
   /* The estimated bottle neck rate from there to here (in bits/s)    */
-  WebRtc_Word32  rec_bw;
+  int32_t  rec_bw;
   float   rec_bw_inv;
   float   rec_bw_avg;
   float   rec_bw_avg_Q;
@@ -212,18 +212,18 @@
   // been detected upstream
   int hsn_detect_snd;
 
-  WebRtc_UWord32 start_wait_period;
+  uint32_t start_wait_period;
 
   int in_wait_period;
 
   int change_to_WB;
 
-  WebRtc_UWord32                 senderTimestamp;
-  WebRtc_UWord32                 receiverTimestamp;
+  uint32_t                 senderTimestamp;
+  uint32_t                 receiverTimestamp;
   //enum IsacSamplingRate incomingStreamSampFreq;
-  WebRtc_UWord16                 numConsecLatePkts;
+  uint16_t                 numConsecLatePkts;
   float                        consecLatency;
-  WebRtc_Word16                  inWaitLatePkts;
+  int16_t                  inWaitLatePkts;
 } BwEstimatorstr;
 
 
@@ -268,7 +268,7 @@
   int         startIdx;
 
   /* Frame length in samples */
-  WebRtc_Word16 framelength;
+  int16_t framelength;
 
   /* Pitch Gain */
   int         pitchGain_index[2];
@@ -284,9 +284,9 @@
   double      LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
 
   /* Encode Spec */
-  WebRtc_Word16 fre[FRAMESAMPLES];
-  WebRtc_Word16 fim[FRAMESAMPLES];
-  WebRtc_Word16 AvgPitchGain[2];
+  int16_t fre[FRAMESAMPLES];
+  int16_t fim[FRAMESAMPLES];
+  int16_t AvgPitchGain[2];
 
   /* Used in adaptive mode only */
   int         minBytes;
@@ -302,8 +302,8 @@
 
   Bitstr      bitStreamObj;
 
-  WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
-  WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
 } ISACUBSaveEncDataStruct;
 
 
@@ -319,29 +319,29 @@
   ISAC_SaveEncData_t  SaveEnc_obj;
 
   int                 buffer_index;
-  WebRtc_Word16         current_framesamples;
+  int16_t         current_framesamples;
 
   float               data_buffer_float[FRAMESAMPLES_30ms];
 
   int                 frame_nb;
   double              bottleneck;
-  WebRtc_Word16         new_framelength;
+  int16_t         new_framelength;
   double              s2nr;
 
   /* Maximum allowed number of bits for a 30 msec packet */
-  WebRtc_Word16         payloadLimitBytes30;
+  int16_t         payloadLimitBytes30;
   /* Maximum allowed number of bits for a 30 msec packet */
-  WebRtc_Word16         payloadLimitBytes60;
+  int16_t         payloadLimitBytes60;
   /* Maximum allowed number of bits for both 30 and 60 msec packet */
-  WebRtc_Word16         maxPayloadBytes;
+  int16_t         maxPayloadBytes;
   /* Maximum allowed rate in bytes per 30 msec packet */
-  WebRtc_Word16         maxRateInBytes;
+  int16_t         maxRateInBytes;
 
   /*---
     If set to 1 iSAC will not addapt the frame-size, if used in
     channel-adaptive mode. The initial value will be used for all rates.
     ---*/
-  WebRtc_Word16         enforceFrameSize;
+  int16_t         enforceFrameSize;
 
   /*-----
     This records the BWE index the encoder injected into the bit-stream.
@@ -350,7 +350,7 @@
     a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
     called only once per each encode.
     -----*/
-  WebRtc_Word16         lastBWIdx;
+  int16_t         lastBWIdx;
 } ISACLBEncStruct;
 
 typedef struct {
@@ -366,14 +366,14 @@
                                             LB_TOTAL_DELAY_SAMPLES];
   double                  bottleneck;
   /* Maximum allowed number of bits for a 30 msec packet */
-  //WebRtc_Word16        payloadLimitBytes30;
+  //int16_t        payloadLimitBytes30;
   /* Maximum allowed number of bits for both 30 and 60 msec packet */
-  //WebRtc_Word16        maxPayloadBytes;
-  WebRtc_Word16             maxPayloadSizeBytes;
+  //int16_t        maxPayloadBytes;
+  int16_t             maxPayloadSizeBytes;
 
   double                  lastLPCVec[UB_LPC_ORDER];
-  WebRtc_Word16             numBytesUsed;
-  WebRtc_Word16             lastJitterInfo;
+  int16_t             numBytesUsed;
+  int16_t             lastJitterInfo;
 } ISACUBEncStruct;
 
 
@@ -422,11 +422,11 @@
   double       loFiltGain[SUBFRAMES];
   double       hiFiltGain[SUBFRAMES];
   /* Upper boundary of interval W */
-  WebRtc_UWord32 W_upper;
-  WebRtc_UWord32 streamval;
+  uint32_t W_upper;
+  uint32_t streamval;
   /* Index to the current position in bytestream */
-  WebRtc_UWord32 stream_index;
-  WebRtc_UWord8  stream[3];
+  uint32_t stream_index;
+  uint8_t  stream[3];
 } transcode_obj;
 
 
@@ -442,19 +442,19 @@
   double                    MaxDelay;
 
   /* 0 = adaptive; 1 = instantaneous */
-  WebRtc_Word16               codingMode;
+  int16_t               codingMode;
 
   // overall bottleneck of the codec
-  WebRtc_Word32               bottleneck;
+  int32_t               bottleneck;
 
   // QMF Filter state
-  WebRtc_Word32               analysisFBState1[FB_STATE_SIZE_WORD32];
-  WebRtc_Word32               analysisFBState2[FB_STATE_SIZE_WORD32];
-  WebRtc_Word32               synthesisFBState1[FB_STATE_SIZE_WORD32];
-  WebRtc_Word32               synthesisFBState2[FB_STATE_SIZE_WORD32];
+  int32_t               analysisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t               analysisFBState2[FB_STATE_SIZE_WORD32];
+  int32_t               synthesisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t               synthesisFBState2[FB_STATE_SIZE_WORD32];
 
   // Error Code
-  WebRtc_Word16               errorCode;
+  int16_t               errorCode;
 
   // bandwidth of the encoded audio 8, 12 or 16 kHz
   enum ISACBandwidth        bandwidthKHz;
@@ -463,19 +463,19 @@
   enum IsacSamplingRate decoderSamplingRateKHz;
   // Flag to keep track of initializations, lower & upper-band
   // encoder and decoder.
-  WebRtc_Word16               initFlag;
+  int16_t               initFlag;
 
   // Flag to to indicate signal bandwidth switch
-  WebRtc_Word16               resetFlag_8kHz;
+  int16_t               resetFlag_8kHz;
 
   // Maximum allowed rate, measured in Bytes per 30 ms.
-  WebRtc_Word16               maxRateBytesPer30Ms;
+  int16_t               maxRateBytesPer30Ms;
   // Maximum allowed payload-size, measured in Bytes.
-  WebRtc_Word16               maxPayloadSizeBytes;
+  int16_t               maxPayloadSizeBytes;
   /* The expected sampling rate of the input signal. Valid values are 16000,
    * 32000 and 48000. This is not the operation sampling rate of the codec.
    * Input signals at 48 kHz are resampled to 32 kHz, then encoded. */
-  WebRtc_UWord16 in_sample_rate_hz;
+  uint16_t in_sample_rate_hz;
   /* State for the input-resampler. It is only used for 48 kHz input signals. */
   int16_t state_in_resampler[SIZE_RESAMPLER_STATE];
 } ISACMainStruct;
diff --git a/modules/audio_coding/codecs/isac/main/source/transform.c b/modules/audio_coding/codecs/isac/main/source/transform.c
index 97b801a..ea6b579 100644
--- a/modules/audio_coding/codecs/isac/main/source/transform.c
+++ b/modules/audio_coding/codecs/isac/main/source/transform.c
@@ -44,8 +44,8 @@
 
 void WebRtcIsac_Time2Spec(double *inre1,
                          double *inre2,
-                         WebRtc_Word16 *outreQ7,
-                         WebRtc_Word16 *outimQ7,
+                         int16_t *outreQ7,
+                         int16_t *outimQ7,
                          FFTstr *fftstr_obj)
 {
 
@@ -80,10 +80,10 @@
 
     tmp1r = costab2[k];
     tmp1i = sintab2[k];
-    outreQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0);
-    outimQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0);
-    outreQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0);
-    outimQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1r + yi * tmp1i) * 128.0);
+    outreQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0);
+    outimQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0);
+    outreQ7[FRAMESAMPLES_HALF - 1 - k] = (int16_t)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0);
+    outimQ7[FRAMESAMPLES_HALF - 1 - k] = (int16_t)WebRtcIsac_lrint((-yr * tmp1r + yi * tmp1i) * 128.0);
   }
 }
 
diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index db34602..5506bd4 100644
--- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -48,32 +48,32 @@
 	int framecnt, endfile;
 
 	int i, errtype, VADusage = 0, packetLossPercent = 0;
-	WebRtc_Word16 CodingMode;
-	WebRtc_Word32 bottleneck;
-	WebRtc_Word16 framesize = 30;           /* ms */
+	int16_t CodingMode;
+	int32_t bottleneck;
+	int16_t framesize = 30;           /* ms */
 	int cur_framesmpls, err;
 
 	/* Runtime statistics */
 	double starttime, runtime, length_file;
 
-	WebRtc_Word16 stream_len = 0;
-	WebRtc_Word16 declen, lostFrame = 0, declenTC = 0;
+	int16_t stream_len = 0;
+	int16_t declen, lostFrame = 0, declenTC = 0;
 
-	WebRtc_Word16 shortdata[SWBFRAMESAMPLES_10ms];
-	WebRtc_Word16 vaddata[SWBFRAMESAMPLES_10ms*3];
-	WebRtc_Word16 decoded[MAX_FRAMESAMPLES << 1];
-	WebRtc_Word16 decodedTC[MAX_FRAMESAMPLES << 1];
-	WebRtc_UWord16 streamdata[500];
-	WebRtc_Word16	speechType[1];
-    WebRtc_Word16 rateBPS = 0;
-    WebRtc_Word16 fixedFL = 0;
-    WebRtc_Word16 payloadSize = 0;
-    WebRtc_Word32 payloadRate = 0;
+	int16_t shortdata[SWBFRAMESAMPLES_10ms];
+	int16_t vaddata[SWBFRAMESAMPLES_10ms*3];
+	int16_t decoded[MAX_FRAMESAMPLES << 1];
+	int16_t decodedTC[MAX_FRAMESAMPLES << 1];
+	uint16_t streamdata[500];
+	int16_t	speechType[1];
+    int16_t rateBPS = 0;
+    int16_t fixedFL = 0;
+    int16_t payloadSize = 0;
+    int32_t payloadRate = 0;
     int setControlBWE = 0;
     short FL, testNum;
 	char version_number[20];
     FILE  *plFile;
-    WebRtc_Word32 sendBN;
+    int32_t sendBN;
 
 #ifdef _DEBUG
 	FILE *fy;
@@ -92,12 +92,12 @@
     short useAssign = 0;
     //FILE logFile;
     bool doTransCoding = false;
-    WebRtc_Word32 rateTransCoding = 0;
-    WebRtc_UWord16 streamDataTransCoding[600];
-    WebRtc_Word16 streamLenTransCoding = 0;
+    int32_t rateTransCoding = 0;
+    uint16_t streamDataTransCoding[600];
+    int16_t streamLenTransCoding = 0;
     FILE* transCodingFile = NULL;
     FILE* transcodingBitstream = NULL;
-    WebRtc_UWord32 numTransCodingBytes = 0;
+    uint32_t numTransCodingBytes = 0;
 
 	/* only one structure used for ISAC encoder */
 	ISACStruct* ISAC_main_inst = NULL;
@@ -180,14 +180,14 @@
     useAssign = 0;
     //logFile = NULL;
     char transCodingFileName[500];
-    WebRtc_Word16 totFileLoop = 0;
-    WebRtc_Word16 numFileLoop = 0;
+    int16_t totFileLoop = 0;
+    int16_t numFileLoop = 0;
 	for (i = 1; i < argc-2;i++)
     {
         if(!strcmp("-LOOP", argv[i]))
         {
             i++;
-            totFileLoop = (WebRtc_Word16)atol(argv[i]);
+            totFileLoop = (int16_t)atol(argv[i]);
             if(totFileLoop <= 0)
             {
                 fprintf(stderr, "Invalid number of runs for the given input file, %d.", totFileLoop);
@@ -609,8 +609,8 @@
     cout << "\n" << flush;
 
     length_file = 0;
-    WebRtc_Word16 bnIdxTC;
-    WebRtc_Word16 jitterInfoTC;
+    int16_t bnIdxTC;
+    int16_t jitterInfoTC;
     while (endfile == 0)
     {
         /* Call init functions at random, fault test number 7 */
@@ -663,7 +663,7 @@
             {
                 stream_len = WebRtcIsac_Encode(ISAC_main_inst,
                     shortdata,
-                    (WebRtc_Word16*)streamdata);
+                    (int16_t*)streamdata);
                 if((payloadSize != 0) && (stream_len > payloadSize))
                 {
                     if(testNum == 0)
@@ -681,40 +681,40 @@
                 {
                     if(doTransCoding)
                     {
-                        WebRtc_Word16 indexStream;
-                        WebRtc_UWord8 auxUW8;
+                        int16_t indexStream;
+                        uint8_t auxUW8;
 
                         /************************* Main Transcoding stream *******************************/
                         WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, &jitterInfoTC);
                         streamLenTransCoding = WebRtcIsac_GetNewBitStream(
                             ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
-                            (WebRtc_Word16*)streamDataTransCoding, false);
+                            (int16_t*)streamDataTransCoding, false);
                         if(streamLenTransCoding < 0)
                         {
                             fprintf(stderr, "Error in trans-coding\n");
                             exit(0);
                         }
-                        auxUW8 = (WebRtc_UWord8)(((streamLenTransCoding & 0xFF00) >> 8) &  0x00FF);
-                        if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1,
+                        auxUW8 = (uint8_t)(((streamLenTransCoding & 0xFF00) >> 8) &  0x00FF);
+                        if (fwrite(&auxUW8, sizeof(uint8_t), 1,
                                    transcodingBitstream) != 1) {
                           return -1;
                         }
 
-                        auxUW8 = (WebRtc_UWord8)(streamLenTransCoding & 0x00FF);
-                        if (fwrite(&auxUW8, sizeof(WebRtc_UWord8),
+                        auxUW8 = (uint8_t)(streamLenTransCoding & 0x00FF);
+                        if (fwrite(&auxUW8, sizeof(uint8_t),
                                    1, transcodingBitstream) != 1) {
                           return -1;
                         }
 
-                        if (fwrite((WebRtc_UWord8*)streamDataTransCoding,
-                                   sizeof(WebRtc_UWord8),
+                        if (fwrite((uint8_t*)streamDataTransCoding,
+                                   sizeof(uint8_t),
                                    streamLenTransCoding,
                                    transcodingBitstream) !=
                             static_cast<size_t>(streamLenTransCoding)) {
                           return -1;
                         }
 
-                        WebRtcIsac_ReadBwIndex((WebRtc_Word16*)streamDataTransCoding, &indexStream);
+                        WebRtcIsac_ReadBwIndex((int16_t*)streamDataTransCoding, &indexStream);
                         if(indexStream != bnIdxTC)
                         {
                             fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n");
@@ -781,13 +781,13 @@
         if(lostFrame)
         {
             stream_len = WebRtcIsac_GetRedPayload(ISAC_main_inst,
-                (WebRtc_Word16*)streamdata);
+                (int16_t*)streamdata);
 
             if(doTransCoding)
             {
                 streamLenTransCoding = WebRtcIsac_GetNewBitStream(
                     ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
-                    (WebRtc_Word16*)streamDataTransCoding, true);
+                    (int16_t*)streamDataTransCoding, true);
                 if(streamLenTransCoding < 0)
                 {
                     fprintf(stderr, "Error in RED trans-coding\n");
@@ -873,7 +873,7 @@
 
             /* Call getFramelen, only used here for function test */
             err = WebRtcIsac_ReadFrameLen(ISAC_main_inst,
-                (WebRtc_Word16*)streamdata, &FL);
+                (int16_t*)streamdata, &FL);
             if(err < 0)
             {
                 /* exit if returned with error */
@@ -951,7 +951,7 @@
         /* Write decoded speech frame to file */
         if((declen > 0) && (numFileLoop == 0))
         {
-          if (fwrite(decoded, sizeof(WebRtc_Word16), declen,
+          if (fwrite(decoded, sizeof(int16_t), declen,
                      outp) != static_cast<size_t>(declen)) {
             return -1;
           }
@@ -959,7 +959,7 @@
 
         if((declenTC > 0) && (numFileLoop == 0))
         {
-          if (fwrite(decodedTC, sizeof(WebRtc_Word16), declen,
+          if (fwrite(decodedTC, sizeof(int16_t), declen,
                      transCodingFile) != static_cast<size_t>(declen)) {
             return -1;
           }
diff --git a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 196eb11..fd70eca 100644
--- a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -36,7 +36,7 @@
   FILE* outFile[MAX_NUM_CLIENTS];
 
   ISACStruct* codecInstance[MAX_NUM_CLIENTS];
-  WebRtc_Word32 resamplerState[MAX_NUM_CLIENTS][8];
+  int32_t resamplerState[MAX_NUM_CLIENTS][8];
 
   int encoderSampRate[MAX_NUM_CLIENTS];
 
@@ -141,7 +141,7 @@
       return -1;
     }
     memset(packetData[clientCntr], 0, sizeof(BottleNeckModel));
-    memset(resamplerState[clientCntr], 0, sizeof(WebRtc_Word32) * 8);
+    memset(resamplerState[clientCntr], 0, sizeof(int32_t) * 8);
   }
 
   for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
@@ -284,7 +284,7 @@
 
       streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
                                     audioBuff10ms, (short*)bitStream);
-      WebRtc_Word16 ggg;
+      int16_t ggg;
       if (streamLen > 0) {
         if((  WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],
                                       (short *) bitStream, &ggg))<0)
@@ -319,12 +319,12 @@
 
           if(codingMode == 0)
           {
-            WebRtc_Word32 bn;
+            int32_t bn;
             WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn);
             printf("[%d] ", bn);
           }
-          //WebRtc_Word16 rateIndexLB;
-          //WebRtc_Word16 rateIndexUB;
+          //int16_t rateIndexLB;
+          //int16_t rateIndexUB;
           //WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx],
           //    &rateIndexLB, &rateIndexUB);
           //printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB);
diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index f1b78c2..4175890 100644
--- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -62,21 +62,21 @@
     unsigned long totalBitsRCU = 0;
     unsigned long totalsmpls =0;
 
-    WebRtc_Word32   bottleneck = 39;
-    WebRtc_Word16   frameSize = 30;           /* ms */
-    WebRtc_Word16   codingMode = 1;
-    WebRtc_Word16   shortdata[FRAMESAMPLES_SWB_10ms];
-    WebRtc_Word16   decoded[MAX_FRAMESAMPLES_SWB];
-    //WebRtc_UWord16  streamdata[1000];
-    WebRtc_Word16   speechType[1];
-    WebRtc_Word16   payloadLimit;
-    WebRtc_Word32   rateLimit;
+    int32_t   bottleneck = 39;
+    int16_t   frameSize = 30;           /* ms */
+    int16_t   codingMode = 1;
+    int16_t   shortdata[FRAMESAMPLES_SWB_10ms];
+    int16_t   decoded[MAX_FRAMESAMPLES_SWB];
+    //uint16_t  streamdata[1000];
+    int16_t   speechType[1];
+    int16_t   payloadLimit;
+    int32_t   rateLimit;
     ISACStruct*   ISAC_main_inst;
 
-    WebRtc_Word16   stream_len = 0;
-    WebRtc_Word16   declen;
-    WebRtc_Word16   err;
-    WebRtc_Word16   cur_framesmpls;
+    int16_t   stream_len = 0;
+    int16_t   declen;
+    int16_t   err;
+    int16_t   cur_framesmpls;
     int           endfile;
 #ifdef WIN32
     double        length_file;
@@ -95,17 +95,17 @@
     FILE*         averageFile;
     int           sampFreqKHz;
     int           samplesIn10Ms;
-    WebRtc_Word16   maxStreamLen = 0;
+    int16_t   maxStreamLen = 0;
     char          histFileName[500];
     char          averageFileName[500];
     unsigned int  hist[600];
     unsigned int  tmpSumStreamLen = 0;
     unsigned int  packetCntr = 0;
     unsigned int  lostPacketCntr = 0;
-    WebRtc_UWord16  payload[600];
-    WebRtc_UWord16  payloadRCU[600];
-    WebRtc_UWord16  packetLossPercent = 0;
-    WebRtc_Word16   rcuStreamLen = 0;
+    uint16_t  payload[600];
+    uint16_t  payloadRCU[600];
+    uint16_t  packetLossPercent = 0;
+    int16_t   rcuStreamLen = 0;
 	int onlyEncode;
 	int onlyDecode;
 
@@ -164,7 +164,7 @@
     sscanf(argv[1], "%s", inname);
     sscanf(argv[2], "%s", outname);
     codingMode = readSwitch(argc, argv, "-I");
-    sampFreqKHz = (WebRtc_Word16)readParamInt(argc, argv, "-fs", 32);
+    sampFreqKHz = (int16_t)readParamInt(argc, argv, "-fs", 32);
     if(readParamString(argc, argv, "-h", histFileName, 500) > 0)
     {
         histFile = fopen(histFileName, "a");
@@ -218,7 +218,7 @@
 valid values are 8 and 16.\n", sampFreqKHz);
         exit(-1);
     }
-    payloadLimit = (WebRtc_Word16)readParamInt(argc, argv, "-plim", 400);
+    payloadLimit = (int16_t)readParamInt(argc, argv, "-plim", 400);
     rateLimit = readParamInt(argc, argv, "-rlim", 106800);
 
     if ((inp = fopen(inname,"rb")) == NULL) {
@@ -279,7 +279,7 @@
     }
 
     //{
-    //    WebRtc_Word32 b1, b2;
+    //    int32_t b1, b2;
     //    FILE* fileID = fopen("GetBNTest.txt", "w");
     //    b2 = 32100;
     //    while(b2 <= 52000)
@@ -341,18 +341,18 @@
 
 		if(onlyDecode)
 		{
-			WebRtc_UWord8 auxUW8;
+			uint8_t auxUW8;
                         size_t auxSizet;
-			if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
+			if(fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1)
 			{
 				break;
 			}
-			stream_len = ((WebRtc_UWord8)auxUW8) << 8;
-			if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
+			stream_len = ((uint8_t)auxUW8) << 8;
+			if(fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1)
 			{
 				break;
 			}
-			stream_len |= (WebRtc_UWord16)auxUW8;
+			stream_len |= (uint16_t)auxUW8;
                         auxSizet = (size_t)stream_len;
                         if(fread(payload, 1, auxSizet, inp) < auxSizet)
 			{
@@ -374,7 +374,7 @@
 
 				//-------- iSAC encoding ---------
 				stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata,
-					(WebRtc_Word16*)payload);
+					(int16_t*)payload);
 
 				if(stream_len < 0)
 				{
@@ -393,7 +393,7 @@
 				break;
 			}
 
-			rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, (WebRtc_Word16*)payloadRCU);
+			rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, (int16_t*)payloadRCU);
 
 			get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
 				sampFreqKHz * 1000, sampFreqKHz * 1000);
@@ -430,14 +430,14 @@
 
 		if(onlyEncode)
 		{
-                  WebRtc_UWord8 auxUW8;
-                  auxUW8 = (WebRtc_UWord8)(((stream_len & 0x7F00) >> 8) & 0xFF);
-                  if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp) != 1) {
+                  uint8_t auxUW8;
+                  auxUW8 = (uint8_t)(((stream_len & 0x7F00) >> 8) & 0xFF);
+                  if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
                     return -1;
                   }
 
-                  auxUW8 = (WebRtc_UWord8)(stream_len & 0xFF);
-                  if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp) != 1) {
+                  auxUW8 = (uint8_t)(stream_len & 0xFF);
+                  if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
                     return -1;
                   }
                   if (fwrite(payload, 1, stream_len,
@@ -470,7 +470,7 @@
 			}
 
 			// Write decoded speech frame to file
-                        if (fwrite(decoded, sizeof(WebRtc_Word16),
+                        if (fwrite(decoded, sizeof(int16_t),
                                    declen, outp) != (size_t)declen) {
                           return -1;
                         }
diff --git a/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
index e3cac4d..76ba2ff 100644
--- a/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
@@ -35,9 +35,9 @@
  * Returned value		: Size in bytes of speechOut16b
  */
 
-WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *speechOut16b);
+int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
+                               int16_t len,
+                               int16_t *speechOut16b);
 
 /****************************************************************************
  * WebRtcPcm16b_Encode(...)
@@ -54,9 +54,9 @@
  * Returned value		: Size in bytes of speech8b
  */
 
-WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
-                                  WebRtc_Word16 len,
-                                  unsigned char *speech8b);
+int16_t WebRtcPcm16b_Encode(int16_t *speech16b,
+                            int16_t len,
+                            unsigned char *speech8b);
 
 /****************************************************************************
  * WebRtcPcm16b_DecodeW16(...)
@@ -73,11 +73,11 @@
  * Returned value		: Samples in speechOut16b
  */
 
-WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
-                                     WebRtc_Word16 *speechIn16b,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *speechOut16b,
-                                     WebRtc_Word16* speechType);
+int16_t WebRtcPcm16b_DecodeW16(void *inst,
+                               int16_t *speechIn16b,
+                               int16_t len,
+                               int16_t *speechOut16b,
+                               int16_t* speechType);
 
 /****************************************************************************
  * WebRtcPcm16b_Decode(...)
@@ -95,9 +95,9 @@
  */
 
 
-WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
-                                  WebRtc_Word16 len,
-                                  WebRtc_Word16 *speech16b);
+int16_t WebRtcPcm16b_Decode(unsigned char *speech8b,
+                            int16_t len,
+                            int16_t *speech16b);
 
 #ifdef __cplusplus
 }
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b.c b/modules/audio_coding/codecs/pcm16b/pcm16b.c
index 0cff5dd..04814b7 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -24,17 +24,17 @@
 
 
 
-/* Encoder with WebRtc_Word16 Output */
-WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *speechOut16b)
+/* Encoder with int16_t Output */
+int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
+                               int16_t len,
+                               int16_t *speechOut16b)
 {
 #ifdef WEBRTC_BIG_ENDIAN
     WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);
 #else
     int i;
     for (i=0;i<len;i++) {
-        speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|((((WebRtc_UWord16)speechIn16b[i])<<8)&0xFF00);
+        speechOut16b[i]=(((uint16_t)speechIn16b[i])>>8)|((((uint16_t)speechIn16b[i])<<8)&0xFF00);
     }
 #endif
     return(len<<1);
@@ -42,14 +42,14 @@
 
 
 /* Encoder with char Output (old version) */
-WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
-                                  WebRtc_Word16 len,
-                                  unsigned char *speech8b)
+int16_t WebRtcPcm16b_Encode(int16_t *speech16b,
+                            int16_t len,
+                            unsigned char *speech8b)
 {
-    WebRtc_Word16 samples=len*2;
-    WebRtc_Word16 pos;
-    WebRtc_Word16 short1;
-    WebRtc_Word16 short2;
+    int16_t samples=len*2;
+    int16_t pos;
+    int16_t short1;
+    int16_t short2;
     for (pos=0;pos<len;pos++) {
         short1=HIGHEND & speech16b[pos];
         short2=LOWEND & speech16b[pos];
@@ -61,21 +61,21 @@
 }
 
 
-/* Decoder with WebRtc_Word16 Input instead of char when the WebRtc_Word16 Encoder is used */
-WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
-                                     WebRtc_Word16 *speechIn16b,
-                                     WebRtc_Word16 len,
-                                     WebRtc_Word16 *speechOut16b,
-                                     WebRtc_Word16* speechType)
+/* Decoder with int16_t Input instead of char when the int16_t Encoder is used */
+int16_t WebRtcPcm16b_DecodeW16(void *inst,
+                               int16_t *speechIn16b,
+                               int16_t len,
+                               int16_t *speechOut16b,
+                               int16_t* speechType)
 {
 #ifdef WEBRTC_BIG_ENDIAN
-    WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(WebRtc_Word16)+1)>>1));
+    WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(int16_t)+1)>>1));
 #else
     int i;
     int samples=len>>1;
 
     for (i=0;i<samples;i++) {
-        speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|(((WebRtc_UWord16)(speechIn16b[i]&0xFF))<<8);
+        speechOut16b[i]=(((uint16_t)speechIn16b[i])>>8)|(((uint16_t)(speechIn16b[i]&0xFF))<<8);
     }
 #endif
 
@@ -88,13 +88,13 @@
 }
 
 /* "old" version of the decoder that uses char as input (not used in NetEq any more) */
-WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
-                                  WebRtc_Word16 len,
-                                  WebRtc_Word16 *speech16b)
+int16_t WebRtcPcm16b_Decode(unsigned char *speech8b,
+                            int16_t len,
+                            int16_t *speech16b)
 {
-    WebRtc_Word16 samples=len>>1;
-    WebRtc_Word16 pos;
-    WebRtc_Word16 shortval;
+    int16_t samples=len>>1;
+    int16_t pos;
+    int16_t shortval;
     for (pos=0;pos<samples;pos++) {
         shortval=((unsigned short) speech8b[pos*2]);
         shortval=(shortval<<8)&HIGHEND;
diff --git a/modules/audio_coding/main/interface/audio_coding_module.h b/modules/audio_coding/main/interface/audio_coding_module.h
index 11c2556..a2e7efe 100644
--- a/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/modules/audio_coding/main/interface/audio_coding_module.h
@@ -31,12 +31,12 @@
  public:
   virtual ~AudioPacketizationCallback() {}
 
-  virtual WebRtc_Word32 SendData(
+  virtual int32_t SendData(
       FrameType frame_type,
-      WebRtc_UWord8 payload_type,
-      WebRtc_UWord32 timestamp,
-      const WebRtc_UWord8* payload_data,
-      WebRtc_UWord16 payload_len_bytes,
+      uint8_t payload_type,
+      uint32_t timestamp,
+      const uint8_t* payload_data,
+      uint16_t payload_len_bytes,
       const RTPFragmentationHeader* fragmentation) = 0;
 };
 
@@ -45,8 +45,8 @@
  public:
   virtual ~AudioCodingFeedback() {}
 
-  virtual WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digit_dtmf,
-                                     const bool end) = 0;
+  virtual int32_t IncomingDtmf(const uint8_t digit_dtmf,
+                               const bool end) = 0;
 };
 
 // Callback class used for reporting VAD decision
@@ -54,7 +54,7 @@
  public:
   virtual ~ACMVADCallback() {}
 
-  virtual WebRtc_Word32 InFrameType(WebRtc_Word16 frameType) = 0;
+  virtual int32_t InFrameType(int16_t frameType) = 0;
 };
 
 // Callback class used for reporting receiver statistics
@@ -62,12 +62,12 @@
  public:
   virtual ~ACMVQMonCallback() {}
 
-  virtual WebRtc_Word32 NetEqStatistics(
-      const WebRtc_Word32 id, // current ACM id
-      const WebRtc_UWord16 MIUsValid, // valid voice duration in ms
-      const WebRtc_UWord16 MIUsReplaced, // concealed voice duration in ms
-      const WebRtc_UWord8 eventFlags, // concealed voice flags
-      const WebRtc_UWord16 delayMS) = 0; // average delay in ms
+  virtual int32_t NetEqStatistics(
+      const int32_t id, // current ACM id
+      const uint16_t MIUsValid, // valid voice duration in ms
+      const uint16_t MIUsReplaced, // concealed voice duration in ms
+      const uint8_t eventFlags, // concealed voice flags
+      const uint16_t delayMS) = 0; // average delay in ms
 };
 
 class AudioCodingModule: public Module {
@@ -79,7 +79,7 @@
   ///////////////////////////////////////////////////////////////////////////
   // Creation and destruction of a ACM
   //
-  static AudioCodingModule* Create(const WebRtc_Word32 id);
+  static AudioCodingModule* Create(const int32_t id);
 
   static void Destroy(AudioCodingModule* module);
 
@@ -88,16 +88,16 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_UWord8 NumberOfCodecs()
+  // uint8_t NumberOfCodecs()
   // Returns number of supported codecs.
   //
   // Return value:
   //   number of supported codecs.
   ///
-  static WebRtc_UWord8 NumberOfCodecs();
+  static uint8_t NumberOfCodecs();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Codec()
+  // int32_t Codec()
   // Get supported codec with list number.
   //
   // Input:
@@ -111,10 +111,10 @@
   //   -1 if the list number (list_id) is invalid.
   //    0 if succeeded.
   //
-  static WebRtc_Word32 Codec(WebRtc_UWord8 list_id, CodecInst* codec);
+  static int32_t Codec(uint8_t list_id, CodecInst* codec);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Codec()
+  // int32_t Codec()
   // Get supported codec with the given codec name, sampling frequency, and
   // a given number of channels.
   //
@@ -132,11 +132,11 @@
   //   -1 if no codec matches the given parameters.
   //    0 if succeeded.
   //
-  static WebRtc_Word32 Codec(const char* payload_name, CodecInst* codec,
-                             int sampling_freq_hz, int channels);
+  static int32_t Codec(const char* payload_name, CodecInst* codec,
+                       int sampling_freq_hz, int channels);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Codec()
+  // int32_t Codec()
   //
   // Returns the list number of the given codec name, sampling frequency, and
   // a given number of channels.
@@ -151,7 +151,7 @@
   //   if the codec is found, the index of the codec in the list,
   //   -1 if the codec is not found.
   //
-  static WebRtc_Word32 Codec(const char* payload_name, int sampling_freq_hz,
+  static int32_t Codec(const char* payload_name, int sampling_freq_hz,
                              int channels);
 
   ///////////////////////////////////////////////////////////////////////////
@@ -173,7 +173,7 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 InitializeSender()
+  // int32_t InitializeSender()
   // Any encoder-related state of ACM will be initialized to the
   // same state when ACM is created. This will not interrupt or
   // effect decoding functionality of ACM. ACM will lose all the
@@ -184,10 +184,10 @@
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 InitializeSender() = 0;
+  virtual int32_t InitializeSender() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ResetEncoder()
+  // int32_t ResetEncoder()
   // This API resets the states of encoder. All the encoder settings, such as
   // send-codec or VAD/DTX, will be preserved.
   //
@@ -195,10 +195,10 @@
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 ResetEncoder() = 0;
+  virtual int32_t ResetEncoder() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterSendCodec()
+  // int32_t RegisterSendCodec()
   // Registers a codec, specified by |send_codec|, as sending codec.
   // This API can be called multiple of times to register Codec. The last codec
   // registered overwrites the previous ones.
@@ -224,7 +224,7 @@
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 RegisterSendCodec(const CodecInst& send_codec) = 0;
+  virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int RegisterSecondarySendCodec()
@@ -254,7 +254,7 @@
   virtual void UnregisterSecondarySendCodec() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SendCodec()
+  // int32_t SendCodec()
   // Get parameters for the codec currently registered as send codec.
   //
   // Output:
@@ -264,7 +264,7 @@
   //   -1 if failed to get send codec,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 SendCodec(CodecInst* current_send_codec) const = 0;
+  virtual int32_t SendCodec(CodecInst* current_send_codec) const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int SecondarySendCodec()
@@ -280,27 +280,27 @@
   virtual int SecondarySendCodec(CodecInst* secondary_codec) const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SendFrequency()
+  // int32_t SendFrequency()
   // Get the sampling frequency of the current encoder in Hertz.
   //
   // Return value:
   //   positive; sampling frequency [Hz] of the current encoder.
   //   -1 if an error has happened.
   //
-  virtual WebRtc_Word32 SendFrequency() const = 0;
+  virtual int32_t SendFrequency() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Bitrate()
+  // int32_t Bitrate()
   // Get encoding bit-rate in bits per second.
   //
   // Return value:
   //   positive; encoding rate in bits/sec,
   //   -1 if an error is happened.
   //
-  virtual WebRtc_Word32 SendBitrate() const = 0;
+  virtual int32_t SendBitrate() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetReceivedEstimatedBandwidth()
+  // int32_t SetReceivedEstimatedBandwidth()
   // Set available bandwidth [bits/sec] of the up-link channel.
   // This information is used for traffic shaping, and is currently only
   // supported if iSAC is the send codec.
@@ -312,11 +312,11 @@
   //   -1 if error occurred in setting the bandwidth,
   //    0 bandwidth is set successfully.
   //
-  virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
-      const WebRtc_Word32 bw) = 0;
+  virtual int32_t SetReceivedEstimatedBandwidth(
+      const int32_t bw) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterTransportCallback()
+  // int32_t RegisterTransportCallback()
   // Register a transport callback which will be called to deliver
   // the encoded buffers whenever Process() is called and a
   // bit-stream is ready.
@@ -331,11 +331,11 @@
   //   -1 if the transport callback could not be registered
   //    0 if registration is successful.
   //
-  virtual WebRtc_Word32 RegisterTransportCallback(
+  virtual int32_t RegisterTransportCallback(
       AudioPacketizationCallback* transport) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Add10MsData()
+  // int32_t Add10MsData()
   // Add 10MS of raw (PCM) audio data to the encoder. If the sampling
   // frequency of the audio does not match the sampling frequency of the
   // current encoder ACM will resample the audio.
@@ -352,14 +352,14 @@
   //   < -1   to add the frame to the buffer n samples had to be
   //          overwritten, -n is the return value in this case.
   //
-  virtual WebRtc_Word32 Add10MsData(const AudioFrame& audio_frame) = 0;
+  virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // (FEC) Forward Error Correction
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetFECStatus(const bool enable)
+  // int32_t SetFECStatus(const bool enable)
   // configure FEC status i.e. on/off.
   //
   // RFC 2198 describes a solution which has a single payload type which
@@ -380,7 +380,7 @@
   //   -1 if failed to set FEC status,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 SetFECStatus(const bool enable_fec) = 0;
+  virtual int32_t SetFECStatus(const bool enable_fec) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // bool FECStatus()
@@ -397,7 +397,7 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetVAD()
+  // int32_t SetVAD()
   // If DTX is enabled & the codec does not have internal DTX/VAD
   // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
   //
@@ -422,12 +422,12 @@
   //   -1 if failed to set up VAD/DTX,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 SetVAD(const bool enable_dtx = true,
+  virtual int32_t SetVAD(const bool enable_dtx = true,
                                const bool enable_vad = false,
                                const ACMVADMode vad_mode = VADNormal) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 VAD()
+  // int32_t VAD()
   // Get VAD status.
   //
   // Outputs:
@@ -441,11 +441,11 @@
   //   -1 if fails to retrieve the setting of DTX/VAD,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 VAD(bool* dtx_enabled, bool* vad_enabled,
+  virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
                             ACMVADMode* vad_mode) const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ReplaceInternalDTXWithWebRtc()
+  // int32_t ReplaceInternalDTXWithWebRtc()
   // Used to replace codec internal DTX scheme with WebRtc. This is only
   // supported for G729, where this call replaces AnnexB with WebRtc DTX.
   //
@@ -458,11 +458,11 @@
   //   -1 if failed to replace codec internal DTX with WebRtc,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
+  virtual int32_t ReplaceInternalDTXWithWebRtc(
       const bool use_webrtc_dtx = false) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 IsInternalDTXReplacedWithWebRtc()
+  // int32_t IsInternalDTXReplacedWithWebRtc()
   // Get status if the codec internal DTX (when such exists) is replaced with
   // WebRtc DTX. This is only supported for G729.
   //
@@ -475,11 +475,11 @@
   //   -1 if failed to determine if codec internal DTX is replaced with WebRtc,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
+  virtual int32_t IsInternalDTXReplacedWithWebRtc(
       bool* uses_webrtc_dtx) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterVADCallback()
+  // int32_t RegisterVADCallback()
   // Call this method to register a callback function which is called
   // any time that ACM encounters an empty frame. That is a frame which is
   // recognized inactive. Depending on the codec WebRtc VAD or internal codec
@@ -492,14 +492,14 @@
   //   -1 if failed to register the callback function.
   //    0 if the callback function is registered successfully.
   //
-  virtual WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vad_callback) = 0;
+  virtual int32_t RegisterVADCallback(ACMVADCallback* vad_callback) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   //   Receiver
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 InitializeReceiver()
+  // int32_t InitializeReceiver()
   // Any decoder-related state of ACM will be initialized to the
   // same state when ACM is created. This will not interrupt or
   // effect encoding functionality of ACM. ACM would lose all the
@@ -511,10 +511,10 @@
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 InitializeReceiver() = 0;
+  virtual int32_t InitializeReceiver() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ResetDecoder()
+  // int32_t ResetDecoder()
   // This API resets the states of decoders. ACM will not lose any
   // decoder-related settings, such as registered codecs.
   //
@@ -522,29 +522,29 @@
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word32 ResetDecoder() = 0;
+  virtual int32_t ResetDecoder() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ReceiveFrequency()
+  // int32_t ReceiveFrequency()
   // Get sampling frequency of the last received payload.
   //
   // Return value:
   //   non-negative the sampling frequency in Hertz.
   //   -1 if an error has occurred.
   //
-  virtual WebRtc_Word32 ReceiveFrequency() const = 0;
+  virtual int32_t ReceiveFrequency() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 PlayoutFrequency()
+  // int32_t PlayoutFrequency()
   // Get sampling frequency of audio played out.
   //
   // Return value:
   //   the sampling frequency in Hertz.
   //
-  virtual WebRtc_Word32 PlayoutFrequency() const = 0;
+  virtual int32_t PlayoutFrequency() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterReceiveCodec()
+  // int32_t RegisterReceiveCodec()
   // Register possible decoders, can be called multiple times for
   // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
   //
@@ -557,11 +557,11 @@
   //   -1 if failed to register the codec
   //    0 if the codec registered successfully.
   //
-  virtual WebRtc_Word32 RegisterReceiveCodec(
+  virtual int32_t RegisterReceiveCodec(
       const CodecInst& receive_codec) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 UnregisterReceiveCodec()
+  // int32_t UnregisterReceiveCodec()
   // Unregister the codec currently registered with a specific payload type
   // from the list of possible receive codecs.
   //
@@ -573,11 +573,11 @@
   //   -1 if fails to unregister.
   //    0 if the given codec is successfully unregistered.
   //
-  virtual WebRtc_Word32 UnregisterReceiveCodec(
-      const WebRtc_Word16 payload_type) = 0;
+  virtual int32_t UnregisterReceiveCodec(
+      const int16_t payload_type) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ReceiveCodec()
+  // int32_t ReceiveCodec()
   // Get the codec associated with last received payload.
   //
   // Output:
@@ -589,10 +589,10 @@
   //   -1 if failed to retrieve the codec,
   //    0 if the codec is successfully retrieved.
   //
-  virtual WebRtc_Word32 ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
+  virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 IncomingPacket()
+  // int32_t IncomingPacket()
   // Call this function to insert a parsed RTP packet into ACM.
   //
   // Inputs:
@@ -605,12 +605,12 @@
   //   -1 if failed to push in the payload
   //    0 if payload is successfully pushed in.
   //
-  virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
-                                       const WebRtc_Word32 payload_len_bytes,
+  virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
+                                       const int32_t payload_len_bytes,
                                        const WebRtcRTPHeader& rtp_info) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 IncomingPayload()
+  // int32_t IncomingPayload()
   // Call this API to push incoming payloads when there is no rtp-info.
   // The rtp-info will be created in ACM. One usage for this API is when
   // pre-encoded files are pushed in ACM
@@ -633,13 +633,13 @@
   //   -1 if failed to push in the payload
   //    0 if payload is successfully pushed in.
   //
-  virtual WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incoming_payload,
-                                        const WebRtc_Word32 payload_len_byte,
-                                        const WebRtc_UWord8 payload_type,
-                                        const WebRtc_UWord32 timestamp = 0) = 0;
+  virtual int32_t IncomingPayload(const uint8_t* incoming_payload,
+                                        const int32_t payload_len_byte,
+                                        const uint8_t payload_type,
+                                        const uint32_t timestamp = 0) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetMinimumPlayoutDelay()
+  // int32_t SetMinimumPlayoutDelay()
   // Set Minimum playout delay, used for lip-sync.
   //
   // Input:
@@ -649,10 +649,10 @@
   //   -1 if failed to set the delay,
   //    0 if the minimum delay is set.
   //
-  virtual WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 time_ms) = 0;
+  virtual int32_t SetMinimumPlayoutDelay(const int32_t time_ms) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterIncomingMessagesCallback()
+  // int32_t RegisterIncomingMessagesCallback()
   // Used by the module to deliver messages to the codec module/application
   // when a DTMF tone is detected, as well as when it stopped.
   //
@@ -669,13 +669,13 @@
   //   -1 if the message callback could not be registered
   //    0 if registration is successful.
   //
-  virtual WebRtc_Word32
+  virtual int32_t
       RegisterIncomingMessagesCallback(
           AudioCodingFeedback* in_message_callback,
           const ACMCountries cpt = ACMDisableCountryDetection) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetDtmfPlayoutStatus()
+  // int32_t SetDtmfPlayoutStatus()
   // Configure DTMF playout, i.e. whether out-of-band
   // DTMF tones are played or not.
   //
@@ -687,7 +687,7 @@
   //   -1 if the method fails, e.g. DTMF playout is not supported.
   //    0 if the status is set successfully.
   //
-  virtual WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable) = 0;
+  virtual int32_t SetDtmfPlayoutStatus(const bool enable) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // bool DtmfPlayoutStatus()
@@ -700,7 +700,7 @@
   virtual bool DtmfPlayoutStatus() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetBackgroundNoiseMode()
+  // int32_t SetBackgroundNoiseMode()
   // Sets the mode of the background noise playout in an event of long
   // packet loss burst. For the valid modes see the declaration of
   // ACMBackgroundNoiseMode in audio_coding_module_typedefs.h.
@@ -712,11 +712,11 @@
   //   -1 if failed to set the mode.
   //    0 if succeeded in setting the mode.
   //
-  virtual WebRtc_Word32 SetBackgroundNoiseMode(
+  virtual int32_t SetBackgroundNoiseMode(
       const ACMBackgroundNoiseMode mode) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 BackgroundNoiseMode()
+  // int32_t BackgroundNoiseMode()
   // Call this method to get the mode of the background noise playout.
   // Playout of background noise is a result of a long packet loss burst.
   // See ACMBackgroundNoiseMode in audio_coding_module_typedefs.h for
@@ -730,26 +730,26 @@
   //   -1 if ACM failed to output a valid mode.
   //
   // TODO(tlegrand): Change function to return the mode.
-  virtual WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode* mode) = 0;
+  virtual int32_t BackgroundNoiseMode(ACMBackgroundNoiseMode* mode) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 PlayoutTimestamp()
+  // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
   // audio of the packet in question. This function returns the timestamp of
   // the latest audio obtained by calling PlayoutData10ms().
   //
   // Input:
-  //   -timestamp          : a reference to a WebRtc_UWord32 to receive the
+  //   -timestamp          : a reference to a uint32_t to receive the
   //                         timestamp.
   // Return value:
   //    0 if the output is a correct timestamp.
   //   -1 if failed to output the correct timestamp.
   //
   // TODO(tlegrand): Change function to return the timestamp.
-  virtual WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32* timestamp) = 0;
+  virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 DecoderEstimatedBandwidth()
+  // int32_t DecoderEstimatedBandwidth()
   // Get the estimate of the Bandwidth, in bits/second, based on the incoming
   // stream. This API is useful in one-way communication scenarios, where
   // the bandwidth information is sent in an out-of-band fashion.
@@ -759,10 +759,10 @@
   //   >0 bandwidth in bits/second.
   //   -1 if failed to get a bandwidth estimate.
   //
-  virtual WebRtc_Word32 DecoderEstimatedBandwidth() const = 0;
+  virtual int32_t DecoderEstimatedBandwidth() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetPlayoutMode()
+  // int32_t SetPlayoutMode()
   // Call this API to set the playout mode. Playout mode could be optimized
   // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
   // optimized to deliver highest audio quality while maintaining a minimum
@@ -781,7 +781,7 @@
   //   -1 if failed to set the mode,
   //    0 if succeeding.
   //
-  virtual WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode) = 0;
+  virtual int32_t SetPlayoutMode(const AudioPlayoutMode mode) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // AudioPlayoutMode PlayoutMode()
@@ -799,7 +799,7 @@
   virtual AudioPlayoutMode PlayoutMode() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 PlayoutData10Ms(
+  // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling
   // frequency. ACM will perform a resampling if required.
   //
@@ -818,7 +818,7 @@
   //   -1 if the function fails,
   //    0 if the function succeeds.
   //
-  virtual WebRtc_Word32 PlayoutData10Ms(WebRtc_Word32 desired_freq_hz,
+  virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
                                         AudioFrame* audio_frame) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
@@ -827,7 +827,7 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 SetReceiveVADMode()
+  // int16_t SetReceiveVADMode()
   // Configure VAD aggressiveness on the incoming stream.
   //
   // Input:
@@ -840,7 +840,7 @@
   //   -1 if fails to set the mode,
   //    0 if the mode is set successfully.
   //
-  virtual WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode) = 0;
+  virtual int16_t SetReceiveVADMode(const ACMVADMode mode) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // ACMVADMode ReceiveVADMode()
@@ -859,7 +859,7 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetISACMaxRate()
+  // int32_t SetISACMaxRate()
   // Set the maximum instantaneous rate of iSAC. For a payload of B bits
   // with a frame-size of T sec the instantaneous rate is B/T bits per
   // second. Therefore, (B/T < |max_rate_bps|) and
@@ -873,11 +873,11 @@
   //   -1 if failed to set the maximum rate.
   //    0 if the maximum rate is set successfully.
   //
-  virtual WebRtc_Word32 SetISACMaxRate(
-      const WebRtc_UWord32 max_rate_bps) = 0;
+  virtual int32_t SetISACMaxRate(
+      const uint32_t max_rate_bps) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetISACMaxPayloadSize()
+  // int32_t SetISACMaxPayloadSize()
   // Set the maximum payload size of iSAC packets. No iSAC payload,
   // regardless of its frame-size, may exceed the given limit. For
   // an iSAC payload of size B bits and frame-size T seconds we have;
@@ -891,11 +891,11 @@
   //   -1 if failed to set the maximum  payload-size.
   //    0 if the given length is set successfully.
   //
-  virtual WebRtc_Word32 SetISACMaxPayloadSize(
-      const WebRtc_UWord16 max_payload_len_bytes) = 0;
+  virtual int32_t SetISACMaxPayloadSize(
+      const uint16_t max_payload_len_bytes) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ConfigISACBandwidthEstimator()
+  // int32_t ConfigISACBandwidthEstimator()
   // Call this function to configure the bandwidth estimator of ISAC.
   // During the adaptation of bit-rate, iSAC automatically adjusts the
   // frame-size (either 30 or 60 ms) to save on RTP header. The initial
@@ -918,9 +918,9 @@
   //   -1 if failed to configure the bandwidth estimator,
   //    0 if the configuration was successfully applied.
   //
-  virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
-      const WebRtc_UWord8 init_frame_size_ms,
-      const WebRtc_UWord16 init_rate_bps,
+  virtual int32_t ConfigISACBandwidthEstimator(
+      const uint8_t init_frame_size_ms,
+      const uint16_t init_rate_bps,
       const bool enforce_frame_size = false) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
@@ -928,7 +928,7 @@
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32  NetworkStatistics()
+  // int32_t  NetworkStatistics()
   // Get network statistics.
   //
   // Input:
@@ -938,7 +938,7 @@
   //   -1 if failed to set the network statistics,
   //    0 if statistics are set successfully.
   //
-  virtual WebRtc_Word32 NetworkStatistics(
+  virtual int32_t NetworkStatistics(
       ACMNetworkStatistics* network_statistics) const = 0;
 
   //
diff --git a/modules/audio_coding/main/interface/audio_coding_module_typedefs.h b/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
index dace4b8..03a7df8 100644
--- a/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
@@ -167,14 +167,14 @@
 // -maxWaitingTimeMs       : max packet waiting time in the buffer
 // -addedSamples           : samples inserted because of packet loss in off mode
 typedef struct {
-  WebRtc_UWord16 currentBufferSize;
-  WebRtc_UWord16 preferredBufferSize;
+  uint16_t currentBufferSize;
+  uint16_t preferredBufferSize;
   bool jitterPeaksFound;
-  WebRtc_UWord16 currentPacketLossRate;
-  WebRtc_UWord16 currentDiscardRate;
-  WebRtc_UWord16 currentExpandRate;
-  WebRtc_UWord16 currentPreemptiveRate;
-  WebRtc_UWord16 currentAccelerateRate;
+  uint16_t currentPacketLossRate;
+  uint16_t currentDiscardRate;
+  uint16_t currentExpandRate;
+  uint16_t currentPreemptiveRate;
+  uint16_t currentAccelerateRate;
   int32_t clockDriftPPM;
   int meanWaitingTimeMs;
   int medianWaitingTimeMs;
diff --git a/modules/audio_coding/main/source/acm_amr.cc b/modules/audio_coding/main/source/acm_amr.cc
index 249fe7b..8e8d6d5 100644
--- a/modules/audio_coding/main/source/acm_amr.cc
+++ b/modules/audio_coding/main/source/acm_amr.cc
@@ -50,7 +50,7 @@
 namespace webrtc {
 
 #ifndef WEBRTC_CODEC_AMR
-ACMAMR::ACMAMR(WebRtc_Word16 /* codec_id */)
+ACMAMR::ACMAMR(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       encoding_mode_(-1),  // Invalid value.
@@ -64,39 +64,39 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* /* bitstream */,
-                                     WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMAMR::InternalEncode(uint8_t* /* bitstream */,
+                               int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                 WebRtc_Word16 /* bitstream_len_byte */,
-                                 WebRtc_Word16* /* audio */,
-                                 WebRtc_Word16* /* audio_samples */,
-                                 WebRtc_Word8* /* speech_type */) {
+int16_t ACMAMR::DecodeSafe(uint8_t* /* bitstream */,
+                           int16_t /* bitstream_len_byte */,
+                           int16_t* /* audio */,
+                           int16_t* /* audio_samples */,
+                           int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMR::EnableDTX() {
+int16_t ACMAMR::EnableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMAMR::DisableDTX() {
+int16_t ACMAMR::DisableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMAMR::InternalInitEncoder(
+int16_t ACMAMR::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMR::InternalInitDecoder(
+int16_t ACMAMR::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                               const CodecInst& /* codec_inst */) {
+int32_t ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                         const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -104,7 +104,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
+int16_t ACMAMR::InternalCreateEncoder() {
   return -1;
 }
 
@@ -112,7 +112,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
+int16_t ACMAMR::InternalCreateDecoder() {
   return -1;
 }
 
@@ -120,7 +120,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+int16_t ACMAMR::SetBitRateSafe(const int32_t /* rate */) {
   return -1;
 }
 
@@ -128,7 +128,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
+int16_t ACMAMR::SetAMREncoderPackingFormat(
     ACMAMRPackingFormat /* packing_format */) {
   return -1;
 }
@@ -137,7 +137,7 @@
   return AMRUndefined;
 }
 
-WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
+int16_t ACMAMR::SetAMRDecoderPackingFormat(
     ACMAMRPackingFormat /* packing_format */) {
   return -1;
 }
@@ -157,7 +157,7 @@
 #define WEBRTC_AMR_MR102  6
 #define WEBRTC_AMR_MR122  7
 
-ACMAMR::ACMAMR(WebRtc_Word16 codec_id)
+ACMAMR::ACMAMR(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       encoding_mode_(-1),  // invalid value
@@ -181,9 +181,9 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* bitstream,
-                                     WebRtc_Word16* bitstream_len_byte) {
-  WebRtc_Word16 vad_decision = 1;
+int16_t ACMAMR::InternalEncode(uint8_t* bitstream,
+                               int16_t* bitstream_len_byte) {
+  int16_t vad_decision = 1;
   // sanity check, if the rate is set correctly. we might skip this
   // sanity check. if rate is not set correctly, initialization flag
   // should be false and should not be here.
@@ -195,7 +195,7 @@
   *bitstream_len_byte = WebRtcAmr_Encode(encoder_inst_ptr_,
                                          &in_audio_[in_audio_ix_read_],
                                          frame_len_smpl_,
-                                         (WebRtc_Word16*)bitstream,
+                                         (int16_t*)bitstream,
                                          encoding_mode_);
 
   // Update VAD, if internal DTX is used
@@ -203,7 +203,7 @@
     if (*bitstream_len_byte <= (7 * frame_len_smpl_ / 160)) {
       vad_decision = 0;
     }
-    for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+    for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
       vad_label_[n] = vad_decision;
     }
   }
@@ -212,15 +212,15 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                 WebRtc_Word16 /* bitstream_len_byte */,
-                                 WebRtc_Word16* /* audio */,
-                                 WebRtc_Word16* /* audio_samples */,
-                                 WebRtc_Word8* /* speech_type */) {
+int16_t ACMAMR::DecodeSafe(uint8_t* /* bitstream */,
+                           int16_t /* bitstream_len_byte */,
+                           int16_t* /* audio */,
+                           int16_t* /* audio_samples */,
+                           int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMAMR::EnableDTX() {
+int16_t ACMAMR::EnableDTX() {
   if (dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -235,7 +235,7 @@
   }
 }
 
-WebRtc_Word16 ACMAMR::DisableDTX() {
+int16_t ACMAMR::DisableDTX() {
   if (!dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -251,8 +251,8 @@
   }
 }
 
-WebRtc_Word16 ACMAMR::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
-  WebRtc_Word16 status = SetBitRateSafe((codec_params->codec_inst).rate);
+int16_t ACMAMR::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+  int16_t status = SetBitRateSafe((codec_params->codec_inst).rate);
   status += (WebRtcAmr_EncoderInit(
       encoder_inst_ptr_, ((codec_params->enable_dtx) ? 1 : 0)) < 0) ? -1 : 0;
   status += (WebRtcAmr_EncodeBitmode(
@@ -260,16 +260,16 @@
   return (status < 0) ? -1 : 0;
 }
 
-WebRtc_Word16 ACMAMR::InternalInitDecoder(
+int16_t ACMAMR::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
-  WebRtc_Word16 status =
+  int16_t status =
       ((WebRtcAmr_DecoderInit(decoder_inst_ptr_) < 0) ? -1 : 0);
   status += WebRtcAmr_DecodeBitmode(decoder_inst_ptr_, decoder_packing_format_);
   return (status < 0) ? -1 : 0;
 }
 
-WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                               const CodecInst& codec_inst) {
+int32_t ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                         const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // Todo:
     // log error
@@ -289,7 +289,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
+int16_t ACMAMR::InternalCreateEncoder() {
   return WebRtcAmr_CreateEnc(&encoder_inst_ptr_);
 }
 
@@ -305,7 +305,7 @@
   encoding_rate_ = 0;  // invalid value
 }
 
-WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
+int16_t ACMAMR::InternalCreateDecoder() {
   return WebRtcAmr_CreateDec(&decoder_inst_ptr_);
 }
 
@@ -319,7 +319,7 @@
   decoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 rate) {
+int16_t ACMAMR::SetBitRateSafe(const int32_t rate) {
   switch (rate) {
     case 4750: {
       encoding_mode_ = WEBRTC_AMR_MR475;
@@ -376,7 +376,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
+int16_t ACMAMR::SetAMREncoderPackingFormat(
     ACMAMRPackingFormat packing_format) {
   if ((packing_format != AMRBandwidthEfficient) &&
       (packing_format != AMROctetAlligned) &&
@@ -398,7 +398,7 @@
   return encoder_packing_format_;
 }
 
-WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
+int16_t ACMAMR::SetAMRDecoderPackingFormat(
     ACMAMRPackingFormat packing_format) {
   if ((packing_format != AMRBandwidthEfficient) &&
       (packing_format != AMROctetAlligned) &&
diff --git a/modules/audio_coding/main/source/acm_amr.h b/modules/audio_coding/main/source/acm_amr.h
index ebff0bb..9c87434 100644
--- a/modules/audio_coding/main/source/acm_amr.h
+++ b/modules/audio_coding/main/source/acm_amr.h
@@ -23,58 +23,58 @@
 
 class ACMAMR: public ACMGenericCodec {
  public:
-  explicit ACMAMR(WebRtc_Word16 codec_id);
+  explicit ACMAMR(int16_t codec_id);
   ~ACMAMR();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 SetAMREncoderPackingFormat(
+  int16_t SetAMREncoderPackingFormat(
       const ACMAMRPackingFormat packing_format);
 
   ACMAMRPackingFormat AMREncoderPackingFormat() const;
 
-  WebRtc_Word16 SetAMRDecoderPackingFormat(
+  int16_t SetAMRDecoderPackingFormat(
       const ACMAMRPackingFormat packing_format);
 
   ACMAMRPackingFormat AMRDecoderPackingFormat() const;
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+  int16_t SetBitRateSafe(const int32_t rate);
 
-  WebRtc_Word16 EnableDTX();
+  int16_t EnableDTX();
 
-  WebRtc_Word16 DisableDTX();
+  int16_t DisableDTX();
 
   AMR_encinst_t_* encoder_inst_ptr_;
   AMR_decinst_t_* decoder_inst_ptr_;
-  WebRtc_Word16 encoding_mode_;
-  WebRtc_Word16 encoding_rate_;
+  int16_t encoding_mode_;
+  int16_t encoding_rate_;
   ACMAMRPackingFormat encoder_packing_format_;
   ACMAMRPackingFormat decoder_packing_format_;
 };
diff --git a/modules/audio_coding/main/source/acm_amrwb.cc b/modules/audio_coding/main/source/acm_amrwb.cc
index caa9494..fb86a3b 100644
--- a/modules/audio_coding/main/source/acm_amrwb.cc
+++ b/modules/audio_coding/main/source/acm_amrwb.cc
@@ -47,7 +47,7 @@
 namespace webrtc {
 
 #ifndef WEBRTC_CODEC_AMRWB
-ACMAMRwb::ACMAMRwb(WebRtc_Word16 /* codec_id */)
+ACMAMRwb::ACMAMRwb(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       encoding_mode_(-1),  // invalid value
@@ -59,40 +59,40 @@
 ACMAMRwb::~ACMAMRwb() {
 }
 
-WebRtc_Word16 ACMAMRwb::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMAMRwb::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMAMRwb::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMRwb::EnableDTX() {
+int16_t ACMAMRwb::EnableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMAMRwb::DisableDTX() {
+int16_t ACMAMRwb::DisableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalInitEncoder(
+int16_t ACMAMRwb::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
+int16_t ACMAMRwb::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                 const CodecInst& /* codec_inst */) {
+int32_t ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                           const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -101,7 +101,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
+int16_t ACMAMRwb::InternalCreateEncoder() {
   return -1;
 }
 
@@ -109,7 +109,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
+int16_t ACMAMRwb::InternalCreateDecoder() {
   return -1;
 }
 
@@ -117,7 +117,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+int16_t ACMAMRwb::SetBitRateSafe(const int32_t /* rate */) {
   return -1;
 }
 
@@ -125,7 +125,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
+int16_t ACMAMRwb::SetAMRwbEncoderPackingFormat(
     ACMAMRPackingFormat /* packing_format */) {
   return -1;
 }
@@ -134,7 +134,7 @@
   return AMRUndefined;
 }
 
-WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
+int16_t ACMAMRwb::SetAMRwbDecoderPackingFormat(
     ACMAMRPackingFormat /* packing_format */) {
   return -1;
 }
@@ -155,7 +155,7 @@
 #define AMRWB_MODE_23k      7
 #define AMRWB_MODE_24k      8
 
-ACMAMRwb::ACMAMRwb(WebRtc_Word16 codec_id)
+ACMAMRwb::ACMAMRwb(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       encoding_mode_(-1),  // invalid value
@@ -179,9 +179,9 @@
   return;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalEncode(WebRtc_UWord8* bitstream,
-                                       WebRtc_Word16* bitstream_len_byte) {
-  WebRtc_Word16 vad_decision = 1;
+int16_t ACMAMRwb::InternalEncode(uint8_t* bitstream,
+                                 int16_t* bitstream_len_byte) {
+  int16_t vad_decision = 1;
   // sanity check, if the rate is set correctly. we might skip this
   // sanity check. if rate is not set correctly, initialization flag
   // should be false and should not be here.
@@ -192,7 +192,7 @@
   *bitstream_len_byte = WebRtcAmrWb_Encode(encoder_inst_ptr_,
                                            &in_audio_[in_audio_ix_read_],
                                            frame_len_smpl_,
-                                           (WebRtc_Word16*)bitstream,
+                                           (int16_t*)bitstream,
                                            encoding_mode_);
 
   // Update VAD, if internal DTX is used
@@ -200,7 +200,7 @@
     if (*bitstream_len_byte <= (7 * frame_len_smpl_ / 160)) {
       vad_decision = 0;
     }
-    for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+    for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
       vad_label_[n] = vad_decision;
     }
   }
@@ -210,15 +210,15 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMAMRwb::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMAMRwb::EnableDTX() {
+int16_t ACMAMRwb::EnableDTX() {
   if (dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -233,7 +233,7 @@
   }
 }
 
-WebRtc_Word16 ACMAMRwb::DisableDTX() {
+int16_t ACMAMRwb::DisableDTX() {
   if (!dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -249,14 +249,14 @@
   }
 }
 
-WebRtc_Word16 ACMAMRwb::InternalInitEncoder(
+int16_t ACMAMRwb::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
   // sanity check
   if (encoder_inst_ptr_ == NULL) {
     return -1;
   }
 
-  WebRtc_Word16 status = SetBitRateSafe((codec_params->codec_inst).rate);
+  int16_t status = SetBitRateSafe((codec_params->codec_inst).rate);
   status += (WebRtcAmrWb_EncoderInit(
       encoder_inst_ptr_, ((codec_params->enable_dtx) ? 1 : 0)) < 0) ? -1 : 0;
   status += (WebRtcAmrWb_EncodeBitmode(
@@ -264,16 +264,16 @@
   return (status < 0) ? -1 : 0;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
+int16_t ACMAMRwb::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
-  WebRtc_Word16 status = WebRtcAmrWb_DecodeBitmode(decoder_inst_ptr_,
+  int16_t status = WebRtcAmrWb_DecodeBitmode(decoder_inst_ptr_,
                                                    decoder_packing_format_);
   status += ((WebRtcAmrWb_DecoderInit(decoder_inst_ptr_) < 0) ? -1 : 0);
   return (status < 0) ? -1 : 0;
 }
 
-WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                 const CodecInst& codec_inst) {
+int32_t ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                           const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     return -1;
   }
@@ -292,7 +292,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
+int16_t ACMAMRwb::InternalCreateEncoder() {
   return WebRtcAmrWb_CreateEnc(&encoder_inst_ptr_);
 }
 
@@ -308,7 +308,7 @@
   encoding_rate_ = 0;
 }
 
-WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
+int16_t ACMAMRwb::InternalCreateDecoder() {
   return WebRtcAmrWb_CreateDec(&decoder_inst_ptr_);
 }
 
@@ -322,7 +322,7 @@
   decoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 rate) {
+int16_t ACMAMRwb::SetBitRateSafe(const int32_t rate) {
   switch (rate) {
     case 7000: {
       encoding_mode_ = AMRWB_MODE_7k;
@@ -383,7 +383,7 @@
   return;
 }
 
-WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
+int16_t ACMAMRwb::SetAMRwbEncoderPackingFormat(
     ACMAMRPackingFormat packing_format) {
   if ((packing_format != AMRBandwidthEfficient) &&
       (packing_format != AMROctetAlligned) &&
@@ -405,7 +405,7 @@
   return encoder_packing_format_;
 }
 
-WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
+int16_t ACMAMRwb::SetAMRwbDecoderPackingFormat(
     ACMAMRPackingFormat packing_format) {
   if ((packing_format != AMRBandwidthEfficient) &&
       (packing_format != AMROctetAlligned) &&
diff --git a/modules/audio_coding/main/source/acm_amrwb.h b/modules/audio_coding/main/source/acm_amrwb.h
index 0f8d0bb..2cd301a 100644
--- a/modules/audio_coding/main/source/acm_amrwb.h
+++ b/modules/audio_coding/main/source/acm_amrwb.h
@@ -21,59 +21,59 @@
 
 class ACMAMRwb: public ACMGenericCodec {
  public:
-  explicit ACMAMRwb(WebRtc_Word16 codec_id);
+  explicit ACMAMRwb(int16_t codec_id);
   ~ACMAMRwb();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams* codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams* codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params);
 
-  WebRtc_Word16 SetAMRwbEncoderPackingFormat(
+  int16_t SetAMRwbEncoderPackingFormat(
       const ACMAMRPackingFormat packing_format);
 
   ACMAMRPackingFormat AMRwbEncoderPackingFormat() const;
 
-  WebRtc_Word16 SetAMRwbDecoderPackingFormat(
+  int16_t SetAMRwbDecoderPackingFormat(
       const ACMAMRPackingFormat packing_format);
 
   ACMAMRPackingFormat AMRwbDecoderPackingFormat() const;
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+  int16_t SetBitRateSafe(const int32_t rate);
 
-  WebRtc_Word16 EnableDTX();
+  int16_t EnableDTX();
 
-  WebRtc_Word16 DisableDTX();
+  int16_t DisableDTX();
 
   AMRWB_encinst_t_* encoder_inst_ptr_;
   AMRWB_decinst_t_* decoder_inst_ptr_;
 
-  WebRtc_Word16 encoding_mode_;
-  WebRtc_Word16 encoding_rate_;
+  int16_t encoding_mode_;
+  int16_t encoding_rate_;
   ACMAMRPackingFormat encoder_packing_format_;
   ACMAMRPackingFormat decoder_packing_format_;
 };
diff --git a/modules/audio_coding/main/source/acm_celt.cc b/modules/audio_coding/main/source/acm_celt.cc
index e6ceda4..31d9e37 100644
--- a/modules/audio_coding/main/source/acm_celt.cc
+++ b/modules/audio_coding/main/source/acm_celt.cc
@@ -49,7 +49,7 @@
                             int16_t /* bitstream_len_byte */,
                             int16_t* /* audio */,
                             int16_t* /* audio_samples */,
-                            WebRtc_Word8* /* speech_type */) {
+                            int8_t* /* speech_type */) {
   return -1;
 }
 
@@ -156,7 +156,7 @@
                             int16_t /* bitstream_len_byte */,
                             int16_t* /* audio */,
                             int16_t* /* audio_samples */,
-                            WebRtc_Word8* /* speech_type */) {
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
diff --git a/modules/audio_coding/main/source/acm_celt.h b/modules/audio_coding/main/source/acm_celt.h
index 4f6953e..ee68044 100644
--- a/modules/audio_coding/main/source/acm_celt.h
+++ b/modules/audio_coding/main/source/acm_celt.h
@@ -33,14 +33,12 @@
   int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(
+  int16_t DecodeSafe(
       uint8_t* /* bitstream */,
       int16_t /* bitstream_len_byte */,
       int16_t* /* audio */,
       int16_t* /* audio_samples */,
-      // TODO(leozwang): use int8_t here when WebRtc_Word8 is properly typed.
-      // http://code.google.com/p/webrtc/issues/detail?id=311
-      WebRtc_Word8* /* speech_type */);
+      int8_t* /* speech_type */);
 
   int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
                    const CodecInst& codec_inst);
diff --git a/modules/audio_coding/main/source/acm_cng.cc b/modules/audio_coding/main/source/acm_cng.cc
index ff8cea0..3549122 100644
--- a/modules/audio_coding/main/source/acm_cng.cc
+++ b/modules/audio_coding/main/source/acm_cng.cc
@@ -20,7 +20,7 @@
 
 namespace webrtc {
 
-ACMCNG::ACMCNG(WebRtc_Word16 codec_id) {
+ACMCNG::ACMCNG(int16_t codec_id) {
   encoder_inst_ptr_ = NULL;
   decoder_inst_ptr_ = NULL;
   codec_id_ = codec_id;
@@ -44,16 +44,16 @@
 // should not be called normally
 // instead the following function is called from inside
 // ACMGenericCodec::ProcessFrameVADDTX
-WebRtc_Word16 ACMCNG::InternalEncode(WebRtc_UWord8* /* bitstream */,
-                                     WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMCNG::InternalEncode(uint8_t* /* bitstream */,
+                               int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMCNG::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                 WebRtc_Word16 /* bitstream_len_byte */,
-                                 WebRtc_Word16* /* audio */,
-                                 WebRtc_Word16* /* audio_samples */,
-                                 WebRtc_Word8* /* speech_type */) {
+int16_t ACMCNG::DecodeSafe(uint8_t* /* bitstream */,
+                           int16_t /* bitstream_len_byte */,
+                           int16_t* /* audio */,
+                           int16_t* /* audio_samples */,
+                           int8_t* /* speech_type */) {
   return 0;
 }
 
@@ -61,18 +61,18 @@
 // this function should not be called normally
 // instead the following function is called from inside
 // ACMGenericCodec::ProcessFrameVADDTX
-WebRtc_Word16 ACMCNG::InternalInitEncoder(
+int16_t ACMCNG::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMCNG::InternalInitDecoder(
+int16_t ACMCNG::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return WebRtcCng_InitDec(decoder_inst_ptr_);
 }
 
-WebRtc_Word32 ACMCNG::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                               const CodecInst& codec_inst) {
+int32_t ACMCNG::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                         const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // TODO(tlegrand): log error
     return -1;
@@ -97,7 +97,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMCNG::InternalCreateEncoder() {
+int16_t ACMCNG::InternalCreateEncoder() {
   if (WebRtcCng_CreateEnc(&encoder_inst_ptr_) < 0) {
     encoder_inst_ptr_ = NULL;
     return -1;
@@ -115,7 +115,7 @@
   encoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMCNG::InternalCreateDecoder() {
+int16_t ACMCNG::InternalCreateDecoder() {
   if (WebRtcCng_CreateDec(&decoder_inst_ptr_) < 0) {
     decoder_inst_ptr_ = NULL;
     return -1;
diff --git a/modules/audio_coding/main/source/acm_cng.h b/modules/audio_coding/main/source/acm_cng.h
index 3f77a85..7221fbe 100644
--- a/modules/audio_coding/main/source/acm_cng.h
+++ b/modules/audio_coding/main/source/acm_cng.h
@@ -21,49 +21,49 @@
 
 class ACMCNG: public ACMGenericCodec {
  public:
-  explicit ACMCNG(WebRtc_Word16 codec_id);
+  explicit ACMCNG(int16_t codec_id);
   ~ACMCNG();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 EnableDTX() {
+  int16_t EnableDTX() {
     return -1;
   }
 
-  WebRtc_Word16 DisableDTX() {
+  int16_t DisableDTX() {
     return -1;
   }
 
   WebRtcCngEncInst* encoder_inst_ptr_;
   WebRtcCngDecInst* decoder_inst_ptr_;
-  WebRtc_UWord16 samp_freq_hz_;
+  uint16_t samp_freq_hz_;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/main/source/acm_common_defs.h b/modules/audio_coding/main/source/acm_common_defs.h
index 61fe279..db901c1 100644
--- a/modules/audio_coding/main/source/acm_common_defs.h
+++ b/modules/audio_coding/main/source/acm_common_defs.h
@@ -103,13 +103,13 @@
 //   last_in_timestamp: same as AudioCodingModuleImpl::last_in_timestamp_
 //
 struct WebRtcACMAudioBuff {
-  WebRtc_Word16 in_audio[AUDIO_BUFFER_SIZE_W16];
-  WebRtc_Word16 in_audio_ix_read;
-  WebRtc_Word16 in_audio_ix_write;
-  WebRtc_UWord32 in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
-  WebRtc_Word16 in_timestamp_ix_write;
-  WebRtc_UWord32 last_timestamp;
-  WebRtc_UWord32 last_in_timestamp;
+  int16_t in_audio[AUDIO_BUFFER_SIZE_W16];
+  int16_t in_audio_ix_read;
+  int16_t in_audio_ix_write;
+  uint32_t in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
+  int16_t in_timestamp_ix_write;
+  uint32_t last_timestamp;
+  uint32_t last_in_timestamp;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/main/source/acm_dtmf_detection.cc b/modules/audio_coding/main/source/acm_dtmf_detection.cc
index 6271eae..5820bc4 100644
--- a/modules/audio_coding/main/source/acm_dtmf_detection.cc
+++ b/modules/audio_coding/main/source/acm_dtmf_detection.cc
@@ -18,20 +18,20 @@
 
 ACMDTMFDetection::~ACMDTMFDetection() {}
 
-WebRtc_Word16 ACMDTMFDetection::Enable(ACMCountries /* cpt */) {
+int16_t ACMDTMFDetection::Enable(ACMCountries /* cpt */) {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFDetection::Disable() {
+int16_t ACMDTMFDetection::Disable() {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFDetection::Detect(
-    const WebRtc_Word16* /* in_audio_buff */,
-    const WebRtc_UWord16 /* in_buff_len_word16 */,
-    const WebRtc_Word32 /* in_freq_hz */,
+int16_t ACMDTMFDetection::Detect(
+    const int16_t* /* in_audio_buff */,
+    const uint16_t /* in_buff_len_word16 */,
+    const int32_t /* in_freq_hz */,
     bool& /* tone_detected */,
-    WebRtc_Word16& /* tone  */) {
+    int16_t& /* tone  */) {
   return -1;
 }
 
diff --git a/modules/audio_coding/main/source/acm_dtmf_detection.h b/modules/audio_coding/main/source/acm_dtmf_detection.h
index 4251d00..cd265fe 100644
--- a/modules/audio_coding/main/source/acm_dtmf_detection.h
+++ b/modules/audio_coding/main/source/acm_dtmf_detection.h
@@ -21,13 +21,13 @@
  public:
   ACMDTMFDetection();
   ~ACMDTMFDetection();
-  WebRtc_Word16 Enable(ACMCountries cpt = ACMDisableCountryDetection);
-  WebRtc_Word16 Disable();
-  WebRtc_Word16 Detect(const WebRtc_Word16* in_audio_buff,
-                       const WebRtc_UWord16 in_buff_len_word16,
-                       const WebRtc_Word32 in_freq_hz,
-                       bool& tone_detected,
-                       WebRtc_Word16& tone);
+  int16_t Enable(ACMCountries cpt = ACMDisableCountryDetection);
+  int16_t Disable();
+  int16_t Detect(const int16_t* in_audio_buff,
+                 const uint16_t in_buff_len_word16,
+                 const int32_t in_freq_hz,
+                 bool& tone_detected,
+                 int16_t& tone);
 
  private:
   ACMResampler resampler_;
diff --git a/modules/audio_coding/main/source/acm_dtmf_playout.cc b/modules/audio_coding/main/source/acm_dtmf_playout.cc
index bf4f63b..6b91db9 100644
--- a/modules/audio_coding/main/source/acm_dtmf_playout.cc
+++ b/modules/audio_coding/main/source/acm_dtmf_playout.cc
@@ -21,7 +21,7 @@
 #ifndef WEBRTC_CODEC_AVT
 
 ACMDTMFPlayout::ACMDTMFPlayout(
-    WebRtc_Word16 /* codec_id */) {
+    int16_t /* codec_id */) {
   return;
 }
 
@@ -29,33 +29,33 @@
   return;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMDTMFPlayout::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16 /* bitstream_len_byte */,
-    WebRtc_Word16* /* audio */,
-    WebRtc_Word16* /* audio_samples */,
-    WebRtc_Word8* /* speech_type */) {
+int16_t ACMDTMFPlayout::DecodeSafe(
+    uint8_t* /* bitstream */,
+    int16_t /* bitstream_len_byte */,
+    int16_t* /* audio */,
+    int16_t* /* audio_samples */,
+    int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
+int16_t ACMDTMFPlayout::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
+int16_t ACMDTMFPlayout::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                       const CodecInst& /* codec_inst */) {
+int32_t ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                                 const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -63,11 +63,11 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
+int16_t ACMDTMFPlayout::InternalCreateEncoder() {
   return -1;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
+int16_t ACMDTMFPlayout::InternalCreateDecoder() {
   return -1;
 }
 
@@ -85,7 +85,7 @@
 
 #else     //===================== Actual Implementation =======================
 
-ACMDTMFPlayout::ACMDTMFPlayout(WebRtc_Word16 codec_id) {
+ACMDTMFPlayout::ACMDTMFPlayout(int16_t codec_id) {
   codec_id_ = codec_id;
 }
 
@@ -93,37 +93,37 @@
   return;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMDTMFPlayout::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return 0;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16 /* bitstream_len_byte */,
-    WebRtc_Word16* /* audio */,
-    WebRtc_Word16* /* audio_samples */,
-    WebRtc_Word8* /* speech_type */) {
+int16_t ACMDTMFPlayout::DecodeSafe(
+    uint8_t* /* bitstream */,
+    int16_t /* bitstream_len_byte */,
+    int16_t* /* audio */,
+    int16_t* /* audio_samples */,
+    int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
+int16_t ACMDTMFPlayout::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization,
   // DTMFPlayout has no instance
   return 0;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
+int16_t ACMDTMFPlayout::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization,
   // DTMFPlayout has no instance
   return 0;
 }
 
-WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                       const CodecInst& codec_inst) {
+int32_t ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                                 const CodecInst& codec_inst) {
   // Fill up the structure by calling
   // "SET_CODEC_PAR" & "SET_AVT_FUNCTION."
   // Then call NetEQ to add the codec to it's
@@ -137,12 +137,12 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
+int16_t ACMDTMFPlayout::InternalCreateEncoder() {
   // DTMFPlayout has no instance
   return 0;
 }
 
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
+int16_t ACMDTMFPlayout::InternalCreateDecoder() {
   // DTMFPlayout has no instance
   return 0;
 }
diff --git a/modules/audio_coding/main/source/acm_dtmf_playout.h b/modules/audio_coding/main/source/acm_dtmf_playout.h
index 2607a2f..d07da47 100644
--- a/modules/audio_coding/main/source/acm_dtmf_playout.h
+++ b/modules/audio_coding/main/source/acm_dtmf_playout.h
@@ -17,35 +17,34 @@
 
 class ACMDTMFPlayout: public ACMGenericCodec {
  public:
-  explicit ACMDTMFPlayout(WebRtc_Word16 codec_id);
+  explicit ACMDTMFPlayout(int16_t codec_id);
   ~ACMDTMFPlayout();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 };
diff --git a/modules/audio_coding/main/source/acm_g722.cc b/modules/audio_coding/main/source/acm_g722.cc
index cc93801..1a023db 100644
--- a/modules/audio_coding/main/source/acm_g722.cc
+++ b/modules/audio_coding/main/source/acm_g722.cc
@@ -22,7 +22,7 @@
 
 #ifndef WEBRTC_CODEC_G722
 
-ACMG722::ACMG722(WebRtc_Word16 /* codec_id */)
+ACMG722::ACMG722(int16_t /* codec_id */)
     : ptr_enc_str_(NULL),
       ptr_dec_str_(NULL),
       encoder_inst_ptr_(NULL),
@@ -31,40 +31,40 @@
 
 ACMG722::~ACMG722() {}
 
-WebRtc_Word32 ACMG722::Add10MsDataSafe(
-    const WebRtc_UWord32 /* timestamp */,
-    const WebRtc_Word16* /* data */,
-    const WebRtc_UWord16 /* length_smpl */,
-    const WebRtc_UWord8 /* audio_channel */) {
+int32_t ACMG722::Add10MsDataSafe(
+    const uint32_t /* timestamp */,
+    const int16_t* /* data */,
+    const uint16_t /* length_smpl */,
+    const uint8_t /* audio_channel */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMG722::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722::InternalInitEncoder(
+int16_t ACMG722::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722::InternalInitDecoder(
+int16_t ACMG722::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                const CodecInst& /* codec_inst */) {
+int32_t ACMG722::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                          const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -72,7 +72,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722::InternalCreateEncoder() {
+int16_t ACMG722::InternalCreateEncoder() {
   return -1;
 }
 
@@ -80,7 +80,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG722::InternalCreateDecoder() {
+int16_t ACMG722::InternalCreateDecoder() {
   return -1;
 }
 
@@ -107,7 +107,7 @@
   G722DecInst* inst_right;  // instance for right channel in case of stereo
 };
 
-ACMG722::ACMG722(WebRtc_Word16 codec_id)
+ACMG722::ACMG722(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       encoder_inst_ptr_right_(NULL),
       decoder_inst_ptr_(NULL) {
@@ -157,33 +157,33 @@
   return;
 }
 
-WebRtc_Word32 ACMG722::Add10MsDataSafe(const WebRtc_UWord32 timestamp,
-                                       const WebRtc_Word16* data,
-                                       const WebRtc_UWord16 length_smpl,
-                                       const WebRtc_UWord8 audio_channel) {
+int32_t ACMG722::Add10MsDataSafe(const uint32_t timestamp,
+                                 const int16_t* data,
+                                 const uint16_t length_smpl,
+                                 const uint8_t audio_channel) {
   return ACMGenericCodec::Add10MsDataSafe((timestamp >> 1), data, length_smpl,
                                           audio_channel);
 }
 
-WebRtc_Word16 ACMG722::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMG722::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
   // If stereo, split input signal in left and right channel before encoding
   if (num_channels_ == 2) {
-    WebRtc_Word16 left_channel[960];
-    WebRtc_Word16 right_channel[960];
-    WebRtc_UWord8 out_left[480];
-    WebRtc_UWord8 out_right[480];
-    WebRtc_Word16 len_in_bytes;
+    int16_t left_channel[960];
+    int16_t right_channel[960];
+    uint8_t out_left[480];
+    uint8_t out_right[480];
+    int16_t len_in_bytes;
     for (int i = 0, j = 0; i < frame_len_smpl_ * 2; i += 2, j++) {
       left_channel[j] = in_audio_[in_audio_ix_read_ + i];
       right_channel[j] = in_audio_[in_audio_ix_read_ + i + 1];
     }
     len_in_bytes = WebRtcG722_Encode(encoder_inst_ptr_, left_channel,
                                      frame_len_smpl_,
-                                     (WebRtc_Word16*)out_left);
+                                     (int16_t*)out_left);
     len_in_bytes += WebRtcG722_Encode(encoder_inst_ptr_right_, right_channel,
                                       frame_len_smpl_,
-                                      (WebRtc_Word16*)out_right);
+                                      (int16_t*)out_right);
     *bitstream_len_byte = len_in_bytes;
 
     // Interleave the 4 bits per sample from left and right channel
@@ -195,7 +195,7 @@
     *bitstream_len_byte = WebRtcG722_Encode(encoder_inst_ptr_,
                                             &in_audio_[in_audio_ix_read_],
                                             frame_len_smpl_,
-                                            (WebRtc_Word16*)bitstream);
+                                            (int16_t*)bitstream);
   }
 
   // increment the read index this tell the caller how far
@@ -204,15 +204,15 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMG722::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMG722::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
   if (codec_params->codec_inst.channels == 2) {
     // Create codec struct for right channel
     if (ptr_enc_str_->inst_right == NULL) {
@@ -230,13 +230,13 @@
   return WebRtcG722_EncoderInit(encoder_inst_ptr_);
 }
 
-WebRtc_Word16 ACMG722::InternalInitDecoder(
+int16_t ACMG722::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return WebRtcG722_DecoderInit(decoder_inst_ptr_);
 }
 
-WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMG722::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // TODO(turajs): log error
     return -1;
@@ -260,7 +260,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722::InternalCreateEncoder() {
+int16_t ACMG722::InternalCreateEncoder() {
   if (ptr_enc_str_ == NULL) {
     // this structure must be created at the costructor
     // if it is still NULL then there is a probelm and
@@ -286,7 +286,7 @@
   encoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMG722::InternalCreateDecoder() {
+int16_t ACMG722::InternalCreateDecoder() {
   if (ptr_dec_str_ == NULL) {
     // this structure must be created at the costructor
     // if it is still NULL then there is a probelm and
diff --git a/modules/audio_coding/main/source/acm_g722.h b/modules/audio_coding/main/source/acm_g722.h
index 60ea2b4..d4273ab 100644
--- a/modules/audio_coding/main/source/acm_g722.h
+++ b/modules/audio_coding/main/source/acm_g722.h
@@ -24,40 +24,39 @@
 
 class ACMG722: public ACMGenericCodec {
  public:
-  explicit ACMG722(WebRtc_Word16 codec_id);
+  explicit ACMG722(int16_t codec_id);
   ~ACMG722();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
-  WebRtc_Word32 Add10MsDataSafe(const WebRtc_UWord32 timestamp,
-                                const WebRtc_Word16* data,
-                                const WebRtc_UWord16 length_smpl,
-                                const WebRtc_UWord8 audio_channel);
+  int32_t Add10MsDataSafe(const uint32_t timestamp,
+                          const int16_t* data,
+                          const uint16_t length_smpl,
+                          const uint8_t audio_channel);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
diff --git a/modules/audio_coding/main/source/acm_g7221.cc b/modules/audio_coding/main/source/acm_g7221.cc
index 48c058c..f784b62 100644
--- a/modules/audio_coding/main/source/acm_g7221.cc
+++ b/modules/audio_coding/main/source/acm_g7221.cc
@@ -88,7 +88,7 @@
 
 #ifndef WEBRTC_CODEC_G722_1
 
-ACMG722_1::ACMG722_1(WebRtc_Word16 /* codec_id */)
+ACMG722_1::ACMG722_1(int16_t /* codec_id */)
     : operational_rate_(-1),
       encoder_inst_ptr_(NULL),
       encoder_inst_ptr_right_(NULL),
@@ -109,32 +109,32 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMG722_1::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitstream  */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722_1::DecodeSafe(uint8_t* /* bitstream  */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1::InternalInitEncoder(
+int16_t ACMG722_1::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1::InternalInitDecoder(
+int16_t ACMG722_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
-                                  const CodecInst& /* codec_inst */) {
+int32_t ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
+                            const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -142,7 +142,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
+int16_t ACMG722_1::InternalCreateEncoder() {
   return -1;
 }
 
@@ -150,7 +150,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
+int16_t ACMG722_1::InternalCreateDecoder() {
   return -1;
 }
 
@@ -163,7 +163,7 @@
 }
 
 #else     //===================== Actual Implementation =======================
-ACMG722_1::ACMG722_1(WebRtc_Word16 codec_id)
+ACMG722_1::ACMG722_1(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       encoder_inst_ptr_right_(NULL),
       decoder_inst_ptr_(NULL),
@@ -229,12 +229,12 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1::InternalEncode(WebRtc_UWord8* bitstream,
-                                        WebRtc_Word16* bitstream_len_byte) {
-  WebRtc_Word16 left_channel[320];
-  WebRtc_Word16 right_channel[320];
-  WebRtc_Word16 len_in_bytes;
-  WebRtc_Word16 out_bits[160];
+int16_t ACMG722_1::InternalEncode(uint8_t* bitstream,
+                                  int16_t* bitstream_len_byte) {
+  int16_t left_channel[320];
+  int16_t right_channel[320];
+  int16_t len_in_bytes;
+  int16_t out_bits[160];
 
   // If stereo, split input signal in left and right channel before encoding
   if (num_channels_ == 2) {
@@ -292,17 +292,17 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722_1::DecodeSafe(uint8_t* /* bitstream */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMG722_1::InternalInitEncoder(
+int16_t ACMG722_1::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
-  WebRtc_Word16 ret;
+  int16_t ret;
 
   switch (operational_rate_) {
     case 16000: {
@@ -334,7 +334,7 @@
   }
 }
 
-WebRtc_Word16 ACMG722_1::InternalInitDecoder(
+int16_t ACMG722_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   switch (operational_rate_) {
     case 16000: {
@@ -354,8 +354,8 @@
   }
 }
 
-WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                  const CodecInst& codec_inst) {
+int32_t ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                            const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // Todo:
     // log error
@@ -401,7 +401,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
+int16_t ACMG722_1::InternalCreateEncoder() {
   if ((encoder_inst_ptr_ == NULL) || (encoder_inst_ptr_right_ == NULL)) {
     return -1;
   }
@@ -446,7 +446,7 @@
   encoder_inst32_ptr_ = NULL;
 }
 
-WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
+int16_t ACMG722_1::InternalCreateDecoder() {
   if (decoder_inst_ptr_ == NULL) {
     return -1;
   }
diff --git a/modules/audio_coding/main/source/acm_g7221.h b/modules/audio_coding/main/source/acm_g7221.h
index 9b4cb91..4e35476 100644
--- a/modules/audio_coding/main/source/acm_g7221.h
+++ b/modules/audio_coding/main/source/acm_g7221.h
@@ -26,39 +26,38 @@
 
 class ACMG722_1: public ACMGenericCodec {
  public:
-  explicit ACMG722_1(WebRtc_Word16 codec_id);
+  explicit ACMG722_1(int16_t codec_id);
   ~ACMG722_1();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio, WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio, int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word32 operational_rate_;
+  int32_t operational_rate_;
 
   G722_1_Inst_t_* encoder_inst_ptr_;
   G722_1_Inst_t_* encoder_inst_ptr_right_;  // Used in stereo mode
diff --git a/modules/audio_coding/main/source/acm_g7221c.cc b/modules/audio_coding/main/source/acm_g7221c.cc
index 3930ffa..a0d9483 100644
--- a/modules/audio_coding/main/source/acm_g7221c.cc
+++ b/modules/audio_coding/main/source/acm_g7221c.cc
@@ -89,7 +89,7 @@
 
 #ifndef WEBRTC_CODEC_G722_1C
 
-ACMG722_1C::ACMG722_1C(WebRtc_Word16 /* codec_id */)
+ACMG722_1C::ACMG722_1C(int16_t /* codec_id */)
     : operational_rate_(-1),
       encoder_inst_ptr_(NULL),
       encoder_inst_ptr_right_(NULL),
@@ -110,32 +110,32 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMG722_1C::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                     WebRtc_Word16 /* bitstream_len_byte */,
-                                     WebRtc_Word16* /* audio */,
-                                     WebRtc_Word16* /* audio_samples */,
-                                     WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722_1C::DecodeSafe(uint8_t* /* bitstream */,
+                               int16_t /* bitstream_len_byte */,
+                               int16_t* /* audio */,
+                               int16_t* /* audio_samples */,
+                               int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
+int16_t ACMG722_1C::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
+int16_t ACMG722_1C::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                   const CodecInst& /* codec_inst */) {
+int32_t ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                             const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -143,7 +143,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
+int16_t ACMG722_1C::InternalCreateEncoder() {
   return -1;
 }
 
@@ -151,7 +151,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
+int16_t ACMG722_1C::InternalCreateDecoder() {
   return -1;
 }
 
@@ -164,7 +164,7 @@
 }
 
 #else     //===================== Actual Implementation =======================
-ACMG722_1C::ACMG722_1C(WebRtc_Word16 codec_id)
+ACMG722_1C::ACMG722_1C(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       encoder_inst_ptr_right_(NULL),
       decoder_inst_ptr_(NULL),
@@ -234,12 +234,12 @@
   return;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalEncode(WebRtc_UWord8* bitstream,
-                                         WebRtc_Word16* bitstream_len_byte) {
-  WebRtc_Word16 left_channel[640];
-  WebRtc_Word16 right_channel[640];
-  WebRtc_Word16 len_in_bytes;
-  WebRtc_Word16 out_bits[240];
+int16_t ACMG722_1C::InternalEncode(uint8_t* bitstream,
+                                   int16_t* bitstream_len_byte) {
+  int16_t left_channel[640];
+  int16_t right_channel[640];
+  int16_t len_in_bytes;
+  int16_t out_bits[240];
 
   // If stereo, split input signal in left and right channel before encoding
   if (num_channels_ == 2) {
@@ -299,17 +299,17 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                     WebRtc_Word16 /* bitstream_len_byte */,
-                                     WebRtc_Word16* /* audio */,
-                                     WebRtc_Word16* /* audio_samples */,
-                                     WebRtc_Word8* /* speech_type */) {
+int16_t ACMG722_1C::DecodeSafe(uint8_t* /* bitstream */,
+                               int16_t /* bitstream_len_byte */,
+                               int16_t* /* audio */,
+                               int16_t* /* audio_samples */,
+                               int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
+int16_t ACMG722_1C::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
-  WebRtc_Word16 ret;
+  int16_t ret;
 
   switch (operational_rate_) {
     case 24000: {
@@ -341,7 +341,7 @@
   }
 }
 
-WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
+int16_t ACMG722_1C::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   switch (operational_rate_) {
     case 24000: {
@@ -361,8 +361,8 @@
   }
 }
 
-WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                   const CodecInst& codec_inst) {
+int32_t ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                             const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "CodeDef: decoder not initialized for G722_1c");
@@ -409,7 +409,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
+int16_t ACMG722_1C::InternalCreateEncoder() {
   if ((encoder_inst_ptr_ == NULL) || (encoder_inst_ptr_right_ == NULL)) {
     return -1;
   }
@@ -454,7 +454,7 @@
   encoder_inst48_ptr_ = NULL;
 }
 
-WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
+int16_t ACMG722_1C::InternalCreateDecoder() {
   if (decoder_inst_ptr_ == NULL) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateEncoder: cannot create decoder");
diff --git a/modules/audio_coding/main/source/acm_g7221c.h b/modules/audio_coding/main/source/acm_g7221c.h
index 4ce2d84..1b4e756 100644
--- a/modules/audio_coding/main/source/acm_g7221c.h
+++ b/modules/audio_coding/main/source/acm_g7221c.h
@@ -26,31 +26,31 @@
 
 class ACMG722_1C : public ACMGenericCodec {
  public:
-  explicit ACMG722_1C(WebRtc_Word16 codec_id);
+  explicit ACMG722_1C(int16_t codec_id);
   ~ACMG722_1C();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(
-      WebRtc_UWord8* bitstream,
-      WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(
+      uint8_t* bitstream,
+      int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(
+  int16_t InternalInitEncoder(
       WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(
+  int16_t InternalInitDecoder(
       WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(
-      WebRtc_UWord8* bitstream,
-      WebRtc_Word16 bitstream_len_byte,
-      WebRtc_Word16* audio,
-      WebRtc_Word16* audio_samples,
-      WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(
+      uint8_t* bitstream,
+      int16_t bitstream_len_byte,
+      int16_t* audio,
+      int16_t* audio_samples,
+      int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(
+  int32_t CodecDef(
       WebRtcNetEQ_CodecDef& codec_def,
       const CodecInst& codec_inst);
 
@@ -58,14 +58,14 @@
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(
       void* ptr_inst);
 
-  WebRtc_Word32 operational_rate_;
+  int32_t operational_rate_;
 
   G722_1_Inst_t_* encoder_inst_ptr_;
   G722_1_Inst_t_* encoder_inst_ptr_right_;  // Used in stereo mode
diff --git a/modules/audio_coding/main/source/acm_g729.cc b/modules/audio_coding/main/source/acm_g729.cc
index 031ccb7..67611cb 100644
--- a/modules/audio_coding/main/source/acm_g729.cc
+++ b/modules/audio_coding/main/source/acm_g729.cc
@@ -27,7 +27,7 @@
 
 #ifndef WEBRTC_CODEC_G729
 
-ACMG729::ACMG729(WebRtc_Word16 /* codec_id */)
+ACMG729::ACMG729(int16_t /* codec_id */)
 : encoder_inst_ptr_(NULL),
   decoder_inst_ptr_(NULL) {
   return;
@@ -37,50 +37,50 @@
   return;
 }
 
-WebRtc_Word16 ACMG729::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMG729::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729::EnableDTX() {
+int16_t ACMG729::EnableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMG729::DisableDTX() {
+int16_t ACMG729::DisableDTX() {
   return -1;
 }
 
-WebRtc_Word32 ACMG729::ReplaceInternalDTXSafe(
+int32_t ACMG729::ReplaceInternalDTXSafe(
     const bool /*replace_internal_dtx */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG729::IsInternalDTXReplacedSafe(
+int32_t ACMG729::IsInternalDTXReplacedSafe(
     bool* /* internal_dtx_replaced */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMG729::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729::InternalInitEncoder(
+int16_t ACMG729::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729::InternalInitDecoder(
+int16_t ACMG729::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG729::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
-                                const CodecInst& /* codec_inst */) {
+int32_t ACMG729::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
+                          const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -88,7 +88,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG729::InternalCreateEncoder() {
+int16_t ACMG729::InternalCreateEncoder() {
   return -1;
 }
 
@@ -96,7 +96,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG729::InternalCreateDecoder() {
+int16_t ACMG729::InternalCreateDecoder() {
   return -1;
 }
 
@@ -109,7 +109,7 @@
 }
 
 #else     //===================== Actual Implementation =======================
-ACMG729::ACMG729(WebRtc_Word16 codec_id)
+ACMG729::ACMG729(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   codec_id_ = codec_id;
@@ -131,19 +131,19 @@
   return;
 }
 
-WebRtc_Word16 ACMG729::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMG729::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
   // Initialize before entering the loop
-  WebRtc_Word16 num_encoded_samples = 0;
-  WebRtc_Word16 tmp_len_byte = 0;
-  WebRtc_Word16 vad_decision = 0;
+  int16_t num_encoded_samples = 0;
+  int16_t tmp_len_byte = 0;
+  int16_t vad_decision = 0;
   *bitstream_len_byte = 0;
   while (num_encoded_samples < frame_len_smpl_) {
     // Call G.729 encoder with pointer to encoder memory, input
     // audio, number of samples and bitsream
     tmp_len_byte = WebRtcG729_Encode(
         encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], 80,
-        (WebRtc_Word16*)(&(bitstream[*bitstream_len_byte])));
+        (int16_t*)(&(bitstream[*bitstream_len_byte])));
 
     // increment the read index this tell the caller that how far
     // we have gone forward in reading the audio buffer
@@ -173,7 +173,7 @@
         // check if G.729 internal DTX is enabled
         if (has_internal_dtx_ && dtx_enabled_) {
           vad_decision = 0;
-          for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+          for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
             vad_label_[n] = vad_decision;
           }
         }
@@ -197,7 +197,7 @@
 
   // update VAD decision vector
   if (has_internal_dtx_ && !vad_decision && dtx_enabled_) {
-    for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+    for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
       vad_label_[n] = vad_decision;
     }
   }
@@ -206,7 +206,7 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMG729::EnableDTX() {
+int16_t ACMG729::EnableDTX() {
   if (dtx_enabled_) {
     // DTX already enabled, do nothing
     return 0;
@@ -222,7 +222,7 @@
   }
 }
 
-WebRtc_Word16 ACMG729::DisableDTX() {
+int16_t ACMG729::DisableDTX() {
   if (!dtx_enabled_) {
     // DTX already dissabled, do nothing
     return 0;
@@ -239,7 +239,7 @@
   }
 }
 
-WebRtc_Word32 ACMG729::ReplaceInternalDTXSafe(const bool replace_internal_dtx) {
+int32_t ACMG729::ReplaceInternalDTXSafe(const bool replace_internal_dtx) {
   // This function is used to disable the G.729 built in DTX and use an
   // external instead.
 
@@ -256,7 +256,7 @@
       ACMGenericCodec::DisableDTX();
     }
     has_internal_dtx_ = !replace_internal_dtx;
-    WebRtc_Word16 status = SetVADSafe(old_enable_dtx, old_enable_vad, old_mode);
+    int16_t status = SetVADSafe(old_enable_dtx, old_enable_vad, old_mode);
     // Check if VAD status has changed from inactive to active, or if error was
     // reported
     if (status == 1) {
@@ -270,35 +270,35 @@
   return 0;
 }
 
-WebRtc_Word32 ACMG729::IsInternalDTXReplacedSafe(bool* internal_dtx_replaced) {
+int32_t ACMG729::IsInternalDTXReplacedSafe(bool* internal_dtx_replaced) {
   // Get status of wether DTX is replaced or not
   *internal_dtx_replaced = !has_internal_dtx_;
   return 0;
 }
 
-WebRtc_Word16 ACMG729::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMG729::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   // This function is not used. G.729 decoder is called from inside NetEQ
   return 0;
 }
 
-WebRtc_Word16 ACMG729::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMG729::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
   // Init G.729 encoder
   return WebRtcG729_EncoderInit(encoder_inst_ptr_,
                                 ((codec_params->enable_dtx) ? 1 : 0));
 }
 
-WebRtc_Word16 ACMG729::InternalInitDecoder(
+int16_t ACMG729::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // Init G.729 decoder
   return WebRtcG729_DecoderInit(decoder_inst_ptr_);
 }
 
-WebRtc_Word32 ACMG729::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMG729::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // Todo:
     // log error
@@ -320,7 +320,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG729::InternalCreateEncoder() {
+int16_t ACMG729::InternalCreateEncoder() {
   // Create encoder memory
   return WebRtcG729_CreateEnc(&encoder_inst_ptr_);
 }
@@ -335,7 +335,7 @@
   }
 }
 
-WebRtc_Word16 ACMG729::InternalCreateDecoder() {
+int16_t ACMG729::InternalCreateDecoder() {
   // Create decoder memory
   return WebRtcG729_CreateDec(&decoder_inst_ptr_);
 }
diff --git a/modules/audio_coding/main/source/acm_g729.h b/modules/audio_coding/main/source/acm_g729.h
index e923a32..d50aa5f 100644
--- a/modules/audio_coding/main/source/acm_g729.h
+++ b/modules/audio_coding/main/source/acm_g729.h
@@ -21,46 +21,46 @@
 
 class ACMG729 : public ACMGenericCodec {
  public:
-  explicit ACMG729(WebRtc_Word16 codec_id);
+  explicit ACMG729(int16_t codec_id);
   ~ACMG729();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 EnableDTX();
+  int16_t EnableDTX();
 
-  WebRtc_Word16 DisableDTX();
+  int16_t DisableDTX();
 
-  WebRtc_Word32 ReplaceInternalDTXSafe(const bool replace_internal_dtx);
+  int32_t ReplaceInternalDTXSafe(const bool replace_internal_dtx);
 
-  WebRtc_Word32 IsInternalDTXReplacedSafe(bool* internal_dtx_replaced);
+  int32_t IsInternalDTXReplacedSafe(bool* internal_dtx_replaced);
 
   G729_encinst_t_* encoder_inst_ptr_;
   G729_decinst_t_* decoder_inst_ptr_;
diff --git a/modules/audio_coding/main/source/acm_g7291.cc b/modules/audio_coding/main/source/acm_g7291.cc
index 89c984a..da473ca 100644
--- a/modules/audio_coding/main/source/acm_g7291.cc
+++ b/modules/audio_coding/main/source/acm_g7291.cc
@@ -26,7 +26,7 @@
 
 #ifndef WEBRTC_CODEC_G729_1
 
-ACMG729_1::ACMG729_1(WebRtc_Word16 /* codec_id */)
+ACMG729_1::ACMG729_1(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       my_rate_(32000),
@@ -39,32 +39,32 @@
   return;
 }
 
-WebRtc_Word16 ACMG729_1::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMG729_1::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729_1::InternalInitEncoder(
+int16_t ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMG729_1::InternalInitDecoder(
+int16_t ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                  const CodecInst& /* codec_inst */) {
+int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                            const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -72,7 +72,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG729_1::InternalCreateEncoder() {
+int16_t ACMG729_1::InternalCreateEncoder() {
   return -1;
 }
 
@@ -80,7 +80,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG729_1::InternalCreateDecoder() {
+int16_t ACMG729_1::InternalCreateDecoder() {
   return -1;
 }
 
@@ -92,7 +92,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG729_1::SetBitRateSafe(const WebRtc_Word32 /*rate*/) {
+int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) {
   return -1;
 }
 
@@ -100,7 +100,7 @@
 
 struct G729_1_inst_t_;
 
-ACMG729_1::ACMG729_1(WebRtc_Word16 codec_id)
+ACMG729_1::ACMG729_1(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       my_rate_(32000),  // Default rate.
@@ -124,18 +124,18 @@
   return;
 }
 
-WebRtc_Word16 ACMG729_1::InternalEncode(WebRtc_UWord8* bitstream,
-                                        WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMG729_1::InternalEncode(uint8_t* bitstream,
+                                  int16_t* bitstream_len_byte) {
 
   // Initialize before entering the loop
-  WebRtc_Word16 num_encoded_samples = 0;
+  int16_t num_encoded_samples = 0;
   *bitstream_len_byte = 0;
 
-  WebRtc_Word16 byte_length_frame = 0;
+  int16_t byte_length_frame = 0;
 
   // Derive number of 20ms frames per encoded packet.
   // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
-  WebRtc_Word16 num_20ms_frames = (frame_len_smpl_ / 320);
+  int16_t num_20ms_frames = (frame_len_smpl_ / 320);
   // Byte length for the frame. +1 is for rate information.
   byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 -
       flag_g729_mode_);
@@ -144,7 +144,7 @@
   do {
     *bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_,
                                              &in_audio_[in_audio_ix_read_],
-                                             (WebRtc_Word16*) bitstream,
+                                             (int16_t*) bitstream,
                                              my_rate_, num_20ms_frames);
 
     // increment the read index this tell the caller that how far
@@ -179,22 +179,22 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMG729_1::InternalInitEncoder(
+int16_t ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
   //set the bit rate and initialize
   my_rate_ = codec_params->codec_inst.rate;
-  return SetBitRateSafe((WebRtc_UWord32) my_rate_);
+  return SetBitRateSafe((uint32_t) my_rate_);
 }
 
-WebRtc_Word16 ACMG729_1::InternalInitDecoder(
+int16_t ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
@@ -204,8 +204,8 @@
   return 0;
 }
 
-WebRtc_Word32 ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                  const CodecInst& codec_inst) {
+int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                            const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "CodeDef: Decoder uninitialized for G729_1");
@@ -226,7 +226,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMG729_1::InternalCreateEncoder() {
+int16_t ACMG729_1::InternalCreateEncoder() {
   if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateEncoder: create encoder failed for G729_1");
@@ -244,7 +244,7 @@
   }
 }
 
-WebRtc_Word16 ACMG729_1::InternalCreateDecoder() {
+int16_t ACMG729_1::InternalCreateDecoder() {
   if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateDecoder: create decoder failed for G729_1");
@@ -269,7 +269,7 @@
   return;
 }
 
-WebRtc_Word16 ACMG729_1::SetBitRateSafe(const WebRtc_Word32 rate) {
+int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
   // allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
   //                22000, 24000, 26000, 28000, 30000, 32000};
   // TODO(tlegrand): This check exists in one other place two. Should be
diff --git a/modules/audio_coding/main/source/acm_g7291.h b/modules/audio_coding/main/source/acm_g7291.h
index d55c51b..433b2fd 100644
--- a/modules/audio_coding/main/source/acm_g7291.h
+++ b/modules/audio_coding/main/source/acm_g7291.h
@@ -21,47 +21,46 @@
 
 class ACMG729_1 : public ACMGenericCodec {
  public:
-  explicit ACMG729_1(WebRtc_Word16 codec_id);
+  explicit ACMG729_1(int16_t codec_id);
   ~ACMG729_1();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+  int16_t SetBitRateSafe(const int32_t rate);
 
   G729_1_inst_t_* encoder_inst_ptr_;
   G729_1_inst_t_* decoder_inst_ptr_;
 
-  WebRtc_UWord16 my_rate_;
-  WebRtc_Word16 flag_8khz_;
-  WebRtc_Word16 flag_g729_mode_;
+  uint16_t my_rate_;
+  int16_t flag_8khz_;
+  int16_t flag_g729_mode_;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/main/source/acm_generic_codec.h b/modules/audio_coding/main/source/acm_generic_codec.h
index 681f56e..945967e 100644
--- a/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/modules/audio_coding/main/source/acm_generic_codec.h
@@ -48,7 +48,7 @@
   virtual ACMGenericCodec* CreateInstance() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 Encode()
+  // int16_t Encode()
   // The function is called to perform an encoding of the audio stored in
   // audio buffer. An encoding is performed only if enough audio, i.e. equal
   // to the frame-size of the codec, exist. The audio frame will be processed
@@ -100,13 +100,13 @@
   //   -1 if error is occurred, otherwise the length of the bit-stream in
   //      bytes.
   //
-  WebRtc_Word16 Encode(WebRtc_UWord8* bitstream,
-                       WebRtc_Word16* bitstream_len_byte,
-                       WebRtc_UWord32* timestamp,
-                       WebRtcACMEncodingType* encoding_type);
+  int16_t Encode(uint8_t* bitstream,
+                 int16_t* bitstream_len_byte,
+                 uint32_t* timestamp,
+                 WebRtcACMEncodingType* encoding_type);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 Decode()
+  // int16_t Decode()
   // This function is used to decode a given bit-stream, without engaging
   // NetEQ.
   //
@@ -127,11 +127,11 @@
   //   -1 if failed to decode,
   //    0 if succeeded.
   //
-  WebRtc_Word16 Decode(WebRtc_UWord8* bitstream,
-                       WebRtc_Word16 bitstream_len_byte,
-                       WebRtc_Word16* audio,
-                       WebRtc_Word16* audio_samples,
-                       WebRtc_Word8* speech_type);
+  int16_t Decode(uint8_t* bitstream,
+                 int16_t bitstream_len_byte,
+                 int16_t* audio,
+                 int16_t* audio_samples,
+                 int8_t* speech_type);
 
   ///////////////////////////////////////////////////////////////////////////
   // void SplitStereoPacket()
@@ -149,8 +149,8 @@
   //                          we simply copy the data and return it both for
   //                          left channel and right channel decoding.
   //
-  virtual void SplitStereoPacket(WebRtc_UWord8* /* payload */,
-                                 WebRtc_Word32* /* payload_length */) {}
+  virtual void SplitStereoPacket(uint8_t* /* payload */,
+                                 int32_t* /* payload_length */) {}
 
   ///////////////////////////////////////////////////////////////////////////
   // bool EncoderInitialized();
@@ -171,7 +171,7 @@
   bool DecoderInitialized();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 EncoderParams()
+  // int16_t EncoderParams()
   // It is called to get encoder parameters. It will call
   // EncoderParamsSafe() in turn.
   //
@@ -184,10 +184,10 @@
   //   -1 if the encoder is not initialized,
   //    0 otherwise.
   //
-  WebRtc_Word16 EncoderParams(WebRtcACMCodecParams *enc_params);
+  int16_t EncoderParams(WebRtcACMCodecParams *enc_params);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 DecoderParams(...)
+  // int16_t DecoderParams(...)
   // It is called to get decoder parameters. It will call DecoderParamsSafe()
   // in turn.
   //
@@ -202,10 +202,10 @@
   //
   //
   bool DecoderParams(WebRtcACMCodecParams *dec_params,
-                     const WebRtc_UWord8 payload_type);
+                     const uint8_t payload_type);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InitEncoder(...)
+  // int16_t InitEncoder(...)
   // This function is called to initialize the encoder with the given
   // parameters.
   //
@@ -220,11 +220,11 @@
   //  -1 if failed to initialize.
   //
   //
-  WebRtc_Word16 InitEncoder(WebRtcACMCodecParams* codec_params,
-                            bool force_initialization);
+  int16_t InitEncoder(WebRtcACMCodecParams* codec_params,
+                      bool force_initialization);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InitDecoder()
+  // int16_t InitDecoder()
   // This function is called to initialize the decoder with the given
   // parameters. (c.f. acm_common_defs.h & common_types.h for the
   // definition of the structure)
@@ -240,11 +240,11 @@
   //  -1 if failed to initialize.
   //
   //
-  WebRtc_Word16 InitDecoder(WebRtcACMCodecParams* codec_params,
-                            bool force_initialization);
+  int16_t InitDecoder(WebRtcACMCodecParams* codec_params,
+                      bool force_initialization);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 RegisterInNetEq(...)
+  // int32_t RegisterInNetEq(...)
   // This function is called to register the decoder in NetEq, with the given
   // payload type.
   //
@@ -256,10 +256,10 @@
   //   -1 if failed to register,
   //    0 if successfully initialized.
   //
-  WebRtc_Word32 RegisterInNetEq(ACMNetEQ* neteq, const CodecInst& codec_inst);
+  int32_t RegisterInNetEq(ACMNetEQ* neteq, const CodecInst& codec_inst);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 Add10MsData(...)
+  // int32_t Add10MsData(...)
   // This function is called to add 10 ms of audio to the audio buffer of
   // the codec.
   //
@@ -277,13 +277,13 @@
   //   -1 if failed
   //    0 otherwise.
   //
-  WebRtc_Word32 Add10MsData(const WebRtc_UWord32 timestamp,
-                            const WebRtc_Word16* data,
-                            const WebRtc_UWord16 length,
-                            const WebRtc_UWord8 audio_channel);
+  int32_t Add10MsData(const uint32_t timestamp,
+                      const int16_t* data,
+                      const uint16_t length,
+                      const uint8_t audio_channel);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_UWord32 NoMissedSamples()
+  // uint32_t NoMissedSamples()
   // This function returns the number of samples which are overwritten in
   // the audio buffer. The audio samples are overwritten if the input audio
   // buffer is full, but Add10MsData() is called. (We might remove this
@@ -292,7 +292,7 @@
   // Return Value:
   //   Number of samples which are overwritten.
   //
-  WebRtc_UWord32 NoMissedSamples() const;
+  uint32_t NoMissedSamples() const;
 
   ///////////////////////////////////////////////////////////////////////////
   // void ResetNoMissedSamples()
@@ -302,7 +302,7 @@
   void ResetNoMissedSamples();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 SetBitRate()
+  // int16_t SetBitRate()
   // The function is called to set the encoding rate.
   //
   // Input:
@@ -313,7 +313,7 @@
   //      codec is not rate-adjustable.
   //    0 if the rate is adjusted successfully
   //
-  WebRtc_Word16 SetBitRate(const WebRtc_Word32 bitrate_bps);
+  int16_t SetBitRate(const int32_t bitrate_bps);
 
   ///////////////////////////////////////////////////////////////////////////
   // DestructEncoderInst()
@@ -328,7 +328,7 @@
   void DestructEncoderInst(void* ptr_inst);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 AudioBuffer()
+  // int16_t AudioBuffer()
   // This is used when synchronization of codecs is required. There are cases
   // that the audio buffers of two codecs have to be synched. By calling this
   // function on can get the audio buffer and other related parameters, such
@@ -342,20 +342,20 @@
   //   -1 if fails to copy the audio buffer,
   //    0 if succeeded.
   //
-  WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audio_buff);
+  int16_t AudioBuffer(WebRtcACMAudioBuff& audio_buff);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_UWord32 EarliestTimestamp()
+  // uint32_t EarliestTimestamp()
   // Returns the timestamp of the first 10 ms in audio buffer. This is used
   // to identify if a synchronization of two encoders is required.
   //
   // Return value:
   //   timestamp of the first 10 ms audio in the audio buffer.
   //
-  WebRtc_UWord32 EarliestTimestamp() const;
+  uint32_t EarliestTimestamp() const;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 SetAudioBuffer()
+  // int16_t SetAudioBuffer()
   // This function is called to set the audio buffer and the associated
   // parameters to a given value.
   //
@@ -363,10 +363,10 @@
   //   -1 if fails to copy the audio buffer,
   //    0 if succeeded.
   //
-  WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audio_buff);
+  int16_t SetAudioBuffer(WebRtcACMAudioBuff& audio_buff);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 SetVAD()
+  // int16_t SetVAD()
   // This is called to set VAD & DTX. If the codec has internal DTX that will
   // be used. If DTX is enabled and the codec does not have internal DTX,
   // WebRtc-VAD will be used to decide if the frame is active. If DTX is
@@ -392,12 +392,12 @@
   //   -1 if failed to set DTX & VAD as specified,
   //    0 if succeeded.
   //
-  WebRtc_Word16 SetVAD(const bool enable_dtx = true,
-                       const bool enable_vad = false,
-                       const ACMVADMode mode = VADNormal);
+  int16_t SetVAD(const bool enable_dtx = true,
+                 const bool enable_vad = false,
+                 const ACMVADMode mode = VADNormal);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ReplaceInternalDTX()
+  // int32_t ReplaceInternalDTX()
   // This is called to replace the codec internal DTX with WebRtc DTX.
   // This is only valid for G729 where the user has possibility to replace
   // AnnexB with WebRtc DTX. For other codecs this function has no effect.
@@ -409,10 +409,10 @@
   //   -1 if failed to replace internal DTX,
   //    0 if succeeded.
   //
-  WebRtc_Word32 ReplaceInternalDTX(const bool replace_internal_dtx);
+  int32_t ReplaceInternalDTX(const bool replace_internal_dtx);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 IsInternalDTXReplaced()
+  // int32_t IsInternalDTXReplaced()
   // This is called to check if the codec internal DTX is replaced by WebRtc
   // DTX. This is only valid for G729 where the user has possibility to replace
   // AnnexB with WebRtc DTX. For other codecs this function has no effect.
@@ -424,7 +424,7 @@
   //   -1 if failed to check
   //    0 if succeeded.
   //
-  WebRtc_Word32 IsInternalDTXReplaced(bool* internal_dtx_replaced);
+  int32_t IsInternalDTXReplaced(bool* internal_dtx_replaced);
 
   ///////////////////////////////////////////////////////////////////////////
   // void SetNetEqDecodeLock()
@@ -450,7 +450,7 @@
   }
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 GetEstimatedBandwidth()
+  // int32_t GetEstimatedBandwidth()
   // Used to get decoder estimated bandwidth. Only iSAC will provide a value.
   //
   //
@@ -458,10 +458,10 @@
   //   -1 if fails to get decoder estimated bandwidth,
   //    >0 estimated bandwidth in bits/sec.
   //
-  WebRtc_Word32 GetEstimatedBandwidth();
+  int32_t GetEstimatedBandwidth();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 SetEstimatedBandwidth()
+  // int32_t SetEstimatedBandwidth()
   // Used to set estiamted bandwidth sent out of band from other side. Only
   // iSAC will have use for the value.
   //
@@ -472,10 +472,10 @@
   //   -1 if fails to set estimated bandwidth,
   //    0 on success.
   //
-  WebRtc_Word32 SetEstimatedBandwidth(WebRtc_Word32 estimated_bandwidth);
+  int32_t SetEstimatedBandwidth(int32_t estimated_bandwidth);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 GetRedPayload()
+  // int32_t GetRedPayload()
   // Used to get codec specific RED payload (if such is implemented).
   // Currently only done in iSAC.
   //
@@ -487,11 +487,11 @@
   //   -1 if fails to get codec specific RED,
   //    0 if succeeded.
   //
-  WebRtc_Word32 GetRedPayload(WebRtc_UWord8* red_payload,
-                              WebRtc_Word16* payload_bytes);
+  int32_t GetRedPayload(uint8_t* red_payload,
+                        int16_t* payload_bytes);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 ResetEncoder()
+  // int16_t ResetEncoder()
   // By calling this function you would re-initialize the encoder with the
   // current parameters. All the settings, e.g. VAD/DTX, frame-size... should
   // remain unchanged. (In case of iSAC we don't want to lose BWE history.)
@@ -500,10 +500,10 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 ResetEncoder();
+  int16_t ResetEncoder();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 ResetEncoder()
+  // int16_t ResetEncoder()
   // By calling this function you would re-initialize the decoder with the
   // current parameters.
   //
@@ -511,7 +511,7 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 ResetDecoder(WebRtc_Word16 payload_type);
+  int16_t ResetDecoder(int16_t payload_type);
 
   ///////////////////////////////////////////////////////////////////////////
   // void DestructEncoder()
@@ -533,22 +533,22 @@
   void DestructDecoder();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 SamplesLeftToEncode()
+  // int16_t SamplesLeftToEncode()
   // Returns the number of samples required to be able to do encoding.
   //
   // Return value:
   //   Number of samples.
   //
-  WebRtc_Word16 SamplesLeftToEncode();
+  int16_t SamplesLeftToEncode();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_UWord32 LastEncodedTimestamp()
+  // uint32_t LastEncodedTimestamp()
   // Returns the timestamp of the last frame it encoded.
   //
   // Return value:
   //   Timestamp.
   //
-  WebRtc_UWord32 LastEncodedTimestamp() const;
+  uint32_t LastEncodedTimestamp() const;
 
   ///////////////////////////////////////////////////////////////////////////
   // SetUniqueID()
@@ -557,7 +557,7 @@
   // Input
   //   -id                 : A number to identify the codec.
   //
-  void SetUniqueID(const WebRtc_UWord32 id);
+  void SetUniqueID(const uint32_t id);
 
   ///////////////////////////////////////////////////////////////////////////
   // IsAudioBufferFresh()
@@ -594,7 +594,7 @@
   //    0 if succeeded in updating the decoder.
   //   -1 if failed to update.
   //
-  virtual WebRtc_Word16 UpdateDecoderSampFreq(WebRtc_Word16 /* codec_id */) {
+  virtual int16_t UpdateDecoderSampFreq(int16_t /* codec_id */) {
     return 0;
   }
 
@@ -614,8 +614,8 @@
   //   -1 if failed, or if this is meaningless for the given codec.
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 UpdateEncoderSampFreq(
-      WebRtc_UWord16 samp_freq_hz);
+  virtual int16_t UpdateEncoderSampFreq(
+      uint16_t samp_freq_hz);
 
   ///////////////////////////////////////////////////////////////////////////
   // EncoderSampFreq()
@@ -629,10 +629,10 @@
   //   -1 if failed to output sampling rate.
   //    0 if the sample rate is returned successfully.
   //
-  virtual WebRtc_Word16 EncoderSampFreq(WebRtc_UWord16& samp_freq_hz);
+  virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word32 ConfigISACBandwidthEstimator()
+  // int32_t ConfigISACBandwidthEstimator()
   // Call this function to configure the bandwidth estimator of ISAC.
   // During the adaptation of bit-rate, iSAC automatically adjusts the
   // frame-size (either 30 or 60 ms) to save on RTP header. The initial
@@ -655,9 +655,9 @@
   //   -1 if failed to configure the bandwidth estimator,
   //    0 if the configuration was successfully applied.
   //
-  virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
-      const WebRtc_UWord8 init_frame_size_msec,
-      const WebRtc_UWord16 init_rate_bps,
+  virtual int32_t ConfigISACBandwidthEstimator(
+      const uint8_t init_frame_size_msec,
+      const uint16_t init_rate_bps,
       const bool enforce_frame_size);
 
   ///////////////////////////////////////////////////////////////////////////
@@ -675,8 +675,8 @@
   //   -1 if failed to set the maximum  payload-size.
   //    0 if the given length is set successfully.
   //
-  virtual WebRtc_Word32 SetISACMaxPayloadSize(
-      const WebRtc_UWord16 max_payload_len_bytes);
+  virtual int32_t SetISACMaxPayloadSize(
+      const uint16_t max_payload_len_bytes);
 
   ///////////////////////////////////////////////////////////////////////////
   // SetISACMaxRate()
@@ -693,7 +693,7 @@
   //   -1 if failed to set the maximum rate.
   //    0 if the maximum rate is set successfully.
   //
-  virtual WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_rate_bps);
+  virtual int32_t SetISACMaxRate(const uint32_t max_rate_bps);
 
   ///////////////////////////////////////////////////////////////////////////
   // SaveDecoderParamS()
@@ -705,7 +705,7 @@
   //
   void SaveDecoderParam(const WebRtcACMCodecParams* codec_params);
 
-  WebRtc_Word32 FrameSize() {
+  int32_t FrameSize() {
     return frame_len_smpl_;
   }
 
@@ -734,10 +734,10 @@
   //   -1 if an error occurs, otherwise the length of the payload (in Bytes)
   //   is returned.
   //
-  virtual WebRtc_Word16 REDPayloadISAC(const WebRtc_Word32 isac_rate,
-                                       const WebRtc_Word16 isac_bw_estimate,
-                                       WebRtc_UWord8* payload,
-                                       WebRtc_Word16* payload_len_bytes);
+  virtual int16_t REDPayloadISAC(const int32_t isac_rate,
+                                 const int16_t isac_bw_estimate,
+                                 uint8_t* payload,
+                                 int16_t* payload_len_bytes);
 
   ///////////////////////////////////////////////////////////////////////////
   // IsTrueStereoCodec()
@@ -771,33 +771,33 @@
   // See Decode() for the description of function, input(s)/output(s) and
   // return value.
   //
-  virtual WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                                   WebRtc_Word16 bitstream_len_byte,
-                                   WebRtc_Word16* audio,
-                                   WebRtc_Word16* audio_samples,
-                                   WebRtc_Word8* speech_type) = 0;
+  virtual int16_t DecodeSafe(uint8_t* bitstream,
+                             int16_t bitstream_len_byte,
+                             int16_t* audio,
+                             int16_t* audio_samples,
+                             int8_t* speech_type) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // See Add10MsSafe() for the description of function, input(s)/output(s)
   // and return value.
   //
-  virtual WebRtc_Word32 Add10MsDataSafe(const WebRtc_UWord32 timestamp,
-                                        const WebRtc_Word16* data,
-                                        const WebRtc_UWord16 length,
-                                        const WebRtc_UWord8 audio_channel);
+  virtual int32_t Add10MsDataSafe(const uint32_t timestamp,
+                                  const int16_t* data,
+                                  const uint16_t length,
+                                  const uint8_t audio_channel);
 
   ///////////////////////////////////////////////////////////////////////////
   // See RegisterInNetEq() for the description of function,
   // input(s)/output(s) and  return value.
   //
-  virtual WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                 const CodecInst& codec_inst) = 0;
+  virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                           const CodecInst& codec_inst) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // See EncoderParam() for the description of function, input(s)/output(s)
   // and return value.
   //
-  WebRtc_Word16 EncoderParamsSafe(WebRtcACMCodecParams *enc_params);
+  int16_t EncoderParamsSafe(WebRtcACMCodecParams *enc_params);
 
   ///////////////////////////////////////////////////////////////////////////
   // See DecoderParam for the description of function, input(s)/output(s)
@@ -814,33 +814,33 @@
   // with a single codec instance.
   //
   virtual bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params,
-                                 const WebRtc_UWord8 payload_type);
+                                 const uint8_t payload_type);
 
   ///////////////////////////////////////////////////////////////////////////
   // See ResetEncoder() for the description of function, input(s)/output(s)
   // and return value.
   //
-  WebRtc_Word16 ResetEncoderSafe();
+  int16_t ResetEncoderSafe();
 
   ///////////////////////////////////////////////////////////////////////////
   // See InitEncoder() for the description of function, input(s)/output(s)
   // and return value.
   //
-  WebRtc_Word16 InitEncoderSafe(WebRtcACMCodecParams *codec_params,
-                                bool force_initialization);
+  int16_t InitEncoderSafe(WebRtcACMCodecParams *codec_params,
+                          bool force_initialization);
 
   ///////////////////////////////////////////////////////////////////////////
   // See InitDecoder() for the description of function, input(s)/output(s)
   // and return value.
   //
-  WebRtc_Word16 InitDecoderSafe(WebRtcACMCodecParams *codec_params,
-                                bool force_initialization);
+  int16_t InitDecoderSafe(WebRtcACMCodecParams *codec_params,
+                          bool force_initialization);
 
   ///////////////////////////////////////////////////////////////////////////
   // See ResetDecoder() for the description of function, input(s)/output(s)
   // and return value.
   //
-  WebRtc_Word16 ResetDecoderSafe(WebRtc_Word16 payload_type);
+  int16_t ResetDecoderSafe(int16_t payload_type);
 
   ///////////////////////////////////////////////////////////////////////////
   // See DestructEncoder() for the description of function,
@@ -860,70 +860,70 @@
   //
   // Any codec that can change the bit-rate has to implement this.
   //
-  virtual WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 bitrate_bps);
+  virtual int16_t SetBitRateSafe(const int32_t bitrate_bps);
 
   ///////////////////////////////////////////////////////////////////////////
   // See GetEstimatedBandwidth() for the description of function,
   // input(s)/output(s) and return value.
   //
-  virtual WebRtc_Word32 GetEstimatedBandwidthSafe();
+  virtual int32_t GetEstimatedBandwidthSafe();
 
   ///////////////////////////////////////////////////////////////////////////
   // See SetEstimatedBandwidth() for the description of function,
   // input(s)/output(s) and return value.
   //
-  virtual WebRtc_Word32 SetEstimatedBandwidthSafe(
-      WebRtc_Word32 estimated_bandwidth);
+  virtual int32_t SetEstimatedBandwidthSafe(
+      int32_t estimated_bandwidth);
 
   ///////////////////////////////////////////////////////////////////////////
   // See GetRedPayload() for the description of function, input(s)/output(s)
   // and return value.
   //
-  virtual WebRtc_Word32 GetRedPayloadSafe(WebRtc_UWord8* red_payload,
-                                          WebRtc_Word16* payload_bytes);
+  virtual int32_t GetRedPayloadSafe(uint8_t* red_payload,
+                                    int16_t* payload_bytes);
 
   ///////////////////////////////////////////////////////////////////////////
   // See SetVAD() for the description of function, input(s)/output(s) and
   // return value.
   //
-  WebRtc_Word16 SetVADSafe(const bool enable_dtx = true,
-                           const bool enable_vad = false,
-                           const ACMVADMode mode = VADNormal);
+  int16_t SetVADSafe(const bool enable_dtx = true,
+                     const bool enable_vad = false,
+                     const ACMVADMode mode = VADNormal);
 
   ///////////////////////////////////////////////////////////////////////////
   // See ReplaceInternalDTX() for the description of function, input and
   // return value.
   //
-  virtual WebRtc_Word32 ReplaceInternalDTXSafe(const bool replace_internal_dtx);
+  virtual int32_t ReplaceInternalDTXSafe(const bool replace_internal_dtx);
 
   ///////////////////////////////////////////////////////////////////////////
   // See IsInternalDTXReplaced() for the description of function, input and
   // return value.
   //
-  virtual WebRtc_Word32 IsInternalDTXReplacedSafe(bool* internal_dtx_replaced);
+  virtual int32_t IsInternalDTXReplacedSafe(bool* internal_dtx_replaced);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 CreateEncoder()
+  // int16_t CreateEncoder()
   // Creates the encoder instance.
   //
   // Return value:
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 CreateEncoder();
+  int16_t CreateEncoder();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 CreateDecoder()
+  // int16_t CreateDecoder()
   // Creates the decoder instance.
   //
   // Return value:
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 CreateDecoder();
+  int16_t CreateDecoder();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 EnableVAD();
+  // int16_t EnableVAD();
   // Enables VAD with the given mode. The VAD instance will be created if
   // it does not exists.
   //
@@ -935,20 +935,20 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 EnableVAD(ACMVADMode mode);
+  int16_t EnableVAD(ACMVADMode mode);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 DisableVAD()
+  // int16_t DisableVAD()
   // Disables VAD.
   //
   // Return value:
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 DisableVAD();
+  int16_t DisableVAD();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 EnableDTX()
+  // int16_t EnableDTX()
   // Enables DTX. This method should be overwritten for codecs which have
   // internal DTX.
   //
@@ -956,10 +956,10 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 EnableDTX();
+  virtual int16_t EnableDTX();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 DisableDTX()
+  // int16_t DisableDTX()
   // Disables usage of DTX. This method should be overwritten for codecs which
   // have internal DTX.
   //
@@ -967,10 +967,10 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 DisableDTX();
+  virtual int16_t DisableDTX();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalEncode()
+  // int16_t InternalEncode()
   // This is a codec-specific function called in EncodeSafe() to actually
   // encode a frame of audio.
   //
@@ -984,11 +984,11 @@
   //   -1 if failed,
   //   otherwise the length of the bit-stream is returned.
   //
-  virtual WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                                       WebRtc_Word16* bitstream_len_byte) = 0;
+  virtual int16_t InternalEncode(uint8_t* bitstream,
+                                 int16_t* bitstream_len_byte) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalInitEncoder()
+  // int16_t InternalInitEncoder()
   // This is a codec-specific function called in InitEncoderSafe(), it has to
   // do all codec-specific operation to initialize the encoder given the
   // encoder parameters.
@@ -1006,11 +1006,11 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 InternalInitEncoder(
+  virtual int16_t InternalInitEncoder(
       WebRtcACMCodecParams *codec_params) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalInitDecoder()
+  // int16_t InternalInitDecoder()
   // This is a codec-specific function called in InitDecoderSafe(), it has to
   // do all codec-specific operation to initialize the decoder given the
   // decoder parameters.
@@ -1023,7 +1023,7 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 InternalInitDecoder(
+  virtual int16_t InternalInitDecoder(
       WebRtcACMCodecParams *codec_params) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
@@ -1035,10 +1035,10 @@
   //   -num_samples        : the number of overwritten samples is incremented
   //                         by this value.
   //
-  void IncreaseNoMissedSamples(const WebRtc_Word16 num_samples);
+  void IncreaseNoMissedSamples(const int16_t num_samples);
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalCreateEncoder()
+  // int16_t InternalCreateEncoder()
   // This is a codec-specific method called in CreateEncoderSafe() it is
   // supposed to perform all codec-specific operations to create encoder
   // instance.
@@ -1047,10 +1047,10 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 InternalCreateEncoder() = 0;
+  virtual int16_t InternalCreateEncoder() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalCreateDecoder()
+  // int16_t InternalCreateDecoder()
   // This is a codec-specific method called in CreateDecoderSafe() it is
   // supposed to perform all codec-specific operations to create decoder
   // instance.
@@ -1059,7 +1059,7 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 InternalCreateDecoder() = 0;
+  virtual int16_t InternalCreateDecoder() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // void InternalDestructEncoderInst()
@@ -1079,7 +1079,7 @@
   virtual void InternalDestructEncoderInst(void* ptr_inst) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 InternalResetEncoder()
+  // int16_t InternalResetEncoder()
   // This method is called to reset the states of encoder. However, the
   // current parameters, e.g. frame-length, should remain as they are. For
   // most of the codecs a re-initialization of the encoder is what needs to
@@ -1091,10 +1091,10 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual WebRtc_Word16 InternalResetEncoder();
+  virtual int16_t InternalResetEncoder();
 
   ///////////////////////////////////////////////////////////////////////////
-  // WebRtc_Word16 ProcessFrameVADDTX()
+  // int16_t ProcessFrameVADDTX()
   // This function is called when a full frame of audio is available. It will
   // break the audio frame into blocks such that each block could be processed
   // by VAD & CN/DTX. If a frame is divided into two blocks then there are two
@@ -1123,9 +1123,9 @@
   //   -1 if failed,
   //    0 if succeeded.
   //
-  WebRtc_Word16 ProcessFrameVADDTX(WebRtc_UWord8* bitstream,
-                                   WebRtc_Word16* bitstream_len_byte,
-                                   WebRtc_Word16* samples_processed);
+  int16_t ProcessFrameVADDTX(uint8_t* bitstream,
+                             int16_t* bitstream_len_byte,
+                             int16_t* samples_processed);
 
   ///////////////////////////////////////////////////////////////////////////
   // CanChangeEncodingParam()
@@ -1148,7 +1148,7 @@
   // Output:
   //   -rate_bps           : the current target rate of the codec.
   //
-  virtual void CurrentRate(WebRtc_Word32& /* rate_bps */) {
+  virtual void CurrentRate(int32_t& /* rate_bps */) {
     return;
   }
 
@@ -1156,30 +1156,30 @@
 
   // &in_audio_[in_audio_ix_write_] always point to where new audio can be
   // written to
-  WebRtc_Word16 in_audio_ix_write_;
+  int16_t in_audio_ix_write_;
 
   // &in_audio_[in_audio_ix_read_] points to where audio has to be read from
-  WebRtc_Word16 in_audio_ix_read_;
+  int16_t in_audio_ix_read_;
 
-  WebRtc_Word16 in_timestamp_ix_write_;
+  int16_t in_timestamp_ix_write_;
 
   // Where the audio is stored before encoding,
   // To save memory the following buffer can be allocated
   // dynamically for 80 ms depending on the sampling frequency
   // of the codec.
-  WebRtc_Word16* in_audio_;
-  WebRtc_UWord32* in_timestamp_;
+  int16_t* in_audio_;
+  uint32_t* in_timestamp_;
 
-  WebRtc_Word16 frame_len_smpl_;
-  WebRtc_UWord16 num_channels_;
+  int16_t frame_len_smpl_;
+  uint16_t num_channels_;
 
   // This will point to a static database of the supported codecs
-  WebRtc_Word16 codec_id_;
+  int16_t codec_id_;
 
   // This will account for the number of samples  were not encoded
   // the case is rare, either samples are missed due to overwrite
   // at input buffer or due to encoding error
-  WebRtc_UWord32 num_missed_samples_;
+  uint32_t num_missed_samples_;
 
   // True if the encoder instance created
   bool encoder_exist_;
@@ -1195,10 +1195,10 @@
   WebRtcVadInst* ptr_vad_inst_;
   bool vad_enabled_;
   ACMVADMode vad_mode_;
-  WebRtc_Word16 vad_label_[MAX_FRAME_SIZE_10MSEC];
+  int16_t vad_label_[MAX_FRAME_SIZE_10MSEC];
   bool dtx_enabled_;
   WebRtcCngEncInst* ptr_dtx_inst_;
-  WebRtc_UWord8 num_lpc_params_;
+  uint8_t num_lpc_params_;
   bool sent_cn_previous_;
   bool is_master_;
   int16_t prev_frame_cng_;
@@ -1213,10 +1213,10 @@
   // such as buffers and state variables.
   RWLockWrapper& codec_wrapper_lock_;
 
-  WebRtc_UWord32 last_encoded_timestamp_;
-  WebRtc_UWord32 last_timestamp_;
+  uint32_t last_encoded_timestamp_;
+  uint32_t last_timestamp_;
   bool is_audio_buff_fresh_;
-  WebRtc_UWord32 unique_id_;
+  uint32_t unique_id_;
 };
 
 }  // namespace webrt
diff --git a/modules/audio_coding/main/source/acm_gsmfr.cc b/modules/audio_coding/main/source/acm_gsmfr.cc
index 5b59ed3..22bbbd8 100644
--- a/modules/audio_coding/main/source/acm_gsmfr.cc
+++ b/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -26,7 +26,7 @@
 
 #ifndef WEBRTC_CODEC_GSMFR
 
-ACMGSMFR::ACMGSMFR(WebRtc_Word16 /* codec_id */)
+ACMGSMFR::ACMGSMFR(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   return;
@@ -36,40 +36,40 @@
   return;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMGSMFR::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMGSMFR::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMGSMFR::EnableDTX() {
+int16_t ACMGSMFR::EnableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMGSMFR::DisableDTX() {
+int16_t ACMGSMFR::DisableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalInitEncoder(
+int16_t ACMGSMFR::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
+int16_t ACMGSMFR::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                 const CodecInst& /* codec_inst */) {
+int32_t ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                           const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -77,7 +77,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
+int16_t ACMGSMFR::InternalCreateEncoder() {
   return -1;
 }
 
@@ -85,7 +85,7 @@
   return;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
+int16_t ACMGSMFR::InternalCreateDecoder() {
   return -1;
 }
 
@@ -99,7 +99,7 @@
 
 #else     //===================== Actual Implementation =======================
 
-ACMGSMFR::ACMGSMFR(WebRtc_Word16 codec_id)
+ACMGSMFR::ACMGSMFR(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   codec_id_ = codec_id;
@@ -119,27 +119,27 @@
   return;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalEncode(WebRtc_UWord8* bitstream,
-                                       WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMGSMFR::InternalEncode(uint8_t* bitstream,
+                                 int16_t* bitstream_len_byte) {
   *bitstream_len_byte = WebRtcGSMFR_Encode(encoder_inst_ptr_,
                                            &in_audio_[in_audio_ix_read_],
                                            frame_len_smpl_,
-                                           (WebRtc_Word16*)bitstream);
+                                           (int16_t*)bitstream);
   // increment the read index this tell the caller that how far
   // we have gone forward in reading the audio buffer
   in_audio_ix_read_ += frame_len_smpl_;
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMGSMFR::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMGSMFR::EnableDTX() {
+int16_t ACMGSMFR::EnableDTX() {
   if (dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {
@@ -155,7 +155,7 @@
   }
 }
 
-WebRtc_Word16 ACMGSMFR::DisableDTX() {
+int16_t ACMGSMFR::DisableDTX() {
   if (!dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {
@@ -172,7 +172,7 @@
   }
 }
 
-WebRtc_Word16 ACMGSMFR::InternalInitEncoder(
+int16_t ACMGSMFR::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
   if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_,
                               ((codec_params->enable_dtx) ? 1 : 0)) < 0) {
@@ -182,7 +182,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
+int16_t ACMGSMFR::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   if (WebRtcGSMFR_DecoderInit(decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
@@ -192,8 +192,8 @@
   return 0;
 }
 
-WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                 const CodecInst& codec_inst) {
+int32_t ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                           const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "CodecDef: decoder is not initialized for GSMFR");
@@ -213,7 +213,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
+int16_t ACMGSMFR::InternalCreateEncoder() {
   if (WebRtcGSMFR_CreateEnc(&encoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateEncoder: cannot create instance for GSMFR "
@@ -232,7 +232,7 @@
   encoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
+int16_t ACMGSMFR::InternalCreateDecoder() {
   if (WebRtcGSMFR_CreateDec(&decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateDecoder: cannot create instance for GSMFR "
diff --git a/modules/audio_coding/main/source/acm_gsmfr.h b/modules/audio_coding/main/source/acm_gsmfr.h
index b841711..61f5768 100644
--- a/modules/audio_coding/main/source/acm_gsmfr.h
+++ b/modules/audio_coding/main/source/acm_gsmfr.h
@@ -21,42 +21,42 @@
 
 class ACMGSMFR : public ACMGenericCodec {
  public:
-  explicit ACMGSMFR(WebRtc_Word16 codec_id);
+  explicit ACMGSMFR(int16_t codec_id);
   ~ACMGSMFR();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 EnableDTX();
+  int16_t EnableDTX();
 
-  WebRtc_Word16 DisableDTX();
+  int16_t DisableDTX();
 
   GSMFR_encinst_t_* encoder_inst_ptr_;
   GSMFR_decinst_t_* decoder_inst_ptr_;
diff --git a/modules/audio_coding/main/source/acm_ilbc.cc b/modules/audio_coding/main/source/acm_ilbc.cc
index 963201b..a2a294e 100644
--- a/modules/audio_coding/main/source/acm_ilbc.cc
+++ b/modules/audio_coding/main/source/acm_ilbc.cc
@@ -23,7 +23,7 @@
 
 #ifndef WEBRTC_CODEC_ILBC
 
-ACMILBC::ACMILBC(WebRtc_Word16 /* codec_id */)
+ACMILBC::ACMILBC(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   return;
@@ -33,32 +33,32 @@
   return;
 }
 
-WebRtc_Word16 ACMILBC::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMILBC::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMILBC::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMILBC::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMILBC::InternalInitEncoder(
+int16_t ACMILBC::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMILBC::InternalInitDecoder(
+int16_t ACMILBC::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
-                                const CodecInst& /* codec_inst */) {
+int32_t ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def  */,
+                          const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -66,7 +66,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMILBC::InternalCreateEncoder() {
+int16_t ACMILBC::InternalCreateEncoder() {
   return -1;
 }
 
@@ -74,7 +74,7 @@
   return;
 }
 
-WebRtc_Word16 ACMILBC::InternalCreateDecoder() {
+int16_t ACMILBC::InternalCreateDecoder() {
   return -1;
 }
 
@@ -86,13 +86,13 @@
   return;
 }
 
-WebRtc_Word16 ACMILBC::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+int16_t ACMILBC::SetBitRateSafe(const int32_t /* rate */) {
   return -1;
 }
 
 #else  //===================== Actual Implementation =======================
 
-ACMILBC::ACMILBC(WebRtc_Word16 codec_id)
+ACMILBC::ACMILBC(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   codec_id_ = codec_id;
@@ -111,12 +111,12 @@
   return;
 }
 
-WebRtc_Word16 ACMILBC::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMILBC::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
   *bitstream_len_byte = WebRtcIlbcfix_Encode(encoder_inst_ptr_,
                                              &in_audio_[in_audio_ix_read_],
                                              frame_len_smpl_,
-                                             (WebRtc_Word16*)bitstream);
+                                             (int16_t*)bitstream);
   if (*bitstream_len_byte < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalEncode: error in encode for ILBC");
@@ -128,15 +128,15 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMILBC::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMILBC::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMILBC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMILBC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
   // initialize with a correct processing block length
   if ((160 == (codec_params->codec_inst).pacsize) ||
       (320 == (codec_params->codec_inst).pacsize)) {
@@ -153,7 +153,7 @@
   }
 }
 
-WebRtc_Word16 ACMILBC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMILBC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
   // initialize with a correct processing block length
   if ((160 == (codec_params->codec_inst).pacsize) ||
       (320 == (codec_params->codec_inst).pacsize)) {
@@ -170,8 +170,8 @@
   }
 }
 
-WebRtc_Word32 ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "CodeDef: decoder not initialized for ILBC");
@@ -191,7 +191,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMILBC::InternalCreateEncoder() {
+int16_t ACMILBC::InternalCreateEncoder() {
   if (WebRtcIlbcfix_EncoderCreate(&encoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateEncoder: cannot create instance for ILBC "
@@ -210,7 +210,7 @@
   }
 }
 
-WebRtc_Word16 ACMILBC::InternalCreateDecoder() {
+int16_t ACMILBC::InternalCreateDecoder() {
   if (WebRtcIlbcfix_DecoderCreate(&decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateDecoder: cannot create instance for ILBC "
@@ -236,7 +236,7 @@
   return;
 }
 
-WebRtc_Word16 ACMILBC::SetBitRateSafe(const WebRtc_Word32 rate) {
+int16_t ACMILBC::SetBitRateSafe(const int32_t rate) {
   // Check that rate is valid. No need to store the value
   if (rate == 13300) {
     WebRtcIlbcfix_EncoderInit(encoder_inst_ptr_, 30);
diff --git a/modules/audio_coding/main/source/acm_ilbc.h b/modules/audio_coding/main/source/acm_ilbc.h
index a61e34e..eb619f0 100644
--- a/modules/audio_coding/main/source/acm_ilbc.h
+++ b/modules/audio_coding/main/source/acm_ilbc.h
@@ -21,38 +21,38 @@
 
 class ACMILBC : public ACMGenericCodec {
  public:
-  explicit ACMILBC(WebRtc_Word16 codec_id);
+  explicit ACMILBC(int16_t codec_id);
   ~ACMILBC();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+  int16_t SetBitRateSafe(const int32_t rate);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
diff --git a/modules/audio_coding/main/source/acm_isac.cc b/modules/audio_coding/main/source/acm_isac.cc
index defcb8b..e22d3f6 100644
--- a/modules/audio_coding/main/source/acm_isac.cc
+++ b/modules/audio_coding/main/source/acm_isac.cc
@@ -41,14 +41,14 @@
 
 // Tables for bandwidth estimates
 #define NR_ISAC_BANDWIDTHS 24
-static const WebRtc_Word32 kIsacRatesWb[NR_ISAC_BANDWIDTHS] = {
+static const int32_t kIsacRatesWb[NR_ISAC_BANDWIDTHS] = {
     10000, 11100, 12300, 13700, 15200, 16900,
     18800, 20900, 23300, 25900, 28700, 31900,
     10100, 11200, 12400, 13800, 15300, 17000,
     18900, 21000, 23400, 26000, 28800, 32000
 };
 
-static const WebRtc_Word32 kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = {
+static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = {
     10000, 11000, 12400, 13800, 15300, 17000,
     18900, 21000, 23200, 25400, 27600, 29800,
     32000, 34100, 36300, 38500, 40700, 42900,
@@ -57,7 +57,7 @@
 
 #if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))
 
-ACMISAC::ACMISAC(WebRtc_Word16 /* codec_id */)
+ACMISAC::ACMISAC(int16_t /* codec_id */)
     : codec_inst_ptr_(NULL),
       is_enc_initialized_(false),
       isac_coding_mode_(CHANNEL_INDEPENDENT),
@@ -79,31 +79,31 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMISAC::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMISAC::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMISAC::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMISAC::InternalInitEncoder(
+int16_t ACMISAC::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::InternalInitDecoder(
+int16_t ACMISAC::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::InternalCreateDecoder() {
+int16_t ACMISAC::InternalCreateDecoder() {
   return -1;
 }
 
@@ -111,7 +111,7 @@
   return;
 }
 
-WebRtc_Word16 ACMISAC::InternalCreateEncoder() {
+int16_t ACMISAC::InternalCreateEncoder() {
   return -1;
 }
 
@@ -119,8 +119,8 @@
   return;
 }
 
-WebRtc_Word32 ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                const CodecInst& /* codec_inst */) {
+int32_t ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                          const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -128,69 +128,69 @@
   return;
 }
 
-WebRtc_Word16 ACMISAC::DeliverCachedIsacData(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */,
-    WebRtc_UWord32* /* timestamp */,
+int16_t ACMISAC::DeliverCachedIsacData(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */,
+    uint32_t* /* timestamp */,
     WebRtcACMEncodingType* /* encoding_type */,
-    const WebRtc_UWord16 /* isac_rate */,
-    const WebRtc_UWord8 /* isac_bw_estimate */) {
+    const uint16_t /* isac_rate */,
+    const uint8_t /* isac_bw_estimate */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::Transcode(WebRtc_UWord8* /* bitstream */,
-                                 WebRtc_Word16* /* bitstream_len_byte */,
-                                 WebRtc_Word16 /* q_bwe */,
-                                 WebRtc_Word32 /* scale */,
-                                 bool /* is_red */) {
+int16_t ACMISAC::Transcode(uint8_t* /* bitstream */,
+                           int16_t* /* bitstream_len_byte */,
+                           int16_t /* q_bwe */,
+                           int32_t /* scale */,
+                           bool /* is_red */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::SetBitRateSafe(WebRtc_Word32 /* bit_rate */) {
+int16_t ACMISAC::SetBitRateSafe(int32_t /* bit_rate */) {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
+int32_t ACMISAC::GetEstimatedBandwidthSafe() {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::SetEstimatedBandwidthSafe(
-    WebRtc_Word32 /* estimated_bandwidth */) {
+int32_t ACMISAC::SetEstimatedBandwidthSafe(
+    int32_t /* estimated_bandwidth */) {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::GetRedPayloadSafe(WebRtc_UWord8* /* red_payload */,
-                                         WebRtc_Word16* /* payload_bytes */) {
+int32_t ACMISAC::GetRedPayloadSafe(uint8_t* /* red_payload */,
+                                   int16_t* /* payload_bytes */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::UpdateDecoderSampFreq(WebRtc_Word16 /* codec_id */) {
+int16_t ACMISAC::UpdateDecoderSampFreq(int16_t /* codec_id */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
-    WebRtc_UWord16 /* encoder_samp_freq_hz */) {
+int16_t ACMISAC::UpdateEncoderSampFreq(
+    uint16_t /* encoder_samp_freq_hz */) {
   return -1;
 }
 
-WebRtc_Word16 ACMISAC::EncoderSampFreq(WebRtc_UWord16& /* samp_freq_hz */) {
+int16_t ACMISAC::EncoderSampFreq(uint16_t& /* samp_freq_hz */) {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::ConfigISACBandwidthEstimator(
-    const WebRtc_UWord8 /* init_frame_size_msec */,
-    const WebRtc_UWord16 /* init_rate_bit_per_sec */,
+int32_t ACMISAC::ConfigISACBandwidthEstimator(
+    const uint8_t /* init_frame_size_msec */,
+    const uint16_t /* init_rate_bit_per_sec */,
     const bool /* enforce_frame_size  */) {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::SetISACMaxPayloadSize(
-    const WebRtc_UWord16 /* max_payload_len_bytes */) {
+int32_t ACMISAC::SetISACMaxPayloadSize(
+    const uint16_t /* max_payload_len_bytes */) {
   return -1;
 }
 
-WebRtc_Word32 ACMISAC::SetISACMaxRate(
-    const WebRtc_UWord32 /* max_rate_bit_per_sec */) {
+int32_t ACMISAC::SetISACMaxRate(
+    const uint32_t /* max_rate_bit_per_sec */) {
   return -1;
 }
 
@@ -198,14 +198,14 @@
   return;
 }
 
-void ACMISAC::CurrentRate(WebRtc_Word32& /*rate_bit_per_sec */) {
+void ACMISAC::CurrentRate(int32_t& /*rate_bit_per_sec */) {
   return;
 }
 
 bool
 ACMISAC::DecoderParamsSafe(
     WebRtcACMCodecParams* /* dec_params */,
-    const WebRtc_UWord8   /* payload_type */) {
+    const uint8_t   /* payload_type */) {
   return false;
 }
 
@@ -215,11 +215,11 @@
   return;
 }
 
-WebRtc_Word16 ACMISAC::REDPayloadISAC(
-    const WebRtc_Word32 /* isac_rate */,
-    const WebRtc_Word16 /* isac_bw_estimate */,
-    WebRtc_UWord8* /* payload */,
-    WebRtc_Word16* /* payload_len_bytes */) {
+int16_t ACMISAC::REDPayloadISAC(
+    const int32_t /* isac_rate */,
+    const int16_t /* isac_bw_estimate */,
+    uint8_t* /* payload */,
+    int16_t* /* payload_len_bytes */) {
   return -1;
 }
 
@@ -243,7 +243,7 @@
 
 #define ISAC_NUM_SUPPORTED_RATES 9
 
-static const WebRtc_UWord16 kIsacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
+static const uint16_t kIsacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
     32000,    30000,    26000,   23000,   21000,
     19000,    17000,   15000,    12000
 };
@@ -258,11 +258,11 @@
   kIsacSuperWideband = 32
 };
 
-static float ACMISACFixTranscodingScale(WebRtc_UWord16 rate) {
+static float ACMISACFixTranscodingScale(uint16_t rate) {
   // find the scale for transcoding, the scale is rounded
   // downward
   float scale = -1;
-  for (WebRtc_Word16 n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) {
+  for (int16_t n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) {
     if (rate >= kIsacSuportedRates[n]) {
       scale = kIsacScale[n];
       break;
@@ -272,29 +272,29 @@
 }
 
 static void ACMISACFixGetSendBitrate(ACM_ISAC_STRUCT* inst,
-                                     WebRtc_Word32* bottleneck) {
+                                     int32_t* bottleneck) {
   *bottleneck = WebRtcIsacfix_GetUplinkBw(inst);
 }
 
-static WebRtc_Word16 ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
-                                               WebRtc_Word16 bwe_index,
-                                               WebRtc_Word16 /* jitter_index */,
-                                               WebRtc_Word32 rate,
-                                               WebRtc_Word16* bitstream,
+static int16_t ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
+                                         int16_t bwe_index,
+                                         int16_t /* jitter_index */,
+                                         int32_t rate,
+                                         int16_t* bitstream,
                                                bool is_red) {
   if (is_red) {
     // RED not supported with iSACFIX
     return -1;
   }
-  float scale = ACMISACFixTranscodingScale((WebRtc_UWord16) rate);
+  float scale = ACMISACFixTranscodingScale((uint16_t) rate);
   return WebRtcIsacfix_GetNewBitStream(inst, bwe_index, scale, bitstream);
 }
 
-static WebRtc_Word16 ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst,
-                                          WebRtc_Word16* rate_index,
-                                          WebRtc_Word16* /* dummy */) {
-  WebRtc_Word16 local_rate_index;
-  WebRtc_Word16 status = WebRtcIsacfix_GetDownLinkBwIndex(inst,
+static int16_t ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst,
+                                    int16_t* rate_index,
+                                    int16_t* /* dummy */) {
+  int16_t local_rate_index;
+  int16_t status = WebRtcIsacfix_GetDownLinkBwIndex(inst,
                                                           &local_rate_index);
   if (status < 0) {
     return -1;
@@ -304,34 +304,34 @@
   }
 }
 
-static WebRtc_Word16 ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst,
-                                          WebRtc_Word32 rate_bps,
-                                          WebRtc_Word16 frame_size_ms,
-                                          WebRtc_Word16 enforce_frame_size) {
-  return WebRtcIsacfix_ControlBwe(inst, (WebRtc_Word16) rate_bps, frame_size_ms,
+static int16_t ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst,
+                                    int32_t rate_bps,
+                                    int16_t frame_size_ms,
+                                    int16_t enforce_frame_size) {
+  return WebRtcIsacfix_ControlBwe(inst, (int16_t) rate_bps, frame_size_ms,
                                   enforce_frame_size);
 }
 
-static WebRtc_Word16 ACMISACFixControl(ACM_ISAC_STRUCT* inst,
-                                       WebRtc_Word32 rate_bps,
-                                       WebRtc_Word16 frame_size_ms) {
-  return WebRtcIsacfix_Control(inst, (WebRtc_Word16) rate_bps, frame_size_ms);
+static int16_t ACMISACFixControl(ACM_ISAC_STRUCT* inst,
+                                 int32_t rate_bps,
+                                 int16_t frame_size_ms) {
+  return WebRtcIsacfix_Control(inst, (int16_t) rate_bps, frame_size_ms);
 }
 
 // The following two function should have the same signature as their counter
 // part in iSAC floating-point, i.e. WebRtcIsac_EncSampRate &
 // WebRtcIsac_DecSampRate.
-static WebRtc_UWord16 ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) {
+static uint16_t ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) {
   return 16000;
 }
 
-static WebRtc_UWord16 ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) {
+static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) {
   return 16000;
 }
 
 #endif
 
-ACMISAC::ACMISAC(WebRtc_Word16 codec_id)
+ACMISAC::ACMISAC(int16_t codec_id)
     : is_enc_initialized_(false),
       isac_coding_mode_(CHANNEL_INDEPENDENT),
       enforce_frame_size_(false),
@@ -372,8 +372,8 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMISAC::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMISAC::InternalEncode(uint8_t* bitstream,
+                               int16_t* bitstream_len_byte) {
   // ISAC takes 10ms audio everytime we call encoder, therefor,
   // it should be treated like codecs with 'basic coding block'
   // non-zero, and the following 'while-loop' should not be necessary.
@@ -396,7 +396,7 @@
     }
     *bitstream_len_byte = ACM_ISAC_ENCODE(codec_inst_ptr_->inst,
                                            &in_audio_[in_audio_ix_read_],
-                                           (WebRtc_Word16*)bitstream);
+                                           (int16_t*)bitstream);
     // increment the read index this tell the caller that how far
     // we have gone forward in reading the audio buffer
     in_audio_ix_read_ += samples_in_10ms_audio_;
@@ -417,15 +417,15 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMISAC::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_sample */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMISAC::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_sample */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
   // if rate is set to -1 then iSAC has to be in adaptive mode
   if (codec_params->codec_inst.rate == -1) {
     isac_coding_mode_ = ADAPTIVE;
@@ -439,7 +439,7 @@
   }
 
   // we need to set the encoder sampling frequency.
-  if (UpdateEncoderSampFreq((WebRtc_UWord16) codec_params->codec_inst.plfreq)
+  if (UpdateEncoderSampFreq((uint16_t) codec_params->codec_inst.plfreq)
       < 0) {
     return -1;
   }
@@ -465,7 +465,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
+int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
   if (codec_inst_ptr_ == NULL) {
     return -1;
   }
@@ -496,11 +496,11 @@
   return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst);
 }
 
-WebRtc_Word16 ACMISAC::InternalCreateDecoder() {
+int16_t ACMISAC::InternalCreateDecoder() {
   if (codec_inst_ptr_ == NULL) {
     return -1;
   }
-  WebRtc_Word16 status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
+  int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
 
   // specific to codecs with one instance for encoding and decoding
   encoder_initialized_ = false;
@@ -518,11 +518,11 @@
   return;
 }
 
-WebRtc_Word16 ACMISAC::InternalCreateEncoder() {
+int16_t ACMISAC::InternalCreateEncoder() {
   if (codec_inst_ptr_ == NULL) {
     return -1;
   }
-  WebRtc_Word16 status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
+  int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
 
   // specific to codecs with one instance for encoding and decoding
   decoder_initialized_ = false;
@@ -540,8 +540,8 @@
   return;
 }
 
-WebRtc_Word32 ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   // Sanity checks
   if (codec_inst_ptr_ == NULL) {
     return -1;
@@ -588,12 +588,12 @@
   return;
 }
 
-WebRtc_Word16 ACMISAC::Transcode(WebRtc_UWord8* bitstream,
-                                 WebRtc_Word16* bitstream_len_byte,
-                                 WebRtc_Word16 q_bwe,
-                                 WebRtc_Word32 rate,
-                                 bool is_red) {
-  WebRtc_Word16 jitter_info = 0;
+int16_t ACMISAC::Transcode(uint8_t* bitstream,
+                           int16_t* bitstream_len_byte,
+                           int16_t q_bwe,
+                           int32_t rate,
+                           bool is_red) {
+  int16_t jitter_info = 0;
   // transcode from a higher rate to lower rate sanity check
   if (codec_inst_ptr_ == NULL) {
     return -1;
@@ -601,7 +601,7 @@
 
   *bitstream_len_byte = ACM_ISAC_GETNEWBITSTREAM(codec_inst_ptr_->inst, q_bwe,
                                                  jitter_info, rate,
-                                                 (WebRtc_Word16*)bitstream,
+                                                 (int16_t*)bitstream,
                                                  (is_red) ? 1 : 0);
 
   if (*bitstream_len_byte < 0) {
@@ -613,11 +613,11 @@
   }
 }
 
-WebRtc_Word16 ACMISAC::SetBitRateSafe(WebRtc_Word32 bit_rate) {
+int16_t ACMISAC::SetBitRateSafe(int32_t bit_rate) {
   if (codec_inst_ptr_ == NULL) {
     return -1;
   }
-  WebRtc_UWord16 encoder_samp_freq;
+  uint16_t encoder_samp_freq;
   EncoderSampFreq(encoder_samp_freq);
   bool reinit = false;
   // change the BN of iSAC
@@ -640,13 +640,13 @@
       reinit = true;
     }
     // store the bottleneck
-    isac_current_bn_ = (WebRtc_UWord16) bit_rate;
+    isac_current_bn_ = (uint16_t) bit_rate;
   } else {
     // invlaid rate
     return -1;
   }
 
-  WebRtc_Word16 status = 0;
+  int16_t status = 0;
   if (reinit) {
     // initialize and check if it is successful
     if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) {
@@ -671,9 +671,9 @@
   return status;
 }
 
-WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
-  WebRtc_Word16 bandwidth_index = 0;
-  WebRtc_Word16 delay_index = 0;
+int32_t ACMISAC::GetEstimatedBandwidthSafe() {
+  int16_t bandwidth_index = 0;
+  int16_t delay_index = 0;
   int samp_rate;
 
   // Get bandwidth information
@@ -693,10 +693,10 @@
   }
 }
 
-WebRtc_Word32 ACMISAC::SetEstimatedBandwidthSafe(
-    WebRtc_Word32 estimated_bandwidth) {
+int32_t ACMISAC::SetEstimatedBandwidthSafe(
+    int32_t estimated_bandwidth) {
   int samp_rate;
-  WebRtc_Word16 bandwidth_index;
+  int16_t bandwidth_index;
 
   // Check sample frequency and choose appropriate table
   samp_rate = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst);
@@ -734,14 +734,14 @@
   return 0;
 }
 
-WebRtc_Word32 ACMISAC::GetRedPayloadSafe(
+int32_t ACMISAC::GetRedPayloadSafe(
 #if (!defined(WEBRTC_CODEC_ISAC))
-    WebRtc_UWord8* /* red_payload */, WebRtc_Word16* /* payload_bytes */) {
+    uint8_t* /* red_payload */, int16_t* /* payload_bytes */) {
   return -1;
 #else
-    WebRtc_UWord8* red_payload, WebRtc_Word16* payload_bytes) {
-  WebRtc_Word16 bytes = WebRtcIsac_GetRedPayload(codec_inst_ptr_->inst,
-                                                 (WebRtc_Word16*)red_payload);
+    uint8_t* red_payload, int16_t* payload_bytes) {
+  int16_t bytes = WebRtcIsac_GetRedPayload(codec_inst_ptr_->inst,
+                                                 (int16_t*)red_payload);
   if (bytes < 0) {
     return -1;
   }
@@ -750,9 +750,9 @@
 #endif
 }
 
-WebRtc_Word16 ACMISAC::UpdateDecoderSampFreq(
+int16_t ACMISAC::UpdateDecoderSampFreq(
 #ifdef WEBRTC_CODEC_ISAC
-    WebRtc_Word16 codec_id) {
+    int16_t codec_id) {
     // The decoder supports only wideband and super-wideband.
   if (ACMCodecDB::kISAC == codec_id) {
     return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 16000);
@@ -763,15 +763,15 @@
     return -1;
   }
 #else
-    WebRtc_Word16 /* codec_id */) {
+    int16_t /* codec_id */) {
   return 0;
 #endif
 }
 
-WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
+int16_t ACMISAC::UpdateEncoderSampFreq(
 #ifdef WEBRTC_CODEC_ISAC
-    WebRtc_UWord16 encoder_samp_freq_hz) {
-  WebRtc_UWord16 current_samp_rate_hz;
+    uint16_t encoder_samp_freq_hz) {
+  uint16_t current_samp_rate_hz;
   EncoderSampFreq(current_samp_rate_hz);
 
   if (current_samp_rate_hz != encoder_samp_freq_hz) {
@@ -795,23 +795,23 @@
     }
   }
 #else
-    WebRtc_UWord16 /* codec_id */) {
+    uint16_t /* codec_id */) {
 #endif
   return 0;
 }
 
-WebRtc_Word16 ACMISAC::EncoderSampFreq(WebRtc_UWord16& samp_freq_hz) {
+int16_t ACMISAC::EncoderSampFreq(uint16_t& samp_freq_hz) {
   samp_freq_hz = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst);
   return 0;
 }
 
-WebRtc_Word32 ACMISAC::ConfigISACBandwidthEstimator(
-    const WebRtc_UWord8 init_frame_size_msec,
-    const WebRtc_UWord16 init_rate_bit_per_sec,
+int32_t ACMISAC::ConfigISACBandwidthEstimator(
+    const uint8_t init_frame_size_msec,
+    const uint16_t init_rate_bit_per_sec,
     const bool enforce_frame_size) {
-  WebRtc_Word16 status;
+  int16_t status;
   {
-    WebRtc_UWord16 samp_freq_hz;
+    uint16_t samp_freq_hz;
     EncoderSampFreq(samp_freq_hz);
     // TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce
     // the frame-size otherwise we might get error. Revise if
@@ -836,14 +836,14 @@
   return 0;
 }
 
-WebRtc_Word32 ACMISAC::SetISACMaxPayloadSize(
-    const WebRtc_UWord16 max_payload_len_bytes) {
+int32_t ACMISAC::SetISACMaxPayloadSize(
+    const uint16_t max_payload_len_bytes) {
   return ACM_ISAC_SETMAXPAYLOADSIZE(codec_inst_ptr_->inst,
                                     max_payload_len_bytes);
 }
 
-WebRtc_Word32 ACMISAC::SetISACMaxRate(
-    const WebRtc_UWord32 max_rate_bit_per_sec) {
+int32_t ACMISAC::SetISACMaxRate(
+    const uint32_t max_rate_bit_per_sec) {
   return ACM_ISAC_SETMAXRATE(codec_inst_ptr_->inst, max_rate_bit_per_sec);
 }
 
@@ -852,14 +852,14 @@
   encoder_params_.codec_inst.pacsize = frame_len_smpl_;
 }
 
-void ACMISAC::CurrentRate(WebRtc_Word32& rate_bit_per_sec) {
+void ACMISAC::CurrentRate(int32_t& rate_bit_per_sec) {
   if (isac_coding_mode_ == ADAPTIVE) {
     ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &rate_bit_per_sec);
   }
 }
 
 bool ACMISAC::DecoderParamsSafe(WebRtcACMCodecParams* dec_params,
-                                const WebRtc_UWord8 payload_type) {
+                                const uint8_t payload_type) {
   if (decoder_initialized_) {
     if (payload_type == decoder_params_.codec_inst.pltype) {
       memcpy(dec_params, &decoder_params_, sizeof(WebRtcACMCodecParams));
@@ -883,11 +883,11 @@
   }
 }
 
-WebRtc_Word16 ACMISAC::REDPayloadISAC(const WebRtc_Word32 isac_rate,
-                                      const WebRtc_Word16 isac_bw_estimate,
-                                      WebRtc_UWord8* payload,
-                                      WebRtc_Word16* payload_len_bytes) {
-  WebRtc_Word16 status;
+int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate,
+                                const int16_t isac_bw_estimate,
+                                uint8_t* payload,
+                                int16_t* payload_len_bytes) {
+  int16_t status;
   ReadLockScoped rl(codec_wrapper_lock_);
   status = Transcode(payload, payload_len_bytes, isac_bw_estimate, isac_rate,
                      true);
diff --git a/modules/audio_coding/main/source/acm_isac.h b/modules/audio_coding/main/source/acm_isac.h
index 1724cf8..6255477 100644
--- a/modules/audio_coding/main/source/acm_isac.h
+++ b/modules/audio_coding/main/source/acm_isac.h
@@ -24,95 +24,95 @@
 
 class ACMISAC : public ACMGenericCodec {
  public:
-  explicit ACMISAC(WebRtc_Word16 codec_id);
+  explicit ACMISAC(int16_t codec_id);
   ~ACMISAC();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 DeliverCachedIsacData(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte,
-                                      WebRtc_UWord32* timestamp,
-                                      WebRtcACMEncodingType* encoding_type,
-                                      const WebRtc_UWord16 isac_rate,
-                                      const WebRtc_UWord8 isac_bwestimate);
+  int16_t DeliverCachedIsacData(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte,
+                                uint32_t* timestamp,
+                                WebRtcACMEncodingType* encoding_type,
+                                const uint16_t isac_rate,
+                                const uint8_t isac_bwestimate);
 
-  WebRtc_Word16 DeliverCachedData(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16* /* bitstream_len_byte */,
-                                  WebRtc_UWord32* /* timestamp */,
-                                  WebRtcACMEncodingType* /* encoding_type */) {
+  int16_t DeliverCachedData(uint8_t* /* bitstream */,
+                            int16_t* /* bitstream_len_byte */,
+                            uint32_t* /* timestamp */,
+                            WebRtcACMEncodingType* /* encoding_type */) {
     return -1;
   }
 
-  WebRtc_Word16 UpdateDecoderSampFreq(WebRtc_Word16 codec_id);
+  int16_t UpdateDecoderSampFreq(int16_t codec_id);
 
-  WebRtc_Word16 UpdateEncoderSampFreq(WebRtc_UWord16 samp_freq_hz);
+  int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz);
 
-  WebRtc_Word16 EncoderSampFreq(WebRtc_UWord16& samp_freq_hz);
+  int16_t EncoderSampFreq(uint16_t& samp_freq_hz);
 
-  WebRtc_Word32 ConfigISACBandwidthEstimator(
-      const WebRtc_UWord8 init_frame_size_msec,
-      const WebRtc_UWord16 init_rate_bit_per_sec,
+  int32_t ConfigISACBandwidthEstimator(
+      const uint8_t init_frame_size_msec,
+      const uint16_t init_rate_bit_per_sec,
       const bool enforce_frame_size);
 
-  WebRtc_Word32 SetISACMaxPayloadSize(
-      const WebRtc_UWord16 max_payload_len_bytes);
+  int32_t SetISACMaxPayloadSize(
+      const uint16_t max_payload_len_bytes);
 
-  WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_rate_bit_per_sec);
+  int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec);
 
-  WebRtc_Word16 REDPayloadISAC(const WebRtc_Word32 isac_rate,
-                               const WebRtc_Word16 isac_bw_estimate,
-                               WebRtc_UWord8* payload,
-                               WebRtc_Word16* payload_len_bytes);
+  int16_t REDPayloadISAC(const int32_t isac_rate,
+                         const int16_t isac_bw_estimate,
+                         uint8_t* payload,
+                         int16_t* payload_len_bytes);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 bit_rate);
+  int16_t SetBitRateSafe(const int32_t bit_rate);
 
-  WebRtc_Word32 GetEstimatedBandwidthSafe();
+  int32_t GetEstimatedBandwidthSafe();
 
-  WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimated_bandwidth);
+  int32_t SetEstimatedBandwidthSafe(int32_t estimated_bandwidth);
 
-  WebRtc_Word32 GetRedPayloadSafe(WebRtc_UWord8* red_payload,
-                                  WebRtc_Word16* payload_bytes);
+  int32_t GetRedPayloadSafe(uint8_t* red_payload,
+                            int16_t* payload_bytes);
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 Transcode(WebRtc_UWord8* bitstream,
-                          WebRtc_Word16* bitstream_len_byte,
-                          WebRtc_Word16 q_bwe,
-                          WebRtc_Word32 rate,
-                          bool is_red);
+  int16_t Transcode(uint8_t* bitstream,
+                    int16_t* bitstream_len_byte,
+                    int16_t q_bwe,
+                    int32_t rate,
+                    bool is_red);
 
-  void CurrentRate(WebRtc_Word32& rate_bit_per_sec);
+  void CurrentRate(int32_t& rate_bit_per_sec);
 
   void UpdateFrameLen();
 
   bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params,
-                         const WebRtc_UWord8 payload_type);
+                         const uint8_t payload_type);
 
   void SaveDecoderParamSafe(const WebRtcACMCodecParams* codec_params);
 
@@ -120,8 +120,8 @@
   bool is_enc_initialized_;
   IsacCodingMode isac_coding_mode_;
   bool enforce_frame_size_;
-  WebRtc_Word32 isac_current_bn_;
-  WebRtc_UWord16 samples_in_10ms_audio_;
+  int32_t isac_current_bn_;
+  uint16_t samples_in_10ms_audio_;
   WebRtcACMCodecParams decoder_params_32khz_;
 };
 
diff --git a/modules/audio_coding/main/source/acm_neteq.cc b/modules/audio_coding/main/source/acm_neteq.cc
index ce80184..75e73eb 100644
--- a/modules/audio_coding/main/source/acm_neteq.cc
+++ b/modules/audio_coding/main/source/acm_neteq.cc
@@ -77,10 +77,10 @@
   }
 }
 
-WebRtc_Word32 ACMNetEQ::Init() {
+int32_t ACMNetEQ::Init() {
   CriticalSectionScoped lock(neteq_crit_sect_);
 
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (InitByIdxSafe(idx) < 0) {
       return -1;
     }
@@ -117,7 +117,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::InitByIdxSafe(const WebRtc_Word16 idx) {
+int16_t ACMNetEQ::InitByIdxSafe(const int16_t idx) {
   int memory_size_bytes;
   if (WebRtcNetEQ_AssignSize(&memory_size_bytes) != 0) {
     LogError("AssignSize", idx);
@@ -163,7 +163,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::EnableVADByIdxSafe(const WebRtc_Word16 idx) {
+int16_t ACMNetEQ::EnableVADByIdxSafe(const int16_t idx) {
   if (ptr_vadinst_[idx] == NULL) {
     if (WebRtcVad_Create(&ptr_vadinst_[idx]) < 0) {
       ptr_vadinst_[idx] = NULL;
@@ -196,15 +196,15 @@
   return 0;
 }
 
-WebRtc_Word32 ACMNetEQ::AllocatePacketBuffer(
+int32_t ACMNetEQ::AllocatePacketBuffer(
     const WebRtcNetEQDecoder* used_codecs,
-    WebRtc_Word16 num_codecs) {
+    int16_t num_codecs) {
   // Due to WebRtcNetEQ_GetRecommendedBufferSize
   // the following has to be int otherwise we will have compiler error
   // if not casted
 
   CriticalSectionScoped lock(neteq_crit_sect_);
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (AllocatePacketBufferByIdxSafe(used_codecs, num_codecs, idx) < 0) {
       return -1;
     }
@@ -212,10 +212,10 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::AllocatePacketBufferByIdxSafe(
+int16_t ACMNetEQ::AllocatePacketBufferByIdxSafe(
     const WebRtcNetEQDecoder* used_codecs,
-    WebRtc_Word16 num_codecs,
-    const WebRtc_Word16 idx) {
+    int16_t num_codecs,
+    const int16_t idx) {
   int max_num_packets;
   int buffer_size_in_bytes;
   int per_packet_overhead_bytes;
@@ -249,7 +249,7 @@
     neteq_packet_buffer_[idx] = NULL;
   }
 
-  neteq_packet_buffer_[idx] = (WebRtc_Word16 *) malloc(buffer_size_in_bytes);
+  neteq_packet_buffer_[idx] = (int16_t *) malloc(buffer_size_in_bytes);
   if (neteq_packet_buffer_[idx] == NULL) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "AllocatePacketBufferByIdxSafe: NetEq Initialization error: "
@@ -269,10 +269,10 @@
   return 0;
 }
 
-WebRtc_Word32 ACMNetEQ::SetExtraDelay(const WebRtc_Word32 delay_in_ms) {
+int32_t ACMNetEQ::SetExtraDelay(const int32_t delay_in_ms) {
   CriticalSectionScoped lock(neteq_crit_sect_);
 
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (!is_initialized_[idx]) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "SetExtraDelay: NetEq is not initialized.");
@@ -287,10 +287,10 @@
   return 0;
 }
 
-WebRtc_Word32 ACMNetEQ::SetAVTPlayout(const bool enable) {
+int32_t ACMNetEQ::SetAVTPlayout(const bool enable) {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (avt_playout_ != enable) {
-    for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+    for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
       if (!is_initialized_[idx]) {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                      "SetAVTPlayout: NetEq is not initialized.");
@@ -311,17 +311,17 @@
   return avt_playout_;
 }
 
-WebRtc_Word32 ACMNetEQ::CurrentSampFreqHz() const {
+int32_t ACMNetEQ::CurrentSampFreqHz() const {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (!is_initialized_[0]) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "CurrentSampFreqHz: NetEq is not initialized.");
     return -1;
   }
-  return (WebRtc_Word32)(1000 * current_samp_freq_khz_);
+  return (int32_t)(1000 * current_samp_freq_khz_);
 }
 
-WebRtc_Word32 ACMNetEQ::SetPlayoutMode(const AudioPlayoutMode mode) {
+int32_t ACMNetEQ::SetPlayoutMode(const AudioPlayoutMode mode) {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (playout_mode_ == mode)
     return 0;
@@ -375,7 +375,7 @@
   return playout_mode_;
 }
 
-WebRtc_Word32 ACMNetEQ::NetworkStatistics(
+int32_t ACMNetEQ::NetworkStatistics(
     ACMNetworkStatistics* statistics) const {
   WebRtcNetEQ_NetworkStatistics stats;
   CriticalSectionScoped lock(neteq_crit_sect_);
@@ -435,10 +435,10 @@
   return 0;
 }
 
-WebRtc_Word32 ACMNetEQ::RecIn(const WebRtc_UWord8* incoming_payload,
-                              const WebRtc_Word32 length_payload,
-                              const WebRtcRTPHeader& rtp_info) {
-  WebRtc_Word16 payload_length = static_cast<WebRtc_Word16>(length_payload);
+int32_t ACMNetEQ::RecIn(const uint8_t* incoming_payload,
+                        const int32_t length_payload,
+                        const WebRtcRTPHeader& rtp_info) {
+  int16_t payload_length = static_cast<int16_t>(length_payload);
 
   // translate to NetEq struct
   WebRtcNetEQ_RTPInfo neteq_rtpinfo;
@@ -453,10 +453,10 @@
   // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
   // we masked 6 most significant bits of 32-bit so we don't loose resolution
   // when do the following multiplication.
-  const WebRtc_UWord32 now_in_ms =
-      static_cast<WebRtc_UWord32>(
+  const uint32_t now_in_ms =
+      static_cast<uint32_t>(
           TickTime::MillisecondTimestamp() & 0x03ffffff);
-  WebRtc_UWord32 recv_timestamp = static_cast<WebRtc_UWord32>(
+  uint32_t recv_timestamp = static_cast<uint32_t>(
       current_samp_freq_khz_ * now_in_ms);
 
   int status;
@@ -505,13 +505,13 @@
   return 0;
 }
 
-WebRtc_Word32 ACMNetEQ::RecOut(AudioFrame& audio_frame) {
+int32_t ACMNetEQ::RecOut(AudioFrame& audio_frame) {
   enum WebRtcNetEQOutputType type;
-  WebRtc_Word16 payload_len_sample;
+  int16_t payload_len_sample;
   enum WebRtcNetEQOutputType type_master;
   enum WebRtcNetEQOutputType type_slave;
 
-  WebRtc_Word16 payload_len_sample_slave;
+  int16_t payload_len_sample_slave;
 
   CriticalSectionScoped lockNetEq(neteq_crit_sect_);
 
@@ -546,8 +546,8 @@
                    "RecOut: NetEq is not initialized.");
       return -1;
     }
-    WebRtc_Word16 payload_master[480];
-    WebRtc_Word16 payload_slave[480];
+    int16_t payload_master[480];
+    int16_t payload_slave[480];
     {
       WriteLockScoped lockCodec(*decode_lock_);
       if (WebRtcNetEQ_RecOutMasterSlave(inst_[0], payload_master,
@@ -590,11 +590,11 @@
       if (payload_len_sample > payload_len_sample_slave) {
         memset(&payload_slave[payload_len_sample_slave], 0,
                (payload_len_sample - payload_len_sample_slave) *
-               sizeof(WebRtc_Word16));
+               sizeof(int16_t));
       }
     }
 
-    for (WebRtc_Word16 n = 0; n < payload_len_sample; n++) {
+    for (int16_t n = 0; n < payload_len_sample; n++) {
       audio_frame.data_[n << 1] = payload_master[n];
       audio_frame.data_[(n << 1) + 1] = payload_slave[n];
     }
@@ -610,7 +610,7 @@
   }
 
   audio_frame.samples_per_channel_ =
-      static_cast<WebRtc_UWord16>(payload_len_sample);
+      static_cast<uint16_t>(payload_len_sample);
   // NetEq always returns 10 ms of audio.
   current_samp_freq_khz_ =
       static_cast<float>(audio_frame.samples_per_channel_) / 10.0f;
@@ -660,8 +660,8 @@
 
 // When ACMGenericCodec has set the codec specific parameters in codec_def
 // it calls AddCodec() to add the new codec to the NetEQ database.
-WebRtc_Word32 ACMNetEQ::AddCodec(WebRtcNetEQ_CodecDef* codec_def,
-                                 bool to_master) {
+int32_t ACMNetEQ::AddCodec(WebRtcNetEQ_CodecDef* codec_def,
+                           bool to_master) {
   if (codec_def == NULL) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "ACMNetEQ::AddCodec: error, codec_def is NULL");
@@ -669,7 +669,7 @@
   }
   CriticalSectionScoped lock(neteq_crit_sect_);
 
-  WebRtc_Word16 idx;
+  int16_t idx;
   if (to_master) {
     idx = 0;
   } else {
@@ -693,11 +693,11 @@
 
 // Creates a Word16 RTP packet out of a Word8 payload and an rtp info struct.
 // Must be byte order safe.
-void ACMNetEQ::RTPPack(WebRtc_Word16* rtp_packet, const WebRtc_Word8* payload,
-                       const WebRtc_Word32 payload_length_bytes,
+void ACMNetEQ::RTPPack(int16_t* rtp_packet, const int8_t* payload,
+                       const int32_t payload_length_bytes,
                        const WebRtcRTPHeader& rtp_info) {
-  WebRtc_Word32 idx = 0;
-  WEBRTC_SPL_SET_BYTE(rtp_packet, (WebRtc_Word8) 0x80, idx);
+  int32_t idx = 0;
+  WEBRTC_SPL_SET_BYTE(rtp_packet, (int8_t) 0x80, idx);
   idx++;
   WEBRTC_SPL_SET_BYTE(rtp_packet, rtp_info.header.payloadType, idx);
   idx++;
@@ -737,7 +737,7 @@
   WEBRTC_SPL_SET_BYTE(rtp_packet, WEBRTC_SPL_GET_BYTE(&(rtp_info.header.ssrc),
                                                       0), idx);
   idx++;
-  for (WebRtc_Word16 i = 0; i < payload_length_bytes; i++) {
+  for (int16_t i = 0; i < payload_length_bytes; i++) {
     WEBRTC_SPL_SET_BYTE(rtp_packet, payload[i], idx);
     idx++;
   }
@@ -748,12 +748,12 @@
   }
 }
 
-WebRtc_Word16 ACMNetEQ::EnableVAD() {
+int16_t ACMNetEQ::EnableVAD() {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (vad_status_) {
     return 0;
   }
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (!is_initialized_[idx]) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "SetVADStatus: NetEq is not initialized.");
@@ -776,7 +776,7 @@
   return vad_mode_;
 }
 
-WebRtc_Word16 ACMNetEQ::SetVADMode(const ACMVADMode mode) {
+int16_t ACMNetEQ::SetVADMode(const ACMVADMode mode) {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if ((mode < VADNormal) || (mode > VADVeryAggr)) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -784,7 +784,7 @@
                  "supported");
     return -1;
   } else {
-    for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+    for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
       if (!is_initialized_[idx]) {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                      "SetVADMode: NetEq is not initialized.");
@@ -800,9 +800,9 @@
   }
 }
 
-WebRtc_Word32 ACMNetEQ::FlushBuffers() {
+int32_t ACMNetEQ::FlushBuffers() {
   CriticalSectionScoped lock(neteq_crit_sect_);
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (!is_initialized_[idx]) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "FlushBuffers: NetEq is not initialized.");
@@ -816,8 +816,8 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::RemoveCodec(WebRtcNetEQDecoder codec_idx,
-                                    bool is_stereo) {
+int16_t ACMNetEQ::RemoveCodec(WebRtcNetEQDecoder codec_idx,
+                              bool is_stereo) {
   // sanity check
   if ((codec_idx <= kDecoderReservedStart) ||
       (codec_idx >= kDecoderReservedEnd)) {
@@ -848,10 +848,10 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::SetBackgroundNoiseMode(
+int16_t ACMNetEQ::SetBackgroundNoiseMode(
     const ACMBackgroundNoiseMode mode) {
   CriticalSectionScoped lock(neteq_crit_sect_);
-  for (WebRtc_Word16 idx = 0; idx < num_slaves_ + 1; idx++) {
+  for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) {
     if (!is_initialized_[idx]) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "SetBackgroundNoiseMode: NetEq is not initialized.");
@@ -865,7 +865,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMNetEQ::BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) {
+int16_t ACMNetEQ::BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) {
   WebRtcNetEQBGNMode my_mode;
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (!is_initialized_[0]) {
@@ -882,13 +882,13 @@
   return 0;
 }
 
-void ACMNetEQ::set_id(WebRtc_Word32 id) {
+void ACMNetEQ::set_id(int32_t id) {
   CriticalSectionScoped lock(neteq_crit_sect_);
   id_ = id;
 }
 
 void ACMNetEQ::LogError(const char* neteq_func_name,
-                        const WebRtc_Word16 idx) const {
+                        const int16_t idx) const {
   char error_name[NETEQ_ERR_MSG_LEN_BYTE];
   char my_func_name[50];
   int neteq_error_code = WebRtcNetEQ_GetErrorCode(inst_[idx]);
@@ -902,7 +902,7 @@
                " %s", idx, my_func_name, neteq_error_code, error_name);
 }
 
-WebRtc_Word32 ACMNetEQ::PlayoutTimestamp(WebRtc_UWord32& timestamp) {
+int32_t ACMNetEQ::PlayoutTimestamp(uint32_t& timestamp) {
   CriticalSectionScoped lock(neteq_crit_sect_);
   if (WebRtcNetEQ_GetSpeechTimeStamp(inst_[0], &timestamp) < 0) {
     LogError("GetSpeechTimeStamp", 0);
@@ -944,10 +944,10 @@
   }
 }
 
-WebRtc_Word16 ACMNetEQ::AddSlave(const WebRtcNetEQDecoder* used_codecs,
-                                 WebRtc_Word16 num_codecs) {
+int16_t ACMNetEQ::AddSlave(const WebRtcNetEQDecoder* used_codecs,
+                           int16_t num_codecs) {
   CriticalSectionScoped lock(neteq_crit_sect_);
-  const WebRtc_Word16 slave_idx = 1;
+  const int16_t slave_idx = 1;
   if (num_slaves_ < 1) {
     // initialize the receiver, this also sets up VAD.
     if (InitByIdxSafe(slave_idx) < 0) {
@@ -1050,7 +1050,7 @@
   received_stereo_ = received_stereo;
 }
 
-WebRtc_UWord8 ACMNetEQ::num_slaves() {
+uint8_t ACMNetEQ::num_slaves() {
   CriticalSectionScoped lock(neteq_crit_sect_);
   return num_slaves_;
 }
diff --git a/modules/audio_coding/main/source/acm_neteq.h b/modules/audio_coding/main/source/acm_neteq.h
index 06a1148..ac91f9f 100644
--- a/modules/audio_coding/main/source/acm_neteq.h
+++ b/modules/audio_coding/main/source/acm_neteq.h
@@ -48,7 +48,7 @@
   //                           -1 if NetEQ or VAD returned an error or
   //                           if out of memory.
   //
-  WebRtc_Word32 Init();
+  int32_t Init();
 
   //
   // RecIn()
@@ -64,9 +64,9 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 RecIn(const WebRtc_UWord8* incoming_payload,
-                      const WebRtc_Word32 length_payload,
-                      const WebRtcRTPHeader& rtp_info);
+  int32_t RecIn(const uint8_t* incoming_payload,
+                const int32_t length_payload,
+                const WebRtcRTPHeader& rtp_info);
 
   //
   // RecOut()
@@ -79,7 +79,7 @@
   // Return value             : 0 if ok.
   //                           -1 if NetEQ returned an error.
   //
-  WebRtc_Word32 RecOut(AudioFrame& audio_frame);
+  int32_t RecOut(AudioFrame& audio_frame);
 
   //
   // AddCodec()
@@ -94,8 +94,8 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 AddCodec(WebRtcNetEQ_CodecDef *codec_def,
-                         bool to_master = true);
+  int32_t AddCodec(WebRtcNetEQ_CodecDef *codec_def,
+                   bool to_master = true);
 
   //
   // AllocatePacketBuffer()
@@ -108,8 +108,8 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs,
-                                     WebRtc_Word16 num_codecs);
+  int32_t AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs,
+                               int16_t num_codecs);
 
   //
   // SetExtraDelay()
@@ -121,7 +121,7 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 SetExtraDelay(const WebRtc_Word32 delay_in_ms);
+  int32_t SetExtraDelay(const int32_t delay_in_ms);
 
   //
   // SetAVTPlayout()
@@ -133,7 +133,7 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 SetAVTPlayout(const bool enable);
+  int32_t SetAVTPlayout(const bool enable);
 
   //
   // AVTPlayout()
@@ -150,7 +150,7 @@
   //
   // Return value             : Sampling frequency in Hz.
   //
-  WebRtc_Word32 CurrentSampFreqHz() const;
+  int32_t CurrentSampFreqHz() const;
 
   //
   // SetPlayoutMode()
@@ -163,7 +163,7 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode);
+  int32_t SetPlayoutMode(const AudioPlayoutMode mode);
 
   //
   // PlayoutMode()
@@ -183,7 +183,7 @@
   // Return value             : 0 if ok.
   //                           <0 if NetEQ returned an error.
   //
-  WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics* statistics) const;
+  int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const;
 
   //
   // VADMode()
@@ -203,7 +203,7 @@
   // Return value             : 0 if ok.
   //                           -1 if an error occurred.
   //
-  WebRtc_Word16 SetVADMode(const ACMVADMode mode);
+  int16_t SetVADMode(const ACMVADMode mode);
 
   //
   // DecodeLock()
@@ -222,7 +222,7 @@
   // Return value             : 0 if ok.
   //                           -1 if NetEQ returned an error.
   //
-  WebRtc_Word32 FlushBuffers();
+  int32_t FlushBuffers();
 
   //
   // RemoveCodec()
@@ -234,8 +234,8 @@
   // Return value             : 0 if ok.
   //                           -1 if an error occurred.
   //
-  WebRtc_Word16 RemoveCodec(WebRtcNetEQDecoder codec_idx,
-                            bool is_stereo = false);
+  int16_t RemoveCodec(WebRtcNetEQDecoder codec_idx,
+                      bool is_stereo = false);
 
   //
   // SetBackgroundNoiseMode()
@@ -248,7 +248,7 @@
   // Return value             : 0 if succeeded,
   //                           -1 if failed to set the mode.
   //
-  WebRtc_Word16 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
+  int16_t SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
 
   //
   // BackgroundNoiseMode()
@@ -256,21 +256,21 @@
   //
   // Return value             : The mode of background noise.
   //
-  WebRtc_Word16 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
+  int16_t BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
 
-  void set_id(WebRtc_Word32 id);
+  void set_id(int32_t id);
 
-  WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp);
+  int32_t PlayoutTimestamp(uint32_t& timestamp);
 
   void set_received_stereo(bool received_stereo);
 
-  WebRtc_UWord8 num_slaves();
+  uint8_t num_slaves();
 
   // Delete all slaves.
   void RemoveSlaves();
 
-  WebRtc_Word16 AddSlave(const WebRtcNetEQDecoder* used_codecs,
-                         WebRtc_Word16 num_codecs);
+  int16_t AddSlave(const WebRtcNetEQDecoder* used_codecs,
+                   int16_t num_codecs);
 
   void BufferSpec(int& num_packets, int& size_bytes, int& overhead_bytes) {
     num_packets = min_of_max_num_packets_;
@@ -292,13 +292,13 @@
   // Output:
   //   - rtp_packet           : The RTP packet.
   //
-  static void RTPPack(WebRtc_Word16* rtp_packet, const WebRtc_Word8* payload,
-                      const WebRtc_Word32 payload_length_bytes,
+  static void RTPPack(int16_t* rtp_packet, const int8_t* payload,
+                      const int32_t payload_length_bytes,
                       const WebRtcRTPHeader& rtp_info);
 
-  void LogError(const char* neteq_func_name, const WebRtc_Word16 idx) const;
+  void LogError(const char* neteq_func_name, const int16_t idx) const;
 
-  WebRtc_Word16 InitByIdxSafe(const WebRtc_Word16 idx);
+  int16_t InitByIdxSafe(const int16_t idx);
 
   //
   // EnableVAD()
@@ -307,14 +307,14 @@
   // Return value             : 0 if ok.
   //                           -1 if an error occurred.
   //
-  WebRtc_Word16 EnableVAD();
+  int16_t EnableVAD();
 
-  WebRtc_Word16 EnableVADByIdxSafe(const WebRtc_Word16 idx);
+  int16_t EnableVADByIdxSafe(const int16_t idx);
 
-  WebRtc_Word16 AllocatePacketBufferByIdxSafe(
+  int16_t AllocatePacketBufferByIdxSafe(
       const WebRtcNetEQDecoder* used_codecs,
-      WebRtc_Word16 num_codecs,
-      const WebRtc_Word16 idx);
+      int16_t num_codecs,
+      const int16_t idx);
 
   // Delete the NetEQ corresponding to |index|.
   void RemoveNetEQSafe(int index);
@@ -324,9 +324,9 @@
   void* inst_[MAX_NUM_SLAVE_NETEQ + 1];
   void* inst_mem_[MAX_NUM_SLAVE_NETEQ + 1];
 
-  WebRtc_Word16* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1];
+  int16_t* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1];
 
-  WebRtc_Word32 id_;
+  int32_t id_;
   float current_samp_freq_khz_;
   bool avt_playout_;
   AudioPlayoutMode playout_mode_;
@@ -338,11 +338,11 @@
   ACMVADMode vad_mode_;
   RWLockWrapper* decode_lock_;
   bool is_initialized_[MAX_NUM_SLAVE_NETEQ + 1];
-  WebRtc_UWord8 num_slaves_;
+  uint8_t num_slaves_;
   bool received_stereo_;
   void* master_slave_info_;
   AudioFrame::VADActivity previous_audio_activity_;
-  WebRtc_Word32 extra_delay_;
+  int32_t extra_delay_;
 
   CriticalSectionWrapper* callback_crit_sect_;
   // Minimum of "max number of packets," among all NetEq instances.
diff --git a/modules/audio_coding/main/source/acm_neteq_unittest.cc b/modules/audio_coding/main/source/acm_neteq_unittest.cc
index d7f3f83..9cf280b 100644
--- a/modules/audio_coding/main/source/acm_neteq_unittest.cc
+++ b/modules/audio_coding/main/source/acm_neteq_unittest.cc
@@ -68,7 +68,7 @@
   rtp_header.header.payloadType = payload_type;
   rtp_header.header.markerBit = marker_bit;
   rtp_header.type.Audio.channel = 1;
-  ASSERT_EQ(0, neteq_.RecIn(reinterpret_cast<WebRtc_UWord8*>(payload),
+  ASSERT_EQ(0, neteq_.RecIn(reinterpret_cast<uint8_t*>(payload),
                             len_payload_bytes, rtp_header));
 }
 
diff --git a/modules/audio_coding/main/source/acm_pcm16b.cc b/modules/audio_coding/main/source/acm_pcm16b.cc
index f341295..91cb9e0 100644
--- a/modules/audio_coding/main/source/acm_pcm16b.cc
+++ b/modules/audio_coding/main/source/acm_pcm16b.cc
@@ -25,7 +25,7 @@
 
 #ifndef WEBRTC_CODEC_PCM16
 
-ACMPCM16B::ACMPCM16B(WebRtc_Word16 /* codec_id */) {
+ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) {
   return;
 }
 
@@ -33,32 +33,32 @@
   return;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMPCM16B::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMPCM16B::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMPCM16B::DecodeSafe(uint8_t* /* bitstream */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalInitEncoder(
+int16_t ACMPCM16B::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalInitDecoder(
+int16_t ACMPCM16B::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                  const CodecInst& /* codec_inst */) {
+int32_t ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                            const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -66,11 +66,11 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalCreateEncoder() {
+int16_t ACMPCM16B::InternalCreateEncoder() {
   return -1;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalCreateDecoder() {
+int16_t ACMPCM16B::InternalCreateDecoder() {
   return -1;
 }
 
@@ -91,7 +91,7 @@
 }
 
 #else     //===================== Actual Implementation =======================
-ACMPCM16B::ACMPCM16B(WebRtc_Word16 codec_id) {
+ACMPCM16B::ACMPCM16B(int16_t codec_id) {
   codec_id_ = codec_id;
   sampling_freq_hz_ = ACMCodecDB::CodecFreq(codec_id_);
 }
@@ -100,8 +100,8 @@
   return;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalEncode(WebRtc_UWord8* bitstream,
-                                        WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMPCM16B::InternalEncode(uint8_t* bitstream,
+                                  int16_t* bitstream_len_byte) {
   *bitstream_len_byte = WebRtcPcm16b_Encode(&in_audio_[in_audio_ix_read_],
                                             frame_len_smpl_ * num_channels_,
                                             bitstream);
@@ -111,28 +111,28 @@
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMPCM16B::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                    WebRtc_Word16 /* bitstream_len_byte */,
-                                    WebRtc_Word16* /* audio */,
-                                    WebRtc_Word16* /* audio_samples */,
-                                    WebRtc_Word8* /* speech_type */) {
+int16_t ACMPCM16B::DecodeSafe(uint8_t* /* bitstream */,
+                              int16_t /* bitstream_len_byte */,
+                              int16_t* /* audio */,
+                              int16_t* /* audio_samples */,
+                              int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalInitEncoder(
+int16_t ACMPCM16B::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalInitDecoder(
+int16_t ACMPCM16B::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word32 ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                  const CodecInst& codec_inst) {
+int32_t ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                            const CodecInst& codec_inst) {
   // Fill up the structure by calling "SET_CODEC_PAR" & "SET_PCMU_FUNCTION".
   // Then call NetEQ to add the codec to it's database.
   if (codec_inst.channels == 1) {
@@ -190,12 +190,12 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalCreateEncoder() {
+int16_t ACMPCM16B::InternalCreateEncoder() {
   // PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCM16B::InternalCreateDecoder() {
+int16_t ACMPCM16B::InternalCreateDecoder() {
   // PCM has no instance.
   return 0;
 }
diff --git a/modules/audio_coding/main/source/acm_pcm16b.h b/modules/audio_coding/main/source/acm_pcm16b.h
index 314e3bb..5368fe7 100644
--- a/modules/audio_coding/main/source/acm_pcm16b.h
+++ b/modules/audio_coding/main/source/acm_pcm16b.h
@@ -17,42 +17,42 @@
 
 class ACMPCM16B : public ACMGenericCodec {
  public:
-  explicit ACMPCM16B(WebRtc_Word16 codec_id);
+  explicit ACMPCM16B(int16_t codec_id);
   ~ACMPCM16B();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
   void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
 
-  WebRtc_Word32 sampling_freq_hz_;
+  int32_t sampling_freq_hz_;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/main/source/acm_pcma.cc b/modules/audio_coding/main/source/acm_pcma.cc
index 24a6fe2..83c1249 100644
--- a/modules/audio_coding/main/source/acm_pcma.cc
+++ b/modules/audio_coding/main/source/acm_pcma.cc
@@ -21,7 +21,7 @@
 
 namespace webrtc {
 
-ACMPCMA::ACMPCMA(WebRtc_Word16 codec_id) {
+ACMPCMA::ACMPCMA(int16_t codec_id) {
   codec_id_ = codec_id;
 }
 
@@ -29,39 +29,39 @@
   return;
 }
 
-WebRtc_Word16 ACMPCMA::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMPCMA::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
   *bitstream_len_byte = WebRtcG711_EncodeA(NULL, &in_audio_[in_audio_ix_read_],
                                            frame_len_smpl_ * num_channels_,
-                                           (WebRtc_Word16*) bitstream);
+                                           (int16_t*) bitstream);
   // Increment the read index this tell the caller that how far
   // we have gone forward in reading the audio buffer.
   in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMPCMA::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMPCMA::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMPCMA::InternalInitEncoder(
+int16_t ACMPCMA::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCMA::InternalInitDecoder(
+int16_t ACMPCMA::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word32 ACMPCMA::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMPCMA::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   // Fill up the structure by calling
   // "SET_CODEC_PAR" & "SET_PCMA_FUNCTION."
   // Then call NetEQ to add the codec to it's database.
@@ -80,12 +80,12 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMPCMA::InternalCreateEncoder() {
+int16_t ACMPCMA::InternalCreateEncoder() {
   // PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCMA::InternalCreateDecoder() {
+int16_t ACMPCMA::InternalCreateDecoder() {
   // PCM has no instance.
   return 0;
 }
diff --git a/modules/audio_coding/main/source/acm_pcma.h b/modules/audio_coding/main/source/acm_pcma.h
index cf12dca..59c3389 100644
--- a/modules/audio_coding/main/source/acm_pcma.h
+++ b/modules/audio_coding/main/source/acm_pcma.h
@@ -17,36 +17,36 @@
 
 class ACMPCMA : public ACMGenericCodec {
  public:
-  explicit ACMPCMA(WebRtc_Word16 codec_id);
+  explicit ACMPCMA(int16_t codec_id);
   ~ACMPCMA();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
diff --git a/modules/audio_coding/main/source/acm_pcmu.cc b/modules/audio_coding/main/source/acm_pcmu.cc
index 919826d..61a64ac 100644
--- a/modules/audio_coding/main/source/acm_pcmu.cc
+++ b/modules/audio_coding/main/source/acm_pcmu.cc
@@ -21,7 +21,7 @@
 
 namespace webrtc {
 
-ACMPCMU::ACMPCMU(WebRtc_Word16 codec_id) {
+ACMPCMU::ACMPCMU(int16_t codec_id) {
   codec_id_ = codec_id;
 }
 
@@ -29,39 +29,39 @@
   return;
 }
 
-WebRtc_Word16 ACMPCMU::InternalEncode(WebRtc_UWord8* bitstream,
-                                      WebRtc_Word16* bitstream_len_byte) {
+int16_t ACMPCMU::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
   *bitstream_len_byte = WebRtcG711_EncodeU(NULL, &in_audio_[in_audio_ix_read_],
                                            frame_len_smpl_ * num_channels_,
-                                           (WebRtc_Word16*)bitstream);
+                                           (int16_t*)bitstream);
   // Increment the read index this tell the caller that how far
   // we have gone forward in reading the audio buffer.
   in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMPCMU::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                  WebRtc_Word16 /* bitstream_len_byte */,
-                                  WebRtc_Word16* /* audio */,
-                                  WebRtc_Word16* /* audio_samples */,
-                                  WebRtc_Word8* /* speech_type */) {
+int16_t ACMPCMU::DecodeSafe(uint8_t* /* bitstream */,
+                            int16_t /* bitstream_len_byte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audio_samples */,
+                            int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMPCMU::InternalInitEncoder(
+int16_t ACMPCMU::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCMU::InternalInitDecoder(
+int16_t ACMPCMU::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization, PCM has no instance.
   return 0;
 }
 
-WebRtc_Word32 ACMPCMU::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                const CodecInst& codec_inst) {
+int32_t ACMPCMU::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                          const CodecInst& codec_inst) {
   // Fill up the structure by calling
   // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
   // Then call NetEQ to add the codec to it's database.
@@ -80,12 +80,12 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMPCMU::InternalCreateEncoder() {
+int16_t ACMPCMU::InternalCreateEncoder() {
   // PCM has no instance.
   return 0;
 }
 
-WebRtc_Word16 ACMPCMU::InternalCreateDecoder() {
+int16_t ACMPCMU::InternalCreateDecoder() {
   // PCM has no instance.
   return 0;
 }
diff --git a/modules/audio_coding/main/source/acm_pcmu.h b/modules/audio_coding/main/source/acm_pcmu.h
index f75e7aa..5169f13 100644
--- a/modules/audio_coding/main/source/acm_pcmu.h
+++ b/modules/audio_coding/main/source/acm_pcmu.h
@@ -17,36 +17,36 @@
 
 class ACMPCMU : public ACMGenericCodec {
  public:
-  explicit ACMPCMU(WebRtc_Word16 codec_id);
+  explicit ACMPCMU(int16_t codec_id);
   ~ACMPCMU();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
diff --git a/modules/audio_coding/main/source/acm_red.cc b/modules/audio_coding/main/source/acm_red.cc
index a249449..6e7ae9f 100644
--- a/modules/audio_coding/main/source/acm_red.cc
+++ b/modules/audio_coding/main/source/acm_red.cc
@@ -18,7 +18,7 @@
 
 namespace webrtc {
 
-ACMRED::ACMRED(WebRtc_Word16 codec_id) {
+ACMRED::ACMRED(int16_t codec_id) {
   codec_id_ = codec_id;
 }
 
@@ -26,37 +26,37 @@
   return;
 }
 
-WebRtc_Word16 ACMRED::InternalEncode(WebRtc_UWord8* /* bitstream */,
-                                     WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMRED::InternalEncode(uint8_t* /* bitstream */,
+                               int16_t* /* bitstream_len_byte */) {
   // RED is never used as an encoder
   // RED has no instance
   return 0;
 }
 
-WebRtc_Word16 ACMRED::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                 WebRtc_Word16 /* bitstream_len_byte */,
-                                 WebRtc_Word16* /* audio */,
-                                 WebRtc_Word16* /* audio_samples */,
-                                 WebRtc_Word8* /* speech_type */) {
+int16_t ACMRED::DecodeSafe(uint8_t* /* bitstream */,
+                           int16_t /* bitstream_len_byte */,
+                           int16_t* /* audio */,
+                           int16_t* /* audio_samples */,
+                           int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMRED::InternalInitEncoder(
+int16_t ACMRED::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization,
   // RED has no instance
   return 0;
 }
 
-WebRtc_Word16 ACMRED::InternalInitDecoder(
+int16_t ACMRED::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   // This codec does not need initialization,
   // RED has no instance
   return 0;
 }
 
-WebRtc_Word32 ACMRED::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                               const CodecInst& codec_inst) {
+int32_t ACMRED::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                         const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     // Todo:
     // log error
@@ -76,12 +76,12 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMRED::InternalCreateEncoder() {
+int16_t ACMRED::InternalCreateEncoder() {
   // RED has no instance
   return 0;
 }
 
-WebRtc_Word16 ACMRED::InternalCreateDecoder() {
+int16_t ACMRED::InternalCreateDecoder() {
   // RED has no instance
   return 0;
 }
diff --git a/modules/audio_coding/main/source/acm_red.h b/modules/audio_coding/main/source/acm_red.h
index 5f4cb9b..e19e335 100644
--- a/modules/audio_coding/main/source/acm_red.h
+++ b/modules/audio_coding/main/source/acm_red.h
@@ -17,36 +17,36 @@
 
 class ACMRED : public ACMGenericCodec {
  public:
-  explicit ACMRED(WebRtc_Word16 codec_id);
+  explicit ACMRED(int16_t codec_id);
   ~ACMRED();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 };
diff --git a/modules/audio_coding/main/source/acm_resampler.cc b/modules/audio_coding/main/source/acm_resampler.cc
index a776052..2618649 100644
--- a/modules/audio_coding/main/source/acm_resampler.cc
+++ b/modules/audio_coding/main/source/acm_resampler.cc
@@ -27,25 +27,25 @@
   delete resampler_crit_sect_;
 }
 
-WebRtc_Word16 ACMResampler::Resample10Msec(const WebRtc_Word16* in_audio,
-                                           WebRtc_Word32 in_freq_hz,
-                                           WebRtc_Word16* out_audio,
-                                           WebRtc_Word32 out_freq_hz,
-                                           WebRtc_UWord8 num_audio_channels) {
+int16_t ACMResampler::Resample10Msec(const int16_t* in_audio,
+                                     int32_t in_freq_hz,
+                                     int16_t* out_audio,
+                                     int32_t out_freq_hz,
+                                     uint8_t num_audio_channels) {
   CriticalSectionScoped cs(resampler_crit_sect_);
 
   if (in_freq_hz == out_freq_hz) {
     size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
-    memcpy(out_audio, in_audio, length * sizeof(WebRtc_Word16));
-    return static_cast<WebRtc_Word16>(in_freq_hz / 100);
+    memcpy(out_audio, in_audio, length * sizeof(int16_t));
+    return static_cast<int16_t>(in_freq_hz / 100);
   }
 
   // |maxLen| is maximum number of samples for 10ms at 48kHz.
   int max_len = 480 * num_audio_channels;
-  int length_in = (WebRtc_Word16)(in_freq_hz / 100) * num_audio_channels;
+  int length_in = (int16_t)(in_freq_hz / 100) * num_audio_channels;
   int out_len;
 
-  WebRtc_Word32 ret;
+  int32_t ret;
   ResamplerType type;
   type = (num_audio_channels == 1) ? kResamplerSynchronous :
       kResamplerSynchronousStereo;
@@ -64,7 +64,7 @@
     return -1;
   }
 
-  WebRtc_Word16 out_audio_len_smpl = (WebRtc_Word16) out_len /
+  int16_t out_audio_len_smpl = (int16_t) out_len /
       num_audio_channels;
 
   return out_audio_len_smpl;
diff --git a/modules/audio_coding/main/source/acm_resampler.h b/modules/audio_coding/main/source/acm_resampler.h
index aacd222..ddb0094 100644
--- a/modules/audio_coding/main/source/acm_resampler.h
+++ b/modules/audio_coding/main/source/acm_resampler.h
@@ -23,11 +23,11 @@
   ACMResampler();
   ~ACMResampler();
 
-  WebRtc_Word16 Resample10Msec(const WebRtc_Word16* in_audio,
-                               const WebRtc_Word32 in_freq_hz,
-                               WebRtc_Word16* out_audio,
-                               const WebRtc_Word32 out_freq_hz,
-                               WebRtc_UWord8 num_audio_channels);
+  int16_t Resample10Msec(const int16_t* in_audio,
+                         const int32_t in_freq_hz,
+                         int16_t* out_audio,
+                         const int32_t out_freq_hz,
+                         uint8_t num_audio_channels);
 
  private:
   // Use the Resampler class.
diff --git a/modules/audio_coding/main/source/acm_speex.cc b/modules/audio_coding/main/source/acm_speex.cc
index 73ab7fc..ce20526 100644
--- a/modules/audio_coding/main/source/acm_speex.cc
+++ b/modules/audio_coding/main/source/acm_speex.cc
@@ -26,7 +26,7 @@
 namespace webrtc {
 
 #ifndef WEBRTC_CODEC_SPEEX
-ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codec_id */)
+ACMSPEEX::ACMSPEEX(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       compl_mode_(0),
@@ -41,40 +41,40 @@
   return;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalEncode(
-    WebRtc_UWord8* /* bitstream */,
-    WebRtc_Word16* /* bitstream_len_byte */) {
+int16_t ACMSPEEX::InternalEncode(
+    uint8_t* /* bitstream */,
+    int16_t* /* bitstream_len_byte */) {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMSPEEX::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::EnableDTX() {
+int16_t ACMSPEEX::EnableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::DisableDTX() {
+int16_t ACMSPEEX::DisableDTX() {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalInitEncoder(
+int16_t ACMSPEEX::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalInitDecoder(
+int16_t ACMSPEEX::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }
 
-WebRtc_Word32 ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
-                                 const CodecInst& /* codec_inst */) {
+int32_t ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
+                           const CodecInst& /* codec_inst */) {
   return -1;
 }
 
@@ -82,7 +82,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalCreateEncoder() {
+int16_t ACMSPEEX::InternalCreateEncoder() {
   return -1;
 }
 
@@ -90,7 +90,7 @@
   return;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalCreateDecoder() {
+int16_t ACMSPEEX::InternalCreateDecoder() {
   return -1;
 }
 
@@ -98,7 +98,7 @@
   return;
 }
 
-WebRtc_Word16 ACMSPEEX::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+int16_t ACMSPEEX::SetBitRateSafe(const int32_t /* rate */) {
   return -1;
 }
 
@@ -107,22 +107,22 @@
 }
 
 #ifdef UNUSEDSPEEX
-WebRtc_Word16 ACMSPEEX::EnableVBR() {
+int16_t ACMSPEEX::EnableVBR() {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::DisableVBR() {
+int16_t ACMSPEEX::DisableVBR() {
   return -1;
 }
 
-WebRtc_Word16 ACMSPEEX::SetComplMode(WebRtc_Word16 mode) {
+int16_t ACMSPEEX::SetComplMode(int16_t mode) {
   return -1;
 }
 #endif
 
 #else  //===================== Actual Implementation =======================
 
-ACMSPEEX::ACMSPEEX(WebRtc_Word16 codec_id)
+ACMSPEEX::ACMSPEEX(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL) {
   codec_id_ = codec_id;
@@ -165,11 +165,11 @@
   return;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalEncode(WebRtc_UWord8* bitstream,
-                                       WebRtc_Word16* bitstream_len_byte) {
-  WebRtc_Word16 status;
-  WebRtc_Word16 num_encoded_samples = 0;
-  WebRtc_Word16 n = 0;
+int16_t ACMSPEEX::InternalEncode(uint8_t* bitstream,
+                                 int16_t* bitstream_len_byte) {
+  int16_t status;
+  int16_t num_encoded_samples = 0;
+  int16_t n = 0;
 
   while (num_encoded_samples < frame_len_smpl_) {
     status = WebRtcSpeex_Encode(encoder_inst_ptr_,
@@ -196,25 +196,25 @@
       // This frame is detected as inactive. We need send whatever
       // encoded so far.
       *bitstream_len_byte = WebRtcSpeex_GetBitstream(encoder_inst_ptr_,
-                                                     (WebRtc_Word16*)bitstream);
+                                                     (int16_t*)bitstream);
       return *bitstream_len_byte;
     }
   }
 
   *bitstream_len_byte = WebRtcSpeex_GetBitstream(encoder_inst_ptr_,
-                                                 (WebRtc_Word16*)bitstream);
+                                                 (int16_t*)bitstream);
   return *bitstream_len_byte;
 }
 
-WebRtc_Word16 ACMSPEEX::DecodeSafe(WebRtc_UWord8* /* bitstream */,
-                                   WebRtc_Word16 /* bitstream_len_byte */,
-                                   WebRtc_Word16* /* audio */,
-                                   WebRtc_Word16* /* audio_samples */,
-                                   WebRtc_Word8* /* speech_type */) {
+int16_t ACMSPEEX::DecodeSafe(uint8_t* /* bitstream */,
+                             int16_t /* bitstream_len_byte */,
+                             int16_t* /* audio */,
+                             int16_t* /* audio_samples */,
+                             int8_t* /* speech_type */) {
   return 0;
 }
 
-WebRtc_Word16 ACMSPEEX::EnableDTX() {
+int16_t ACMSPEEX::EnableDTX() {
   if (dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -234,7 +234,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMSPEEX::DisableDTX() {
+int16_t ACMSPEEX::DisableDTX() {
   if (!dtx_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -255,7 +255,7 @@
   return 0;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalInitEncoder(
+int16_t ACMSPEEX::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
   // sanity check
   if (encoder_inst_ptr_ == NULL) {
@@ -264,7 +264,7 @@
     return -1;
   }
 
-  WebRtc_Word16 status = SetBitRateSafe((codec_params->codecInstant).rate);
+  int16_t status = SetBitRateSafe((codec_params->codecInstant).rate);
   status +=
       (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, vbr_enabled_, compl_mode_,
                                ((codec_params->enable_dtx) ? 1 : 0)) < 0) ?
@@ -279,9 +279,9 @@
   }
 }
 
-WebRtc_Word16 ACMSPEEX::InternalInitDecoder(
+int16_t ACMSPEEX::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
-  WebRtc_Word16 status;
+  int16_t status;
 
   // sanity check
   if (decoder_inst_ptr_ == NULL) {
@@ -300,8 +300,8 @@
   }
 }
 
-WebRtc_Word32 ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                                 const CodecInst& codec_inst) {
+int32_t ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                           const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "Error, Speex decoder is not initialized");
@@ -340,7 +340,7 @@
   return NULL;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalCreateEncoder() {
+int16_t ACMSPEEX::InternalCreateEncoder() {
   return WebRtcSpeex_CreateEnc(&encoder_inst_ptr_, sampling_frequency_);
 }
 
@@ -355,7 +355,7 @@
   encoding_rate_ = 0;
 }
 
-WebRtc_Word16 ACMSPEEX::InternalCreateDecoder() {
+int16_t ACMSPEEX::InternalCreateDecoder() {
   return WebRtcSpeex_CreateDec(&decoder_inst_ptr_, sampling_frequency_, 1);
 }
 
@@ -369,7 +369,7 @@
   decoder_initialized_ = false;
 }
 
-WebRtc_Word16 ACMSPEEX::SetBitRateSafe(const WebRtc_Word32 rate) {
+int16_t ACMSPEEX::SetBitRateSafe(const int32_t rate) {
   // Check if changed rate
   if (rate == encoding_rate_) {
     return 0;
@@ -397,7 +397,7 @@
 
 // This API is currently not in use. If requested to be able to enable/disable
 // VBR an ACM API need to be added.
-WebRtc_Word16 ACMSPEEX::EnableVBR() {
+int16_t ACMSPEEX::EnableVBR() {
   if (vbr_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -418,7 +418,7 @@
 
 // This API is currently not in use. If requested to be able to enable/disable
 // VBR an ACM API need to be added.
-WebRtc_Word16 ACMSPEEX::DisableVBR() {
+int16_t ACMSPEEX::DisableVBR() {
   if (!vbr_enabled_) {
     return 0;
   } else if (encoder_exist_) {  // check if encoder exist
@@ -440,7 +440,7 @@
 
 // This API is currently not in use. If requested to be able to set complexity
 // an ACM API need to be added.
-WebRtc_Word16 ACMSPEEX::SetComplMode(WebRtc_Word16 mode) {
+int16_t ACMSPEEX::SetComplMode(int16_t mode) {
   // Check if new mode
   if (mode == compl_mode_) {
     return 0;
diff --git a/modules/audio_coding/main/source/acm_speex.h b/modules/audio_coding/main/source/acm_speex.h
index d1cd840..0f62ea3 100644
--- a/modules/audio_coding/main/source/acm_speex.h
+++ b/modules/audio_coding/main/source/acm_speex.h
@@ -21,60 +21,60 @@
 
 class ACMSPEEX : public ACMGenericCodec {
  public:
-  explicit ACMSPEEX(WebRtc_Word16 codec_id);
+  explicit ACMSPEEX(int16_t codec_id);
   ~ACMSPEEX();
 
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
-                               WebRtc_Word16* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte);
 
-  WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
-  WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
-                           WebRtc_Word16 bitstream_len_byte,
-                           WebRtc_Word16* audio,
-                           WebRtc_Word16* audio_samples,
-                           WebRtc_Word8* speech_type);
+  int16_t DecodeSafe(uint8_t* bitstream,
+                     int16_t bitstream_len_byte,
+                     int16_t* audio,
+                     int16_t* audio_samples,
+                     int8_t* speech_type);
 
-  WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
-                         const CodecInst& codec_inst);
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
+                   const CodecInst& codec_inst);
 
   void DestructEncoderSafe();
 
   void DestructDecoderSafe();
 
-  WebRtc_Word16 InternalCreateEncoder();
+  int16_t InternalCreateEncoder();
 
-  WebRtc_Word16 InternalCreateDecoder();
+  int16_t InternalCreateDecoder();
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+  int16_t SetBitRateSafe(const int32_t rate);
 
-  WebRtc_Word16 EnableDTX();
+  int16_t EnableDTX();
 
-  WebRtc_Word16 DisableDTX();
+  int16_t DisableDTX();
 
 #ifdef UNUSEDSPEEX
-  WebRtc_Word16 EnableVBR();
+  int16_t EnableVBR();
 
-  WebRtc_Word16 DisableVBR();
+  int16_t DisableVBR();
 
-  WebRtc_Word16 SetComplMode(WebRtc_Word16 mode);
+  int16_t SetComplMode(int16_t mode);
 #endif
 
   SPEEX_encinst_t_* encoder_inst_ptr_;
   SPEEX_decinst_t_* decoder_inst_ptr_;
-  WebRtc_Word16 compl_mode_;
+  int16_t compl_mode_;
   bool vbr_enabled_;
-  WebRtc_Word32 encoding_rate_;
-  WebRtc_Word16 sampling_frequency_;
-  WebRtc_UWord16 samples_in_20ms_audio_;
+  int32_t encoding_rate_;
+  int16_t sampling_frequency_;
+  uint16_t samples_in_20ms_audio_;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/main/source/audio_coding_module.cc b/modules/audio_coding/main/source/audio_coding_module.cc
index 91620b3..159fbb8 100644
--- a/modules/audio_coding/main/source/audio_coding_module.cc
+++ b/modules/audio_coding/main/source/audio_coding_module.cc
@@ -19,7 +19,7 @@
 namespace webrtc {
 
 // Create module
-AudioCodingModule* AudioCodingModule::Create(const WebRtc_Word32 id) {
+AudioCodingModule* AudioCodingModule::Create(const int32_t id) {
   return new AudioCodingModuleImpl(id);
 }
 
@@ -29,21 +29,21 @@
 }
 
 // Get number of supported codecs
-WebRtc_UWord8 AudioCodingModule::NumberOfCodecs() {
-  return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs);
+uint8_t AudioCodingModule::NumberOfCodecs() {
+  return static_cast<uint8_t>(ACMCodecDB::kNumCodecs);
 }
 
 // Get supported codec param with id
-WebRtc_Word32 AudioCodingModule::Codec(WebRtc_UWord8 list_id,
-                                       CodecInst* codec) {
+int32_t AudioCodingModule::Codec(uint8_t list_id,
+                                 CodecInst* codec) {
   // Get the codec settings for the codec with the given list ID
   return ACMCodecDB::Codec(list_id, codec);
 }
 
 // Get supported codec Param with name, frequency and number of channels.
-WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
-                                       CodecInst* codec, int sampling_freq_hz,
-                                       int channels) {
+int32_t AudioCodingModule::Codec(const char* payload_name,
+                                 CodecInst* codec, int sampling_freq_hz,
+                                 int channels) {
   int codec_id;
 
   // Get the id of the codec from the database.
@@ -70,8 +70,8 @@
 }
 
 // Get supported codec Index with name, frequency and number of channels.
-WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
-                                       int sampling_freq_hz, int channels) {
+int32_t AudioCodingModule::Codec(const char* payload_name,
+                                 int sampling_freq_hz, int channels) {
   return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
 }
 
diff --git a/modules/audio_coding/main/source/audio_coding_module_impl.cc b/modules/audio_coding/main/source/audio_coding_module_impl.cc
index f8e966d..8f16bf3 100644
--- a/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -102,7 +102,7 @@
 
 }  // namespace
 
-AudioCodingModuleImpl::AudioCodingModuleImpl(const WebRtc_Word32 id)
+AudioCodingModuleImpl::AudioCodingModuleImpl(const int32_t id)
     : packetization_callback_(NULL),
       id_(id),
       last_timestamp_(0xD87F3F9F),
@@ -170,7 +170,7 @@
   neteq_.set_id(id_);
 
   // Allocate memory for RED.
-  red_buffer_ = new WebRtc_UWord8[MAX_PAYLOAD_SIZE_BYTE];
+  red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE];
 
   // TODO(turajs): This might not be exactly how this class is supposed to work.
   // The external usage might be that |fragmentationVectorSize| has to match
@@ -270,7 +270,7 @@
                "Destroyed");
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) {
+int32_t AudioCodingModuleImpl::ChangeUniqueId(const int32_t id) {
   {
     CriticalSectionScoped lock(acm_crit_sect_);
     id_ = id;
@@ -288,7 +288,7 @@
 
 // Returns the number of milliseconds until the module want a
 // worker thread to call Process.
-WebRtc_Word32 AudioCodingModuleImpl::TimeUntilNextProcess() {
+int32_t AudioCodingModuleImpl::TimeUntilNextProcess() {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("TimeUntilNextProcess")) {
@@ -298,7 +298,7 @@
       (send_codec_inst_.plfreq / 1000);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::Process() {
+int32_t AudioCodingModuleImpl::Process() {
   bool dual_stream;
   {
     CriticalSectionScoped lock(acm_crit_sect_);
@@ -327,7 +327,7 @@
   fragmentation_.fragmentationLength[fragmentation_index] = len_bytes;
   fragmentation_.fragmentationPlType[fragmentation_index] = payload_type;
   fragmentation_.fragmentationTimeDiff[fragmentation_index] =
-      static_cast<WebRtc_UWord16>(current_timestamp - rtp_timestamp);
+      static_cast<uint16_t>(current_timestamp - rtp_timestamp);
   fragmentation_.fragmentationVectorSize++;
   return len_bytes;
 }
@@ -448,7 +448,7 @@
       fragmentation_.fragmentationPlType[index_previous_secondary] =
           secondary_send_codec_inst_.pltype;
       fragmentation_.fragmentationTimeDiff[index_previous_secondary] =
-          static_cast<WebRtc_UWord16>(current_timestamp - last_fec_timestamp_);
+          static_cast<uint16_t>(current_timestamp - last_fec_timestamp_);
       fragmentation_.fragmentationVectorSize++;
     }
 
@@ -512,14 +512,14 @@
 // Process any pending tasks such as timeouts.
 int AudioCodingModuleImpl::ProcessSingleStream() {
   // Make room for 1 RED payload.
-  WebRtc_UWord8 stream[2 * MAX_PAYLOAD_SIZE_BYTE];
-  WebRtc_Word16 length_bytes = 2 * MAX_PAYLOAD_SIZE_BYTE;
-  WebRtc_Word16 red_length_bytes = length_bytes;
-  WebRtc_UWord32 rtp_timestamp;
-  WebRtc_Word16 status;
+  uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];
+  int16_t length_bytes = 2 * MAX_PAYLOAD_SIZE_BYTE;
+  int16_t red_length_bytes = length_bytes;
+  uint32_t rtp_timestamp;
+  int16_t status;
   WebRtcACMEncodingType encoding_type;
   FrameType frame_type = kAudioFrameSpeech;
-  WebRtc_UWord8 current_payload_type = 0;
+  uint8_t current_payload_type = 0;
   bool has_data_to_send = false;
   bool fec_active = false;
   RTPFragmentationHeader my_fragmentation;
@@ -553,7 +553,7 @@
         }
         case kActiveNormalEncoded:
         case kPassiveNormalEncoded: {
-          current_payload_type = (WebRtc_UWord8) send_codec_inst_.pltype;
+          current_payload_type = (uint8_t) send_codec_inst_.pltype;
           frame_type = kAudioFrameSpeech;
           break;
         }
@@ -641,7 +641,7 @@
                  fragmentation_.fragmentationLength[1]);
           // Update the fragmentation time difference vector, in number of
           // timestamps.
-          WebRtc_UWord16 time_since_last = WebRtc_UWord16(
+          uint16_t time_since_last = uint16_t(
               rtp_timestamp - last_fec_timestamp_);
 
           // Update fragmentation vectors.
@@ -665,7 +665,7 @@
         // RFC 2198 (RTP Payload for Redundant Audio Data) will be used.
         // First fragment is the current data (new).
         // Second fragment is the previous data (old).
-        length_bytes = static_cast<WebRtc_Word16>(
+        length_bytes = static_cast<int16_t>(
             fragmentation_.fragmentationLength[0] +
             fragmentation_.fragmentationLength[1]);
 
@@ -714,7 +714,7 @@
 
     if (vad_callback_ != NULL) {
       // Callback with VAD decision.
-      vad_callback_->InFrameType(((WebRtc_Word16) encoding_type));
+      vad_callback_->InFrameType(((int16_t) encoding_type));
     }
   }
   return length_bytes;
@@ -725,7 +725,7 @@
 //
 
 // Initialize send codec.
-WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() {
+int32_t AudioCodingModuleImpl::InitializeSender() {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   // Start with invalid values.
@@ -756,7 +756,7 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() {
+int32_t AudioCodingModuleImpl::ResetEncoder() {
   CriticalSectionScoped lock(acm_crit_sect_);
   if (!HaveValidEncoder("ResetEncoder")) {
     return -1;
@@ -921,7 +921,7 @@
 }
 
 // Can be called multiple times for Codec, CNG, RED.
-WebRtc_Word32 AudioCodingModuleImpl::RegisterSendCodec(
+int32_t AudioCodingModuleImpl::RegisterSendCodec(
     const CodecInst& send_codec) {
   int mirror_id;
   int codec_id = IsValidSendCodec(send_codec, true, id_, &mirror_id);
@@ -1038,7 +1038,7 @@
     }
 
     ACMGenericCodec* codec_ptr = codecs_[codec_id];
-    WebRtc_Word16 status;
+    int16_t status;
     WebRtcACMCodecParams codec_params;
 
     memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst));
@@ -1171,7 +1171,7 @@
 }
 
 // Get current send codec.
-WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
+int32_t AudioCodingModuleImpl::SendCodec(
     CodecInst* current_codec) const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "SendCodec()");
@@ -1193,7 +1193,7 @@
 }
 
 // Get current send frequency.
-WebRtc_Word32 AudioCodingModuleImpl::SendFrequency() const {
+int32_t AudioCodingModuleImpl::SendFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "SendFrequency()");
   CriticalSectionScoped lock(acm_crit_sect_);
@@ -1211,7 +1211,7 @@
 // Get encode bitrate.
 // Adaptive rate codecs return their current encode target rate, while other
 // codecs return there longterm avarage or their fixed rate.
-WebRtc_Word32 AudioCodingModuleImpl::SendBitrate() const {
+int32_t AudioCodingModuleImpl::SendBitrate() const {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!send_codec_registered_) {
@@ -1229,14 +1229,14 @@
 
 // Set available bandwidth, inform the encoder about the estimated bandwidth
 // received from the remote party.
-WebRtc_Word32 AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
-    const WebRtc_Word32 bw) {
+int32_t AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
+    const int32_t bw) {
   return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw);
 }
 
 // Register a transport callback which will be called to deliver
 // the encoded buffers.
-WebRtc_Word32 AudioCodingModuleImpl::RegisterTransportCallback(
+int32_t AudioCodingModuleImpl::RegisterTransportCallback(
     AudioPacketizationCallback* transport) {
   CriticalSectionScoped lock(callback_crit_sect_);
   packetization_callback_ = transport;
@@ -1245,7 +1245,7 @@
 
 // Used by the module to deliver messages to the codec module/application
 // AVT(DTMF).
-WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
+int32_t AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
 #ifndef WEBRTC_DTMF_DETECTION
     AudioCodingFeedback* /* incoming_message */,
     const ACMCountries /* cpt */) {
@@ -1253,7 +1253,7 @@
 #else
     AudioCodingFeedback* incoming_message,
     const ACMCountries cpt) {
-  WebRtc_Word16 status = 0;
+  int16_t status = 0;
 
   // Enter the critical section for callback.
   {
@@ -1302,7 +1302,7 @@
 }
 
 // Add 10MS of raw (PCM) audio data to the encoder.
-WebRtc_Word32 AudioCodingModuleImpl::Add10MsData(
+int32_t AudioCodingModuleImpl::Add10MsData(
     const AudioFrame& audio_frame) {
   if (audio_frame.samples_per_channel_ <= 0) {
     assert(false);
@@ -1412,7 +1412,7 @@
   assert((secondary_encoder_.get() != NULL) ?
       secondary_send_codec_inst_.plfreq == send_codec_inst_.plfreq : true);
 
-  bool resample = ((WebRtc_Word32) in_frame.sample_rate_hz_
+  bool resample = ((int32_t) in_frame.sample_rate_hz_
       != send_codec_inst_.plfreq);
 
   // This variable is true if primary codec and secondary codec (if exists)
@@ -1465,7 +1465,7 @@
     // Calculate the timestamp of this frame.
     if (last_in_timestamp_ > in_frame.timestamp_) {
       // A wrap around has happened.
-      timestamp_diff = ((WebRtc_UWord32) 0xFFFFFFFF - last_in_timestamp_)
+      timestamp_diff = ((uint32_t) 0xFFFFFFFF - last_in_timestamp_)
                                                  + in_frame.timestamp_;
     } else {
       timestamp_diff = in_frame.timestamp_ - last_in_timestamp_;
@@ -1502,7 +1502,7 @@
 }
 
 // Configure FEC status i.e on/off.
-WebRtc_Word32
+int32_t
 AudioCodingModuleImpl::SetFECStatus(
 #ifdef WEBRTC_CODEC_RED
     const bool enable_fec) {
@@ -1532,9 +1532,9 @@
 /////////////////////////////////////////
 //   (VAD) Voice Activity Detection
 //
-WebRtc_Word32 AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
-                                            const bool enable_vad,
-                                            const ACMVADMode mode) {
+int32_t AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
+                                      const bool enable_vad,
+                                      const ACMVADMode mode) {
   CriticalSectionScoped lock(acm_crit_sect_);
   return SetVADSafe(enable_dtx, enable_vad, mode);
 }
@@ -1569,7 +1569,7 @@
 
   // If a send codec is registered, set VAD/DTX for the codec.
   if (HaveValidEncoder("SetVAD")) {
-    WebRtc_Word16 status = codecs_[current_send_codec_idx_]->SetVAD(enable_dtx,
+    int16_t status = codecs_[current_send_codec_idx_]->SetVAD(enable_dtx,
                                                                     enable_vad,
                                                                     mode);
     if (status == 1) {
@@ -1600,8 +1600,8 @@
 
 // Get VAD/DTX settings.
 // TODO(tlegrand): Change this method to void.
-WebRtc_Word32 AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
-                                         ACMVADMode* mode) const {
+int32_t AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
+                                   ACMVADMode* mode) const {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   *dtx_enabled = dtx_enabled_;
@@ -1615,13 +1615,13 @@
 //   Receiver
 //
 
-WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiver() {
+int32_t AudioCodingModuleImpl::InitializeReceiver() {
   CriticalSectionScoped lock(acm_crit_sect_);
   return InitializeReceiverSafe();
 }
 
 // Initialize receiver, resets codec database etc.
-WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiverSafe() {
+int32_t AudioCodingModuleImpl::InitializeReceiverSafe() {
   initial_delay_ms_ = 0;
   num_packets_accumulated_ = 0;
   num_bytes_accumulated_ = 0;
@@ -1673,7 +1673,7 @@
 }
 
 // Reset the decoder state.
-WebRtc_Word32 AudioCodingModuleImpl::ResetDecoder() {
+int32_t AudioCodingModuleImpl::ResetDecoder() {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
@@ -1689,7 +1689,7 @@
 }
 
 // Get current receive frequency.
-WebRtc_Word32 AudioCodingModuleImpl::ReceiveFrequency() const {
+int32_t AudioCodingModuleImpl::ReceiveFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "ReceiveFrequency()");
   WebRtcACMCodecParams codec_params;
@@ -1706,7 +1706,7 @@
 }
 
 // Get current playout frequency.
-WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
+int32_t AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
 
@@ -1717,7 +1717,7 @@
 
 // Register possible receive codecs, can be called multiple times,
 // for codecs, CNG (NB, WB and SWB), DTMF, RED.
-WebRtc_Word32 AudioCodingModuleImpl::RegisterReceiveCodec(
+int32_t AudioCodingModuleImpl::RegisterReceiveCodec(
     const CodecInst& receive_codec) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
@@ -1859,9 +1859,9 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::RegisterRecCodecMSSafe(
-    const CodecInst& receive_codec, WebRtc_Word16 codec_id,
-    WebRtc_Word16 mirror_id, ACMNetEQ::JitterBuffer jitter_buffer) {
+int32_t AudioCodingModuleImpl::RegisterRecCodecMSSafe(
+    const CodecInst& receive_codec, int16_t codec_id,
+    int16_t mirror_id, ACMNetEQ::JitterBuffer jitter_buffer) {
   ACMGenericCodec** codecs;
   if (jitter_buffer == ACMNetEQ::kMasterJb) {
     codecs = &codecs_[0];
@@ -1896,7 +1896,7 @@
 
   codecs[codec_id]->SetIsMaster(jitter_buffer == ACMNetEQ::kMasterJb);
 
-  WebRtc_Word16 status = 0;
+  int16_t status = 0;
   WebRtcACMCodecParams codec_params;
   memcpy(&(codec_params.codec_inst), &receive_codec, sizeof(CodecInst));
   codec_params.enable_vad = false;
@@ -1933,7 +1933,7 @@
 }
 
 // Get current received codec.
-WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
+int32_t AudioCodingModuleImpl::ReceiveCodec(
     CodecInst* current_codec) const {
   WebRtcACMCodecParams decoder_param;
   CriticalSectionScoped lock(acm_crit_sect_);
@@ -1958,9 +1958,9 @@
 }
 
 // Incoming packet from network parsed and ready for decode.
-WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
-    const WebRtc_UWord8* incoming_payload,
-    const WebRtc_Word32 payload_length,
+int32_t AudioCodingModuleImpl::IncomingPacket(
+    const uint8_t* incoming_payload,
+    const int32_t payload_length,
     const WebRtcRTPHeader& rtp_info) {
   WebRtcRTPHeader rtp_header;
 
@@ -1977,7 +1977,7 @@
     // and "received frequency."
     CriticalSectionScoped lock(acm_crit_sect_);
 
-    WebRtc_UWord8 my_payload_type;
+    uint8_t my_payload_type;
 
     // Check if this is an RED payload.
     if (rtp_info.header.payloadType == receive_red_pltype_) {
@@ -2037,8 +2037,8 @@
   if (expected_channels_ == 2) {
     if (!rtp_info.type.Audio.isCNG) {
       // Create a new vector for the payload, maximum payload size.
-      WebRtc_Word32 length = payload_length;
-      WebRtc_UWord8 payload[kMaxPacketSize];
+      int32_t length = payload_length;
+      uint8_t payload[kMaxPacketSize];
       assert(payload_length <= kMaxPacketSize);
       memcpy(payload, incoming_payload, payload_length);
       codecs_[current_receive_codec_idx_]->SplitStereoPacket(payload, &length);
@@ -2133,8 +2133,8 @@
 }
 
 // Minimum playout delay (Used for lip-sync).
-WebRtc_Word32 AudioCodingModuleImpl::SetMinimumPlayoutDelay(
-    const WebRtc_Word32 time_ms) {
+int32_t AudioCodingModuleImpl::SetMinimumPlayoutDelay(
+    const int32_t time_ms) {
   if ((time_ms < 0) || (time_ms > 10000)) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Delay must be in the range of 0-10000 milliseconds.");
@@ -2160,7 +2160,7 @@
 
 // Configure Dtmf playout status i.e on/off playout the incoming outband
 // Dtmf tone.
-WebRtc_Word32 AudioCodingModuleImpl::SetDtmfPlayoutStatus(
+int32_t AudioCodingModuleImpl::SetDtmfPlayoutStatus(
 #ifndef WEBRTC_CODEC_AVT
     const bool /* enable */) {
   WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
@@ -2175,9 +2175,9 @@
 // Estimate the Bandwidth based on the incoming stream, needed for one way
 // audio where the RTCP send the BW estimate.
 // This is also done in the RTP module.
-WebRtc_Word32 AudioCodingModuleImpl::DecoderEstimatedBandwidth() const {
+int32_t AudioCodingModuleImpl::DecoderEstimatedBandwidth() const {
   CodecInst codec;
-  WebRtc_Word16 codec_id = -1;
+  int16_t codec_id = -1;
   int pltype_wb;
   int pltype_swb;
 
@@ -2212,7 +2212,7 @@
 }
 
 // Set playout mode for: voice, fax, or streaming.
-WebRtc_Word32 AudioCodingModuleImpl::SetPlayoutMode(
+int32_t AudioCodingModuleImpl::SetPlayoutMode(
     const AudioPlayoutMode mode) {
   if ((mode != voice) && (mode != fax) && (mode != streaming) &&
       (mode != off)) {
@@ -2230,10 +2230,9 @@
 
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
-WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
-    WebRtc_Word32 desired_freq_hz, AudioFrame* audio_frame) {
+int32_t AudioCodingModuleImpl::PlayoutData10Ms(
+    int32_t desired_freq_hz, AudioFrame* audio_frame) {
   TRACE_EVENT0("webrtc_voe", "ACM::PlayoutData10Ms");
-
   bool stereo_mode;
 
   if (GetSilence(desired_freq_hz, audio_frame))
@@ -2254,11 +2253,11 @@
   // For stereo playout:
   // Master and Slave samples are interleaved starting with Master.
 
-  const WebRtc_UWord16 receive_freq =
-      static_cast<WebRtc_UWord16>(audio_frame_.sample_rate_hz_);
+  const uint16_t receive_freq =
+      static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
   bool tone_detected = false;
-  WebRtc_Word16 last_detected_tone;
-  WebRtc_Word16 tone;
+  int16_t last_detected_tone;
+  int16_t tone;
 
   // Limit the scope of ACM Critical section.
   {
@@ -2266,7 +2265,7 @@
 
     if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) {
       // Resample payload_data.
-      WebRtc_Word16 temp_len = output_resampler_.Resample10Msec(
+      int16_t temp_len = output_resampler_.Resample10Msec(
           audio_frame_.data_, receive_freq, audio_frame->data_,
           desired_freq_hz, audio_frame_.num_channels_);
 
@@ -2277,13 +2276,13 @@
       }
 
       // Set the payload data length from the resampler.
-      audio_frame->samples_per_channel_ = (WebRtc_UWord16) temp_len;
+      audio_frame->samples_per_channel_ = (uint16_t) temp_len;
       // Set the sampling frequency.
       audio_frame->sample_rate_hz_ = desired_freq_hz;
     } else {
       memcpy(audio_frame->data_, audio_frame_.data_,
              audio_frame_.samples_per_channel_ * audio_frame->num_channels_
-             * sizeof(WebRtc_Word16));
+             * sizeof(int16_t));
       // Set the payload length.
       audio_frame->samples_per_channel_ =
           audio_frame_.samples_per_channel_;
@@ -2304,7 +2303,7 @@
                                  tone);
         } else {
           // We are in 8 kHz so the master channel needs only 80 samples.
-          WebRtc_Word16 master_channel[80];
+          int16_t master_channel[80];
           for (int n = 0; n < 80; n++) {
             master_channel[n] = audio_frame->data_[n << 1];
           }
@@ -2320,7 +2319,7 @@
                                  audio_frame_.samples_per_channel_,
                                  receive_freq, tone_detected, tone);
         } else {
-          WebRtc_Word16 master_channel[WEBRTC_10MS_PCM_AUDIO];
+          int16_t master_channel[WEBRTC_10MS_PCM_AUDIO];
           for (int n = 0; n < audio_frame_.samples_per_channel_; n++) {
             master_channel[n] = audio_frame_.data_[n << 1];
           }
@@ -2348,11 +2347,11 @@
     if (dtmf_callback_ != NULL) {
       if (tone != kACMToneEnd) {
         // just a tone
-        dtmf_callback_->IncomingDtmf((WebRtc_UWord8) tone, false);
+        dtmf_callback_->IncomingDtmf((uint8_t) tone, false);
       } else if ((tone == kACMToneEnd) && (last_detected_tone != kACMToneEnd)) {
         // The tone is "END" and the previously detected tone is
         // not "END," so call fir an end.
-        dtmf_callback_->IncomingDtmf((WebRtc_UWord8) last_detected_tone, true);
+        dtmf_callback_->IncomingDtmf((uint8_t) last_detected_tone, true);
       }
     }
   }
@@ -2375,7 +2374,7 @@
 }
 
 // Configure VAD aggressiveness on the incoming stream.
-WebRtc_Word16 AudioCodingModuleImpl::SetReceiveVADMode(const ACMVADMode mode) {
+int16_t AudioCodingModuleImpl::SetReceiveVADMode(const ACMVADMode mode) {
   return neteq_.SetVADMode(mode);
 }
 
@@ -2383,9 +2382,9 @@
 //   Statistics
 //
 
-WebRtc_Word32 AudioCodingModuleImpl::NetworkStatistics(
+int32_t AudioCodingModuleImpl::NetworkStatistics(
     ACMNetworkStatistics* statistics) const {
-  WebRtc_Word32 status;
+  int32_t status;
   status = neteq_.NetworkStatistics(statistics);
   return status;
 }
@@ -2400,7 +2399,7 @@
   codecs_[current_send_codec_idx_]->DestructEncoderInst(inst);
 }
 
-WebRtc_Word16 AudioCodingModuleImpl::AudioBuffer(
+int16_t AudioCodingModuleImpl::AudioBuffer(
     WebRtcACMAudioBuff& buffer) {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
                "AudioBuffer()");
@@ -2411,7 +2410,7 @@
   return codecs_[current_send_codec_idx_]->AudioBuffer(buffer);
 }
 
-WebRtc_Word16 AudioCodingModuleImpl::SetAudioBuffer(
+int16_t AudioCodingModuleImpl::SetAudioBuffer(
     WebRtcACMAudioBuff& buffer) {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
                "SetAudioBuffer()");
@@ -2421,7 +2420,7 @@
   return codecs_[current_send_codec_idx_]->SetAudioBuffer(buffer);
 }
 
-WebRtc_UWord32 AudioCodingModuleImpl::EarliestTimestamp() const {
+uint32_t AudioCodingModuleImpl::EarliestTimestamp() const {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
                "EarliestTimestamp()");
   if (!HaveValidEncoder("EarliestTimestamp")) {
@@ -2430,7 +2429,7 @@
   return codecs_[current_send_codec_idx_]->EarliestTimestamp();
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::RegisterVADCallback(
+int32_t AudioCodingModuleImpl::RegisterVADCallback(
     ACMVADCallback* vad_callback) {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
                "RegisterVADCallback()");
@@ -2440,9 +2439,9 @@
 }
 
 // TODO(tlegrand): Modify this function to work for stereo, and add tests.
-WebRtc_Word32 AudioCodingModuleImpl::IncomingPayload(
-    const WebRtc_UWord8* incoming_payload, const WebRtc_Word32 payload_length,
-    const WebRtc_UWord8 payload_type, const WebRtc_UWord32 timestamp) {
+int32_t AudioCodingModuleImpl::IncomingPayload(
+    const uint8_t* incoming_payload, const int32_t payload_length,
+    const uint8_t payload_type, const uint32_t timestamp) {
   if (payload_length < 0) {
     // Log error in trace file.
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -2467,8 +2466,8 @@
     // Start with random numbers.
     dummy_rtp_header_->header.sequenceNumber = rand();
     dummy_rtp_header_->header.timestamp =
-        (static_cast<WebRtc_UWord32>(rand()) << 16) +
-        static_cast<WebRtc_UWord32>(rand());
+        (static_cast<uint32_t>(rand()) << 16) +
+        static_cast<uint32_t>(rand());
     dummy_rtp_header_->type.Audio.channel = 1;
 
     if (DecoderParamByPlType(payload_type, codec_params) < 0) {
@@ -2513,11 +2512,11 @@
   return 0;
 }
 
-WebRtc_Word16 AudioCodingModuleImpl::DecoderParamByPlType(
-    const WebRtc_UWord8 payload_type,
+int16_t AudioCodingModuleImpl::DecoderParamByPlType(
+    const uint8_t payload_type,
     WebRtcACMCodecParams& codec_params) const {
   CriticalSectionScoped lock(acm_crit_sect_);
-  for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs;
+  for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs;
       id++) {
     if (codecs_[id] != NULL) {
       if (codecs_[id]->DecoderInitialized()) {
@@ -2537,17 +2536,17 @@
   return -1;
 }
 
-WebRtc_Word16 AudioCodingModuleImpl::DecoderListIDByPlName(
-    const char* name, const WebRtc_UWord16 frequency) const {
+int16_t AudioCodingModuleImpl::DecoderListIDByPlName(
+    const char* name, const uint16_t frequency) const {
   WebRtcACMCodecParams codec_params;
   CriticalSectionScoped lock(acm_crit_sect_);
-  for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
+  for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
     if ((codecs_[id] != NULL)) {
       if (codecs_[id]->DecoderInitialized()) {
         assert(registered_pltypes_[id] >= 0);
         assert(registered_pltypes_[id] <= 255);
         codecs_[id]->DecoderParams(
-            &codec_params, (WebRtc_UWord8) registered_pltypes_[id]);
+            &codec_params, (uint8_t) registered_pltypes_[id]);
         if (!STR_CASE_CMP(codec_params.codec_inst.plname, name)) {
           // Check if the given sampling frequency matches.
           // A zero sampling frequency means we matching the names
@@ -2568,8 +2567,8 @@
   return -1;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(
-    WebRtc_UWord32& timestamp) const {
+int32_t AudioCodingModuleImpl::LastEncodedTimestamp(
+    uint32_t& timestamp) const {
   CriticalSectionScoped lock(acm_crit_sect_);
   if (!HaveValidEncoder("LastEncodedTimestamp")) {
     return -1;
@@ -2578,7 +2577,7 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
+int32_t AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
     bool use_webrtc_dtx) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
@@ -2589,7 +2588,7 @@
     return -1;
   }
 
-  WebRtc_Word32 res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX(
+  int32_t res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX(
       use_webrtc_dtx);
   // Check if VAD is turned on, or if there is any error.
   if (res == 1) {
@@ -2604,7 +2603,7 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
+int32_t AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
     bool* uses_webrtc_dtx) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
@@ -2618,8 +2617,8 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate(
-    const WebRtc_UWord32 max_bit_per_sec) {
+int32_t AudioCodingModuleImpl::SetISACMaxRate(
+    const uint32_t max_bit_per_sec) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("SetISACMaxRate")) {
@@ -2629,8 +2628,8 @@
   return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxPayloadSize(
-    const WebRtc_UWord16 max_size_bytes) {
+int32_t AudioCodingModuleImpl::SetISACMaxPayloadSize(
+    const uint16_t max_size_bytes) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
@@ -2641,9 +2640,9 @@
       max_size_bytes);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
-    const WebRtc_UWord8 frame_size_ms,
-    const WebRtc_UWord16 rate_bit_per_sec,
+int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
+    const uint8_t frame_size_ms,
+    const uint16_t rate_bit_per_sec,
     const bool enforce_frame_size) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
@@ -2655,7 +2654,7 @@
       frame_size_ms, rate_bit_per_sec, enforce_frame_size);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::SetBackgroundNoiseMode(
+int32_t AudioCodingModuleImpl::SetBackgroundNoiseMode(
     const ACMBackgroundNoiseMode mode) {
   if ((mode < On) || (mode > Off)) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -2665,13 +2664,13 @@
   return neteq_.SetBackgroundNoiseMode(mode);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::BackgroundNoiseMode(
+int32_t AudioCodingModuleImpl::BackgroundNoiseMode(
     ACMBackgroundNoiseMode* mode) {
   return neteq_.BackgroundNoiseMode(*mode);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::PlayoutTimestamp(
-    WebRtc_UWord32* timestamp) {
+int32_t AudioCodingModuleImpl::PlayoutTimestamp(
+    uint32_t* timestamp) {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutTimestamp()");
   {
@@ -2705,8 +2704,8 @@
   return true;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::UnregisterReceiveCodec(
-    const WebRtc_Word16 payload_type) {
+int32_t AudioCodingModuleImpl::UnregisterReceiveCodec(
+    const int16_t payload_type) {
   CriticalSectionScoped lock(acm_crit_sect_);
   int id;
 
@@ -2727,10 +2726,10 @@
   return UnregisterReceiveCodecSafe(id);
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::UnregisterReceiveCodecSafe(
-    const WebRtc_Word16 codec_id) {
+int32_t AudioCodingModuleImpl::UnregisterReceiveCodecSafe(
+    const int16_t codec_id) {
   const WebRtcNetEQDecoder *neteq_decoder = ACMCodecDB::NetEQDecoders();
-  WebRtc_Word16 mirror_id = ACMCodecDB::MirrorID(codec_id);
+  int16_t mirror_id = ACMCodecDB::MirrorID(codec_id);
   bool stereo_receiver = false;
 
   if (codecs_[codec_id] != NULL) {
@@ -2798,13 +2797,13 @@
   return 0;
 }
 
-WebRtc_Word32 AudioCodingModuleImpl::REDPayloadISAC(
-    const WebRtc_Word32 isac_rate, const WebRtc_Word16 isac_bw_estimate,
-    WebRtc_UWord8* payload, WebRtc_Word16* length_bytes) {
+int32_t AudioCodingModuleImpl::REDPayloadISAC(
+    const int32_t isac_rate, const int16_t isac_bw_estimate,
+    uint8_t* payload, int16_t* length_bytes) {
   if (!HaveValidEncoder("EncodeData")) {
     return -1;
   }
-  WebRtc_Word16 status;
+  int16_t status;
   status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate,
                                                             isac_bw_estimate,
                                                             payload,
@@ -2823,7 +2822,7 @@
   memset(fragmentation_.fragmentationPlType, 0, kMaxNumFragmentationVectors *
          sizeof(fragmentation_.fragmentationPlType[0]));
   fragmentation_.fragmentationVectorSize =
-      static_cast<WebRtc_UWord16>(vector_size);
+      static_cast<uint16_t>(vector_size);
 }
 
 int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
diff --git a/modules/audio_coding/main/source/audio_coding_module_impl.h b/modules/audio_coding/main/source/audio_coding_module_impl.h
index 6fb40d5..7ce32dc 100644
--- a/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -28,33 +28,33 @@
 class AudioCodingModuleImpl : public AudioCodingModule {
  public:
   // Constructor
-  explicit AudioCodingModuleImpl(const WebRtc_Word32 id);
+  explicit AudioCodingModuleImpl(const int32_t id);
 
   // Destructor
   ~AudioCodingModuleImpl();
 
   // Change the unique identifier of this object.
-  virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+  virtual int32_t ChangeUniqueId(const int32_t id);
 
   // Returns the number of milliseconds until the module want a worker thread
   // to call Process.
-  WebRtc_Word32 TimeUntilNextProcess();
+  int32_t TimeUntilNextProcess();
 
   // Process any pending tasks such as timeouts.
-  WebRtc_Word32 Process();
+  int32_t Process();
 
   /////////////////////////////////////////
   //   Sender
   //
 
   // Initialize send codec.
-  WebRtc_Word32 InitializeSender();
+  int32_t InitializeSender();
 
   // Reset send codec.
-  WebRtc_Word32 ResetEncoder();
+  int32_t ResetEncoder();
 
   // Can be called multiple times for Codec, CNG, RED.
-  WebRtc_Word32 RegisterSendCodec(const CodecInst& send_codec);
+  int32_t RegisterSendCodec(const CodecInst& send_codec);
 
   // Register Secondary codec for dual-streaming. Dual-streaming is activated
   // right after the secondary codec is registered.
@@ -68,45 +68,45 @@
   int SecondarySendCodec(CodecInst* secondary_codec) const;
 
   // Get current send codec.
-  WebRtc_Word32 SendCodec(CodecInst* current_codec) const;
+  int32_t SendCodec(CodecInst* current_codec) const;
 
   // Get current send frequency.
-  WebRtc_Word32 SendFrequency() const;
+  int32_t SendFrequency() const;
 
   // Get encode bit-rate.
   // Adaptive rate codecs return their current encode target rate, while other
   // codecs return there long-term average or their fixed rate.
-  WebRtc_Word32 SendBitrate() const;
+  int32_t SendBitrate() const;
 
   // Set available bandwidth, inform the encoder about the
   // estimated bandwidth received from the remote party.
-  virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(const WebRtc_Word32 bw);
+  virtual int32_t SetReceivedEstimatedBandwidth(const int32_t bw);
 
   // Register a transport callback which will be
   // called to deliver the encoded buffers.
-  WebRtc_Word32 RegisterTransportCallback(
+  int32_t RegisterTransportCallback(
       AudioPacketizationCallback* transport);
 
   // Used by the module to deliver messages to the codec module/application
   // AVT(DTMF).
-  WebRtc_Word32 RegisterIncomingMessagesCallback(
+  int32_t RegisterIncomingMessagesCallback(
       AudioCodingFeedback* incoming_message, const ACMCountries cpt);
 
   // Add 10 ms of raw (PCM) audio data to the encoder.
-  WebRtc_Word32 Add10MsData(const AudioFrame& audio_frame);
+  int32_t Add10MsData(const AudioFrame& audio_frame);
 
   // Set background noise mode for NetEQ, on, off or fade.
-  WebRtc_Word32 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
+  int32_t SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
 
   // Get current background noise mode.
-  WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode* mode);
+  int32_t BackgroundNoiseMode(ACMBackgroundNoiseMode* mode);
 
   /////////////////////////////////////////
   // (FEC) Forward Error Correction
   //
 
   // Configure FEC status i.e on/off.
-  WebRtc_Word32 SetFECStatus(const bool enable_fec);
+  int32_t SetFECStatus(const bool enable_fec);
 
   // Get FEC status.
   bool FECStatus() const;
@@ -117,62 +117,62 @@
   //   (CNG) Comfort Noise Generation
   //
 
-  WebRtc_Word32 SetVAD(const bool enable_dtx = true,
-                       const bool enable_vad = false,
-                       const ACMVADMode mode = VADNormal);
+  int32_t SetVAD(const bool enable_dtx = true,
+                 const bool enable_vad = false,
+                 const ACMVADMode mode = VADNormal);
 
-  WebRtc_Word32 VAD(bool* dtx_enabled, bool* vad_enabled,
-                    ACMVADMode* mode) const;
+  int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
+              ACMVADMode* mode) const;
 
-  WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vad_callback);
+  int32_t RegisterVADCallback(ACMVADCallback* vad_callback);
 
   // Get VAD aggressiveness on the incoming stream.
   ACMVADMode ReceiveVADMode() const;
 
   // Configure VAD aggressiveness on the incoming stream.
-  WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode);
+  int16_t SetReceiveVADMode(const ACMVADMode mode);
 
   /////////////////////////////////////////
   //   Receiver
   //
 
   // Initialize receiver, resets codec database etc.
-  WebRtc_Word32 InitializeReceiver();
+  int32_t InitializeReceiver();
 
   // Reset the decoder state.
-  WebRtc_Word32 ResetDecoder();
+  int32_t ResetDecoder();
 
   // Get current receive frequency.
-  WebRtc_Word32 ReceiveFrequency() const;
+  int32_t ReceiveFrequency() const;
 
   // Get current playout frequency.
-  WebRtc_Word32 PlayoutFrequency() const;
+  int32_t PlayoutFrequency() const;
 
   // Register possible receive codecs, can be called multiple times,
   // for codecs, CNG, DTMF, RED.
-  WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receive_codec);
+  int32_t RegisterReceiveCodec(const CodecInst& receive_codec);
 
   // Get current received codec.
-  WebRtc_Word32 ReceiveCodec(CodecInst* current_codec) const;
+  int32_t ReceiveCodec(CodecInst* current_codec) const;
 
   // Incoming packet from network parsed and ready for decode.
-  WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
-                               const WebRtc_Word32 payload_length,
-                               const WebRtcRTPHeader& rtp_info);
+  int32_t IncomingPacket(const uint8_t* incoming_payload,
+                         const int32_t payload_length,
+                         const WebRtcRTPHeader& rtp_info);
 
   // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
   // One usage for this API is when pre-encoded files are pushed in ACM.
-  WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incoming_payload,
-                                const WebRtc_Word32 payload_length,
-                                const WebRtc_UWord8 payload_type,
-                                const WebRtc_UWord32 timestamp = 0);
+  int32_t IncomingPayload(const uint8_t* incoming_payload,
+                          const int32_t payload_length,
+                          const uint8_t payload_type,
+                          const uint32_t timestamp = 0);
 
   // Minimum playout delay (used for lip-sync).
-  WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 time_ms);
+  int32_t SetMinimumPlayoutDelay(const int32_t time_ms);
 
   // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
   // tone.
-  WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable);
+  int32_t SetDtmfPlayoutStatus(const bool enable);
 
   // Get Dtmf playout status.
   bool DtmfPlayoutStatus() const;
@@ -180,81 +180,81 @@
   // Estimate the Bandwidth based on the incoming stream, needed
   // for one way audio where the RTCP send the BW estimate.
   // This is also done in the RTP module .
-  WebRtc_Word32 DecoderEstimatedBandwidth() const;
+  int32_t DecoderEstimatedBandwidth() const;
 
   // Set playout mode voice, fax.
-  WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode);
+  int32_t SetPlayoutMode(const AudioPlayoutMode mode);
 
   // Get playout mode voice, fax.
   AudioPlayoutMode PlayoutMode() const;
 
   // Get playout timestamp.
-  WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32* timestamp);
+  int32_t PlayoutTimestamp(uint32_t* timestamp);
 
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
-  WebRtc_Word32 PlayoutData10Ms(WebRtc_Word32 desired_freq_hz,
-                                AudioFrame* audio_frame);
+  int32_t PlayoutData10Ms(int32_t desired_freq_hz,
+                          AudioFrame* audio_frame);
 
   /////////////////////////////////////////
   //   Statistics
   //
 
-  WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics* statistics) const;
+  int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const;
 
   void DestructEncoderInst(void* inst);
 
-  WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& buffer);
+  int16_t AudioBuffer(WebRtcACMAudioBuff& buffer);
 
   // GET RED payload for iSAC. The method id called when 'this' ACM is
   // the default ACM.
-  WebRtc_Word32 REDPayloadISAC(const WebRtc_Word32 isac_rate,
-                               const WebRtc_Word16 isac_bw_estimate,
-                               WebRtc_UWord8* payload,
-                               WebRtc_Word16* length_bytes);
+  int32_t REDPayloadISAC(const int32_t isac_rate,
+                         const int16_t isac_bw_estimate,
+                         uint8_t* payload,
+                         int16_t* length_bytes);
 
-  WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& buffer);
+  int16_t SetAudioBuffer(WebRtcACMAudioBuff& buffer);
 
-  WebRtc_UWord32 EarliestTimestamp() const;
+  uint32_t EarliestTimestamp() const;
 
-  WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
+  int32_t LastEncodedTimestamp(uint32_t& timestamp) const;
 
-  WebRtc_Word32 ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx);
+  int32_t ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx);
 
-  WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx);
+  int32_t IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx);
 
-  WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_bit_per_sec);
+  int32_t SetISACMaxRate(const uint32_t max_bit_per_sec);
 
-  WebRtc_Word32 SetISACMaxPayloadSize(const WebRtc_UWord16 max_size_bytes);
+  int32_t SetISACMaxPayloadSize(const uint16_t max_size_bytes);
 
-  WebRtc_Word32 ConfigISACBandwidthEstimator(
-      const WebRtc_UWord8 frame_size_ms,
-      const WebRtc_UWord16 rate_bit_per_sec,
+  int32_t ConfigISACBandwidthEstimator(
+      const uint8_t frame_size_ms,
+      const uint16_t rate_bit_per_sec,
       const bool enforce_frame_size = false);
 
-  WebRtc_Word32 UnregisterReceiveCodec(const WebRtc_Word16 payload_type);
+  int32_t UnregisterReceiveCodec(const int16_t payload_type);
 
  protected:
   void UnregisterSendCodec();
 
-  WebRtc_Word32 UnregisterReceiveCodecSafe(const WebRtc_Word16 id);
+  int32_t UnregisterReceiveCodecSafe(const int16_t id);
 
   ACMGenericCodec* CreateCodec(const CodecInst& codec);
 
-  WebRtc_Word16 DecoderParamByPlType(const WebRtc_UWord8 payload_type,
-                                     WebRtcACMCodecParams& codec_params) const;
+  int16_t DecoderParamByPlType(const uint8_t payload_type,
+                               WebRtcACMCodecParams& codec_params) const;
 
-  WebRtc_Word16 DecoderListIDByPlName(
-      const char* name, const WebRtc_UWord16 frequency = 0) const;
+  int16_t DecoderListIDByPlName(
+      const char* name, const uint16_t frequency = 0) const;
 
-  WebRtc_Word32 InitializeReceiverSafe();
+  int32_t InitializeReceiverSafe();
 
   bool HaveValidEncoder(const char* caller_name) const;
 
-  WebRtc_Word32 RegisterRecCodecMSSafe(const CodecInst& receive_codec,
-                                       WebRtc_Word16 codec_id,
-                                       WebRtc_Word16 mirror_id,
-                                       ACMNetEQ::JitterBuffer jitter_buffer);
+  int32_t RegisterRecCodecMSSafe(const CodecInst& receive_codec,
+                                 int16_t codec_id,
+                                 int16_t mirror_id,
+                                 ACMNetEQ::JitterBuffer jitter_buffer);
 
   // Set VAD/DTX status. This function does not acquire a lock, and it is
   // created to be called only from inside a critical section.
@@ -313,9 +313,9 @@
   bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame);
 
   AudioPacketizationCallback* packetization_callback_;
-  WebRtc_Word32 id_;
-  WebRtc_UWord32 last_timestamp_;
-  WebRtc_UWord32 last_in_timestamp_;
+  int32_t id_;
+  uint32_t last_timestamp_;
+  uint32_t last_in_timestamp_;
   CodecInst send_codec_inst_;
   uint8_t cng_nb_pltype_;
   uint8_t cng_wb_pltype_;
@@ -327,13 +327,13 @@
   ACMVADMode vad_mode_;
   ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs];
   ACMGenericCodec* slave_codecs_[ACMCodecDB::kMaxNumCodecs];
-  WebRtc_Word16 mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs];
+  int16_t mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs];
   bool stereo_receive_[ACMCodecDB::kMaxNumCodecs];
   bool stereo_receive_registered_;
   bool stereo_send_;
   int prev_received_channel_;
   int expected_channels_;
-  WebRtc_Word32 current_send_codec_idx_;
+  int32_t current_send_codec_idx_;
   int current_receive_codec_idx_;
   bool send_codec_registered_;
   ACMResampler input_resampler_;
@@ -341,7 +341,7 @@
   ACMNetEQ neteq_;
   CriticalSectionWrapper* acm_crit_sect_;
   ACMVADCallback* vad_callback_;
-  WebRtc_UWord8 last_recv_audio_codec_pltype_;
+  uint8_t last_recv_audio_codec_pltype_;
 
   // RED/FEC.
   bool is_first_red_;
@@ -350,35 +350,35 @@
   // as pointers and not an array. If concerned about the memory, then make a
   // set-up function to allocate them only when they are going to be used, i.e.
   // FEC or Dual-streaming is enabled.
-  WebRtc_UWord8* red_buffer_;
+  uint8_t* red_buffer_;
   // TODO(turajs): we actually don't need |fragmentation_| as a member variable.
   // It is sufficient to keep the length & payload type of previous payload in
   // member variables.
   RTPFragmentationHeader fragmentation_;
-  WebRtc_UWord32 last_fec_timestamp_;
+  uint32_t last_fec_timestamp_;
   // If no RED is registered as receive codec this
   // will have an invalid value.
-  WebRtc_UWord8 receive_red_pltype_;
+  uint8_t receive_red_pltype_;
 
   // This is to keep track of CN instances where we can send DTMFs.
-  WebRtc_UWord8 previous_pltype_;
+  uint8_t previous_pltype_;
 
   // This keeps track of payload types associated with codecs_[].
   // We define it as signed variable and initialize with -1 to indicate
   // unused elements.
-  WebRtc_Word16 registered_pltypes_[ACMCodecDB::kMaxNumCodecs];
+  int16_t registered_pltypes_[ACMCodecDB::kMaxNumCodecs];
 
   // Used when payloads are pushed into ACM without any RTP info
   // One example is when pre-encoded bit-stream is pushed from
   // a file.
   WebRtcRTPHeader* dummy_rtp_header_;
-  WebRtc_UWord16 recv_pl_frame_size_smpls_;
+  uint16_t recv_pl_frame_size_smpls_;
 
   bool receiver_initialized_;
   ACMDTMFDetection* dtmf_detector_;
 
   AudioCodingFeedback* dtmf_callback_;
-  WebRtc_Word16 last_detected_tone_;
+  int16_t last_detected_tone_;
   CriticalSectionWrapper* callback_crit_sect_;
 
   AudioFrame audio_frame_;
diff --git a/modules/audio_coding/main/test/APITest.cc b/modules/audio_coding/main/test/APITest.cc
index 97376a2..6e4115d 100644
--- a/modules/audio_coding/main/test/APITest.cc
+++ b/modules/audio_coding/main/test/APITest.cc
@@ -50,7 +50,7 @@
 
 
 void
-APITest::Wait(WebRtc_UWord32 waitLengthMs)
+APITest::Wait(uint32_t waitLengthMs)
 {
     if(_randomTest)
     {
@@ -160,19 +160,19 @@
 
 
 
-//WebRtc_Word16
-//APITest::SetInFile(char* fileName, WebRtc_UWord16 frequencyHz)
+//int16_t
+//APITest::SetInFile(char* fileName, uint16_t frequencyHz)
 //{
 //    return _inFile.Open(fileName, frequencyHz, "rb");
 //}
 //
-//WebRtc_Word16
-//APITest::SetOutFile(char* fileName, WebRtc_UWord16 frequencyHz)
+//int16_t
+//APITest::SetOutFile(char* fileName, uint16_t frequencyHz)
 //{
 //    return _outFile.Open(fileName, frequencyHz, "wb");
 //}
 
-WebRtc_Word16
+int16_t
 APITest::SetUp()
 {
     _acmA = AudioCodingModule::Create(1);
@@ -181,8 +181,8 @@
     CodecInst dummyCodec;
     int lastPayloadType = 0;
 
-    WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
-    for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
+    int16_t numCodecs = _acmA->NumberOfCodecs();
+    for(uint8_t n = 0; n < numCodecs; n++)
     {
         AudioCodingModule::Codec(n, &dummyCodec);
         if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
@@ -250,15 +250,15 @@
     _thereIsDecoderB = true;
 
     // Register Send Codec
-    AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, &dummyCodec);
+    AudioCodingModule::Codec((uint8_t)_codecCntrA, &dummyCodec);
     CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
     _thereIsEncoderA = true;
     //
-    AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, &dummyCodec);
+    AudioCodingModule::Codec((uint8_t)_codecCntrB, &dummyCodec);
     CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
     _thereIsEncoderB = true;
 
-    WebRtc_UWord16 frequencyHz;
+    uint16_t frequencyHz;
 
     printf("\n\nAPI Test\n");
     printf("========\n");
@@ -747,8 +747,8 @@
     // Keep main thread waiting for sender/receiver
     // threads to complete
     EventWrapper* completeEvent = EventWrapper::Create();
-    WebRtc_UWord64 startTime = TickTime::MillisecondTimestamp();
-    WebRtc_UWord64 currentTime;
+    uint64_t startTime = TickTime::MillisecondTimestamp();
+    uint64_t currentTime;
     do
     {
         {
@@ -891,11 +891,11 @@
 {
     AudioCodingModule* myACM;
     Channel* myChannel;
-    WebRtc_Word32* myMinDelay;
+    int32_t* myMinDelay;
     EventWrapper* myEvent = EventWrapper::Create();
 
-    WebRtc_UWord32 inTimestamp = 0;
-    WebRtc_UWord32 outTimestamp = 0;
+    uint32_t inTimestamp = 0;
+    uint32_t outTimestamp = 0;
     double estimDelay = 0;
 
     double averageEstimDelay = 0;
@@ -937,7 +937,7 @@
             CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
 
             //std::cout << outTimestamp << std::endl << std::flush;
-            estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
+            estimDelay = (double)((uint32_t)(inTimestamp - outTimestamp)) /
                 ((double)myACM->ReceiveFrequency() / 1000.0);
 
             estimDelayCB.Update(estimDelay);
@@ -1063,7 +1063,7 @@
 
     if(!FixedPayloadTypeCodec(myCodec.plname))
     {
-        WebRtc_Word32 i;
+        int32_t i;
         for(i = 0; i < 32; i++)
         {
             if(!_payloadUsed[i])
@@ -1172,8 +1172,8 @@
             receiveACM = _acmA;
     }
 
-    WebRtc_Word32 receiveFreqHz = receiveACM->ReceiveFrequency();
-    WebRtc_Word32 playoutFreqHz = receiveACM->PlayoutFrequency();
+    int32_t receiveFreqHz = receiveACM->ReceiveFrequency();
+    int32_t playoutFreqHz = receiveACM->PlayoutFrequency();
 
     CHECK_ERROR_MT(receiveFreqHz);
     CHECK_ERROR_MT(playoutFreqHz);
@@ -1437,7 +1437,7 @@
 {
     CodecInst myCodec;
     AudioCodingModule* myACM;
-    WebRtc_UWord8* codecCntr;
+    uint8_t* codecCntr;
     bool* thereIsEncoder;
     bool* vad;
     bool* dtx;
diff --git a/modules/audio_coding/main/test/APITest.h b/modules/audio_coding/main/test/APITest.h
index ee3f5e6..ead3134 100644
--- a/modules/audio_coding/main/test/APITest.h
+++ b/modules/audio_coding/main/test/APITest.h
@@ -29,7 +29,7 @@
 
     void Perform();
 private:
-    WebRtc_Word16 SetUp();
+    int16_t SetUp();
 
     static bool PushAudioThreadA(void* obj);
     static bool PullAudioThreadA(void* obj);
@@ -63,7 +63,7 @@
 
     void ChangeCodec(char side);
 
-    void Wait(WebRtc_UWord32 waitLengthMs);
+    void Wait(uint32_t waitLengthMs);
 
     void LookForDTMF(char side);
 
@@ -99,9 +99,9 @@
 
     //--- I/O params
     // A
-    WebRtc_Word32 _outFreqHzA;
+    int32_t _outFreqHzA;
     // B
-    WebRtc_Word32 _outFreqHzB;
+    int32_t _outFreqHzB;
 
     // Should we write to file.
     // we might skip writing to file if we
@@ -120,8 +120,8 @@
     EventWrapper* _apiEventB;       // API calls
 
     // keep track of the codec in either side.
-    WebRtc_UWord8 _codecCntrA;
-    WebRtc_UWord8 _codecCntrB;
+    uint8_t _codecCntrA;
+    uint8_t _codecCntrB;
 
     // Is set to true if there is no encoder in either side
     bool _thereIsEncoderA;
@@ -137,8 +137,8 @@
     bool             _sendDTXB;
     ACMVADMode       _sendVADModeB;
 
-    WebRtc_Word32    _minDelayA;
-    WebRtc_Word32    _minDelayB;
+    int32_t    _minDelayA;
+    int32_t    _minDelayB;
     bool             _payloadUsed[32];
 
     AudioPlayoutMode    _playoutModeA;
diff --git a/modules/audio_coding/main/test/Channel.cc b/modules/audio_coding/main/test/Channel.cc
index 55ffef5..c06e452 100644
--- a/modules/audio_coding/main/test/Channel.cc
+++ b/modules/audio_coding/main/test/Channel.cc
@@ -19,18 +19,18 @@
 
 namespace webrtc {
 
-WebRtc_Word32
+int32_t
 Channel::SendData(
         const FrameType       frameType,
-        const WebRtc_UWord8   payloadType,
-        const WebRtc_UWord32  timeStamp,
-        const WebRtc_UWord8*  payloadData,
-        const WebRtc_UWord16  payloadSize,
+        const uint8_t   payloadType,
+        const uint32_t  timeStamp,
+        const uint8_t*  payloadData,
+        const uint16_t  payloadSize,
         const RTPFragmentationHeader* fragmentation)
 {
     WebRtcRTPHeader rtpInfo;
-    WebRtc_Word32   status;
-    WebRtc_UWord16  payloadDataSize = payloadSize;
+    int32_t   status;
+    uint16_t  payloadDataSize = payloadSize;
 
     rtpInfo.header.markerBit = false;
     rtpInfo.header.ssrc = 0;
@@ -60,10 +60,10 @@
         {
             // only 0x80 if we have multiple blocks
             _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
-            WebRtc_UWord32 REDheader =  (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
-            _payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
-            _payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
-            _payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);
+            uint32_t REDheader =  (((uint32_t)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
+            _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
+            _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
+            _payloadData[3] = uint8_t(REDheader & 0x000000FF);
 
             _payloadData[4] = fragmentation->fragmentationPlType[0];
             // copy the RED data
@@ -81,7 +81,7 @@
             memcpy(_payloadData,
                 payloadData + fragmentation->fragmentationOffset[0],
                 fragmentation->fragmentationLength[0]);
-            payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
+            payloadDataSize = uint16_t(fragmentation->fragmentationLength[0]);
             rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
         }
     }
@@ -108,7 +108,7 @@
     _channelCritSect->Enter();
     if(_saveBitStream)
     {
-        //fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
+        //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
     }
 
     if(!_isStereo)
@@ -138,7 +138,7 @@
 void
 Channel::CalcStatistics(
     WebRtcRTPHeader& rtpInfo,
-    WebRtc_UWord16   payloadSize)
+    uint16_t   payloadSize)
 {
     int n;
     if((rtpInfo.header.payloadType != _lastPayloadType) &&
@@ -175,8 +175,8 @@
     {
         if(!currentPayloadStr->newPacket)
         {
-            WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
-                (WebRtc_UWord32)currentPayloadStr->lastTimestamp);
+            uint32_t lastFrameSizeSample = (uint32_t)((uint32_t)rtpInfo.header.timestamp -
+                (uint32_t)currentPayloadStr->lastTimestamp);
             assert(lastFrameSizeSample > 0);
             int k = 0;
             while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
@@ -187,7 +187,7 @@
             }
             ACMTestFrameSizeStats* currentFrameSizeStats =
                 &(currentPayloadStr->frameSizeStats[k]);
-            currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;
+            currentFrameSizeStats->frameSizeSample = (int16_t)lastFrameSizeSample;
 
             // increment the number of encoded samples.
             currentFrameSizeStats->totalEncodedSamples +=
@@ -235,7 +235,7 @@
     }
 }
 
-Channel::Channel(WebRtc_Word16 chID) :
+Channel::Channel(int16_t chID) :
 _receiverACM(NULL),
 _seqNo(0),
 _channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
@@ -315,7 +315,7 @@
     _channelCritSect->Leave();
 }
 
-WebRtc_Word16
+int16_t
 Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
 {
     _channelCritSect->Enter();
@@ -355,12 +355,12 @@
 }
 
 void
-Channel::Stats(WebRtc_UWord32* numPackets)
+Channel::Stats(uint32_t* numPackets)
 {
     _channelCritSect->Enter();
     int k;
     int n;
-    memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
+    memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
     for(k = 0; k < MAX_NUM_PAYLOADS; k++)
     {
         if(_payloadStats[k].payloadType == -1)
@@ -382,20 +382,20 @@
 }
 
 void
-Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
+Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte)
 {
     _channelCritSect->Enter();
 
     int k;
     int n;
-    memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
+    memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
     for(k = 0; k < MAX_NUM_PAYLOADS; k++)
     {
         if(_payloadStats[k].payloadType == -1)
         {
             break;
         }
-        payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
+        payloadType[k] = (uint8_t)_payloadStats[k].payloadType;
         payloadLenByte[k] = 0;
         for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
         {
@@ -403,7 +403,7 @@
             {
                 break;
             }
-            payloadLenByte[k] += (WebRtc_UWord16)
+            payloadLenByte[k] += (uint16_t)
                 _payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
         }
     }
@@ -453,10 +453,10 @@
 
 }
 
-WebRtc_UWord32
+uint32_t
 Channel::LastInTimestamp()
 {
-    WebRtc_UWord32 timestamp;
+    uint32_t timestamp;
     _channelCritSect->Enter();
     timestamp = _lastInTimestamp;
     _channelCritSect->Leave();
@@ -467,7 +467,7 @@
 Channel::BitRate()
 {
     double rate;
-    WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
+    uint64_t currTime = TickTime::MillisecondTimestamp();
     _channelCritSect->Enter();
     rate =   ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
     _channelCritSect->Leave();
diff --git a/modules/audio_coding/main/test/Channel.h b/modules/audio_coding/main/test/Channel.h
index 7484584..c0bf7f3 100644
--- a/modules/audio_coding/main/test/Channel.h
+++ b/modules/audio_coding/main/test/Channel.h
@@ -26,11 +26,11 @@
 
 struct ACMTestFrameSizeStats
 {
-    WebRtc_UWord16 frameSizeSample;
-    WebRtc_Word16  maxPayloadLen;
-    WebRtc_UWord32 numPackets;
-    WebRtc_UWord64 totalPayloadLenByte;
-    WebRtc_UWord64 totalEncodedSamples;
+    uint16_t frameSizeSample;
+    int16_t  maxPayloadLen;
+    uint32_t numPackets;
+    uint64_t totalPayloadLenByte;
+    uint64_t totalEncodedSamples;
     double         rateBitPerSec;
     double         usageLenSec;
 
@@ -39,9 +39,9 @@
 struct ACMTestPayloadStats
 {
     bool                  newPacket;
-    WebRtc_Word16         payloadType;
-    WebRtc_Word16         lastPayloadLenByte;
-    WebRtc_UWord32        lastTimestamp;
+    int16_t         payloadType;
+    int16_t         lastPayloadLenByte;
+    uint32_t        lastTimestamp;
     ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
 };
 
@@ -50,15 +50,15 @@
 public:
 
     Channel(
-        WebRtc_Word16 chID = -1);
+        int16_t chID = -1);
     ~Channel();
 
-    WebRtc_Word32 SendData(
+    int32_t SendData(
         const FrameType       frameType,
-        const WebRtc_UWord8   payloadType,
-        const WebRtc_UWord32  timeStamp,
-        const WebRtc_UWord8*  payloadData,
-        const WebRtc_UWord16  payloadSize,
+        const uint8_t   payloadType,
+        const uint32_t  timeStamp,
+        const uint8_t*  payloadData,
+        const uint16_t  payloadSize,
         const RTPFragmentationHeader* fragmentation);
 
     void RegisterReceiverACM(
@@ -66,16 +66,16 @@
 
     void ResetStats();
 
-    WebRtc_Word16 Stats(
+    int16_t Stats(
         CodecInst&           codecInst,
         ACMTestPayloadStats& payloadStats);
 
     void Stats(
-        WebRtc_UWord32* numPackets);
+        uint32_t* numPackets);
 
     void Stats(
-        WebRtc_UWord8*  payloadLenByte,
-        WebRtc_UWord32* payloadType);
+        uint8_t*  payloadLenByte,
+        uint32_t* payloadType);
 
     void PrintStats(
         CodecInst& codecInst);
@@ -85,7 +85,7 @@
         _isStereo = isStereo;
     }
 
-    WebRtc_UWord32 LastInTimestamp();
+    uint32_t LastInTimestamp();
 
     void SetFECTestWithPacketLoss(bool usePacketLoss)
     {
@@ -97,27 +97,27 @@
 private:
     void CalcStatistics(
         WebRtcRTPHeader& rtpInfo,
-        WebRtc_UWord16   payloadSize);
+        uint16_t   payloadSize);
 
     AudioCodingModule*      _receiverACM;
-    WebRtc_UWord16          _seqNo;
+    uint16_t          _seqNo;
     // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
-    WebRtc_UWord8           _payloadData[60 * 32 * 2 * 2];
+    uint8_t           _payloadData[60 * 32 * 2 * 2];
 
     CriticalSectionWrapper* _channelCritSect;
     FILE*                   _bitStreamFile;
     bool                    _saveBitStream;
-    WebRtc_Word16           _lastPayloadType;
+    int16_t           _lastPayloadType;
     ACMTestPayloadStats     _payloadStats[MAX_NUM_PAYLOADS];
     bool                    _isStereo;
     WebRtcRTPHeader         _rtpInfo;
     bool                    _leftChannel;
-    WebRtc_UWord32          _lastInTimestamp;
+    uint32_t          _lastInTimestamp;
     // FEC Test variables
-    WebRtc_Word16           _packetLoss;
+    int16_t           _packetLoss;
     bool                    _useFECTestWithPacketLoss;
-    WebRtc_UWord64          _beginTime;
-    WebRtc_UWord64          _totalBytes;
+    uint64_t          _beginTime;
+    uint64_t          _totalBytes;
 };
 
 } // namespace webrtc
diff --git a/modules/audio_coding/main/test/EncodeDecodeTest.cc b/modules/audio_coding/main/test/EncodeDecodeTest.cc
index c4f9a47..58e6299 100644
--- a/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -28,7 +28,7 @@
 namespace webrtc {
 
 TestPacketization::TestPacketization(RTPStream *rtpStream,
-                                     WebRtc_UWord16 frequency)
+                                     uint16_t frequency)
     : _rtpStream(rtpStream),
       _frequency(frequency),
       _seqNo(0) {
@@ -36,12 +36,12 @@
 
 TestPacketization::~TestPacketization() { }
 
-WebRtc_Word32 TestPacketization::SendData(
+int32_t TestPacketization::SendData(
     const FrameType /* frameType */,
-    const WebRtc_UWord8 payloadType,
-    const WebRtc_UWord32 timeStamp,
-    const WebRtc_UWord8* payloadData,
-    const WebRtc_UWord16 payloadSize,
+    const uint8_t payloadType,
+    const uint32_t timeStamp,
+    const uint8_t* payloadData,
+    const uint16_t payloadSize,
     const RTPFragmentationHeader* /* fragmentation */) {
   _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
                     _frequency);
@@ -103,7 +103,7 @@
 bool Sender::Add10MsData() {
   if (!_pcmFile.EndOfFile()) {
     _pcmFile.Read10MsData(_audioFrame);
-    WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
+    int32_t ok = _acm->Add10MsData(_audioFrame);
     if (ok != 0) {
       printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
       exit(1);
@@ -114,7 +114,7 @@
 }
 
 bool Sender::Process() {
-  WebRtc_Word32 ok = _acm->Process();
+  int32_t ok = _acm->Process();
   if (ok < 0) {
     printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
     exit(1);
@@ -145,7 +145,7 @@
 
   noOfCodecs = acm->NumberOfCodecs();
   for (int i = 0; i < noOfCodecs; i++) {
-    acm->Codec((WebRtc_UWord8) i, &recvCodec);
+    acm->Codec((uint8_t) i, &recvCodec);
     if (acm->RegisterReceiveCodec(recvCodec) != 0) {
       printf("Unable to register codec: for run: codecId: %d\n", codeId);
       exit(1);
@@ -177,7 +177,7 @@
   }
 
   _realPayloadSizeBytes = 0;
-  _playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
+  _playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
   _frequency = playSampFreq;
   _acm = acm;
   _firstTime = true;
@@ -207,7 +207,7 @@
       }
    }
 
-   WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
+   int32_t ok = _acm->IncomingPacket(_incomingPayload,
                                            _realPayloadSizeBytes, _rtpInfo);
    if (ok != 0) {
      printf("Error when inserting packet to ACM, for run: codecId: %d\n",
@@ -239,8 +239,8 @@
 }
 
 void Receiver::Run() {
-  WebRtc_UWord8 counter500Ms = 50;
-  WebRtc_UWord32 clock = 0;
+  uint8_t counter500Ms = 50;
+  uint32_t clock = 0;
 
   while (counter500Ms > 0) {
     if (clock == 0 || clock >= _nextTime) {
diff --git a/modules/audio_coding/main/test/EncodeDecodeTest.h b/modules/audio_coding/main/test/EncodeDecodeTest.h
index f407a6b..9b58d4d 100644
--- a/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -26,22 +26,22 @@
 // TestPacketization callback which writes the encoded payloads to file
 class TestPacketization: public AudioPacketizationCallback {
  public:
-  TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
+  TestPacketization(RTPStream *rtpStream, uint16_t frequency);
   ~TestPacketization();
-  virtual WebRtc_Word32 SendData(const FrameType frameType,
-                                 const WebRtc_UWord8 payloadType,
-                                 const WebRtc_UWord32 timeStamp,
-                                 const WebRtc_UWord8* payloadData,
-                                 const WebRtc_UWord16 payloadSize,
-                                 const RTPFragmentationHeader* fragmentation);
+  virtual int32_t SendData(const FrameType frameType,
+                           const uint8_t payloadType,
+                           const uint32_t timeStamp,
+                           const uint8_t* payloadData,
+                           const uint16_t payloadSize,
+                           const RTPFragmentationHeader* fragmentation);
 
  private:
-  static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
-                            WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
-                            WebRtc_UWord32 ssrc);
+  static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
+                            int16_t seqNo, uint32_t timeStamp,
+                            uint32_t ssrc);
   RTPStream* _rtpStream;
-  WebRtc_Word32 _frequency;
-  WebRtc_Word16 _seqNo;
+  int32_t _frequency;
+  int16_t _seqNo;
 };
 
 class Sender {
@@ -54,8 +54,8 @@
   bool Process();
 
   //for auto_test and logging
-  WebRtc_UWord8 testMode;
-  WebRtc_UWord8 codeId;
+  uint8_t testMode;
+  uint8_t codeId;
 
  private:
   AudioCodingModule* _acm;
@@ -74,22 +74,22 @@
   bool PlayoutData();
 
   //for auto_test and logging
-  WebRtc_UWord8 codeId;
-  WebRtc_UWord8 testMode;
+  uint8_t codeId;
+  uint8_t testMode;
 
  private:
   AudioCodingModule* _acm;
   RTPStream* _rtpStream;
   PCMFile _pcmFile;
-  WebRtc_Word16* _playoutBuffer;
-  WebRtc_UWord16 _playoutLengthSmpls;
-  WebRtc_UWord8 _incomingPayload[MAX_INCOMING_PAYLOAD];
-  WebRtc_UWord16 _payloadSizeBytes;
-  WebRtc_UWord16 _realPayloadSizeBytes;
-  WebRtc_Word32 _frequency;
+  int16_t* _playoutBuffer;
+  uint16_t _playoutLengthSmpls;
+  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+  uint16_t _payloadSizeBytes;
+  uint16_t _realPayloadSizeBytes;
+  int32_t _frequency;
   bool _firstTime;
   WebRtcRTPHeader _rtpInfo;
-  WebRtc_UWord32 _nextTime;
+  uint32_t _nextTime;
 };
 
 class EncodeDecodeTest: public ACMTest {
@@ -98,8 +98,8 @@
   EncodeDecodeTest(int testMode);
   virtual void Perform();
 
-  WebRtc_UWord16 _playoutFreq;
-  WebRtc_UWord8 _testMode;
+  uint16_t _playoutFreq;
+  uint8_t _testMode;
 
  private:
   void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
diff --git a/modules/audio_coding/main/test/PCMFile.cc b/modules/audio_coding/main/test/PCMFile.cc
index fbe73f5..0b61bbb 100644
--- a/modules/audio_coding/main/test/PCMFile.cc
+++ b/modules/audio_coding/main/test/PCMFile.cc
@@ -30,11 +30,11 @@
       rewinded_(false),
       read_stereo_(false),
       save_stereo_(false) {
-  timestamp_ = (((WebRtc_UWord32)rand() & 0x0000FFFF) << 16) |
-      ((WebRtc_UWord32)rand() & 0x0000FFFF);
+  timestamp_ = (((uint32_t)rand() & 0x0000FFFF) << 16) |
+      ((uint32_t)rand() & 0x0000FFFF);
 }
 
-PCMFile::PCMFile(WebRtc_UWord32 timestamp)
+PCMFile::PCMFile(uint32_t timestamp)
     : pcm_file_(NULL),
       samples_10ms_(160),
       frequency_(16000),
@@ -46,13 +46,12 @@
   timestamp_ = timestamp;
 }
 
-WebRtc_Word16 PCMFile::ChooseFile(std::string* file_name,
-                                  WebRtc_Word16 max_len) {
+int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len) {
   char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
 
   EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
   tmp_name[MAX_FILE_NAME_LENGTH_BYTE - 1] = '\0';
-  WebRtc_Word16 n = 0;
+  int16_t n = 0;
 
   // Removing leading spaces.
   while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (tmp_name[n] != 0)
@@ -64,7 +63,7 @@
   }
 
   // Removing trailing spaces.
-  n = (WebRtc_Word16)(strlen(tmp_name) - 1);
+  n = (int16_t)(strlen(tmp_name) - 1);
   if (n >= 0) {
     while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (n >= 0)) {
       n--;
@@ -74,7 +73,7 @@
     tmp_name[n + 1] = '\0';
   }
 
-  WebRtc_Word16 len = (WebRtc_Word16) strlen(tmp_name);
+  int16_t len = (int16_t) strlen(tmp_name);
   if (len > max_len) {
     return -1;
   }
@@ -85,14 +84,14 @@
   return 0;
 }
 
-WebRtc_Word16 PCMFile::ChooseFile(std::string* file_name,
-                                  WebRtc_Word16 max_len,
-                                  WebRtc_UWord16* frequency_hz) {
+int16_t PCMFile::ChooseFile(std::string* file_name,
+                            int16_t max_len,
+                            uint16_t* frequency_hz) {
   char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
 
   EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
   tmp_name[MAX_FILE_NAME_LENGTH_BYTE - 1] = '\0';
-  WebRtc_Word16 n = 0;
+  int16_t n = 0;
 
   // Removing trailing spaces.
   while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (tmp_name[n] != 0)
@@ -104,7 +103,7 @@
   }
 
   // Removing trailing spaces.
-  n = (WebRtc_Word16)(strlen(tmp_name) - 1);
+  n = (int16_t)(strlen(tmp_name) - 1);
   if (n >= 0) {
     while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (n >= 0)) {
       n--;
@@ -114,7 +113,7 @@
     tmp_name[n + 1] = '\0';
   }
 
-  WebRtc_Word16 len = (WebRtc_Word16) strlen(tmp_name);
+  int16_t len = (int16_t) strlen(tmp_name);
   if (len > max_len) {
     return -1;
   }
@@ -125,42 +124,42 @@
   printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
          *frequency_hz);
   EXPECT_TRUE(fgets(tmp_name, 10, stdin) != NULL);
-  WebRtc_UWord16 tmp_frequency = (WebRtc_UWord16) atoi(tmp_name);
+  uint16_t tmp_frequency = (uint16_t) atoi(tmp_name);
   if (tmp_frequency > 0) {
     *frequency_hz = tmp_frequency;
   }
   return 0;
 }
 
-void PCMFile::Open(const std::string& file_name, WebRtc_UWord16 frequency,
+void PCMFile::Open(const std::string& file_name, uint16_t frequency,
                    const char* mode, bool auto_rewind) {
   if ((pcm_file_ = fopen(file_name.c_str(), mode)) == NULL) {
     printf("Cannot open file %s.\n", file_name.c_str());
     ADD_FAILURE() << "Unable to read file";
   }
   frequency_ = frequency;
-  samples_10ms_ = (WebRtc_UWord16)(frequency_ / 100);
+  samples_10ms_ = (uint16_t)(frequency_ / 100);
   auto_rewind_ = auto_rewind;
   end_of_file_ = false;
   rewinded_ = false;
 }
 
-WebRtc_Word32 PCMFile::SamplingFrequency() const {
+int32_t PCMFile::SamplingFrequency() const {
   return frequency_;
 }
 
-WebRtc_UWord16 PCMFile::PayloadLength10Ms() const {
+uint16_t PCMFile::PayloadLength10Ms() const {
   return samples_10ms_;
 }
 
-WebRtc_Word32 PCMFile::Read10MsData(AudioFrame& audio_frame) {
-  WebRtc_UWord16 channels = 1;
+int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
+  uint16_t channels = 1;
   if (read_stereo_) {
     channels = 2;
   }
 
-  WebRtc_Word32 payload_size = (WebRtc_Word32) fread(audio_frame.data_,
-                                                    sizeof(WebRtc_UWord16),
+  int32_t payload_size = (int32_t) fread(audio_frame.data_,
+                                                    sizeof(uint16_t),
                                                     samples_10ms_ * channels,
                                                     pcm_file_);
   if (payload_size < samples_10ms_ * channels) {
@@ -185,20 +184,20 @@
 void PCMFile::Write10MsData(AudioFrame& audio_frame) {
   if (audio_frame.num_channels_ == 1) {
     if (!save_stereo_) {
-      if (fwrite(audio_frame.data_, sizeof(WebRtc_UWord16),
+      if (fwrite(audio_frame.data_, sizeof(uint16_t),
                  audio_frame.samples_per_channel_, pcm_file_) !=
           static_cast<size_t>(audio_frame.samples_per_channel_)) {
         return;
       }
     } else {
-      WebRtc_Word16* stereo_audio =
-          new WebRtc_Word16[2 * audio_frame.samples_per_channel_];
+      int16_t* stereo_audio =
+          new int16_t[2 * audio_frame.samples_per_channel_];
       int k;
       for (k = 0; k < audio_frame.samples_per_channel_; k++) {
         stereo_audio[k << 1] = audio_frame.data_[k];
         stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
       }
-      if (fwrite(stereo_audio, sizeof(WebRtc_Word16),
+      if (fwrite(stereo_audio, sizeof(int16_t),
                  2 * audio_frame.samples_per_channel_, pcm_file_) !=
           static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
         return;
@@ -206,7 +205,7 @@
       delete[] stereo_audio;
     }
   } else {
-    if (fwrite(audio_frame.data_, sizeof(WebRtc_Word16),
+    if (fwrite(audio_frame.data_, sizeof(int16_t),
                audio_frame.num_channels_ * audio_frame.samples_per_channel_,
                pcm_file_) != static_cast<size_t>(
             audio_frame.num_channels_ * audio_frame.samples_per_channel_)) {
@@ -215,9 +214,9 @@
   }
 }
 
-void PCMFile::Write10MsData(WebRtc_Word16* playout_buffer,
-                            WebRtc_UWord16 length_smpls) {
-  if (fwrite(playout_buffer, sizeof(WebRtc_UWord16),
+void PCMFile::Write10MsData(int16_t* playout_buffer,
+                            uint16_t length_smpls) {
+  if (fwrite(playout_buffer, sizeof(uint16_t),
              length_smpls, pcm_file_) != length_smpls) {
     return;
   }
diff --git a/modules/audio_coding/main/test/PCMFile.h b/modules/audio_coding/main/test/PCMFile.h
index 2dbfecd..bd2ce21 100644
--- a/modules/audio_coding/main/test/PCMFile.h
+++ b/modules/audio_coding/main/test/PCMFile.h
@@ -23,45 +23,45 @@
 class PCMFile {
  public:
   PCMFile();
-  PCMFile(WebRtc_UWord32 timestamp);
+  PCMFile(uint32_t timestamp);
   ~PCMFile() {
     if (pcm_file_ != NULL) {
       fclose(pcm_file_);
     }
   }
 
-  void Open(const std::string& filename, WebRtc_UWord16 frequency,
+  void Open(const std::string& filename, uint16_t frequency,
                   const char* mode, bool auto_rewind = false);
 
-  WebRtc_Word32 Read10MsData(AudioFrame& audio_frame);
+  int32_t Read10MsData(AudioFrame& audio_frame);
 
-  void Write10MsData(WebRtc_Word16 *playout_buffer,
-                     WebRtc_UWord16 length_smpls);
+  void Write10MsData(int16_t *playout_buffer,
+                     uint16_t length_smpls);
   void Write10MsData(AudioFrame& audio_frame);
 
-  WebRtc_UWord16 PayloadLength10Ms() const;
-  WebRtc_Word32 SamplingFrequency() const;
+  uint16_t PayloadLength10Ms() const;
+  int32_t SamplingFrequency() const;
   void Close();
   bool EndOfFile() const {
     return end_of_file_;
   }
   void Rewind();
-  static WebRtc_Word16 ChooseFile(std::string* file_name,
-                                  WebRtc_Word16 max_len,
-                                  WebRtc_UWord16* frequency_hz);
-  static WebRtc_Word16 ChooseFile(std::string* file_name,
-                                  WebRtc_Word16 max_len);
+  static int16_t ChooseFile(std::string* file_name,
+                            int16_t max_len,
+                            uint16_t* frequency_hz);
+  static int16_t ChooseFile(std::string* file_name,
+                            int16_t max_len);
   bool Rewinded();
   void SaveStereo(bool is_stereo = true);
   void ReadStereo(bool is_stereo = true);
  private:
   FILE* pcm_file_;
-  WebRtc_UWord16 samples_10ms_;
-  WebRtc_Word32 frequency_;
+  uint16_t samples_10ms_;
+  int32_t frequency_;
   bool end_of_file_;
   bool auto_rewind_;
   bool rewinded_;
-  WebRtc_UWord32 timestamp_;
+  uint32_t timestamp_;
   bool read_stereo_;
   bool save_stereo_;
 };
diff --git a/modules/audio_coding/main/test/RTPFile.cc b/modules/audio_coding/main/test/RTPFile.cc
index 37f9d3c..47850ae 100644
--- a/modules/audio_coding/main/test/RTPFile.cc
+++ b/modules/audio_coding/main/test/RTPFile.cc
@@ -25,23 +25,23 @@
 
 namespace webrtc {
 
-void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
+void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader)
 {
     rtpInfo->header.payloadType = rtpHeader[1];
-    rtpInfo->header.sequenceNumber = (static_cast<WebRtc_UWord16>(rtpHeader[2])<<8) | rtpHeader[3];
-    rtpInfo->header.timestamp = (static_cast<WebRtc_UWord32>(rtpHeader[4])<<24) |
-                         (static_cast<WebRtc_UWord32>(rtpHeader[5])<<16) |
-                         (static_cast<WebRtc_UWord32>(rtpHeader[6])<<8) |
+    rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2])<<8) | rtpHeader[3];
+    rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4])<<24) |
+                         (static_cast<uint32_t>(rtpHeader[5])<<16) |
+                         (static_cast<uint32_t>(rtpHeader[6])<<8) |
                          rtpHeader[7];
-    rtpInfo->header.ssrc = (static_cast<WebRtc_UWord32>(rtpHeader[8])<<24) |
-                    (static_cast<WebRtc_UWord32>(rtpHeader[9])<<16) |
-                    (static_cast<WebRtc_UWord32>(rtpHeader[10])<<8) |
+    rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8])<<24) |
+                    (static_cast<uint32_t>(rtpHeader[9])<<16) |
+                    (static_cast<uint32_t>(rtpHeader[10])<<8) |
                     rtpHeader[11];
 }
 
-void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader, 
-                              WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
-                              WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc)
+void RTPStream::MakeRTPheader(uint8_t* rtpHeader, 
+                              uint8_t payloadType, int16_t seqNo,
+                              uint32_t timeStamp, uint32_t ssrc)
 {
     rtpHeader[0]=(unsigned char)0x80;
     rtpHeader[1]=(unsigned char)(payloadType & 0xFF);
@@ -61,9 +61,9 @@
 }
 
 
-RTPPacket::RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
-                                    WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                    WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp,
+                                    int16_t seqNo, const uint8_t* payloadData,
+                                    uint16_t payloadSize, uint32_t frequency)
                                     :
 payloadType(payloadType),
 timeStamp(timeStamp),
@@ -73,7 +73,7 @@
 {
     if (payloadSize > 0)
     {
-        this->payloadData = new WebRtc_UWord8[payloadSize];
+        this->payloadData = new uint8_t[payloadSize];
         memcpy(this->payloadData, payloadData, payloadSize);
     }
 }
@@ -94,9 +94,9 @@
 }
 
 void
-RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
-                                    const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                    const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
+                                    const int16_t seqNo, const uint8_t* payloadData,
+                                    const uint16_t payloadSize, uint32_t frequency)
 {
     RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, payloadSize, frequency);
     _queueRWLock->AcquireLockExclusive();
@@ -104,11 +104,11 @@
     _queueRWLock->ReleaseLockExclusive();
 }
 
-WebRtc_UWord16
+uint16_t
 RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
-                WebRtc_UWord8* payloadData,
-                WebRtc_UWord16 payloadSize,
-                WebRtc_UWord32* offset)
+                uint8_t* payloadData,
+                uint16_t payloadSize,
+                uint32_t* offset)
 {
     _queueRWLock->AcquireLockShared();
     RTPPacket *packet = _rtpQueue.front();
@@ -165,7 +165,7 @@
 {
     // Write data in a format that NetEQ and RTP Play can parse
     fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
-    WebRtc_UWord32 dummy_variable = 0;
+    uint32_t dummy_variable = 0;
     // should be converted to network endian format, but does not matter when 0
     if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
       return;
@@ -187,8 +187,8 @@
 
 void RTPFile::ReadHeader()
 {
-    WebRtc_UWord32 start_sec, start_usec, source;
-    WebRtc_UWord16 port, padding;
+    uint32_t start_sec, start_usec, source;
+    uint16_t port, padding;
     char fileHeader[40];
     EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
     EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
@@ -203,16 +203,16 @@
     padding=ntohs(padding);
 }
 
-void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
-                    const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                    const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
+                    const int16_t seqNo, const uint8_t* payloadData,
+                    const uint16_t payloadSize, uint32_t frequency)
 {
     /* write RTP packet to file */
-    WebRtc_UWord8 rtpHeader[12];
+    uint8_t rtpHeader[12];
     MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
-    WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
-    WebRtc_UWord16 plen = htons(12 + payloadSize);
-    WebRtc_UWord32 offsetMs;
+    uint16_t lengthBytes = htons(12 + payloadSize + 8);
+    uint16_t plen = htons(12 + payloadSize);
+    uint32_t offsetMs;
 
     offsetMs = (timeStamp/(frequency/1000));
     offsetMs = htonl(offsetMs);
@@ -233,14 +233,14 @@
     }
 }
 
-WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
-                   WebRtc_UWord8* payloadData,
-                   WebRtc_UWord16 payloadSize,
-                   WebRtc_UWord32* offset)
+uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
+                       uint8_t* payloadData,
+                       uint16_t payloadSize,
+                       uint32_t* offset)
 {
-    WebRtc_UWord16 lengthBytes;
-    WebRtc_UWord16 plen;
-    WebRtc_UWord8 rtpHeader[12];
+    uint16_t lengthBytes;
+    uint16_t plen;
+    uint8_t rtpHeader[12];
     size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
     /* Check if we have reached end of file. */
     if ((read_len == 0) && feof(_rtpFile))
diff --git a/modules/audio_coding/main/test/RTPFile.h b/modules/audio_coding/main/test/RTPFile.h
index b5f5299..b6bbf87 100644
--- a/modules/audio_coding/main/test/RTPFile.h
+++ b/modules/audio_coding/main/test/RTPFile.h
@@ -25,38 +25,38 @@
 public:
     virtual ~RTPStream(){}
 
-    virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
-                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency) = 0;
+    virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
+                                     const int16_t seqNo, const uint8_t* payloadData,
+                                     const uint16_t payloadSize, uint32_t frequency) = 0;
 
     // Returns the packet's payload size. Zero should be treated as an
     // end-of-stream (in the case that EndOfFile() is true) or an error.
-    virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
-                    WebRtc_UWord8* payloadData,
-                    WebRtc_UWord16 payloadSize,
-                    WebRtc_UWord32* offset) = 0;
+    virtual uint16_t Read(WebRtcRTPHeader* rtpInfo,
+                          uint8_t* payloadData,
+                          uint16_t payloadSize,
+                          uint32_t* offset) = 0;
     virtual bool EndOfFile() const = 0;
 
 protected:
-    void MakeRTPheader(WebRtc_UWord8* rtpHeader, 
-                                      WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo, 
-                                      WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
-    void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader);
+    void MakeRTPheader(uint8_t* rtpHeader, 
+                                      uint8_t payloadType, int16_t seqNo, 
+                                      uint32_t timeStamp, uint32_t ssrc);
+    void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
 };
 
 class RTPPacket
 {
 public:
-    RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
-                                     WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                     WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+    RTPPacket(uint8_t payloadType, uint32_t timeStamp,
+                                     int16_t seqNo, const uint8_t* payloadData,
+                                     uint16_t payloadSize, uint32_t frequency);
     ~RTPPacket();
-    WebRtc_UWord8 payloadType;
-    WebRtc_UWord32 timeStamp;
-    WebRtc_Word16 seqNo;
-    WebRtc_UWord8* payloadData;
-    WebRtc_UWord16 payloadSize;
-    WebRtc_UWord32 frequency;
+    uint8_t payloadType;
+    uint32_t timeStamp;
+    int16_t seqNo;
+    uint8_t* payloadData;
+    uint16_t payloadSize;
+    uint32_t frequency;
 };
 
 class RTPBuffer : public RTPStream
@@ -64,13 +64,13 @@
 public:
     RTPBuffer();
     ~RTPBuffer();
-    void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
-                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
-    WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
-                    WebRtc_UWord8* payloadData,
-                    WebRtc_UWord16 payloadSize,
-                    WebRtc_UWord32* offset);
+    void Write(const uint8_t payloadType, const uint32_t timeStamp,
+                                     const int16_t seqNo, const uint8_t* payloadData,
+                                     const uint16_t payloadSize, uint32_t frequency);
+    uint16_t Read(WebRtcRTPHeader* rtpInfo,
+                  uint8_t* payloadData,
+                  uint16_t payloadSize,
+                  uint32_t* offset);
     virtual bool EndOfFile() const;
 private:
     RWLockWrapper*             _queueRWLock;
@@ -86,13 +86,13 @@
     void Close();
     void WriteHeader();
     void ReadHeader();
-    void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
-                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
-                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
-    WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
-                    WebRtc_UWord8* payloadData,
-                    WebRtc_UWord16 payloadSize,
-                    WebRtc_UWord32* offset);
+    void Write(const uint8_t payloadType, const uint32_t timeStamp,
+                                     const int16_t seqNo, const uint8_t* payloadData,
+                                     const uint16_t payloadSize, uint32_t frequency);
+    uint16_t Read(WebRtcRTPHeader* rtpInfo,
+                  uint8_t* payloadData,
+                  uint16_t payloadSize,
+                  uint32_t* offset);
     bool EndOfFile() const { return _rtpEOF; }
 private:
     FILE*   _rtpFile;
diff --git a/modules/audio_coding/main/test/SpatialAudio.cc b/modules/audio_coding/main/test/SpatialAudio.cc
index 15875ee..ec8dd1f 100644
--- a/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/modules/audio_coding/main/test/SpatialAudio.cc
@@ -38,7 +38,7 @@
     _outFile.Close();
 }
 
-WebRtc_Word16 
+int16_t 
 SpatialAudio::Setup()
 {
     // Create ACMs and the Channel;
@@ -53,7 +53,7 @@
     // Register the receiver ACM in channel
     _channel->RegisterReceiverACM(_acmReceiver);
 
-    WebRtc_UWord16 sampFreqHz = 32000;
+    uint16_t sampFreqHz = 32000;
 
     const std::string file_name =
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
@@ -79,9 +79,9 @@
     // Register all available codes as receiving codecs.
     CodecInst codecInst;
     int status;
-    WebRtc_UWord8 num_encoders = _acmReceiver->NumberOfCodecs();
+    uint8_t num_encoders = _acmReceiver->NumberOfCodecs();
     // Register all available codes as receiving codecs once more.
-    for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+    for (uint8_t n = 0; n < num_encoders; n++) {
       status = _acmReceiver->Codec(n, &codecInst);
       if (status < 0) {
         printf("Error in Codec(), no matching codec found");
@@ -109,11 +109,11 @@
     Setup();
 
     CodecInst codecInst;
-    _acmLeft->Codec((WebRtc_UWord8)1, &codecInst);
+    _acmLeft->Codec((uint8_t)1, &codecInst);
     CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
     EncodeDecode();
 
-    WebRtc_Word16 pannCntr = 0;
+    int16_t pannCntr = 0;
 
     double leftPanning[NUM_PANN_COEFFS] =  
         {1.00, 0.95, 0.90, 0.85, 0.80, 0.75, 0.70, 0.60, 0.55, 0.50};
@@ -122,7 +122,7 @@
 
     while((pannCntr + 1) < NUM_PANN_COEFFS)
     {
-        _acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
+        _acmLeft->Codec((uint8_t)0, &codecInst);
         codecInst.pacsize = 480;
         CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
         CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@@ -131,7 +131,7 @@
         pannCntr++;
 
         // Change codec    
-        _acmLeft->Codec((WebRtc_UWord8)3, &codecInst);
+        _acmLeft->Codec((uint8_t)3, &codecInst);
         codecInst.pacsize = 320;
         CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
         CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@@ -144,11 +144,11 @@
         }
     }
 
-    _acmLeft->Codec((WebRtc_UWord8)4, &codecInst);
+    _acmLeft->Codec((uint8_t)4, &codecInst);
     CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
     EncodeDecode();
 
-    _acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
+    _acmLeft->Codec((uint8_t)0, &codecInst);
     codecInst.pacsize = 480;
     CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
     CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@@ -174,7 +174,7 @@
     const double rightPanning)
 {
     AudioFrame audioFrame;
-    WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
+    int32_t outFileSampFreq = _outFile.SamplingFrequency();
 
     const double rightToLeftRatio = rightPanning / leftPanning;
 
@@ -185,14 +185,14 @@
         _inFile.Read10MsData(audioFrame);
         for(int n = 0; n < audioFrame.samples_per_channel_; n++)
         {
-            audioFrame.data_[n] = (WebRtc_Word16)floor(
+            audioFrame.data_[n] = (int16_t)floor(
                 audioFrame.data_[n] * leftPanning + 0.5);
         }
         CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
 
         for(int n = 0; n < audioFrame.samples_per_channel_; n++)
         {
-            audioFrame.data_[n] = (WebRtc_Word16)floor(
+            audioFrame.data_[n] = (int16_t)floor(
                 audioFrame.data_[n] * rightToLeftRatio + 0.5);
         }
         CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
@@ -211,7 +211,7 @@
 SpatialAudio::EncodeDecode()
 {
     AudioFrame audioFrame;
-    WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
+    int32_t outFileSampFreq = _outFile.SamplingFrequency();
 
     _channel->SetIsStereo(false);
 
diff --git a/modules/audio_coding/main/test/SpatialAudio.h b/modules/audio_coding/main/test/SpatialAudio.h
index 6a88327..051d596 100644
--- a/modules/audio_coding/main/test/SpatialAudio.h
+++ b/modules/audio_coding/main/test/SpatialAudio.h
@@ -29,7 +29,7 @@
 
     void Perform();
 private:
-    WebRtc_Word16 Setup();
+    int16_t Setup();
     void EncodeDecode(double leftPanning, double rightPanning);
     void EncodeDecode();
 
diff --git a/modules/audio_coding/main/test/TestFEC.cc b/modules/audio_coding/main/test/TestFEC.cc
index 9f5f022..3aad3fe 100644
--- a/modules/audio_coding/main/test/TestFEC.cc
+++ b/modules/audio_coding/main/test/TestFEC.cc
@@ -71,13 +71,13 @@
     _acmA->InitializeReceiver();
     _acmB->InitializeReceiver();
 
-    WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+    uint8_t numEncoders = _acmA->NumberOfCodecs();
     CodecInst myCodecParam;
     if(_testMode != 0)
     {
         printf("Registering codecs at receiver... \n");
     }
-    for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+    for(uint8_t n = 0; n < numEncoders; n++)
     {
         _acmB->Codec(n, &myCodecParam);
         if(_testMode != 0)
@@ -503,19 +503,19 @@
     }
 }
 
-WebRtc_Word32 TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode)
+int32_t TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode)
 {
     if(_testMode != 0)
     {
         printf("DTX %s; VAD %s; VAD-Mode %d\n", 
             enableDTX? "ON":"OFF", 
             enableVAD? "ON":"OFF", 
-            (WebRtc_Word16)vadMode);
+            (int16_t)vadMode);
     }
     return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
 }
 
-WebRtc_Word16 TestFEC::RegisterSendCodec(char side, char* codecName, WebRtc_Word32 samplingFreqHz)
+int16_t TestFEC::RegisterSendCodec(char side, char* codecName, int32_t samplingFreqHz)
 {
     if(_testMode != 0)
     {
@@ -566,9 +566,9 @@
 {
     AudioFrame audioFrame;
 
-    WebRtc_UWord16 msecPassed = 0;
-    WebRtc_UWord32 secPassed  = 0;
-    WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+    uint16_t msecPassed = 0;
+    uint32_t secPassed  = 0;
+    int32_t outFreqHzB = _outFileB.SamplingFrequency();
 
     while(!_inFileA.EndOfFile())
     {
@@ -599,7 +599,7 @@
     _inFileA.Rewind();
 }
 
-void TestFEC::OpenOutFile(WebRtc_Word16 test_number) {
+void TestFEC::OpenOutFile(int16_t test_number) {
   std::string file_name;
   std::stringstream file_stream;
   file_stream << webrtc::test::OutputPath();
diff --git a/modules/audio_coding/main/test/TestFEC.h b/modules/audio_coding/main/test/TestFEC.h
index 00e951f..666058c 100644
--- a/modules/audio_coding/main/test/TestFEC.h
+++ b/modules/audio_coding/main/test/TestFEC.h
@@ -28,11 +28,11 @@
     // The default value of '-1' indicates that the registration is based only on codec name
     // and a sampling frequncy matching is not required. This is useful for codecs which support
     // several sampling frequency.
-    WebRtc_Word16 RegisterSendCodec(char side, char* codecName, WebRtc_Word32 sampFreqHz = -1);
+    int16_t RegisterSendCodec(char side, char* codecName, int32_t sampFreqHz = -1);
     void Run();
-    void OpenOutFile(WebRtc_Word16 testNumber);
+    void OpenOutFile(int16_t testNumber);
     void DisplaySendReceiveCodec();
-    WebRtc_Word32 SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
+    int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
     AudioCodingModule* _acmA;
     AudioCodingModule* _acmB;
 
@@ -40,7 +40,7 @@
 
     PCMFile                _inFileA;
     PCMFile                _outFileB;
-    WebRtc_Word16            _testCntr;
+    int16_t            _testCntr;
     int                    _testMode;
 };
 
diff --git a/modules/audio_coding/main/test/TestStereo.cc b/modules/audio_coding/main/test/TestStereo.cc
index e1186ba..d692703 100644
--- a/modules/audio_coding/main/test/TestStereo.cc
+++ b/modules/audio_coding/main/test/TestStereo.cc
@@ -42,15 +42,15 @@
   return;
 }
 
-WebRtc_Word32 TestPackStereo::SendData(
+int32_t TestPackStereo::SendData(
     const FrameType frame_type,
-    const WebRtc_UWord8 payload_type,
-    const WebRtc_UWord32 timestamp,
-    const WebRtc_UWord8* payload_data,
-    const WebRtc_UWord16 payload_size,
+    const uint8_t payload_type,
+    const uint32_t timestamp,
+    const uint8_t* payload_data,
+    const uint16_t payload_size,
     const RTPFragmentationHeader* fragmentation) {
   WebRtcRTPHeader rtp_info;
-  WebRtc_Word32 status = 0;
+  int32_t status = 0;
 
   rtp_info.header.markerBit = false;
   rtp_info.header.ssrc = 0;
@@ -86,11 +86,11 @@
   return status;
 }
 
-WebRtc_UWord16 TestPackStereo::payload_size() {
+uint16_t TestPackStereo::payload_size() {
   return payload_size_;
 }
 
-WebRtc_UWord32 TestPackStereo::timestamp_diff() {
+uint32_t TestPackStereo::timestamp_diff() {
   return timestamp_diff_;
 }
 
@@ -145,7 +145,7 @@
 }
 
 void TestStereo::Perform() {
-  WebRtc_UWord16 frequency_hz;
+  uint16_t frequency_hz;
   int audio_channels;
   int codec_channels;
   bool dtx;
@@ -179,21 +179,21 @@
   EXPECT_EQ(0, acm_b_->InitializeReceiver());
 
   // Register all available codes as receiving codecs.
-  WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
+  uint8_t num_encoders = acm_a_->NumberOfCodecs();
   CodecInst my_codec_param;
-  for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+  for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
     EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
   }
 
   // Test that unregister all receive codecs works.
-  for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+  for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
     EXPECT_EQ(0, acm_b_->UnregisterReceiveCodec(my_codec_param.pltype));
   }
 
   // Register all available codes as receiving codecs once more.
-  for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+  for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
     EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
   }
@@ -686,7 +686,7 @@
   RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels,
                      opus_pltype_);
   CodecInst opus_codec_param;
-  for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+  for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
     if (!strcmp(opus_codec_param.plname, "opus")) {
       opus_codec_param.channels = 1;
@@ -777,7 +777,7 @@
 //          channels         - number of channels; 1 for mono, 2 for stereo
 //          payload_type     - payload type for the codec
 void TestStereo::RegisterSendCodec(char side, char* codec_name,
-                                   WebRtc_Word32 sampling_freq_hz, int rate,
+                                   int32_t sampling_freq_hz, int rate,
                                    int pack_size, int channels,
                                    int payload_type) {
   if (test_mode_ != 0) {
@@ -793,12 +793,12 @@
   // packet. Add 0.875 to always round up to a whole byte.
   // For Celt the packet size in bytes is already counting the stereo part.
   if (!strcmp(codec_name, "CELT")) {
-    pack_size_bytes_ = (WebRtc_UWord16)(
+    pack_size_bytes_ = (uint16_t)(
         static_cast<float>(pack_size * rate) /
         static_cast<float>(sampling_freq_hz * 8) + 0.875)
         / channels;
   } else {
-    pack_size_bytes_ = (WebRtc_UWord16)(
+    pack_size_bytes_ = (uint16_t)(
         static_cast<float>(pack_size * rate) /
         static_cast<float>(sampling_freq_hz * 8) + 0.875);
   }
@@ -834,9 +834,9 @@
                      int percent_loss) {
   AudioFrame audio_frame;
 
-  WebRtc_Word32 out_freq_hz_b = out_file_.SamplingFrequency();
-  WebRtc_UWord16 rec_size;
-  WebRtc_UWord32 time_stamp_diff;
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  uint16_t rec_size;
+  uint32_t time_stamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
 
@@ -908,7 +908,7 @@
   channel->set_lost_packet(false);
 }
 
-void TestStereo::OpenOutFile(WebRtc_Word16 test_number) {
+void TestStereo::OpenOutFile(int16_t test_number) {
   std::string file_name;
   std::stringstream file_stream;
   file_stream << webrtc::test::OutputPath() << "teststereo_out_"
diff --git a/modules/audio_coding/main/test/TestStereo.h b/modules/audio_coding/main/test/TestStereo.h
index e990515..ecfce5e 100644
--- a/modules/audio_coding/main/test/TestStereo.h
+++ b/modules/audio_coding/main/test/TestStereo.h
@@ -32,26 +32,26 @@
 
   void RegisterReceiverACM(AudioCodingModule* acm);
 
-  virtual WebRtc_Word32 SendData(const FrameType frame_type,
-                                 const WebRtc_UWord8 payload_type,
-                                 const WebRtc_UWord32 timestamp,
-                                 const WebRtc_UWord8* payload_data,
-                                 const WebRtc_UWord16 payload_size,
-                                 const RTPFragmentationHeader* fragmentation);
+  virtual int32_t SendData(const FrameType frame_type,
+                           const uint8_t payload_type,
+                           const uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           const uint16_t payload_size,
+                           const RTPFragmentationHeader* fragmentation);
 
-  WebRtc_UWord16 payload_size();
-  WebRtc_UWord32 timestamp_diff();
+  uint16_t payload_size();
+  uint32_t timestamp_diff();
   void reset_payload_size();
   void set_codec_mode(StereoMonoMode mode);
   void set_lost_packet(bool lost);
 
  private:
   AudioCodingModule* receiver_acm_;
-  WebRtc_Word16 seq_no_;
-  WebRtc_UWord32 timestamp_diff_;
-  WebRtc_UWord32 last_in_timestamp_;
-  WebRtc_UWord64 total_bytes_;
-  WebRtc_UWord16 payload_size_;
+  int16_t seq_no_;
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  uint16_t payload_size_;
   StereoMonoMode codec_mode_;
   // Simulate packet losses
   bool lost_packet_;
@@ -68,20 +68,20 @@
   // codec name and a sampling frequncy matching is not required. This is useful
   // for codecs which support several sampling frequency.
   void RegisterSendCodec(char side, char* codec_name,
-                         WebRtc_Word32 samp_freq_hz, int rate, int pack_size,
+                         int32_t samp_freq_hz, int rate, int pack_size,
                          int channels, int payload_type);
 
   void Run(TestPackStereo* channel, int in_channels, int out_channels,
            int percent_loss = 0);
-  void OpenOutFile(WebRtc_Word16 test_number);
+  void OpenOutFile(int16_t test_number);
   void DisplaySendReceiveCodec();
 
-  WebRtc_Word32 SendData(const FrameType frame_type,
-                         const WebRtc_UWord8 payload_type,
-                         const WebRtc_UWord32 timestamp,
-                         const WebRtc_UWord8* payload_data,
-                         const WebRtc_UWord16 payload_size,
-                         const RTPFragmentationHeader* fragmentation);
+  int32_t SendData(const FrameType frame_type,
+                   const uint8_t payload_type,
+                   const uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   const uint16_t payload_size,
+                   const RTPFragmentationHeader* fragmentation);
 
   int test_mode_;
 
@@ -93,9 +93,9 @@
   PCMFile* in_file_stereo_;
   PCMFile* in_file_mono_;
   PCMFile out_file_;
-  WebRtc_Word16 test_cntr_;
-  WebRtc_UWord16 pack_size_samp_;
-  WebRtc_UWord16 pack_size_bytes_;
+  int16_t test_cntr_;
+  uint16_t pack_size_samp_;
+  uint16_t pack_size_bytes_;
   int counter_;
   char* send_codec_name_;
 
diff --git a/modules/audio_coding/main/test/TestVADDTX.cc b/modules/audio_coding/main/test/TestVADDTX.cc
index 9832565..bd89dd5 100644
--- a/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/modules/audio_coding/main/test/TestVADDTX.cc
@@ -71,13 +71,13 @@
     _acmA->InitializeReceiver();
     _acmB->InitializeReceiver();
 
-    WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+    uint8_t numEncoders = _acmA->NumberOfCodecs();
     CodecInst myCodecParam;
     if(_testMode != 0)
     {
         printf("Registering codecs at receiver... \n");
     }
-    for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+    for(uint8_t n = 0; n < numEncoders; n++)
     {
         _acmB->Codec(n, &myCodecParam);
         if(_testMode != 0)
@@ -99,8 +99,8 @@
     _acmA->RegisterVADCallback(&_monitor);
 
 
-    WebRtc_Word16 testCntr = 1;
-    WebRtc_Word16 testResults = 0;
+    int16_t testCntr = 1;
+    int16_t testResults = 0;
 
 #ifdef WEBRTC_CODEC_ISAC
     // Open outputfile
@@ -232,7 +232,7 @@
     _testResults += VerifyTest();
 }
 
-void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
+void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, int16_t vadMode)
 {
     bool dtxEnabled, vadEnabled;
     ACMVADMode vadModeSet;
@@ -261,7 +261,7 @@
         if(vadModeSet != vadMode)
         {
             printf("VAD mode: %d not the same as requested: %d\n",
-            (WebRtc_Word16)vadModeSet, (WebRtc_Word16)vadMode);
+            (int16_t)vadModeSet, (int16_t)vadMode);
         }
     }
 
@@ -293,10 +293,10 @@
     return retStruct;
 }
 
-WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
-                                          char* codecName,
-                                          WebRtc_Word32 samplingFreqHz,
-                                          WebRtc_Word32 rateKbps)
+int16_t TestVADDTX::RegisterSendCodec(char side,
+                                      char* codecName,
+                                      int32_t samplingFreqHz,
+                                      int32_t rateKbps)
 {
     if(_testMode != 0)
     {
@@ -326,10 +326,10 @@
     }
 
     CodecInst myCodecParam;
-    for(WebRtc_Word16 codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
+    for(int16_t codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
         codecCntr++)
     {
-        CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, &myCodecParam));
+        CHECK_ERROR(myACM->Codec((uint8_t)codecCntr, &myCodecParam));
         if(!STR_CASE_CMP(myCodecParam.plname, codecName))
         {
             if((samplingFreqHz == -1) || (myCodecParam.plfreq == samplingFreqHz))
@@ -354,9 +354,9 @@
 {
     AudioFrame audioFrame;
 
-    WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms();
-    WebRtc_UWord32 timestampA = 1;
-    WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+    uint16_t SamplesIn10MsecA = _inFileA.PayloadLength10Ms();
+    uint32_t timestampA = 1;
+    int32_t outFreqHzB = _outFileB.SamplingFrequency();
 
     while(!_inFileA.EndOfFile())
     {
@@ -378,7 +378,7 @@
     _monitor.ResetStatistics();
 }
 
-void TestVADDTX::OpenOutFile(WebRtc_Word16 test_number) {
+void TestVADDTX::OpenOutFile(int16_t test_number) {
   std::string file_name;
   std::stringstream file_stream;
   file_stream << webrtc::test::OutputPath();
@@ -393,12 +393,12 @@
 }
 
 
-WebRtc_Word16 TestVADDTX::VerifyTest()
+int16_t TestVADDTX::VerifyTest()
 {
     // Verify empty frame result
-    WebRtc_UWord8 statusEF = 0;
-    WebRtc_UWord8 vadPattern = 0;
-    WebRtc_UWord8 emptyFramePattern[6];
+    uint8_t statusEF = 0;
+    uint8_t vadPattern = 0;
+    uint8_t emptyFramePattern[6];
     CodecInst myCodecParam;
     _acmA->SendCodec(&myCodecParam);
     bool dtxInUse = true;
@@ -493,7 +493,7 @@
 {
 }
 
-WebRtc_Word32 ActivityMonitor::InFrameType(WebRtc_Word16 frameType)
+int32_t ActivityMonitor::InFrameType(int16_t frameType)
 {
     _counter[frameType]++;
     return 0;
@@ -522,7 +522,7 @@
     _counter[0] = _counter[1] = _counter[2] = _counter[3] = _counter[4] = _counter[5] = 0;
 }
 
-void ActivityMonitor::GetStatistics(WebRtc_UWord32* getCounter)
+void ActivityMonitor::GetStatistics(uint32_t* getCounter)
 {
     for (int ii = 0; ii < 6; ii++)
     {
diff --git a/modules/audio_coding/main/test/TestVADDTX.h b/modules/audio_coding/main/test/TestVADDTX.h
index e8f9e1e..f85cff3 100644
--- a/modules/audio_coding/main/test/TestVADDTX.h
+++ b/modules/audio_coding/main/test/TestVADDTX.h
@@ -29,10 +29,10 @@
 public:
     ActivityMonitor();
     ~ActivityMonitor();
-    WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
+    int32_t InFrameType(int16_t frameType);
     void PrintStatistics(int testMode);
     void ResetStatistics();
-    void GetStatistics(WebRtc_UWord32* getCounter);
+    void GetStatistics(uint32_t* getCounter);
 private:
     // counting according to
     /*enum WebRtcACMEncodingType
@@ -44,7 +44,7 @@
         kPassiveDTXWB,
         kPassiveDTXSWB
     };*/
-    WebRtc_UWord32 _counter[6];
+    uint32_t _counter[6];
 };
 
 class TestVADDTX : public ACMTest
@@ -57,17 +57,17 @@
 private:
     // Registration can be based on codec name only, codec name and sampling frequency, or 
     // codec name, sampling frequency and rate.
-    WebRtc_Word16 RegisterSendCodec(char side, 
+    int16_t RegisterSendCodec(char side, 
         char* codecName, 
-        WebRtc_Word32 samplingFreqHz = -1,
-        WebRtc_Word32 rateKhz = -1);
+        int32_t samplingFreqHz = -1,
+        int32_t rateKhz = -1);
     void Run();
-    void OpenOutFile(WebRtc_Word16 testNumber);
+    void OpenOutFile(int16_t testNumber);
     void runTestCases();
     void runTestInternalDTX();
-    void SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode);
+    void SetVAD(bool statusDTX, bool statusVAD, int16_t vadMode);
     VADDTXstruct GetVAD();
-    WebRtc_Word16 VerifyTest();//VADDTXstruct setDTX, VADDTXstruct getDTX);
+    int16_t VerifyTest();//VADDTXstruct setDTX, VADDTXstruct getDTX);
     AudioCodingModule* _acmA;
     AudioCodingModule* _acmB;
 
@@ -77,7 +77,7 @@
     PCMFile                _outFileB;
 
     ActivityMonitor        _monitor;
-    WebRtc_UWord32           _statCounter[6];
+    uint32_t           _statCounter[6];
 
     int                    _testMode;
     int                    _testResults;
diff --git a/modules/audio_coding/main/test/TimedTrace.cc b/modules/audio_coding/main/test/TimedTrace.cc
index 6bf301f..dbfe4e7 100644
--- a/modules/audio_coding/main/test/TimedTrace.cc
+++ b/modules/audio_coding/main/test/TimedTrace.cc
@@ -28,7 +28,7 @@
     _timedTraceFile = NULL;
 }
 
-WebRtc_Word16
+int16_t
 TimedTrace::SetUp(char* fileName)
 {
     if(_timedTraceFile == NULL)
@@ -62,8 +62,8 @@
 
 void
 TimedTrace::TimedLogg(char* message)
-{    
-    unsigned int minutes = (WebRtc_UWord32)floor(_timeEllapsedSec / 60.0);
+{
+    unsigned int minutes = (uint32_t)floor(_timeEllapsedSec / 60.0);
     double seconds = _timeEllapsedSec - minutes * 60;
     //char myFormat[100] = "%8.2f, %3u:%05.2f: %s\n";
     if(_timedTraceFile != NULL)
diff --git a/modules/audio_coding/main/test/TimedTrace.h b/modules/audio_coding/main/test/TimedTrace.h
index d37d287..fd0790f 100644
--- a/modules/audio_coding/main/test/TimedTrace.h
+++ b/modules/audio_coding/main/test/TimedTrace.h
@@ -26,7 +26,7 @@
     void SetTimeEllapsed(double myTime);
     double TimeEllapsed();
     void Tick10Msec();
-    WebRtc_Word16 SetUp(char* fileName);
+    int16_t SetUp(char* fileName);
     void TimedLogg(char* message);
 
 private:
diff --git a/modules/audio_coding/main/test/TwoWayCommunication.cc b/modules/audio_coding/main/test/TwoWayCommunication.cc
index 6b569fa..45ae525 100644
--- a/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/main/test/TwoWayCommunication.cc
@@ -67,16 +67,16 @@
 }
 
 
-WebRtc_UWord8
-TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
-                                 WebRtc_UWord8* codecID_B)
+uint8_t
+TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
+                                 uint8_t* codecID_B)
 {
     AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
-    WebRtc_UWord8 noCodec = tmpACM->NumberOfCodecs();
+    uint8_t noCodec = tmpACM->NumberOfCodecs();
     CodecInst codecInst;
     printf("List of Supported Codecs\n");
     printf("========================\n");
-    for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
+    for(uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++)
     {
         tmpACM->Codec(codecCntr, &codecInst);
         printf("%d- %s\n", codecCntr, codecInst.plname);
@@ -84,18 +84,18 @@
     printf("\nChoose a send codec for side A [0]: ");
     char myStr[15] = "";
     EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
-    *codecID_A = (WebRtc_UWord8)atoi(myStr);
+    *codecID_A = (uint8_t)atoi(myStr);
 
     printf("\nChoose a send codec for side B [0]: ");
     EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
-    *codecID_B = (WebRtc_UWord8)atoi(myStr);
+    *codecID_B = (uint8_t)atoi(myStr);
 
     AudioCodingModule::Destroy(tmpACM);
     printf("\n");
     return 0;
 }
 
-WebRtc_Word16 TwoWayCommunication::SetUp()
+int16_t TwoWayCommunication::SetUp()
 {
     _acmA = AudioCodingModule::Create(1);
     _acmB = AudioCodingModule::Create(2);
@@ -103,8 +103,8 @@
     _acmRefA = AudioCodingModule::Create(3);
     _acmRefB = AudioCodingModule::Create(4);
 
-    WebRtc_UWord8 codecID_A;
-    WebRtc_UWord8 codecID_B;
+    uint8_t codecID_A;
+    uint8_t codecID_B;
 
     ChooseCodec(&codecID_A, &codecID_B);
     CodecInst codecInst_A;
@@ -140,8 +140,8 @@
     CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
     CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
 
-    WebRtc_UWord16 frequencyHz;
-    
+    uint16_t frequencyHz;
+
     //--- Input A
     std::string in_file_name =
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
@@ -202,7 +202,7 @@
     return 0;
 }
 
-WebRtc_Word16 TwoWayCommunication::SetUpAutotest()
+int16_t TwoWayCommunication::SetUpAutotest()
 {
     _acmA = AudioCodingModule::Create(1);
     _acmB = AudioCodingModule::Create(2);
@@ -244,7 +244,7 @@
     CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
     CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
 
-    WebRtc_UWord16 frequencyHz;
+    uint16_t frequencyHz;
 
     //--- Input A and B
     std::string in_file_name =
@@ -312,8 +312,8 @@
     unsigned int msecPassed = 0;
     unsigned int secPassed  = 0;
 
-    WebRtc_Word32 outFreqHzA = _outFileA.SamplingFrequency();
-    WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+    int32_t outFreqHzA = _outFileA.SamplingFrequency();
+    int32_t outFreqHzB = _outFileB.SamplingFrequency();
 
     AudioFrame audioFrame;
 
diff --git a/modules/audio_coding/main/test/TwoWayCommunication.h b/modules/audio_coding/main/test/TwoWayCommunication.h
index fe53532..f955f3e 100644
--- a/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/modules/audio_coding/main/test/TwoWayCommunication.h
@@ -27,9 +27,9 @@
 
     void Perform();
 private:
-    WebRtc_UWord8 ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecID_B);
-    WebRtc_Word16 SetUp();
-    WebRtc_Word16 SetUpAutotest();
+    uint8_t ChooseCodec(uint8_t* codecID_A, uint8_t* codecID_B);
+    int16_t SetUp();
+    int16_t SetUpAutotest();
 
     AudioCodingModule* _acmA;
     AudioCodingModule* _acmB;
diff --git a/modules/audio_coding/main/test/delay_test.cc b/modules/audio_coding/main/test/delay_test.cc
index ff63312..cd0367d 100644
--- a/modules/audio_coding/main/test/delay_test.cc
+++ b/modules/audio_coding/main/test/delay_test.cc
@@ -106,7 +106,7 @@
       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay));
     }
 
-    WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
+    uint8_t num_encoders = acm_a_->NumberOfCodecs();
     CodecInst my_codec_param;
     for(int n = 0; n < num_encoders; n++) {
       acm_b_->Codec(n, &my_codec_param);
diff --git a/modules/audio_coding/main/test/dual_stream_unittest.cc b/modules/audio_coding/main/test/dual_stream_unittest.cc
index 1e3d08e..d36e770 100644
--- a/modules/audio_coding/main/test/dual_stream_unittest.cc
+++ b/modules/audio_coding/main/test/dual_stream_unittest.cc
@@ -26,11 +26,11 @@
   DualStreamTest();
   ~DualStreamTest();
 
-  WebRtc_Word32 SendData(FrameType frameType, WebRtc_UWord8 payload_type,
-                         WebRtc_UWord32 timestamp,
-                         const WebRtc_UWord8* payload_data,
-                         WebRtc_UWord16 payload_size,
-                         const RTPFragmentationHeader* fragmentation);
+  int32_t SendData(FrameType frameType, uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   uint16_t payload_size,
+                   const RTPFragmentationHeader* fragmentation);
 
   void Perform(bool start_in_sync, int num_channels_input);
 
@@ -282,9 +282,9 @@
   }
 }
 
-WebRtc_Word32 DualStreamTest::SendData(
-    FrameType frameType, WebRtc_UWord8 payload_type, WebRtc_UWord32 timestamp,
-    const WebRtc_UWord8* payload_data, WebRtc_UWord16 payload_size,
+int32_t DualStreamTest::SendData(
+    FrameType frameType, uint8_t payload_type, uint32_t timestamp,
+    const uint8_t* payload_data, uint16_t payload_size,
     const RTPFragmentationHeader* fragmentation) {
   int position;
   int stream_index;
diff --git a/modules/audio_coding/main/test/iSACTest.cc b/modules/audio_coding/main/test/iSACTest.cc
index a40f2b7..ce1e0ea 100644
--- a/modules/audio_coding/main/test/iSACTest.cc
+++ b/modules/audio_coding/main/test/iSACTest.cc
@@ -47,7 +47,7 @@
 }
 
 
-WebRtc_Word16 SetISAConfig(
+int16_t SetISAConfig(
     ACMTestISACConfig& isacConfig,
     AudioCodingModule* acm,
     int testMode)
@@ -112,8 +112,8 @@
         (isacConfig.initRateBitPerSec != 0))
     {
         CHECK_ERROR(acm->ConfigISACBandwidthEstimator(
-            (WebRtc_UWord8)isacConfig.initFrameSizeInMsec,
-            (WebRtc_UWord16)isacConfig.initRateBitPerSec, 
+            (uint8_t)isacConfig.initFrameSizeInMsec,
+            (uint16_t)isacConfig.initRateBitPerSec, 
             isacConfig.enforceFrameSize));
         if((isacConfig.initFrameSizeInMsec != 0) && (testMode != 0))
         {
@@ -146,7 +146,7 @@
 }
 
 
-WebRtc_Word16
+int16_t
 ISACTest::Setup()
 {
     int codecCntr;
@@ -245,7 +245,7 @@
 
     Setup();
 
-    WebRtc_Word16 testNr = 0;
+    int16_t testNr = 0;
     ACMTestISACConfig wbISACConfig;
     ACMTestISACConfig swbISACConfig;
 
@@ -290,21 +290,21 @@
     SetISACConfigDefault(swbISACConfig);
     testNr++;
     EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-    
+
     int user_input;
     if((_testMode == 0) || (_testMode == 1))
     {
-        swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
-        wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
+        swbISACConfig.maxPayloadSizeByte = (uint16_t)200;
+        wbISACConfig.maxPayloadSizeByte = (uint16_t)200;
     }
     else
     {
         printf("Enter the max payload-size for side A: ");
         CHECK_ERROR(scanf("%d", &user_input));
-        swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
+        swbISACConfig.maxPayloadSizeByte = (uint16_t)user_input;
         printf("Enter the max payload-size for side B: ");
         CHECK_ERROR(scanf("%d", &user_input));
-        wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
+        wbISACConfig.maxPayloadSizeByte = (uint16_t)user_input;
     }
     testNr++;
     EncodeDecode(testNr, wbISACConfig, swbISACConfig);
@@ -316,17 +316,17 @@
 
     if((_testMode == 0) || (_testMode == 1))
     {
-        swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
-        wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
+        swbISACConfig.maxRateBitPerSec = (uint32_t)48000;
+        wbISACConfig.maxRateBitPerSec = (uint32_t)48000;
     }
     else
     {
         printf("Enter the max rate for side A: ");
         CHECK_ERROR(scanf("%d", &user_input));
-        swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
+        swbISACConfig.maxRateBitPerSec = (uint32_t)user_input;
         printf("Enter the max rate for side B: ");
         CHECK_ERROR(scanf("%d", &user_input));
-        wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
+        wbISACConfig.maxRateBitPerSec = (uint32_t)user_input;
     }
  
     testNr++;
diff --git a/modules/audio_coding/main/test/iSACTest.h b/modules/audio_coding/main/test/iSACTest.h
index 96d3fb6..548ecb6 100644
--- a/modules/audio_coding/main/test/iSACTest.h
+++ b/modules/audio_coding/main/test/iSACTest.h
@@ -27,13 +27,13 @@
 
 struct ACMTestISACConfig
 {
-    WebRtc_Word32  currentRateBitPerSec;
-    WebRtc_Word16  currentFrameSizeMsec;
-    WebRtc_UWord32 maxRateBitPerSec;
-    WebRtc_Word16  maxPayloadSizeByte;
-    WebRtc_Word16  encodingMode;
-    WebRtc_UWord32 initRateBitPerSec;
-    WebRtc_Word16  initFrameSizeInMsec;
+    int32_t  currentRateBitPerSec;
+    int16_t  currentFrameSizeMsec;
+    uint32_t maxRateBitPerSec;
+    int16_t  maxPayloadSizeByte;
+    int16_t  encodingMode;
+    uint32_t initRateBitPerSec;
+    int16_t  initFrameSizeInMsec;
     bool           enforceFrameSize;
 };
 
@@ -47,10 +47,10 @@
 
     void Perform();
 private:
-    WebRtc_Word16 Setup();
-    WebRtc_Word16 SetupConference();
-    WebRtc_Word16 RunConference();    
-    
+    int16_t Setup();
+    int16_t SetupConference();
+    int16_t RunConference();    
+
 
     void Run10ms();
 
@@ -78,8 +78,8 @@
     PCMFile _outFileA;
     PCMFile _outFileB;
 
-    WebRtc_UWord8 _idISAC16kHz;
-    WebRtc_UWord8 _idISAC32kHz;
+    uint8_t _idISAC16kHz;
+    uint8_t _idISAC32kHz;
     CodecInst _paramISAC16kHz;
     CodecInst _paramISAC32kHz;
 
diff --git a/modules/audio_coding/main/test/utility.cc b/modules/audio_coding/main/test/utility.cc
index b727ccd..74d99fc 100644
--- a/modules/audio_coding/main/test/utility.cc
+++ b/modules/audio_coding/main/test/utility.cc
@@ -116,15 +116,15 @@
 }
 
 
-WebRtc_Word16
+int16_t
 ChooseCodec(
     CodecInst& codecInst)
 {
 
     PrintCodecs();
     //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
-    WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
-    WebRtc_Word8 codecID;
+    uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+    int8_t codecID;
     bool outOfRange = false;
     char myStr[15] = "";
     do
@@ -139,18 +139,18 @@
         }
     } while(outOfRange);
 
-    CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, &codecInst));
+    CHECK_ERROR(AudioCodingModule::Codec((uint8_t)codecID, &codecInst));
     return 0;
 }
 
 void
 PrintCodecs()
 {
-    WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
-        
+    uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+
     CodecInst codecInst;
     printf("No  Name                [Hz]    [bps]\n");     
-    for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
+    for(uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++)
     {
         AudioCodingModule::Codec(codecCntr, &codecInst);
         printf("%2d- %-18s %5d   %6d\n", 
@@ -159,7 +159,7 @@
 
 }
 
-CircularBuffer::CircularBuffer(WebRtc_UWord32 len):
+CircularBuffer::CircularBuffer(uint32_t len):
 _buff(NULL),
 _idx(0),
 _buffIsFull(false),
@@ -175,7 +175,7 @@
     }
     else
     {
-        for(WebRtc_UWord32 n = 0; n < len; n++)
+        for(uint32_t n = 0; n < len; n++)
         {
             _buff[n] = 0;
         }
@@ -239,7 +239,7 @@
 
     if(enable && !_calcAvg)
     {
-        WebRtc_UWord32 lim;
+        uint32_t lim;
         if(_buffIsFull)
         {
             lim = _buffLen;
@@ -249,7 +249,7 @@
             lim = _idx;
         }
         _sum = 0;
-        for(WebRtc_UWord32 n = 0; n < lim; n++)
+        for(uint32_t n = 0; n < lim; n++)
         {
             _sum += _buff[n];
         }
@@ -265,7 +265,7 @@
 
     if(enable && !_calcVar)
     {
-        WebRtc_UWord32 lim;
+        uint32_t lim;
         if(_buffIsFull)
         {
             lim = _buffLen;
@@ -275,7 +275,7 @@
             lim = _idx;
         }
         _sumSqr = 0;
-        for(WebRtc_UWord32 n = 0; n < lim; n++)
+        for(uint32_t n = 0; n < lim; n++)
         {
             _sumSqr += _buff[n] * _buff[n];
         }
@@ -283,7 +283,7 @@
     _calcAvg = enable;
 }
 
-WebRtc_Word16
+int16_t
 CircularBuffer::ArithMean(double& mean)
 {
     assert(_buffLen > 0);
@@ -309,7 +309,7 @@
     }
 }
 
-WebRtc_Word16
+int16_t
 CircularBuffer::Variance(double& var)
 {
     assert(_buffLen > 0);
@@ -366,7 +366,7 @@
 
 DTMFDetector::DTMFDetector()
 {
-    for(WebRtc_Word16 n = 0; n < 1000; n++)
+    for(int16_t n = 0; n < 1000; n++)
     {
         _toneCntr[n] = 0;
     }
@@ -376,7 +376,7 @@
 {
 }
 
-WebRtc_Word32 DTMFDetector::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool /* toneEnded */)
+int32_t DTMFDetector::IncomingDtmf(const uint8_t digitDtmf, const bool /* toneEnded */)
 {
     fprintf(stdout, "%d-",digitDtmf);
     _toneCntr[digitDtmf]++;
@@ -385,7 +385,7 @@
 
 void DTMFDetector::PrintDetectedDigits()
 {
-    for(WebRtc_Word16 n = 0; n < 1000; n++)
+    for(int16_t n = 0; n < 1000; n++)
     {
         if(_toneCntr[n] > 0)
         {
@@ -424,9 +424,9 @@
     fprintf(stdout, "Passive DTX super-wideband... %d\n", _numFrameTypes[5]);
 }
 
-WebRtc_Word32 
+int32_t 
 VADCallback::InFrameType(
-    WebRtc_Word16 frameType)
+    int16_t frameType)
 {
     _numFrameTypes[frameType]++;
     return 0;
diff --git a/modules/audio_coding/main/test/utility.h b/modules/audio_coding/main/test/utility.h
index 82935a5..a4a89d1 100644
--- a/modules/audio_coding/main/test/utility.h
+++ b/modules/audio_coding/main/test/utility.h
@@ -105,7 +105,7 @@
 class CircularBuffer
 {
 public:
-    CircularBuffer(WebRtc_UWord32 len);
+    CircularBuffer(uint32_t len);
     ~CircularBuffer();
 
     void SetArithMean(
@@ -116,14 +116,14 @@
     void Update(
         const double newVal);
     void IsBufferFull();
-    
-    WebRtc_Word16 Variance(double& var);
-    WebRtc_Word16 ArithMean(double& mean);
+
+    int16_t Variance(double& var);
+    int16_t ArithMean(double& mean);
 
 protected:
     double* _buff;
-    WebRtc_UWord32 _idx;
-    WebRtc_UWord32 _buffLen;
+    uint32_t _idx;
+    uint32_t _buffLen;
 
     bool         _buffIsFull;
     bool         _calcAvg;
@@ -136,7 +136,7 @@
 
 
 
-WebRtc_Word16 ChooseCodec(
+int16_t ChooseCodec(
     CodecInst& codecInst);
 
 void PrintCodecs();
@@ -152,11 +152,11 @@
     DTMFDetector();
     ~DTMFDetector();
     // used for inband DTMF detection
-    WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool toneEnded);
+    int32_t IncomingDtmf(const uint8_t digitDtmf, const bool toneEnded);
     void PrintDetectedDigits();
 
 private:
-    WebRtc_UWord32 _toneCntr[1000];
+    uint32_t _toneCntr[1000];
 
 };
 
@@ -169,14 +169,14 @@
     VADCallback();
     ~VADCallback(){}
 
-    WebRtc_Word32 InFrameType(
-        WebRtc_Word16 frameType);
+    int32_t InFrameType(
+        int16_t frameType);
 
     void PrintFrameTypes();
     void Reset();
 
 private:
-    WebRtc_UWord32 _numFrameTypes[6];
+    uint32_t _numFrameTypes[6];
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/accelerate.c b/modules/audio_coding/neteq/accelerate.c
index 285de4d..ce3f3b2 100644
--- a/modules/audio_coding/neteq/accelerate.c
+++ b/modules/audio_coding/neteq/accelerate.c
@@ -28,9 +28,9 @@
 /* Scratch usage:
 
  Type	        Name                size    startpos    endpos
- WebRtc_Word16  pw16_downSampSpeech 110     0           109
- WebRtc_Word32  pw32_corr           2*50    110         209
- WebRtc_Word16  pw16_corr           50      0           49
+ int16_t  pw16_downSampSpeech 110     0           109
+ int32_t  pw32_corr           2*50    110         209
+ int16_t  pw16_corr           50      0           49
 
  Total: 110+2*50
  */
@@ -67,45 +67,45 @@
 
 int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
 #ifdef SCRATCH
-                           WebRtc_Word16 *pw16_scratchPtr,
+                           int16_t *pw16_scratchPtr,
 #endif
-                           const WebRtc_Word16 *pw16_decoded, int len,
-                           WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                           WebRtc_Word16 BGNonly)
+                           const int16_t *pw16_decoded, int len,
+                           int16_t *pw16_outData, int16_t *pw16_len,
+                           int16_t BGNonly)
 {
 
 #ifdef SCRATCH
     /* Use scratch memory for internal temporary vectors */
-    WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
-    WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
-    WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
+    int16_t *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
+    int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
+    int16_t *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
 #else
     /* Allocate memory for temporary vectors */
-    WebRtc_Word16 pw16_downSampSpeech[ACCELERATE_DOWNSAMPLED_LEN];
-    WebRtc_Word32 pw32_corr[ACCELERATE_CORR_LEN];
-    WebRtc_Word16 pw16_corr[ACCELERATE_CORR_LEN];
+    int16_t pw16_downSampSpeech[ACCELERATE_DOWNSAMPLED_LEN];
+    int32_t pw32_corr[ACCELERATE_CORR_LEN];
+    int16_t pw16_corr[ACCELERATE_CORR_LEN];
 #endif
-    WebRtc_Word16 w16_decodedMax = 0;
-    WebRtc_Word16 w16_tmp;
-    WebRtc_Word16 w16_tmp2;
-    WebRtc_Word32 w32_tmp;
-    WebRtc_Word32 w32_tmp2;
+    int16_t w16_decodedMax = 0;
+    int16_t w16_tmp;
+    int16_t w16_tmp2;
+    int32_t w32_tmp;
+    int32_t w32_tmp2;
 
-    const WebRtc_Word16 w16_startLag = ACCELERATE_MIN_LAG;
-    const WebRtc_Word16 w16_endLag = ACCELERATE_MAX_LAG;
-    const WebRtc_Word16 w16_corrLen = ACCELERATE_CORR_LEN;
-    const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
-    WebRtc_Word16 *pw16_vectmp;
-    WebRtc_Word16 w16_inc, w16_startfact;
-    WebRtc_Word16 w16_bestIndex, w16_bestVal;
-    WebRtc_Word16 w16_VAD = 1;
-    WebRtc_Word16 fsMult;
-    WebRtc_Word16 fsMult120;
-    WebRtc_Word32 w32_en1, w32_en2, w32_cc;
-    WebRtc_Word16 w16_en1, w16_en2;
-    WebRtc_Word16 w16_en1Scale, w16_en2Scale;
-    WebRtc_Word16 w16_sqrtEn1En2;
-    WebRtc_Word16 w16_bestCorr = 0;
+    const int16_t w16_startLag = ACCELERATE_MIN_LAG;
+    const int16_t w16_endLag = ACCELERATE_MAX_LAG;
+    const int16_t w16_corrLen = ACCELERATE_CORR_LEN;
+    const int16_t *pw16_vec1, *pw16_vec2;
+    int16_t *pw16_vectmp;
+    int16_t w16_inc, w16_startfact;
+    int16_t w16_bestIndex, w16_bestVal;
+    int16_t w16_VAD = 1;
+    int16_t fsMult;
+    int16_t fsMult120;
+    int32_t w32_en1, w32_en2, w32_cc;
+    int16_t w16_en1, w16_en2;
+    int16_t w16_en1Scale, w16_en2Scale;
+    int16_t w16_sqrtEn1En2;
+    int16_t w16_bestCorr = 0;
     int ok;
 
 #ifdef NETEQ_STEREO
@@ -115,20 +115,20 @@
     fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
 
     /* Pre-calculate common multiplication with fsMult */
-    fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
+    fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
 
     inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
 
     /* Sanity check for len variable; must be (almost) 30 ms 
      (120*fsMult + max(bestIndex)) */
-    if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult))
+    if (len < (int16_t) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult))
     {
         /* Length of decoded data too short */
         inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
         *pw16_len = len;
 
         /* simply move all data from decoded to outData */
-        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
         return NETEQ_OTHER_ERROR;
     }
@@ -150,7 +150,7 @@
             *pw16_len = len;
 
             /* simply move all data from decoded to outData */
-            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
             return NETEQ_OTHER_ERROR;
         }
@@ -192,7 +192,7 @@
         /****************************************************************/
 
         /* find maximum absolute value */
-        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len);
 
         /* downsample the decoded speech to 4 kHz */
         ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
@@ -203,7 +203,7 @@
             inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
             *pw16_len = len;
             /* simply move all data from decoded to outData */
-            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
             return NETEQ_OTHER_ERROR;
         }
 
@@ -218,9 +218,9 @@
         WebRtcNetEQ_CrossCorr(
             pw32_corr, &pw16_downSampSpeech[w16_endLag],
             &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
-            (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
+            (int16_t) (w16_endLag - w16_startLag), w16_tmp, -1);
 
-        /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+        /* Normalize correlation to 14 bits and put in a int16_t vector */
         w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
         w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
         w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
@@ -233,7 +233,7 @@
     if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
     {
         /* Find the strongest correlation peak by using the parabolic fit method */
-        WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
+        WebRtcNetEQ_PeakDetection(pw16_corr, (int16_t) w16_corrLen, 1, fsMult,
             &w16_bestIndex, &w16_bestVal);
         /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
 
@@ -265,7 +265,7 @@
 #else /* NETEQ_STEREO */
 
     /* Find the strongest correlation peak by using the parabolic fit method */
-    WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
+    WebRtcNetEQ_PeakDetection(pw16_corr, (int16_t) w16_corrLen, 1, fsMult,
         &w16_bestIndex, &w16_bestVal);
     /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
 
@@ -304,13 +304,13 @@
         pw16_vec2 = &pw16_decoded[fsMult120];
 
         /* Calculate energies for vec1 and vec2 */
-        w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
-            (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
-        w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
-            (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
+        w32_en1 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1,
+            (int16_t*) pw16_vec1, w16_bestIndex, w16_tmp);
+        w32_en2 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec2,
+            (int16_t*) pw16_vec2, w16_bestIndex, w16_tmp);
 
         /* Calculate cross-correlation at the found lag */
-        w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
+        w32_cc = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, (int16_t*) pw16_vec2,
             w16_bestIndex, w16_tmp);
 
         /* Check VAD constraint 
@@ -328,7 +328,7 @@
         w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
         w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
         w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
-        w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
+        w16_tmp2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
         w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
 
         /* Scale w32_tmp properly before comparing with w32_tmp2 */
@@ -336,7 +336,7 @@
         if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
         {
             /* Cannot scale only w32_tmp, must scale w32_temp2 too */
-            WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
+            int16_t tempshift = WebRtcSpl_NormW32(w32_tmp);
             w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
             w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
                 WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
@@ -372,21 +372,21 @@
                 w16_en1Scale += 1;
             }
 
-            /* Convert energies to WebRtc_Word16 */
-            w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
-            w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+            /* Convert energies to int16_t */
+            w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+            w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
 
             /* Calculate energy product */
             w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
 
             /* Calculate square-root of energy product */
-            w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
+            w16_sqrtEn1En2 = (int16_t) WebRtcSpl_SqrtFloor(w32_tmp);
 
             /* Calculate cc/sqrt(en1*en2) in Q14 */
             w16_tmp = 14 - WEBRTC_SPL_RSHIFT_W16(w16_en1Scale+w16_en2Scale, 1);
             w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
             w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
-            w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
+            w16_bestCorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
             w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
         }
 
@@ -414,8 +414,8 @@
          * Calculate cross-fading slope so that the fading factor goes from
          * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
          */
-        w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
-            (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
+        w16_inc = (int16_t) WebRtcSpl_DivW32W16((int32_t) 16384,
+            (int16_t) (w16_bestIndex + 1)); /* in Q14 */
 
         /* Initiate fading factor */
         w16_startfact = 16384 - w16_inc;
@@ -432,14 +432,14 @@
         /* Generate interpolated part of length bestIndex (1 pitch period) */
         pw16_vectmp = pw16_outData + w16_tmp; /* start of interpolation output */
         /* Reuse mixing function from Expand */
-        WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec1,
-            (WebRtc_Word16*) pw16_vec2, &w16_startfact, w16_inc, w16_bestIndex);
+        WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (int16_t*) pw16_vec1,
+            (int16_t*) pw16_vec2, &w16_startfact, w16_inc, w16_bestIndex);
 
         /* Move the last part (also unmodified) */
         /* Take from decoded at 15 ms + 1 pitch period */
         pw16_vec2 = &pw16_decoded[fsMult120 + w16_bestIndex];
         WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[fsMult120], pw16_vec2,
-            (WebRtc_Word16) (len - fsMult120 - w16_bestIndex));
+            (int16_t) (len - fsMult120 - w16_bestIndex));
 
         /* Set the mode flag */
         if (w16_VAD)
@@ -478,7 +478,7 @@
         *pw16_len = len;
 
         /* Simply move all data from decoded to outData */
-        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
         return 0;
     }
diff --git a/modules/audio_coding/neteq/automode.c b/modules/audio_coding/neteq/automode.c
index 78933cc..edee98e 100644
--- a/modules/audio_coding/neteq/automode.c
+++ b/modules/audio_coding/neteq/automode.c
@@ -30,15 +30,15 @@
 
 
 int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
-                                    WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
-                                    WebRtc_Word32 fsHz, int mdCodec, int streamingMode)
+                                    uint16_t seqNumber, uint32_t timeStamp,
+                                    int32_t fsHz, int mdCodec, int streamingMode)
 {
-    WebRtc_UWord32 timeIat; /* inter-arrival time */
+    uint32_t timeIat; /* inter-arrival time */
     int i;
-    WebRtc_Word32 tempsum = 0; /* temp summation */
-    WebRtc_Word32 tempvar; /* temporary variable */
+    int32_t tempsum = 0; /* temp summation */
+    int32_t tempvar; /* temporary variable */
     int retval = 0; /* return value */
-    WebRtc_Word16 packetLenSamp; /* packet speech length in samples */
+    int16_t packetLenSamp; /* packet speech length in samples */
 
     /****************/
     /* Sanity check */
@@ -63,7 +63,7 @@
     else
     {
         /* calculate timestamps per packet */
-        packetLenSamp = (WebRtc_Word16) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp,
+        packetLenSamp = (int16_t) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp,
             seqNumber - inst->lastSeqNo);
     }
 
@@ -81,7 +81,7 @@
              * Calculate IAT in Q8, including fractions of a packet (i.e., more accurate
              * than timeIat).
              */
-            WebRtc_Word16 timeIatQ8 = (WebRtc_Word16) WebRtcSpl_DivW32W16(
+            int16_t timeIatQ8 = (int16_t) WebRtcSpl_DivW32W16(
                 WEBRTC_SPL_LSHIFT_W32(inst->packetIatCountSamp, 8), packetLenSamp);
 
             /*
@@ -105,7 +105,7 @@
             }
 
             /* too long since the last maximum was observed; decrease max value */
-            if (inst->maxCSumUpdateTimer > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
+            if (inst->maxCSumUpdateTimer > (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz,
                 MAX_STREAMING_PEAK_PERIOD))
             {
                 inst->maxCSumIatQ8 -= 4; /* remove 1000*4/256 = 15.6 ms/s */
@@ -120,12 +120,12 @@
              * the IAT is not negative.
              */
             timeIat -= WEBRTC_SPL_MIN(timeIat,
-                (WebRtc_UWord32) (seqNumber - inst->lastSeqNo - 1));
+                (uint32_t) (seqNumber - inst->lastSeqNo - 1));
         }
         else if (seqNumber < inst->lastSeqNo)
         {
             /* compensate for re-ordering */
-            timeIat += (WebRtc_UWord32) (inst->lastSeqNo + 1 - seqNumber);
+            timeIat += (uint32_t) (inst->lastSeqNo + 1 - seqNumber);
         }
 
         /* saturate IAT at maximum value */
@@ -134,7 +134,7 @@
         /* update iatProb = forgetting_factor * iatProb for all elements */
         for (i = 0; i <= MAX_IAT; i++)
         {
-            WebRtc_Word32 tempHi, tempLo; /* Temporary variables */
+            int32_t tempHi, tempLo; /* Temporary variables */
 
             /*
              * Multiply iatProbFact (Q15) with iatProb (Q30) and right-shift 15 steps
@@ -143,12 +143,12 @@
 
             /*
              * 1) Multiply the high 16 bits (15 bits + sign) of iatProb. Shift iatProb
-             * 16 steps right to get the high 16 bits in a WebRtc_Word16 prior to
+             * 16 steps right to get the high 16 bits in a int16_t prior to
              * multiplication, and left-shift with 1 afterwards to come back to
              * Q30 = (Q15 * (Q30>>16)) << 1.
              */
             tempHi = WEBRTC_SPL_MUL_16_16(inst->iatProbFact,
-                (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16));
+                (int16_t) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16));
             tempHi = WEBRTC_SPL_LSHIFT_W32(tempHi, 1); /* left-shift 1 step */
 
             /*
@@ -157,7 +157,7 @@
              */
             tempLo = inst->iatProb[i] & 0x0000FFFF; /* sift out the 16 low bits */
             tempLo = WEBRTC_SPL_MUL_16_U16(inst->iatProbFact,
-                (WebRtc_UWord16) tempLo);
+                (uint16_t) tempLo);
             tempLo = WEBRTC_SPL_RSHIFT_W32(tempLo, 15);
 
             /* Finally, add the high and low parts */
@@ -212,7 +212,7 @@
         }
 
         /* Calculate optimal buffer level based on updated statistics */
-        tempvar = (WebRtc_Word32) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat,
+        tempvar = (int32_t) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat,
             streamingMode);
         if (tempvar > 0)
         {
@@ -254,8 +254,8 @@
 
     /* Calculate inter-arrival time in ms = packetIatCountSamp / (fsHz / 1000) */
     timeIat = WEBRTC_SPL_UDIV(
-        WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (WebRtc_Word16) 1000),
-        (WebRtc_UWord32) fsHz);
+        WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (int16_t) 1000),
+        (uint32_t) fsHz);
 
     /* Increase counter corresponding to current inter-arrival time */
     if (timeIat > 2000)
@@ -291,16 +291,16 @@
 }
 
 
-WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
-                                            int mdCodec, WebRtc_UWord32 timeIatPkts,
-                                            int streamingMode)
+int16_t WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, int32_t fsHz,
+                                      int mdCodec, uint32_t timeIatPkts,
+                                      int streamingMode)
 {
 
-    WebRtc_Word32 sum1 = 1 << 30; /* assign to 1 in Q30 */
-    WebRtc_Word16 B;
-    WebRtc_UWord16 Bopt;
+    int32_t sum1 = 1 << 30; /* assign to 1 in Q30 */
+    int16_t B;
+    uint16_t Bopt;
     int i;
-    WebRtc_Word32 betaInv; /* optimization parameter */
+    int32_t betaInv; /* optimization parameter */
 
 #ifdef NETEQ_DELAY_LOGGING
     /* special code for offline delay logging */
@@ -362,7 +362,7 @@
          * Do not have to re-calculate all points, just back off a few steps from
          * previous value of B.
          */
-        WebRtc_Word32 sum2 = sum1; /* copy sum1 */
+        int32_t sum2 = sum1; /* copy sum1 */
 
         while ((sum2 <= betaInv + inst->iatProb[Bopt]) && (Bopt > 0))
         {
@@ -426,8 +426,8 @@
      * If IAT > optimal level + threshold (+1 for MD codecs)
      * or if IAT > 2 * optimal level (note: optimal level is in Q8):
      */
-    if (timeIatPkts > (WebRtc_UWord32) (Bopt + inst->peakThresholdPkt + (mdCodec != 0))
-        || timeIatPkts > (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_U16(Bopt, 1))
+    if (timeIatPkts > (uint32_t) (Bopt + inst->peakThresholdPkt + (mdCodec != 0))
+        || timeIatPkts > (uint32_t) WEBRTC_SPL_LSHIFT_U16(Bopt, 1))
     {
         /* A peak is observed */
 
@@ -440,7 +440,7 @@
         }
         else if (inst->peakIatCountSamp
             <=
-            (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD))
+            (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD))
         {
             /* This is not the first peak and the period time is valid */
 
@@ -450,7 +450,7 @@
             /* saturate height to 16 bits */
             inst->peakHeightPkt[inst->peakIndex]
                 =
-                (WebRtc_Word16) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX);
+                (int16_t) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX);
 
             /* increment peakIndex and wrap/modulo */
             inst->peakIndex = (inst->peakIndex + 1) & PEAK_INDEX_MASK;
@@ -472,7 +472,7 @@
             inst->peakModeDisabled >>= 1; /* decrease mode-disable "counter" */
 
         }
-        else if (inst->peakIatCountSamp > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
+        else if (inst->peakIatCountSamp > (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz,
             WEBRTC_SPL_LSHIFT_W16(MAX_PEAK_PERIOD, 1)))
         {
             /*
@@ -535,11 +535,11 @@
 }
 
 
-int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
-                                  int sampPerCall, WebRtc_Word16 fsMult)
+int WebRtcNetEQ_BufferLevelFilter(int32_t curSizeMs8, AutomodeInst_t *inst,
+                                  int sampPerCall, int16_t fsMult)
 {
 
-    WebRtc_Word16 curSizeFrames;
+    int16_t curSizeFrames;
 
     /****************/
     /* Sanity check */
@@ -558,7 +558,7 @@
          * Current buffer level in packet lengths
          * = (curSizeMs8 * fsMult) / packetSpeechLenSamp
          */
-        curSizeFrames = (WebRtc_Word16) WebRtcSpl_DivW32W16(
+        curSizeFrames = (int16_t) WebRtcSpl_DivW32W16(
             WEBRTC_SPL_MUL_32_16(curSizeMs8, fsMult), inst->packetSpeechLenSamp);
     }
     else
@@ -613,8 +613,8 @@
 }
 
 
-int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
-                                   WebRtc_Word32 fsHz)
+int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, int16_t newLenSamp,
+                                   int32_t fsHz)
 {
 
     /* Sanity check for newLenSamp and fsHz */
@@ -635,9 +635,9 @@
      * the (fractional) number of packets that corresponds to PEAK_HEIGHT
      * (in Q8 seconds). That is, threshold = PEAK_HEIGHT/256 * fsHz / packLen.
      */
-    inst->peakThresholdPkt = (WebRtc_UWord16) WebRtcSpl_DivW32W16ResW16(
+    inst->peakThresholdPkt = (uint16_t) WebRtcSpl_DivW32W16ResW16(
         WEBRTC_SPL_MUL_16_16_RSFT(PEAK_HEIGHT,
-            (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp);
+            (int16_t) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp);
 
     return 0;
 }
@@ -647,7 +647,7 @@
 {
 
     int i;
-    WebRtc_UWord16 tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */
+    uint16_t tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */
 
     /* Sanity check for maxBufLenPackets */
     if (maxBufLenPackets <= 1)
@@ -690,7 +690,7 @@
         /* iatProb[i] = 0.5^(i+1) = iatProb[i-1] / 2 */
         tempprob = WEBRTC_SPL_RSHIFT_U16(tempprob, 1);
         /* store in PDF vector */
-        inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) tempprob, 16);
+        inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((int32_t) tempprob, 16);
     }
 
     /*
diff --git a/modules/audio_coding/neteq/automode.h b/modules/audio_coding/neteq/automode.h
index 2e6b514..5996a51 100644
--- a/modules/audio_coding/neteq/automode.h
+++ b/modules/audio_coding/neteq/automode.h
@@ -64,57 +64,57 @@
 {
 
     /* Filtered current buffer level */
-    WebRtc_UWord16 levelFiltFact; /* filter forgetting factor in Q8 */
+    uint16_t levelFiltFact; /* filter forgetting factor in Q8 */
     int buffLevelFilt; /* filtered buffer level in Q8 */
 
     /* Inter-arrival time (iat) statistics */
-    WebRtc_Word32 iatProb[MAX_IAT + 1]; /* iat probabilities in Q30 */
-    WebRtc_Word16 iatProbFact; /* iat forgetting factor in Q15 */
-    WebRtc_UWord32 packetIatCountSamp; /* time (in timestamps) elapsed since last
+    int32_t iatProb[MAX_IAT + 1]; /* iat probabilities in Q30 */
+    int16_t iatProbFact; /* iat forgetting factor in Q15 */
+    uint32_t packetIatCountSamp; /* time (in timestamps) elapsed since last
      packet arrival, based on RecOut calls */
     int optBufLevel; /* current optimal buffer level in Q8 */
 
     /* Packet related information */
-    WebRtc_Word16 packetSpeechLenSamp; /* speech samples per incoming packet */
-    WebRtc_Word16 lastPackCNGorDTMF; /* indicates that the last received packet
+    int16_t packetSpeechLenSamp; /* speech samples per incoming packet */
+    int16_t lastPackCNGorDTMF; /* indicates that the last received packet
      contained special information */
-    WebRtc_UWord16 lastSeqNo; /* sequence number for last packet received */
-    WebRtc_UWord32 lastTimeStamp; /* timestamp for the last packet received */
-    WebRtc_Word32 sampleMemory; /* memory position for keeping track of how many
+    uint16_t lastSeqNo; /* sequence number for last packet received */
+    uint32_t lastTimeStamp; /* timestamp for the last packet received */
+    int32_t sampleMemory; /* memory position for keeping track of how many
      samples we cut during expand */
-    WebRtc_Word16 prevTimeScale; /* indicates that the last mode was an accelerate
+    int16_t prevTimeScale; /* indicates that the last mode was an accelerate
      or pre-emptive expand operation */
-    WebRtc_UWord32 timescaleHoldOff; /* counter that is shifted one step right each
+    uint32_t timescaleHoldOff; /* counter that is shifted one step right each
      RecOut call; time-scaling allowed when it has
      reached 0 */
-    WebRtc_Word16 extraDelayMs; /* extra delay for sync with video */
+    int16_t extraDelayMs; /* extra delay for sync with video */
 
     /* Peak-detection */
     /* vector with the latest peak periods (peak spacing in samples) */
-    WebRtc_UWord32 peakPeriodSamp[NUM_PEAKS];
+    uint32_t peakPeriodSamp[NUM_PEAKS];
     /* vector with the latest peak heights (in packets) */
-    WebRtc_Word16 peakHeightPkt[NUM_PEAKS];
-    WebRtc_Word16 peakIndex; /* index for the vectors peakPeriodSamp and peakHeightPkt;
+    int16_t peakHeightPkt[NUM_PEAKS];
+    int16_t peakIndex; /* index for the vectors peakPeriodSamp and peakHeightPkt;
      -1 if still waiting for first peak */
-    WebRtc_UWord16 peakThresholdPkt; /* definition of peak (in packets);
+    uint16_t peakThresholdPkt; /* definition of peak (in packets);
      calculated from PEAK_HEIGHT */
-    WebRtc_UWord32 peakIatCountSamp; /* samples elapsed since last peak was observed */
-    WebRtc_UWord32 curPeakPeriod; /* current maximum of peakPeriodSamp vector */
-    WebRtc_Word16 curPeakHeight; /* derived from peakHeightPkt vector;
+    uint32_t peakIatCountSamp; /* samples elapsed since last peak was observed */
+    uint32_t curPeakPeriod; /* current maximum of peakPeriodSamp vector */
+    int16_t curPeakHeight; /* derived from peakHeightPkt vector;
      used as optimal buffer level in peak mode */
-    WebRtc_Word16 peakModeDisabled; /* ==0 if peak mode can be engaged; >0 if not */
+    int16_t peakModeDisabled; /* ==0 if peak mode can be engaged; >0 if not */
     uint16_t peakFound; /* 1 if peaks are detected and extra delay is applied;
                         * 0 otherwise. */
 
     /* Post-call statistics */
-    WebRtc_UWord32 countIAT500ms; /* number of times we got small network outage */
-    WebRtc_UWord32 countIAT1000ms; /* number of times we got medium network outage */
-    WebRtc_UWord32 countIAT2000ms; /* number of times we got large network outage */
-    WebRtc_UWord32 longestIATms; /* mSec duration of longest network outage */
+    uint32_t countIAT500ms; /* number of times we got small network outage */
+    uint32_t countIAT1000ms; /* number of times we got medium network outage */
+    uint32_t countIAT2000ms; /* number of times we got large network outage */
+    uint32_t longestIATms; /* mSec duration of longest network outage */
 
-    WebRtc_Word16 cSumIatQ8; /* cumulative sum of inter-arrival times */
-    WebRtc_Word16 maxCSumIatQ8; /* max cumulative sum IAT */
-    WebRtc_UWord32 maxCSumUpdateTimer;/* time elapsed since maximum was observed */
+    int16_t cSumIatQ8; /* cumulative sum of inter-arrival times */
+    int16_t maxCSumIatQ8; /* max cumulative sum IAT */
+    uint32_t maxCSumUpdateTimer;/* time elapsed since maximum was observed */
 
 } AutomodeInst_t;
 
@@ -148,8 +148,8 @@
  */
 
 int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
-                                    WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
-                                    WebRtc_Word32 fsHz, int mdCodec, int streamingMode);
+                                    uint16_t seqNumber, uint32_t timeStamp,
+                                    int32_t fsHz, int mdCodec, int streamingMode);
 
 /****************************************************************************
  * WebRtcNetEQ_CalcOptimalBufLvl(...)
@@ -172,9 +172,9 @@
  *                        <0 - Error
  */
 
-WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
-                                            int mdCodec, WebRtc_UWord32 timeIatPkts,
-                                            int streamingMode);
+int16_t WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, int32_t fsHz,
+                                      int mdCodec, uint32_t timeIatPkts,
+                                      int streamingMode);
 
 /****************************************************************************
  * WebRtcNetEQ_BufferLevelFilter(...)
@@ -197,8 +197,8 @@
  *                      : <0 - Error
  */
 
-int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
-                                  int sampPerCall, WebRtc_Word16 fsMult);
+int WebRtcNetEQ_BufferLevelFilter(int32_t curSizeMs8, AutomodeInst_t *inst,
+                                  int sampPerCall, int16_t fsMult);
 
 /****************************************************************************
  * WebRtcNetEQ_SetPacketSpeechLen(...)
@@ -220,8 +220,8 @@
  *                        <0 - Error
  */
 
-int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
-                                   WebRtc_Word32 fsHz);
+int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, int16_t newLenSamp,
+                                   int32_t fsHz);
 
 /****************************************************************************
  * WebRtcNetEQ_ResetAutomode(...)
diff --git a/modules/audio_coding/neteq/bgn_update.c b/modules/audio_coding/neteq/bgn_update.c
index 05956c2..4d660ff 100644
--- a/modules/audio_coding/neteq/bgn_update.c
+++ b/modules/audio_coding/neteq/bgn_update.c
@@ -22,10 +22,10 @@
  Designed for BGN_LPC_ORDER <= 10
 
  Type           Name            size   startpos  endpos
- WebRtc_Word32  pw32_autoCorr   22     0         21  (Length (BGN_LPC_ORDER + 1)*2)
- WebRtc_Word16  pw16_tempVec    10     22        31	(Length BGN_LPC_ORDER)
- WebRtc_Word16  pw16_rc         10     32        41	(Length BGN_LPC_ORDER)
- WebRtc_Word16  pw16_outVec     74     0         73  (Length BGN_LPC_ORDER + 64)
+ int32_t  pw32_autoCorr   22     0         21  (Length (BGN_LPC_ORDER + 1)*2)
+ int16_t  pw16_tempVec    10     22        31	(Length BGN_LPC_ORDER)
+ int16_t  pw16_rc         10     32        41	(Length BGN_LPC_ORDER)
+ int16_t  pw16_outVec     74     0         73  (Length BGN_LPC_ORDER + 64)
 
  Total: 74
  */
@@ -58,34 +58,34 @@
 
 void WebRtcNetEQ_BGNUpdate(
 #ifdef SCRATCH
-                           DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
+                           DSPInst_t *inst, int16_t *pw16_scratchPtr
 #else
                            DSPInst_t *inst
 #endif
 )
 {
-    const WebRtc_Word16 w16_vecLen = 256;
+    const int16_t w16_vecLen = 256;
     BGNInst_t *BGN_Inst = &(inst->BGNInst);
 #ifdef SCRATCH
-    WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_AUTO_CORR);
-    WebRtc_Word16 *pw16_tempVec = pw16_scratchPtr + SCRATCH_PW16_TEMP_VEC;
-    WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
-    WebRtc_Word16 *pw16_outVec = pw16_scratchPtr + SCRATCH_PW16_OUT_VEC;
+    int32_t *pw32_autoCorr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_AUTO_CORR);
+    int16_t *pw16_tempVec = pw16_scratchPtr + SCRATCH_PW16_TEMP_VEC;
+    int16_t *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
+    int16_t *pw16_outVec = pw16_scratchPtr + SCRATCH_PW16_OUT_VEC;
 #else
-    WebRtc_Word32 pw32_autoCorr[BGN_LPC_ORDER + 1];
-    WebRtc_Word16 pw16_tempVec[BGN_LPC_ORDER];
-    WebRtc_Word16 pw16_outVec[BGN_LPC_ORDER + 64];
-    WebRtc_Word16 pw16_rc[BGN_LPC_ORDER];
+    int32_t pw32_autoCorr[BGN_LPC_ORDER + 1];
+    int16_t pw16_tempVec[BGN_LPC_ORDER];
+    int16_t pw16_outVec[BGN_LPC_ORDER + 64];
+    int16_t pw16_rc[BGN_LPC_ORDER];
 #endif
-    WebRtc_Word16 pw16_A[BGN_LPC_ORDER + 1];
-    WebRtc_Word32 w32_tmp;
-    WebRtc_Word16 *pw16_vec;
-    WebRtc_Word16 w16_maxSample;
-    WebRtc_Word16 w16_tmp, w16_tmp2;
-    WebRtc_Word16 w16_enSampleShift;
-    WebRtc_Word32 w32_en, w32_enBGN;
-    WebRtc_Word32 w32_enUpdateThreashold;
-    WebRtc_Word16 stability;
+    int16_t pw16_A[BGN_LPC_ORDER + 1];
+    int32_t w32_tmp;
+    int16_t *pw16_vec;
+    int16_t w16_maxSample;
+    int16_t w16_tmp, w16_tmp2;
+    int16_t w16_enSampleShift;
+    int32_t w32_en, w32_enBGN;
+    int32_t w32_enUpdateThreashold;
+    int16_t stability;
 
     pw16_vec = inst->pw16_speechHistory + inst->w16_speechHistoryLen - w16_vecLen;
 
@@ -188,7 +188,7 @@
             w32_enBGN = WEBRTC_SPL_SHIFT_W32(w32_enBGN, w16_tmp2);
 
             /* Calculate scale and shift factor */
-            BGN_Inst->w16_scale = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_enBGN);
+            BGN_Inst->w16_scale = (int16_t) WebRtcSpl_SqrtFloor(w32_enBGN);
             BGN_Inst->w16_scaleShift = 13 + ((6 + w16_tmp2) >> 1); /* RANDN table is in Q13, */
             /* 6=log2(64) */
 
@@ -207,13 +207,13 @@
         w32_tmp = WEBRTC_SPL_MUL_16_16_RSFT(NETEQFIX_BGNFRAQINCQ16,
             BGN_Inst->w32_energyUpdateLow, 16);
         w32_tmp += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
-            (WebRtc_Word16)(BGN_Inst->w32_energyUpdate & 0xFF));
+            (int16_t)(BGN_Inst->w32_energyUpdate & 0xFF));
         w32_tmp += (WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
-            (WebRtc_Word16)((BGN_Inst->w32_energyUpdate>>8) & 0xFF)) << 8);
+            (int16_t)((BGN_Inst->w32_energyUpdate>>8) & 0xFF)) << 8);
         BGN_Inst->w32_energyUpdateLow += w32_tmp;
 
         BGN_Inst->w32_energyUpdate += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
-            (WebRtc_Word16)(BGN_Inst->w32_energyUpdate>>16));
+            (int16_t)(BGN_Inst->w32_energyUpdate>>16));
         BGN_Inst->w32_energyUpdate += BGN_Inst->w32_energyUpdateLow >> 16;
         BGN_Inst->w32_energyUpdateLow = (BGN_Inst->w32_energyUpdateLow & 0x0FFFF);
 
diff --git a/modules/audio_coding/neteq/buffer_stats.h b/modules/audio_coding/neteq/buffer_stats.h
index 9820519..722f477 100644
--- a/modules/audio_coding/neteq/buffer_stats.h
+++ b/modules/audio_coding/neteq/buffer_stats.h
@@ -51,13 +51,13 @@
 {
 
     /* store statistical data here */
-    WebRtc_Word16 w16_cngOn; /* remember if CNG is interrupted by other event (e.g. DTMF) */
-    WebRtc_Word16 w16_noExpand;
-    WebRtc_Word32 uw32_CNGplayedTS;
+    int16_t w16_cngOn; /* remember if CNG is interrupted by other event (e.g. DTMF) */
+    int16_t w16_noExpand;
+    int32_t uw32_CNGplayedTS;
 
     /* VQmon data */
-    WebRtc_UWord16 avgDelayMsQ8;
-    WebRtc_Word16 maxDelayMs;
+    uint16_t avgDelayMsQ8;
+    int16_t maxDelayMs;
 
     AutomodeInst_t Automode_inst;
 
@@ -83,13 +83,13 @@
  *
  */
 
-WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
-                                            WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
-                                            WebRtc_UWord32 availableTS, int noPacket,
-                                            int cngPacket, int prevPlayMode,
-                                            enum WebRtcNetEQPlayoutMode playoutMode,
-                                            int timestampsPerCall, int NoOfExpandCalls,
-                                            WebRtc_Word16 fs_mult,
-                                            WebRtc_Word16 lastModeBGNonly, int playDtmf);
+uint16_t WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, int16_t frameSize,
+                                      int32_t cur_size, uint32_t targetTS,
+                                      uint32_t availableTS, int noPacket,
+                                      int cngPacket, int prevPlayMode,
+                                      enum WebRtcNetEQPlayoutMode playoutMode,
+                                      int timestampsPerCall, int NoOfExpandCalls,
+                                      int16_t fs_mult,
+                                      int16_t lastModeBGNonly, int playDtmf);
 
 #endif
diff --git a/modules/audio_coding/neteq/bufstats_decision.c b/modules/audio_coding/neteq/bufstats_decision.c
index cf7c0b0..352e050 100644
--- a/modules/audio_coding/neteq/bufstats_decision.c
+++ b/modules/audio_coding/neteq/bufstats_decision.c
@@ -26,22 +26,22 @@
 
 #define NETEQ_BUFSTAT_20MS_Q7 2560 /* = 20 ms in Q7  */
 
-WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
-                                            WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
-                                            WebRtc_UWord32 availableTS, int noPacket,
-                                            int cngPacket, int prevPlayMode,
-                                            enum WebRtcNetEQPlayoutMode playoutMode,
-                                            int timestampsPerCall, int NoOfExpandCalls,
-                                            WebRtc_Word16 fs_mult,
-                                            WebRtc_Word16 lastModeBGNonly, int playDtmf)
+uint16_t WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, int16_t frameSize,
+                                      int32_t cur_size, uint32_t targetTS,
+                                      uint32_t availableTS, int noPacket,
+                                      int cngPacket, int prevPlayMode,
+                                      enum WebRtcNetEQPlayoutMode playoutMode,
+                                      int timestampsPerCall, int NoOfExpandCalls,
+                                      int16_t fs_mult,
+                                      int16_t lastModeBGNonly, int playDtmf)
 {
 
     int currentDelayMs;
-    WebRtc_Word32 currSizeSamples = cur_size;
+    int32_t currSizeSamples = cur_size;
     int extraDelayPacketsQ8 = 0;
 
     /* Avoid overflow if the buffer size should be really large (cur_size is limited 256ms) */
-    WebRtc_Word32 curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4);
+    int32_t curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4);
     int level_limit_hi, level_limit_lo;
 
     inst->Automode_inst.prevTimeScale &= (prevPlayMode == MODE_SUCCESS_ACCELERATE
@@ -68,7 +68,7 @@
 
     /* Calculate VQmon related variables */
     /* avgDelay = avgDelay*(511/512) + currentDelay*(1/512) (sample ms delay in Q8) */
-    inst->avgDelayMsQ8 = (WebRtc_Word16) (WEBRTC_SPL_MUL_16_16_RSFT(inst->avgDelayMsQ8,511,9)
+    inst->avgDelayMsQ8 = (int16_t) (WEBRTC_SPL_MUL_16_16_RSFT(inst->avgDelayMsQ8,511,9)
         + (cur_size >> 9));
 
     /* Update maximum delay if needed */
@@ -106,7 +106,7 @@
         if (cngPacket)
         {
             /* signed difference between wanted and available TS */
-            WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
+            int32_t diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
             int32_t optimal_level_samp = (inst->Automode_inst.optBufLevel *
                 inst->Automode_inst.packetSpeechLenSamp) >> 8;
             int32_t excess_waiting_time_samp = -diffTS - optimal_level_samp;
@@ -225,13 +225,13 @@
             /* Check that we do not play a packet "too early" */
             if ((prevPlayMode == MODE_EXPAND)
                 && (availableTS - targetTS
-                    < (WebRtc_UWord32) WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
-                        (WebRtc_Word16)REINIT_AFTER_EXPANDS))
+                    < (uint32_t) WEBRTC_SPL_MUL_16_16((int16_t)timestampsPerCall,
+                        (int16_t)REINIT_AFTER_EXPANDS))
                 && (NoOfExpandCalls < MAX_WAIT_FOR_PACKET)
                 && (availableTS
                     > targetTS
-                        + WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
-                            (WebRtc_Word16)NoOfExpandCalls))
+                        + WEBRTC_SPL_MUL_16_16((int16_t)timestampsPerCall,
+                            (int16_t)NoOfExpandCalls))
                 && (inst->Automode_inst.buffLevelFilt <= inst->Automode_inst.optBufLevel
                     + extraDelayPacketsQ8))
             {
@@ -256,7 +256,7 @@
                  * precaution), but make sure that the number of samples in buffer is no
                  * higher than 4 times the optimal level.
                  */
-                WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
+                int32_t diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
                 int val = ((inst->Automode_inst.optBufLevel +
                     extraDelayPacketsQ8) *
                     inst->Automode_inst.packetSpeechLenSamp) >> 6;
@@ -310,7 +310,7 @@
     { /* kPlayoutOff or kPlayoutFax */
         if (cngPacket)
         {
-            if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
+            if (((int32_t) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
             {
                 /* time to play this packet now */
                 return BUFSTATS_DO_RFC3389CNG_PACKET;
@@ -363,7 +363,7 @@
         }
         else
         {
-            if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
+            if (((int32_t) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
             {
                 return BUFSTATS_DO_NORMAL;
             }
diff --git a/modules/audio_coding/neteq/cng_internal.c b/modules/audio_coding/neteq/cng_internal.c
index f3a10dc..cb4878f 100644
--- a/modules/audio_coding/neteq/cng_internal.c
+++ b/modules/audio_coding/neteq/cng_internal.c
@@ -43,12 +43,12 @@
 #ifdef NETEQ_CNG_CODEC
 /* Must compile NetEQ with CNG support to enable this function */
 
-int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len)
+int WebRtcNetEQ_Cng(DSPInst_t *inst, int16_t *pw16_outData, int len)
 {
-    WebRtc_Word16 w16_winMute = 0; /* mixing factor for overlap data */
-    WebRtc_Word16 w16_winUnMute = 0; /* mixing factor for comfort noise */
-    WebRtc_Word16 w16_winMuteInc = 0; /* mixing factor increment (negative) */
-    WebRtc_Word16 w16_winUnMuteInc = 0; /* mixing factor increment */
+    int16_t w16_winMute = 0; /* mixing factor for overlap data */
+    int16_t w16_winUnMute = 0; /* mixing factor for comfort noise */
+    int16_t w16_winMuteInc = 0; /* mixing factor increment (negative) */
+    int16_t w16_winUnMuteInc = 0; /* mixing factor increment */
     int i;
 
     /*
@@ -61,7 +61,7 @@
 
         /* Generate len samples + overlap */
         if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData,
-            (WebRtc_Word16) (len + inst->ExpandInst.w16_overlap), 1) < 0)
+            (int16_t) (len + inst->ExpandInst.w16_overlap), 1) < 0)
         {
             /* error returned */
             return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
@@ -116,7 +116,7 @@
         for (i = 0; i < inst->ExpandInst.w16_overlap; i++)
         {
             /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */
-            inst->ExpandInst.pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+            inst->ExpandInst.pw16_overlapVec[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
                 WEBRTC_SPL_MUL_16_16(
                     inst->ExpandInst.pw16_overlapVec[i], w16_winMute) +
                 WEBRTC_SPL_MUL_16_16(pw16_outData[i], w16_winUnMute)
@@ -140,7 +140,7 @@
         /* This is a subsequent CNG call; no special overlap needed */
 
         /* Generate len samples */
-        if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, (WebRtc_Word16) len, 0) < 0)
+        if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, (int16_t) len, 0) < 0)
         {
             /* error returned */
             return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
diff --git a/modules/audio_coding/neteq/codec_db.c b/modules/audio_coding/neteq/codec_db.c
index c3fc940..8d8389a 100644
--- a/modules/audio_coding/neteq/codec_db.c
+++ b/modules/audio_coding/neteq/codec_db.c
@@ -28,8 +28,8 @@
 {
     int i;
 
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) inst, 0,
-        sizeof(CodecDbInst_t) / sizeof(WebRtc_Word16));
+    WebRtcSpl_MemSetW16((int16_t*) inst, 0,
+        sizeof(CodecDbInst_t) / sizeof(int16_t));
 
     for (i = 0; i < NUM_TOTAL_CODECS; i++)
     {
@@ -54,13 +54,13 @@
  */
 
 int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
-                      WebRtc_Word16 payloadType, FuncDecode funcDecode,
+                      int16_t payloadType, FuncDecode funcDecode,
                       FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
                       FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
                       FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
                       FuncUpdBWEst funcUpdBWEst, FuncDurationEst funcDurationEst,
                       FuncGetErrorCode funcGetErrorCode, void* codec_state,
-                      WebRtc_UWord16 codec_fs)
+                      uint16_t codec_fs)
 {
 
     int temp;
@@ -404,8 +404,8 @@
     }
     else
     {
-        WebRtcSpl_MemSetW16((WebRtc_Word16*) ptr_inst, 0,
-            sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
+        WebRtcSpl_MemSetW16((int16_t*) ptr_inst, 0,
+            sizeof(CodecFuncInst_t) / sizeof(int16_t));
         return CODEC_DB_NOT_EXIST1;
     }
 }
@@ -723,7 +723,7 @@
 /*
  * Return the sample rate for the codec with the given payload type, 0 if error
  */
-WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType)
+uint16_t WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType)
 {
     int i;
     CodecFuncInst_t codecInst;
diff --git a/modules/audio_coding/neteq/codec_db.h b/modules/audio_coding/neteq/codec_db.h
index 102dec8..cc4b48e 100644
--- a/modules/audio_coding/neteq/codec_db.h
+++ b/modules/audio_coding/neteq/codec_db.h
@@ -34,10 +34,10 @@
 typedef struct
 {
 
-    WebRtc_Word16 position[NUM_TOTAL_CODECS];
-    WebRtc_Word16 nrOfCodecs;
+    int16_t position[NUM_TOTAL_CODECS];
+    int16_t nrOfCodecs;
 
-    WebRtc_Word16 payloadType[NUM_CODECS];
+    int16_t payloadType[NUM_CODECS];
     FuncDecode funcDecode[NUM_CODECS];
     FuncDecode funcDecodeRCU[NUM_CODECS];
     FuncDecodePLC funcDecodePLC[NUM_CODECS];
@@ -49,8 +49,8 @@
     FuncDurationEst funcDurationEst[NUM_CODECS];
     FuncGetErrorCode funcGetErrorCode[NUM_CODECS];
     void * codec_state[NUM_CODECS];
-    WebRtc_UWord16 codec_fs[NUM_CODECS];
-    WebRtc_Word16 CNGpayloadType[NUM_CNG_CODECS];
+    uint16_t codec_fs[NUM_CODECS];
+    int16_t CNGpayloadType[NUM_CNG_CODECS];
 
 } CodecDbInst_t;
 
@@ -58,8 +58,8 @@
 
 typedef struct
 {
-    WebRtc_Word16 deltaBytes;
-    WebRtc_Word16 deltaTime;
+    int16_t deltaBytes;
+    int16_t deltaTime;
 } SplitInfo_t;
 
 /*
@@ -71,13 +71,13 @@
  * Adds a new codec to the database.
  */
 int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
-                      WebRtc_Word16 payloadType, FuncDecode funcDecode,
+                      int16_t payloadType, FuncDecode funcDecode,
                       FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
                       FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
                       FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
                       FuncUpdBWEst funcUpdBWEst, FuncDurationEst funcDurationEst,
                       FuncGetErrorCode funcGetErrorCode, void* codec_state,
-                      WebRtc_UWord16 codec_fs);
+                      uint16_t codec_fs);
 
 /*
  * Removes a codec from the database.
@@ -122,7 +122,7 @@
 /*
  * Return the sample rate for the codec with the given payload type, 0 if error.
  */
-WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType);
+uint16_t WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType);
 
 #endif
 
diff --git a/modules/audio_coding/neteq/codec_db_defines.h b/modules/audio_coding/neteq/codec_db_defines.h
index ffe1e85..d97306a 100644
--- a/modules/audio_coding/neteq/codec_db_defines.h
+++ b/modules/audio_coding/neteq/codec_db_defines.h
@@ -23,45 +23,45 @@
 /*
  * Pointer to decoder function.
  */
-typedef WebRtc_Word16 (*FuncDecode)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len,
-                                    WebRtc_Word16* decoded, WebRtc_Word16* speechType);
+typedef int16_t (*FuncDecode)(void* state, int16_t* encoded, int16_t len,
+                                    int16_t* decoded, int16_t* speechType);
 
 /*
  * Pointer to PLC function.
  */
-typedef WebRtc_Word16 (*FuncDecodePLC)(void* state, WebRtc_Word16* decodec,
-                                       WebRtc_Word16 frames);
+typedef int16_t (*FuncDecodePLC)(void* state, int16_t* decodec,
+                                       int16_t frames);
 
 /*
  * Pointer to decoder init function.
  */
-typedef WebRtc_Word16 (*FuncDecodeInit)(void* state);
+typedef int16_t (*FuncDecodeInit)(void* state);
 
 /*
  * Pointer to add late packet function.
  */
-typedef WebRtc_Word16
-                (*FuncAddLatePkt)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len);
+typedef int16_t
+                (*FuncAddLatePkt)(void* state, int16_t* encoded, int16_t len);
 
 /*
  * Pointer to get MD infofunction.
  */
-typedef WebRtc_Word16 (*FuncGetMDinfo)(void* state);
+typedef int16_t (*FuncGetMDinfo)(void* state);
 
 /*
  * Pointer to pitch info function.
  * Return 0 for unvoiced, -1 if pitch not availiable.
  */
-typedef WebRtc_Word16 (*FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
-                                          WebRtc_Word16* length);
+typedef int16_t (*FuncGetPitchInfo)(void* state, int16_t* encoded,
+                                          int16_t* length);
 
 /*
  *  Pointer to the update bandwidth estimate function
  */
-typedef WebRtc_Word16 (*FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
-                                      WebRtc_Word32 packet_size,
-                                      WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts,
-                                      WebRtc_UWord32 arr_ts);
+typedef int16_t (*FuncUpdBWEst)(void* state, const uint16_t *encoded,
+                                      int32_t packet_size,
+                                      uint16_t rtp_seq_number, uint32_t send_ts,
+                                      uint32_t arr_ts);
 
 /*
  *  Pointer to the frame size estimate function.
@@ -73,7 +73,7 @@
 /*
  *  Pointer to error code function
  */
-typedef WebRtc_Word16 (*FuncGetErrorCode)(void* state);
+typedef int16_t (*FuncGetErrorCode)(void* state);
 
 typedef struct CodecFuncInst_t_
 {
@@ -88,8 +88,8 @@
     FuncDurationEst funcDurationEst;
     FuncGetErrorCode funcGetErrorCode;
     void * codec_state;
-    WebRtc_UWord16 codec_fs;
-    WebRtc_UWord32 timeStamp;
+    uint16_t codec_fs;
+    uint32_t timeStamp;
 
 } CodecFuncInst_t;
 
diff --git a/modules/audio_coding/neteq/correlator.c b/modules/audio_coding/neteq/correlator.c
index 97c41da..0a4404a 100644
--- a/modules/audio_coding/neteq/correlator.c
+++ b/modules/audio_coding/neteq/correlator.c
@@ -17,9 +17,9 @@
 /* Scratch usage:
 
  Type           Name                size  startpos  endpos
- WebRtc_Word16  pw16_corrVec        62    0         61
- WebRtc_Word16  pw16_data_ds        124   0         123
- WebRtc_Word32  pw32_corr           2*54  124       231
+ int16_t  pw16_corrVec        62    0         61
+ int16_t  pw16_data_ds        124   0         123
+ int32_t  pw32_corr           2*54  124       231
 
  Total:  232
  */
@@ -30,48 +30,48 @@
 
 #define NETEQ_CORRELATOR_DSVECLEN 		124	/* 124 = 60 + 10 + 54 */
 
-WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
+int16_t WebRtcNetEQ_Correlator(DSPInst_t *inst,
 #ifdef SCRATCH
-                                     WebRtc_Word16 *pw16_scratchPtr,
+                               int16_t *pw16_scratchPtr,
 #endif
-                                     WebRtc_Word16 *pw16_data,
-                                     WebRtc_Word16 w16_dataLen,
-                                     WebRtc_Word16 *pw16_corrOut,
-                                     WebRtc_Word16 *pw16_corrScale)
+                               int16_t *pw16_data,
+                               int16_t w16_dataLen,
+                               int16_t *pw16_corrOut,
+                               int16_t *pw16_corrScale)
 {
-    WebRtc_Word16 w16_corrLen = 60;
+    int16_t w16_corrLen = 60;
 #ifdef SCRATCH
-    WebRtc_Word16 *pw16_data_ds = pw16_scratchPtr + SCRATCH_pw16_corrVec;
-    WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
-    /*	WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;*/
+    int16_t *pw16_data_ds = pw16_scratchPtr + SCRATCH_pw16_corrVec;
+    int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_pw32_corr);
+    /*	int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;*/
 #else
-    WebRtc_Word16 pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN];
-    WebRtc_Word32 pw32_corr[54];
-    /*	WebRtc_Word16 pw16_corrVec[4+54+4];*/
+    int16_t pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN];
+    int32_t pw32_corr[54];
+    /*	int16_t pw16_corrVec[4+54+4];*/
 #endif
-    /*	WebRtc_Word16 *pw16_corr=&pw16_corrVec[4];*/
-    WebRtc_Word16 w16_maxVal;
-    WebRtc_Word32 w32_maxVal;
-    WebRtc_Word16 w16_normVal;
-    WebRtc_Word16 w16_normVal2;
-    /*	WebRtc_Word16 w16_corrUpsLen;*/
-    WebRtc_Word16 *pw16_B = NULL;
-    WebRtc_Word16 w16_Blen = 0;
-    WebRtc_Word16 w16_factor = 0;
+    /*	int16_t *pw16_corr=&pw16_corrVec[4];*/
+    int16_t w16_maxVal;
+    int32_t w32_maxVal;
+    int16_t w16_normVal;
+    int16_t w16_normVal2;
+    /*	int16_t w16_corrUpsLen;*/
+    int16_t *pw16_B = NULL;
+    int16_t w16_Blen = 0;
+    int16_t w16_factor = 0;
 
     /* Set constants depending on frequency used */
     if (inst->fs == 8000)
     {
         w16_Blen = 3;
         w16_factor = 2;
-        pw16_B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
+        pw16_B = (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl;
 #ifdef NETEQ_WIDEBAND
     }
     else if (inst->fs==16000)
     {
         w16_Blen = 5;
         w16_factor = 4;
-        pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl;
+        pw16_B = (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl;
 #endif
 #ifdef NETEQ_32KHZ_WIDEBAND
     }
@@ -79,7 +79,7 @@
     {
         w16_Blen = 7;
         w16_factor = 8;
-        pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl;
+        pw16_B = (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl;
 #endif
 #ifdef NETEQ_48KHZ_WIDEBAND
     }
@@ -87,19 +87,19 @@
     {
         w16_Blen = 7;
         w16_factor = 12;
-        pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl;
+        pw16_B = (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl;
 #endif
     }
 
     /* Downsample data in order to work on a 4 kHz sampled signal */
     WebRtcSpl_DownsampleFast(
         pw16_data + w16_dataLen - (NETEQ_CORRELATOR_DSVECLEN * w16_factor),
-        (WebRtc_Word16) (NETEQ_CORRELATOR_DSVECLEN * w16_factor), pw16_data_ds,
-        NETEQ_CORRELATOR_DSVECLEN, pw16_B, w16_Blen, w16_factor, (WebRtc_Word16) 0);
+        (int16_t) (NETEQ_CORRELATOR_DSVECLEN * w16_factor), pw16_data_ds,
+        NETEQ_CORRELATOR_DSVECLEN, pw16_B, w16_Blen, w16_factor, (int16_t) 0);
 
     /* Normalize downsampled vector to using entire 16 bit */
     w16_maxVal = WebRtcSpl_MaxAbsValueW16(pw16_data_ds, 124);
-    w16_normVal = 16 - WebRtcSpl_NormW32((WebRtc_Word32) w16_maxVal);
+    w16_normVal = 16 - WebRtcSpl_NormW32((int32_t) w16_maxVal);
     WebRtcSpl_VectorBitShiftW16(pw16_data_ds, NETEQ_CORRELATOR_DSVECLEN, pw16_data_ds,
         w16_normVal);
 
diff --git a/modules/audio_coding/neteq/dsp.c b/modules/audio_coding/neteq/dsp.c
index 82c54ea..e840ad1 100644
--- a/modules/audio_coding/neteq/dsp.c
+++ b/modules/audio_coding/neteq/dsp.c
@@ -24,40 +24,40 @@
  Coefficients are in Q12. */
 
 /* {0.3, 0.4, 0.3} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[] = { 1229, 1638, 1229 };
+const int16_t WebRtcNetEQ_kDownsample8kHzTbl[] = { 1229, 1638, 1229 };
 
 #ifdef NETEQ_WIDEBAND
 /* {0.15, 0.2, 0.3, 0.2, 0.15} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[] =
+const int16_t WebRtcNetEQ_kDownsample16kHzTbl[] =
 {   614, 819, 1229, 819, 614};
 #endif
 
 #ifdef NETEQ_32KHZ_WIDEBAND
 /* {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[] =
+const int16_t WebRtcNetEQ_kDownsample32kHzTbl[] =
 {   584, 512, 625, 667, 625, 512, 584};
 #endif
 
 #ifdef NETEQ_48KHZ_WIDEBAND
 /* {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[] =
+const int16_t WebRtcNetEQ_kDownsample48kHzTbl[] =
 {   1019, 390, 427, 440, 427, 390, 1019};
 #endif
 
 /* Constants used in expand function WebRtcNetEQ_Expand */
 
 /* Q12: -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 */
-const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[4] = { -5179, 19931, -16422, 5776 };
+const int16_t WebRtcNetEQ_kMixFractionFuncTbl[4] = { -5179, 19931, -16422, 5776 };
 
 /* Tabulated divisions to save complexity */
 /* 1049/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k1049div[7] = { 0, 1049, 524, 349, 262, 209, 174 };
+const int16_t WebRtcNetEQ_k1049div[7] = { 0, 1049, 524, 349, 262, 209, 174 };
 
 /* 2097/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k2097div[7] = { 0, 2097, 1048, 699, 524, 419, 349 };
+const int16_t WebRtcNetEQ_k2097div[7] = { 0, 2097, 1048, 699, 524, 419, 349 };
 
 /* 5243/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k5243div[7] = { 0, 5243, 2621, 1747, 1310, 1048, 873 };
+const int16_t WebRtcNetEQ_k5243div[7] = { 0, 5243, 2621, 1747, 1310, 1048, 873 };
 
 #ifdef WEBRTC_NETEQ_40BITACC_TEST
 /*
@@ -89,17 +89,17 @@
  *		- crossCorr		: The cross correlation in Q-rShift
  */
 
-void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr,
-    WebRtc_Word16 *seq1,
-    WebRtc_Word16 *seq2,
-    WebRtc_Word16 dimSeq,
-    WebRtc_Word16 dimCrossCorr,
-    WebRtc_Word16 rShift,
-    WebRtc_Word16 step_seq2)
+void WebRtcNetEQ_40BitAccCrossCorr(int32_t *crossCorr,
+    int16_t *seq1,
+    int16_t *seq2,
+    int16_t dimSeq,
+    int16_t dimCrossCorr,
+    int16_t rShift,
+    int16_t step_seq2)
 {
     int i, j;
-    WebRtc_Word16 *seq1Ptr, *seq2Ptr;
-    WebRtc_Word64 acc;
+    int16_t *seq1Ptr, *seq2Ptr;
+    int64_t acc;
 
     for (i = 0; i < dimCrossCorr; i++)
     {
@@ -117,7 +117,7 @@
             seq2Ptr++;
         }
 
-        (*crossCorr) = (WebRtc_Word32) (acc >> rShift);
+        (*crossCorr) = (int32_t) (acc >> rShift);
         crossCorr++;
     }
 }
@@ -125,7 +125,7 @@
 /****************************************************************************
  * WebRtcNetEQ_40BitAccDotW16W16(...)
  *
- * Calculates the dot product between two vectors (WebRtc_Word16)
+ * Calculates the dot product between two vectors (int16_t)
  *
  * Input:
  *		- vector1		: Vector 1
@@ -137,14 +137,14 @@
  * Return value			: The dot product
  */
 
-WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1,
-    WebRtc_Word16 *vector2,
+int32_t WebRtcNetEQ_40BitAccDotW16W16(int16_t *vector1,
+    int16_t *vector2,
     int len,
     int scaling)
 {
-    WebRtc_Word32 sum;
+    int32_t sum;
     int i;
-    WebRtc_Word64 acc;
+    int64_t acc;
 
     acc = 0;
     for (i = 0; i < len; i++)
@@ -152,7 +152,7 @@
         acc += WEBRTC_SPL_MUL_16_16(*vector1++, *vector2++);
     }
 
-    sum = (WebRtc_Word32) (acc >> scaling);
+    sum = (int32_t) (acc >> scaling);
 
     return(sum);
 }
@@ -175,11 +175,11 @@
  *                      : non-zero - error
  */
 
-int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs)
+int WebRtcNetEQ_DSPInit(DSPInst_t *inst, uint16_t fs)
 {
 
     int res = 0;
-    WebRtc_Word16 fs_mult;
+    int16_t fs_mult;
 
     /* Pointers and values to save before clearing the instance */
 #ifdef NETEQ_CNG_CODEC
@@ -193,22 +193,22 @@
     VADInitFunction savedVADinit = inst->VADInst.initFunction;
     VADSetmodeFunction savedVADsetmode = inst->VADInst.setmodeFunction;
     VADFunction savedVADfunc = inst->VADInst.VADFunction;
-    WebRtc_Word16 savedVADEnabled = inst->VADInst.VADEnabled;
+    int16_t savedVADEnabled = inst->VADInst.VADEnabled;
     int savedVADMode = inst->VADInst.VADMode;
 #endif /* NETEQ_VAD */
     DSPStats_t saveStats;
-    WebRtc_Word16 saveMsPerCall = inst->millisecondsPerCall;
+    int16_t saveMsPerCall = inst->millisecondsPerCall;
     enum BGNMode saveBgnMode = inst->BGNInst.bgnMode;
 #ifdef NETEQ_STEREO
     MasterSlaveInfo saveMSinfo;
 #endif
 
     /* copy contents of statInst to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveStats, &(inst->statInst),
-        sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
+        sizeof(DSPStats_t)/sizeof(int16_t));
 
 #ifdef NETEQ_STEREO
     /* copy contents of msInfo to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveMSinfo, &(inst->msInfo),
-        sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
+        sizeof(MasterSlaveInfo)/sizeof(int16_t));
 #endif
 
     /* check that the sample rate is valid */
@@ -232,14 +232,14 @@
     fs_mult = WebRtcSpl_DivW32W16ResW16(fs, 8000);
 
     /* Set everything to zero since most variables should be zero at start */
-    WebRtcSpl_MemSetW16((WebRtc_Word16 *) inst, 0, sizeof(DSPInst_t) / sizeof(WebRtc_Word16));
+    WebRtcSpl_MemSetW16((int16_t *) inst, 0, sizeof(DSPInst_t) / sizeof(int16_t));
 
     /* Restore saved pointers  */
 #ifdef NETEQ_CNG_CODEC
     inst->CNG_Codec_inst = (CNG_dec_inst *)savedPtr1;
 #endif
-    inst->pw16_readAddress = (WebRtc_Word16 *) savedPtr2;
-    inst->pw16_writeAddress = (WebRtc_Word16 *) savedPtr3;
+    inst->pw16_readAddress = (int16_t *) savedPtr2;
+    inst->pw16_writeAddress = (int16_t *) savedPtr3;
     inst->main_inst = savedPtr4;
 #ifdef NETEQ_VAD
     inst->VADInst.VADState = savedVADptr;
@@ -289,11 +289,11 @@
     inst->BGNInst.bgnMode = saveBgnMode;
 
     /* Recreate statistics counters */WEBRTC_SPL_MEMCPY_W16(&(inst->statInst), &saveStats,
-        sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
+        sizeof(DSPStats_t)/sizeof(int16_t));
 
 #ifdef NETEQ_STEREO
     /* Recreate MSinfo */WEBRTC_SPL_MEMCPY_W16(&(inst->msInfo), &saveMSinfo,
-        sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
+        sizeof(MasterSlaveInfo)/sizeof(int16_t));
 #endif
 
 #ifdef NETEQ_CNG_CODEC
@@ -335,8 +335,8 @@
 {
 
     /* set shared-memory addresses in the DSP instance */
-    inst->pw16_readAddress = (WebRtc_Word16 *) data2DspAddress;
-    inst->pw16_writeAddress = (WebRtc_Word16 *) data2McuAddress;
+    inst->pw16_readAddress = (int16_t *) data2DspAddress;
+    inst->pw16_writeAddress = (int16_t *) data2McuAddress;
 
     /* set pointer to main NetEQ instance */
     inst->main_inst = (void *) mainInst;
@@ -414,7 +414,7 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs)
+int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, uint16_t fs)
 {
 
     int res = 0;
@@ -505,7 +505,7 @@
 
 int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst)
 {
-    WebRtc_Word16 fs_mult;
+    int16_t fs_mult;
 
     /* calcualte fs/8000 */
     fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
diff --git a/modules/audio_coding/neteq/dsp.h b/modules/audio_coding/neteq/dsp.h
index 40b7831..5452272 100644
--- a/modules/audio_coding/neteq/dsp.h
+++ b/modules/audio_coding/neteq/dsp.h
@@ -92,15 +92,15 @@
 /* Constant tables */
 /*******************/
 
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_k1049div[];
-extern const WebRtc_Word16 WebRtcNetEQ_k2097div[];
-extern const WebRtc_Word16 WebRtcNetEQ_k5243div[];
+extern const int16_t WebRtcNetEQ_kDownsample8kHzTbl[];
+extern const int16_t WebRtcNetEQ_kDownsample16kHzTbl[];
+extern const int16_t WebRtcNetEQ_kDownsample32kHzTbl[];
+extern const int16_t WebRtcNetEQ_kDownsample48kHzTbl[];
+extern const int16_t WebRtcNetEQ_kRandnTbl[];
+extern const int16_t WebRtcNetEQ_kMixFractionFuncTbl[];
+extern const int16_t WebRtcNetEQ_k1049div[];
+extern const int16_t WebRtcNetEQ_k2097div[];
+extern const int16_t WebRtcNetEQ_k5243div[];
 
 
 
@@ -142,16 +142,16 @@
 typedef struct BGNInst_t_
 {
 
-    WebRtc_Word32 w32_energy;
-    WebRtc_Word32 w32_energyMax;
-    WebRtc_Word32 w32_energyUpdate;
-    WebRtc_Word32 w32_energyUpdateLow;
-    WebRtc_Word16 pw16_filterState[BGN_LPC_ORDER];
-    WebRtc_Word16 pw16_filter[BGN_LPC_ORDER + 1];
-    WebRtc_Word16 w16_mutefactor;
-    WebRtc_Word16 w16_scale;
-    WebRtc_Word16 w16_scaleShift;
-    WebRtc_Word16 w16_initialized;
+    int32_t w32_energy;
+    int32_t w32_energyMax;
+    int32_t w32_energyUpdate;
+    int32_t w32_energyUpdateLow;
+    int16_t pw16_filterState[BGN_LPC_ORDER];
+    int16_t pw16_filter[BGN_LPC_ORDER + 1];
+    int16_t w16_mutefactor;
+    int16_t w16_scale;
+    int16_t w16_scaleShift;
+    int16_t w16_initialized;
     enum BGNMode bgnMode;
 
 } BGNInst_t;
@@ -160,24 +160,24 @@
 typedef struct ExpandInst_t_
 {
 
-    WebRtc_Word16 w16_overlap; /* Constant, 5 for NB and 10 for WB */
-    WebRtc_Word16 w16_consecExp; /* Number of consecutive expand calls */
-    WebRtc_Word16 *pw16_arFilter; /* length [UNVOICED_LPC_ORDER+1]	*/
-    WebRtc_Word16 *pw16_arState; /* length [UNVOICED_LPC_ORDER]		*/
-    WebRtc_Word16 w16_arGain;
-    WebRtc_Word16 w16_arGainScale;
-    WebRtc_Word16 w16_vFraction; /* Q14 */
-    WebRtc_Word16 w16_currentVFraction; /* Q14 */
-    WebRtc_Word16 *pw16_expVecs[2];
-    WebRtc_Word16 w16_lags[3];
-    WebRtc_Word16 w16_maxLag;
-    WebRtc_Word16 *pw16_overlapVec; /* last samples of speech history */
-    WebRtc_Word16 w16_lagsDirection;
-    WebRtc_Word16 w16_lagsPosition;
-    WebRtc_Word16 w16_expandMuteFactor; /* Q14 */
-    WebRtc_Word16 w16_stopMuting;
-    WebRtc_Word16 w16_onset;
-    WebRtc_Word16 w16_muteSlope; /* Q20 */
+    int16_t w16_overlap; /* Constant, 5 for NB and 10 for WB */
+    int16_t w16_consecExp; /* Number of consecutive expand calls */
+    int16_t *pw16_arFilter; /* length [UNVOICED_LPC_ORDER+1]	*/
+    int16_t *pw16_arState; /* length [UNVOICED_LPC_ORDER]		*/
+    int16_t w16_arGain;
+    int16_t w16_arGainScale;
+    int16_t w16_vFraction; /* Q14 */
+    int16_t w16_currentVFraction; /* Q14 */
+    int16_t *pw16_expVecs[2];
+    int16_t w16_lags[3];
+    int16_t w16_maxLag;
+    int16_t *pw16_overlapVec; /* last samples of speech history */
+    int16_t w16_lagsDirection;
+    int16_t w16_lagsPosition;
+    int16_t w16_expandMuteFactor; /* Q14 */
+    int16_t w16_stopMuting;
+    int16_t w16_onset;
+    int16_t w16_muteSlope; /* Q20 */
 
 } ExpandInst_t;
 
@@ -190,7 +190,7 @@
  */
 typedef int (*VADInitFunction)(void *VAD_inst);
 typedef int (*VADSetmodeFunction)(void *VAD_inst, int mode);
-typedef int (*VADFunction)(void *VAD_inst, int fs, WebRtc_Word16 *frame,
+typedef int (*VADFunction)(void *VAD_inst, int fs, int16_t *frame,
                            int frameLen);
 
 /* Post-decode VAD instance (sub-instance of NETEQDSP_inst) */
@@ -199,10 +199,10 @@
 
     void *VADState; /* pointer to a VAD instance */
 
-    WebRtc_Word16 VADEnabled; /* 1 if enabled, 0 if disabled */
+    int16_t VADEnabled; /* 1 if enabled, 0 if disabled */
     int VADMode; /* mode parameter to pass to the VAD function */
     int VADDecision; /* 1 for active, 0 for passive */
-    WebRtc_Word16 SIDintervalCounter; /* reset when decoding CNG/SID frame,
+    int16_t SIDintervalCounter; /* reset when decoding CNG/SID frame,
      increment for each recout call */
 
     /* Function pointers */
@@ -224,13 +224,13 @@
 
     enum MasterSlaveExtraInfo  extraInfo;
 
-    WebRtc_UWord16 instruction;
-    WebRtc_Word16 distLag;
-    WebRtc_Word16 corrLag;
-    WebRtc_Word16 bestIndex;
+    uint16_t instruction;
+    int16_t distLag;
+    int16_t corrLag;
+    int16_t bestIndex;
 
-    WebRtc_UWord32 endTimestamp;
-    WebRtc_UWord16 samplesLeftWithOverlap;
+    uint32_t endTimestamp;
+    uint16_t samplesLeftWithOverlap;
 
 } MasterSlaveInfo;
 #endif
@@ -241,13 +241,13 @@
 {
 
     /* MCU/DSP Communication layer */
-    WebRtc_Word16 *pw16_readAddress;
-    WebRtc_Word16 *pw16_writeAddress;
+    int16_t *pw16_readAddress;
+    int16_t *pw16_writeAddress;
     void *main_inst;
 
     /* Output frame size in ms and samples */
-    WebRtc_Word16 millisecondsPerCall;
-    WebRtc_Word16 timestampsPerCall;
+    int16_t millisecondsPerCall;
+    int16_t timestampsPerCall;
 
     /*
      *	Example of speech buffer
@@ -263,27 +263,27 @@
      *      new data at the end.
      */
 
-    WebRtc_Word16 speechBuffer[SPEECH_BUF_SIZE]; /* History/future speech buffer */
+    int16_t speechBuffer[SPEECH_BUF_SIZE]; /* History/future speech buffer */
     int curPosition; /* Next sample to play */
     int endPosition; /* Position that ends future data */
-    WebRtc_UWord32 endTimestamp; /* Timestamp value at end of future data */
-    WebRtc_UWord32 videoSyncTimestamp; /* (Estimated) timestamp of the last
+    uint32_t endTimestamp; /* Timestamp value at end of future data */
+    uint32_t videoSyncTimestamp; /* (Estimated) timestamp of the last
      played sample (usually same as
      endTimestamp-(endPosition-curPosition)
      except during Expand and CNG) */
-    WebRtc_UWord16 fs; /* sample rate in Hz */
-    WebRtc_Word16 w16_frameLen; /* decoder frame length in samples */
-    WebRtc_Word16 w16_mode; /* operation used during last RecOut call */
-    WebRtc_Word16 w16_muteFactor; /* speech mute factor in Q14 */
-    WebRtc_Word16 *pw16_speechHistory; /* beginning of speech history during Expand */
-    WebRtc_Word16 w16_speechHistoryLen; /* 256 for NB and 512 for WB */
+    uint16_t fs; /* sample rate in Hz */
+    int16_t w16_frameLen; /* decoder frame length in samples */
+    int16_t w16_mode; /* operation used during last RecOut call */
+    int16_t w16_muteFactor; /* speech mute factor in Q14 */
+    int16_t *pw16_speechHistory; /* beginning of speech history during Expand */
+    int16_t w16_speechHistoryLen; /* 256 for NB and 512 for WB */
 
     /* random noise seed parameters */
-    WebRtc_Word16 w16_seedInc;
-    WebRtc_UWord32 uw16_seed;
+    int16_t w16_seedInc;
+    uint32_t uw16_seed;
 
     /* VQmon related variable */
-    WebRtc_Word16 w16_concealedTS;
+    int16_t w16_concealedTS;
 
     /*****************/
     /* Sub-instances */
@@ -344,7 +344,7 @@
  *                      : non-zero - error
  */
 
-int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs);
+int WebRtcNetEQ_DSPInit(DSPInst_t *inst, uint16_t fs);
 
 /****************************************************************************
  * WebRtcNetEQ_AddressInit(...)
@@ -420,8 +420,8 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                       WebRtc_Word16 BGNonly);
+int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, int16_t *pw16_outData, int16_t *pw16_len,
+                       int16_t BGNonly);
 
 /****************************************************************************
  * WebRtcNetEQ_Normal(...)
@@ -447,10 +447,10 @@
 
 int WebRtcNetEQ_Normal(DSPInst_t *inst,
 #ifdef SCRATCH
-                       WebRtc_Word16 *pw16_scratchPtr,
+                       int16_t *pw16_scratchPtr,
 #endif
-                       WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
-                       WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
+                       int16_t *pw16_decoded, int16_t len,
+                       int16_t *pw16_outData, int16_t *pw16_len);
 
 /****************************************************************************
  * WebRtcNetEQ_Expand(...)
@@ -477,10 +477,10 @@
 
 int WebRtcNetEQ_Expand(DSPInst_t *inst,
 #ifdef SCRATCH
-                       WebRtc_Word16 *pw16_scratchPtr,
+                       int16_t *pw16_scratchPtr,
 #endif
-                       WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                       WebRtc_Word16 BGNonly);
+                       int16_t *pw16_outData, int16_t *pw16_len,
+                       int16_t BGNonly);
 
 /****************************************************************************
  * WebRtcNetEQ_GenerateBGN(...)
@@ -506,9 +506,9 @@
 
 int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
 #ifdef SCRATCH
-                            WebRtc_Word16 *pw16_scratchPtr,
+                            int16_t *pw16_scratchPtr,
 #endif
-                            WebRtc_Word16 *pw16_outData, WebRtc_Word16 len);
+                            int16_t *pw16_outData, int16_t len);
 
 /****************************************************************************
  * WebRtcNetEQ_PreEmptiveExpand(...)
@@ -543,11 +543,11 @@
 
 int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
 #ifdef SCRATCH
-                                 WebRtc_Word16 *pw16_scratchPtr,
+                                 int16_t *pw16_scratchPtr,
 #endif
-                                 const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
-                                 WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                                 WebRtc_Word16 BGNonly);
+                                 const int16_t *pw16_decoded, int len, int oldDataLen,
+                                 int16_t *pw16_outData, int16_t *pw16_len,
+                                 int16_t BGNonly);
 
 /****************************************************************************
  * WebRtcNetEQ_Accelerate(...)
@@ -578,11 +578,11 @@
 
 int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
 #ifdef SCRATCH
-                           WebRtc_Word16 *pw16_scratchPtr,
+                           int16_t *pw16_scratchPtr,
 #endif
-                           const WebRtc_Word16 *pw16_decoded, int len,
-                           WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                           WebRtc_Word16 BGNonly);
+                           const int16_t *pw16_decoded, int len,
+                           int16_t *pw16_outData, int16_t *pw16_len,
+                           int16_t BGNonly);
 
 /****************************************************************************
  * WebRtcNetEQ_Merge(...)
@@ -611,10 +611,10 @@
 
 int WebRtcNetEQ_Merge(DSPInst_t *inst,
 #ifdef SCRATCH
-                      WebRtc_Word16 *pw16_scratchPtr,
+                      int16_t *pw16_scratchPtr,
 #endif
-                      WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
-                      WebRtc_Word16 *pw16_len);
+                      int16_t *pw16_decoded, int len, int16_t *pw16_outData,
+                      int16_t *pw16_len);
 
 /****************************************************************************
  * WebRtcNetEQ_Cng(...)
@@ -635,7 +635,7 @@
 #ifdef NETEQ_CNG_CODEC
 /* Must compile NetEQ with CNG support to enable this function */
 
-int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len);
+int WebRtcNetEQ_Cng(DSPInst_t *inst, int16_t *pw16_outData, int len);
 
 #endif /* NETEQ_CNG_CODEC */
 
@@ -656,7 +656,7 @@
 
 void WebRtcNetEQ_BGNUpdate(
 #ifdef SCRATCH
-                           DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
+                           DSPInst_t *inst, int16_t *pw16_scratchPtr
 #else
                            DSPInst_t *inst
 #endif
@@ -681,7 +681,7 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs);
+int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, uint16_t fs);
 
 /****************************************************************************
  * WebRtcNetEQ_SetVADModeInternal(...)
@@ -760,15 +760,15 @@
  *		- crossCorr		: The cross correlation in Q-rShift
  */
 
-void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr, WebRtc_Word16 *seq1,
-                                   WebRtc_Word16 *seq2, WebRtc_Word16 dimSeq,
-                                   WebRtc_Word16 dimCrossCorr, WebRtc_Word16 rShift,
-                                   WebRtc_Word16 step_seq2);
+void WebRtcNetEQ_40BitAccCrossCorr(int32_t *crossCorr, int16_t *seq1,
+                                   int16_t *seq2, int16_t dimSeq,
+                                   int16_t dimCrossCorr, int16_t rShift,
+                                   int16_t step_seq2);
 
 /****************************************************************************
  * WebRtcNetEQ_40BitAccDotW16W16(...)
  *
- * Calculates the dot product between two vectors (WebRtc_Word16)
+ * Calculates the dot product between two vectors (int16_t)
  *
  * Input:
  *		- vector1		: Vector 1
@@ -780,8 +780,8 @@
  * Return value			: The dot product
  */
 
-WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
-                                            int len, int scaling);
+int32_t WebRtcNetEQ_40BitAccDotW16W16(int16_t *vector1, int16_t *vector2,
+                                      int len, int scaling);
 
 #endif /* WEBRTC_NETEQ_40BITACC_TEST */
 
diff --git a/modules/audio_coding/neteq/dsp_helpfunctions.c b/modules/audio_coding/neteq/dsp_helpfunctions.c
index 6e9a283..ef721d5 100644
--- a/modules/audio_coding/neteq/dsp_helpfunctions.c
+++ b/modules/audio_coding/neteq/dsp_helpfunctions.c
@@ -15,7 +15,7 @@
 #include "dsp_helpfunctions.h"
 
 
-WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz)
+int16_t WebRtcNetEQ_CalcFsMult(uint16_t fsHz)
 {
     switch (fsHz)
     {
@@ -43,13 +43,13 @@
 }
 
 
-int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
-                                 WebRtc_Word16 *out, int outLen, int compensateDelay)
+int WebRtcNetEQ_DownSampleTo4kHz(const int16_t *in, int inLen, uint16_t inFsHz,
+                                 int16_t *out, int outLen, int compensateDelay)
 {
-    WebRtc_Word16 *B; /* filter coefficients */
-    WebRtc_Word16 Blen; /* number of coefficients */
-    WebRtc_Word16 filterDelay; /* phase delay in samples */
-    WebRtc_Word16 factor; /* conversion rate (inFsHz/8000) */
+    int16_t *B; /* filter coefficients */
+    int16_t Blen; /* number of coefficients */
+    int16_t filterDelay; /* phase delay in samples */
+    int16_t factor; /* conversion rate (inFsHz/8000) */
     int ok;
 
     /* Set constants depending on frequency used */
@@ -62,7 +62,7 @@
         {
             Blen = 3;
             factor = 2;
-            B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
+            B = (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl;
             filterDelay = 1 + 1;
             break;
         }
@@ -71,7 +71,7 @@
             {
                 Blen = 5;
                 factor = 4;
-                B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample16kHzTbl;
+                B = (int16_t*) WebRtcNetEQ_kDownsample16kHzTbl;
                 filterDelay = 2 + 1;
                 break;
             }
@@ -81,7 +81,7 @@
             {
                 Blen = 7;
                 factor = 8;
-                B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample32kHzTbl;
+                B = (int16_t*) WebRtcNetEQ_kDownsample32kHzTbl;
                 filterDelay = 3 + 1;
                 break;
             }
@@ -91,7 +91,7 @@
             {
                 Blen = 7;
                 factor = 12;
-                B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample48kHzTbl;
+                B = (int16_t*) WebRtcNetEQ_kDownsample48kHzTbl;
                 filterDelay = 3 + 1;
                 break;
             }
@@ -109,9 +109,9 @@
         filterDelay = 0;
     }
 
-    ok = WebRtcSpl_DownsampleFast((WebRtc_Word16*) &in[Blen - 1],
-        (WebRtc_Word16) (inLen - (Blen - 1)), /* number of input samples */
-        out, (WebRtc_Word16) outLen, /* number of output samples to produce */
+    ok = WebRtcSpl_DownsampleFast((int16_t*) &in[Blen - 1],
+        (int16_t) (inLen - (Blen - 1)), /* number of input samples */
+        out, (int16_t) outLen, /* number of output samples to produce */
         B, Blen, factor, filterDelay); /* filter parameters */
 
     return ok; /* return value is -1 if input signal is too short */
diff --git a/modules/audio_coding/neteq/dsp_helpfunctions.h b/modules/audio_coding/neteq/dsp_helpfunctions.h
index f728c09..11119f1 100644
--- a/modules/audio_coding/neteq/dsp_helpfunctions.h
+++ b/modules/audio_coding/neteq/dsp_helpfunctions.h
@@ -36,13 +36,13 @@
  * Return value         : Length of correlated data
  */
 
-WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
+int16_t WebRtcNetEQ_Correlator(DSPInst_t *inst,
 #ifdef SCRATCH
-                                     WebRtc_Word16 *pw16_scratchPtr,
+                               int16_t *pw16_scratchPtr,
 #endif
-                                     WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
-                                     WebRtc_Word16 *pw16_corrOut,
-                                     WebRtc_Word16 *pw16_corrScale);
+                               int16_t *pw16_data, int16_t w16_dataLen,
+                               int16_t *pw16_corrOut,
+                               int16_t *pw16_corrScale);
 
 /****************************************************************************
  * WebRtcNetEQ_PeakDetection(...)
@@ -62,10 +62,10 @@
  * Return value         : 0 for ok
  */
 
-WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
-                                        WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
-                                        WebRtc_Word16 *pw16_corrIndex,
-                                        WebRtc_Word16 *pw16_winners);
+int16_t WebRtcNetEQ_PeakDetection(int16_t *pw16_data, int16_t w16_dataLen,
+                                  int16_t w16_nmbPeaks, int16_t fs_mult,
+                                  int16_t *pw16_corrIndex,
+                                  int16_t *pw16_winners);
 
 /****************************************************************************
  * WebRtcNetEQ_PrblFit(...)
@@ -83,8 +83,8 @@
  * Return value         : 0 for ok
  */
 
-WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
-                                  WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult);
+int16_t WebRtcNetEQ_PrblFit(int16_t *pw16_3pts, int16_t *pw16_Ind,
+                            int16_t *pw16_outVal, int16_t fs_mult);
 
 /****************************************************************************
  * WebRtcNetEQ_MinDistortion(...)
@@ -104,9 +104,9 @@
  * Return value         : Lag for minimum distortion
  */
 
-WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
-                                        WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
-                                        WebRtc_Word16 len, WebRtc_Word32 *pw16_dist);
+int16_t WebRtcNetEQ_MinDistortion(const int16_t *pw16_data,
+                                  int16_t w16_minLag, int16_t w16_maxLag,
+                                  int16_t len, int32_t *pw16_dist);
 
 /****************************************************************************
  * WebRtcNetEQ_RandomVec(...)
@@ -122,8 +122,8 @@
  *      - randVec       : Generated random vector
  */
 
-void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
-                           WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval);
+void WebRtcNetEQ_RandomVec(uint32_t *w32_seed, int16_t *pw16_randVec,
+                           int16_t w16_len, int16_t w16_incval);
 
 /****************************************************************************
  * WebRtcNetEQ_MixVoiceUnvoice(...)
@@ -141,10 +141,10 @@
  *      - outData           : Mixed signal
  */
 
-void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
-                                 WebRtc_Word16 *pw16_unvoicedVec,
-                                 WebRtc_Word16 *w16_current_vfraction,
-                                 WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N);
+void WebRtcNetEQ_MixVoiceUnvoice(int16_t *pw16_outData, int16_t *pw16_voicedVec,
+                                 int16_t *pw16_unvoicedVec,
+                                 int16_t *w16_current_vfraction,
+                                 int16_t w16_vfraction_change, int16_t N);
 
 /****************************************************************************
  * WebRtcNetEQ_UnmuteSignal(...)
@@ -161,9 +161,9 @@
  *      - outVec        : Output signal
  */
 
-void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
-                              WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
-                              WebRtc_Word16 N);
+void WebRtcNetEQ_UnmuteSignal(int16_t *pw16_inVec, int16_t *startMuteFact,
+                              int16_t *pw16_outVec, int16_t unmuteFact,
+                              int16_t N);
 
 /****************************************************************************
  * WebRtcNetEQ_MuteSignal(...)
@@ -176,8 +176,8 @@
  *      - N             : Number of samples
  */
 
-void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
-                            WebRtc_Word16 N);
+void WebRtcNetEQ_MuteSignal(int16_t *pw16_inout, int16_t muteSlope,
+                            int16_t N);
 
 /****************************************************************************
  * WebRtcNetEQ_CalcFsMult(...)
@@ -190,7 +190,7 @@
  * Return value			: fsHz/8000 for the valid values, 1 for other inputs
  */
 
-WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz);
+int16_t WebRtcNetEQ_CalcFsMult(uint16_t fsHz);
 
 /****************************************************************************
  * WebRtcNetEQ_DownSampleTo4kHz(...)
@@ -213,8 +213,8 @@
  *
  */
 
-int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
-                                 WebRtc_Word16 *out, int outLen, int compensateDelay);
+int WebRtcNetEQ_DownSampleTo4kHz(const int16_t *in, int inLen, uint16_t inFsHz,
+                                 int16_t *out, int outLen, int compensateDelay);
 
 #endif
 
diff --git a/modules/audio_coding/neteq/dtmf_buffer.c b/modules/audio_coding/neteq/dtmf_buffer.c
index f00f9c9..9e32126 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.c
+++ b/modules/audio_coding/neteq/dtmf_buffer.c
@@ -22,7 +22,7 @@
 
 #ifdef NETEQ_ATEVENT_DECODE
 
-WebRtc_Word16 WebRtcNetEQ_DtmfRemoveEvent(dtmf_inst_t *DTMFdec_inst)
+int16_t WebRtcNetEQ_DtmfRemoveEvent(dtmf_inst_t *DTMFdec_inst)
 {
 
     int i;
@@ -44,8 +44,8 @@
     return 0;
 }
 
-WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
-                                          WebRtc_Word16 MaxPLCtime)
+int16_t WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, uint16_t fs,
+                                    int16_t MaxPLCtime)
 {
     int i;
     if (((fs != 8000) && (fs != 16000) && (fs != 32000) && (fs != 48000)) || (MaxPLCtime < 0))
@@ -76,18 +76,18 @@
     return 0;
 }
 
-WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
-                                          const WebRtc_Word16 *encoded, WebRtc_Word16 len,
-                                          WebRtc_UWord32 timeStamp)
+int16_t WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
+                                    const int16_t *encoded, int16_t len,
+                                    uint32_t timeStamp)
 {
 
     int i;
-    WebRtc_Word16 value;
-    const WebRtc_Word16 *EventStart;
-    WebRtc_Word16 endEvent;
-    WebRtc_Word16 Volume;
-    WebRtc_Word16 Duration;
-    WebRtc_Word16 position = -1;
+    int16_t value;
+    const int16_t *EventStart;
+    int16_t endEvent;
+    int16_t Volume;
+    int16_t Duration;
+    int16_t position = -1;
 
     /* Extract event */
     if (len == 4)
@@ -102,8 +102,8 @@
         value = ((*EventStart) & 0xFF);
         endEvent = ((*EventStart) & 0x8000) >> 15;
         Volume = ((*EventStart) & 0x3F00) >> 8;
-        Duration = (((((WebRtc_UWord16) EventStart[1]) >> 8) & 0xFF)
-            | (((WebRtc_UWord16) (EventStart[1] & 0xFF)) << 8));
+        Duration = (((((uint16_t) EventStart[1]) >> 8) & 0xFF)
+            | (((uint16_t) (EventStart[1] & 0xFF)) << 8));
 #endif
         /* Only events between 0-15 are supported (DTMF tones) */
         if ((value < 0) || (value > 15))
@@ -153,8 +153,8 @@
     return DTMF_INSERT_ERROR;
 }
 
-WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
-                                     WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp)
+int16_t WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, int16_t *event,
+                               int16_t *volume, uint32_t currTimeStamp)
 {
 
     if (DTMFdec_inst->EventBufferSize < 1) return 0; /* No events to play */
@@ -204,7 +204,7 @@
         else
         {
             /* Less than frameLen to play and not end of event. */
-            DTMFdec_inst->CurrentPLCtime = (WebRtc_Word16) (currTimeStamp
+            DTMFdec_inst->CurrentPLCtime = (int16_t) (currTimeStamp
                 - DTMFdec_inst->EventQueueEndTime[0]);
 
             if ((DTMFdec_inst->CurrentPLCtime > DTMFdec_inst->MaxPLCtime)
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h
index e185411..99c9e6a 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/modules/audio_coding/neteq/dtmf_buffer.h
@@ -26,15 +26,15 @@
 
 typedef struct dtmf_inst_t_
 {
-    WebRtc_Word16 MaxPLCtime;
-    WebRtc_Word16 CurrentPLCtime;
-    WebRtc_Word16 EventQueue[MAX_DTMF_QUEUE_SIZE];
-    WebRtc_Word16 EventQueueVolume[MAX_DTMF_QUEUE_SIZE];
-    WebRtc_Word16 EventQueueEnded[MAX_DTMF_QUEUE_SIZE];
-    WebRtc_UWord32 EventQueueStartTime[MAX_DTMF_QUEUE_SIZE];
-    WebRtc_UWord32 EventQueueEndTime[MAX_DTMF_QUEUE_SIZE];
-    WebRtc_Word16 EventBufferSize;
-    WebRtc_Word16 framelen;
+    int16_t MaxPLCtime;
+    int16_t CurrentPLCtime;
+    int16_t EventQueue[MAX_DTMF_QUEUE_SIZE];
+    int16_t EventQueueVolume[MAX_DTMF_QUEUE_SIZE];
+    int16_t EventQueueEnded[MAX_DTMF_QUEUE_SIZE];
+    uint32_t EventQueueStartTime[MAX_DTMF_QUEUE_SIZE];
+    uint32_t EventQueueEndTime[MAX_DTMF_QUEUE_SIZE];
+    int16_t EventBufferSize;
+    int16_t framelen;
 } dtmf_inst_t;
 
 /****************************************************************************
@@ -51,8 +51,8 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
-                                          WebRtc_Word16 MaxPLCtime);
+int16_t WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, uint16_t fs,
+                                    int16_t MaxPLCtime);
 
 /****************************************************************************
  * WebRtcNetEQ_DtmfInsertEvent(...)
@@ -69,9 +69,9 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
-                                          const WebRtc_Word16 *encoded, WebRtc_Word16 len,
-                                          WebRtc_UWord32 timeStamp);
+int16_t WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
+                                    const int16_t *encoded, int16_t len,
+                                    uint32_t timeStamp);
 
 /****************************************************************************
  * WebRtcNetEQ_DtmfDecode(...)
@@ -92,8 +92,8 @@
  *                            -1 - Error
  */
 
-WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
-                                     WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp);
+int16_t WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, int16_t *event,
+                               int16_t *volume, uint32_t currTimeStamp);
 
 #endif    /* NETEQ_ATEVENT_DECODE */
 
diff --git a/modules/audio_coding/neteq/dtmf_tonegen.c b/modules/audio_coding/neteq/dtmf_tonegen.c
index c9ad232..8ea413c 100644
--- a/modules/audio_coding/neteq/dtmf_tonegen.c
+++ b/modules/audio_coding/neteq/dtmf_tonegen.c
@@ -65,14 +65,14 @@
  * Table values in Q14.
  */
 
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl8Khz[8] =
+const int16_t WebRtcNetEQ_dtfm_aTbl8Khz[8] =
 {
     27980, 26956, 25701, 24219,
     19073, 16325, 13085, 9315
 };
 
 #ifdef NETEQ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl16Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_aTbl16Khz[8]=
 {
     31548, 31281, 30951, 30556,
     29144, 28361, 27409, 26258
@@ -80,7 +80,7 @@
 #endif
 
 #ifdef NETEQ_32KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl32Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_aTbl32Khz[8]=
 {
     32462, 32394, 32311, 32210,
     31849, 31647, 31400, 31098
@@ -88,7 +88,7 @@
 #endif
 
 #ifdef NETEQ_48KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl48Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_aTbl48Khz[8]=
 {
     32632, 32602, 32564, 32520,
     32359, 32268, 32157, 32022
@@ -100,14 +100,14 @@
  * Table values in Q14.
  */
 
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab8Khz[8] =
+const int16_t WebRtcNetEQ_dtfm_yInitTab8Khz[8] =
 {
     8528, 9315, 10163, 11036,
     13323, 14206,15021, 15708
 };
 
 #ifdef NETEQ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab16Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_yInitTab16Khz[8]=
 {
     4429, 4879, 5380, 5918,
     7490, 8207, 8979, 9801
@@ -115,7 +115,7 @@
 #endif
 
 #ifdef NETEQ_32KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab32Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_yInitTab32Khz[8]=
 {
     2235, 2468, 2728, 3010,
     3853, 4249, 4685, 5164
@@ -123,7 +123,7 @@
 #endif
 
 #ifdef NETEQ_48KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab48Khz[8]=
+const int16_t WebRtcNetEQ_dtfm_yInitTab48Khz[8]=
 {
     1493, 1649, 1823, 2013,
     2582, 2851, 3148, 3476
@@ -135,7 +135,7 @@
  Table entries are in Q14.
  */
 
-const WebRtc_Word16 WebRtcNetEQ_dtfm_dBm0[37] = { 16141, 14386, 12821, 11427, 10184, 9077, 8090,
+const int16_t WebRtcNetEQ_dtfm_dBm0[37] = { 16141, 14386, 12821, 11427, 10184, 9077, 8090,
                                                 7210, 6426, 5727, 5104, 4549, 4054, 3614,
                                                 3221, 2870, 2558, 2280, 2032, 1811, 1614,
                                                 1439, 1282, 1143, 1018, 908, 809, 721, 643,
@@ -163,21 +163,21 @@
  *                      : <0 - error
  */
 
-WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, WebRtc_Word16 value,
-                                       WebRtc_Word16 volume, WebRtc_Word16 *signal,
-                                       WebRtc_UWord16 sampFreq, WebRtc_Word16 extFrameLen)
+int16_t WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, int16_t value,
+                                 int16_t volume, int16_t *signal,
+                                 uint16_t sampFreq, int16_t extFrameLen)
 {
-    const WebRtc_Word16 *aTbl; /* pointer to a-coefficient table */
-    const WebRtc_Word16 *yInitTable; /* pointer to initialization value table */
-    WebRtc_Word16 a1 = 0; /* a-coefficient for first tone (low tone) */
-    WebRtc_Word16 a2 = 0; /* a-coefficient for second tone (high tone) */
+    const int16_t *aTbl; /* pointer to a-coefficient table */
+    const int16_t *yInitTable; /* pointer to initialization value table */
+    int16_t a1 = 0; /* a-coefficient for first tone (low tone) */
+    int16_t a2 = 0; /* a-coefficient for second tone (high tone) */
     int i;
     int frameLen; /* number of samples to generate */
     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;
+    int32_t tempVal;
+    int16_t tempValLow;
+    int16_t tempValHigh;
 
     /* Sanity check for volume */
     if ((volume < 0) || (volume > 36))
@@ -333,10 +333,10 @@
 
         /* Use rescursion formula y[n] = a*y[n-1] - y[n-2] */
         tempValLow
-                        = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a1, DTMFdecInst->oldOutputLow[1])
+                        = (int16_t) (((WEBRTC_SPL_MUL_16_16(a1, DTMFdecInst->oldOutputLow[1])
                                         + 8192) >> 14) - DTMFdecInst->oldOutputLow[0]);
         tempValHigh
-                        = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a2, DTMFdecInst->oldOutputHigh[1])
+                        = (int16_t) (((WEBRTC_SPL_MUL_16_16(a2, DTMFdecInst->oldOutputHigh[1])
                                         + 8192) >> 14) - DTMFdecInst->oldOutputHigh[0]);
 
         /* Update recursion memory */
@@ -348,13 +348,13 @@
         /* scale high tone with 32768 (15 left shifts) 
          and low tone with 23171 (3dB lower than high tone) */
         tempVal = WEBRTC_SPL_MUL_16_16(DTMF_AMP_LOW, tempValLow)
-                        + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tempValHigh, 15);
+                        + WEBRTC_SPL_LSHIFT_W32((int32_t)tempValHigh, 15);
 
         /* Norm the signal to Q14 (with proper rounding) */
         tempVal = (tempVal + 16384) >> 15;
 
         /* Scale the signal to correct dbM0 value */
-        signal[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+        signal[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
                                (WEBRTC_SPL_MUL_16_16(tempVal, WebRtcNetEQ_dtfm_dBm0[volume])
                                + 8192), 14); /* volume value is in Q14; use proper rounding */
     }
diff --git a/modules/audio_coding/neteq/dtmf_tonegen.h b/modules/audio_coding/neteq/dtmf_tonegen.h
index add6eb1..5f44899 100644
--- a/modules/audio_coding/neteq/dtmf_tonegen.h
+++ b/modules/audio_coding/neteq/dtmf_tonegen.h
@@ -28,10 +28,10 @@
 typedef struct dtmf_tone_inst_t_
 {
 
-    WebRtc_Word16 reinit; /* non-zero if the oscillator model should
+    int16_t reinit; /* non-zero if the oscillator model should
      be reinitialized for next event */
-    WebRtc_Word16 oldOutputLow[2]; /* oscillator recursion history (low tone) */
-    WebRtc_Word16 oldOutputHigh[2]; /* oscillator recursion history (high tone) */
+    int16_t oldOutputLow[2]; /* oscillator recursion history (low tone) */
+    int16_t oldOutputHigh[2]; /* oscillator recursion history (high tone) */
 
     int lastDtmfSample; /* index to the first non-DTMF sample in the
      speech history, if non-negative */
@@ -59,12 +59,12 @@
  *                      : <0 - Error
  */
 
-WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst,
-                WebRtc_Word16 value,
-                WebRtc_Word16 volume,
-                WebRtc_Word16 *signal,
-                WebRtc_UWord16 sampFreq,
-                WebRtc_Word16 frameLen
+int16_t WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst,
+                int16_t value,
+                int16_t volume,
+                int16_t *signal,
+                uint16_t sampFreq,
+                int16_t frameLen
 );
 
 #endif /* NETEQ_ATEVENT_DECODE */
diff --git a/modules/audio_coding/neteq/expand.c b/modules/audio_coding/neteq/expand.c
index 3db7a2a..6a69925 100644
--- a/modules/audio_coding/neteq/expand.c
+++ b/modules/audio_coding/neteq/expand.c
@@ -30,22 +30,22 @@
 
  Type            Name                    size            startpos        endpos
  (First part of first expand)
- WebRtc_Word16  pw16_bestCorrIndex      3               0               2
- WebRtc_Word16  pw16_bestCorr           3               3               5
- WebRtc_Word16  pw16_bestDistIndex      3               6               8
- WebRtc_Word16  pw16_bestDist           3               9               11
- WebRtc_Word16  pw16_corrVec            102*fs/8000     12              11+102*fs/8000
+ int16_t  pw16_bestCorrIndex      3               0               2
+ int16_t  pw16_bestCorr           3               3               5
+ int16_t  pw16_bestDistIndex      3               6               8
+ int16_t  pw16_bestDist           3               9               11
+ int16_t  pw16_corrVec            102*fs/8000     12              11+102*fs/8000
  func           WebRtcNetEQ_Correlator  232             12+102*fs/8000  243+102*fs/8000
 
  (Second part of first expand)
- WebRtc_Word32  pw32_corr2              99*fs/8000+1    0               99*fs/8000
- WebRtc_Word32  pw32_autoCorr           2*7             0               13
- WebRtc_Word16  pw16_rc                 6               14              19
+ int32_t  pw32_corr2              99*fs/8000+1    0               99*fs/8000
+ int32_t  pw32_autoCorr           2*7             0               13
+ int16_t  pw16_rc                 6               14              19
 
  Signal combination:
- WebRtc_Word16  pw16_randVec            30+120*fs/8000  0               29+120*fs/8000
- WebRtc_Word16  pw16_scaledRandVec      125*fs/8000     30+120*fs/8000  29+245*fs/8000
- WebRtc_Word16  pw16_unvoicedVecSpace   10+125*fs/8000  30+245*fs/8000  39+370*fs/8000
+ int16_t  pw16_randVec            30+120*fs/8000  0               29+120*fs/8000
+ int16_t  pw16_scaledRandVec      125*fs/8000     30+120*fs/8000  29+245*fs/8000
+ int16_t  pw16_unvoicedVecSpace   10+125*fs/8000  30+245*fs/8000  39+370*fs/8000
 
  Total: 40+370*fs/8000 (size depends on UNVOICED_LPC_ORDER and BGN_LPC_ORDER)
  */
@@ -106,59 +106,59 @@
 
 int WebRtcNetEQ_Expand(DSPInst_t *inst,
 #ifdef SCRATCH
-                       WebRtc_Word16 *pw16_scratchPtr,
+                       int16_t *pw16_scratchPtr,
 #endif
-                       WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                       WebRtc_Word16 BGNonly)
+                       int16_t *pw16_outData, int16_t *pw16_len,
+                       int16_t BGNonly)
 {
 
-    WebRtc_Word16 fs_mult;
+    int16_t fs_mult;
     ExpandInst_t *ExpandState = &(inst->ExpandInst);
     BGNInst_t *BGNState = &(inst->BGNInst);
     int i;
 #ifdef SCRATCH
-    WebRtc_Word16 *pw16_randVec = pw16_scratchPtr + SCRATCH_PW16_RAND_VEC;
-    WebRtc_Word16 *pw16_scaledRandVec = pw16_scratchPtr + SCRATCH_PW16_SCALED_RAND_VEC;
-    WebRtc_Word16 *pw16_unvoicedVecSpace = pw16_scratchPtr + SCRATCH_PW16_UNVOICED_VEC_SPACE;
+    int16_t *pw16_randVec = pw16_scratchPtr + SCRATCH_PW16_RAND_VEC;
+    int16_t *pw16_scaledRandVec = pw16_scratchPtr + SCRATCH_PW16_SCALED_RAND_VEC;
+    int16_t *pw16_unvoicedVecSpace = pw16_scratchPtr + SCRATCH_PW16_UNVOICED_VEC_SPACE;
 #else
-    WebRtc_Word16 pw16_randVec[FSMULT * 120 + 30]; /* 150 for NB and 270 for WB */
-    WebRtc_Word16 pw16_scaledRandVec[FSMULT * 125]; /* 125 for NB and 250 for WB */
-    WebRtc_Word16 pw16_unvoicedVecSpace[BGN_LPC_ORDER + FSMULT * 125];
+    int16_t pw16_randVec[FSMULT * 120 + 30]; /* 150 for NB and 270 for WB */
+    int16_t pw16_scaledRandVec[FSMULT * 125]; /* 125 for NB and 250 for WB */
+    int16_t pw16_unvoicedVecSpace[BGN_LPC_ORDER + FSMULT * 125];
 #endif
     /* 125 for NB and 250 for WB etc. Reuse pw16_outData[] for this vector */
-    WebRtc_Word16 *pw16_voicedVecStorage = pw16_outData;
-    WebRtc_Word16 *pw16_voicedVec = &pw16_voicedVecStorage[ExpandState->w16_overlap];
-    WebRtc_Word16 *pw16_unvoicedVec = pw16_unvoicedVecSpace + UNVOICED_LPC_ORDER;
-    WebRtc_Word16 *pw16_cngVec = pw16_unvoicedVecSpace + BGN_LPC_ORDER;
-    WebRtc_Word16 w16_expVecsLen, w16_lag = 0, w16_expVecPos;
-    WebRtc_Word16 w16_randLen;
-    WebRtc_Word16 w16_vfractionChange; /* in Q14 */
-    WebRtc_Word16 w16_winMute = 0, w16_winMuteInc = 0, w16_winUnMute = 0, w16_winUnMuteInc = 0;
-    WebRtc_Word32 w32_tmp;
-    WebRtc_Word16 w16_tmp, w16_tmp2;
-    WebRtc_Word16 stability;
+    int16_t *pw16_voicedVecStorage = pw16_outData;
+    int16_t *pw16_voicedVec = &pw16_voicedVecStorage[ExpandState->w16_overlap];
+    int16_t *pw16_unvoicedVec = pw16_unvoicedVecSpace + UNVOICED_LPC_ORDER;
+    int16_t *pw16_cngVec = pw16_unvoicedVecSpace + BGN_LPC_ORDER;
+    int16_t w16_expVecsLen, w16_lag = 0, w16_expVecPos;
+    int16_t w16_randLen;
+    int16_t w16_vfractionChange; /* in Q14 */
+    int16_t w16_winMute = 0, w16_winMuteInc = 0, w16_winUnMute = 0, w16_winUnMuteInc = 0;
+    int32_t w32_tmp;
+    int16_t w16_tmp, w16_tmp2;
+    int16_t stability;
     enum BGNMode bgnMode = inst->BGNInst.bgnMode;
 
     /* Pre-calculate common multiplications with fs_mult */
-    WebRtc_Word16 fsMult4;
-    WebRtc_Word16 fsMult20;
-    WebRtc_Word16 fsMult120;
-    WebRtc_Word16 fsMultDistLen;
-    WebRtc_Word16 fsMultLPCAnalasysLen;
+    int16_t fsMult4;
+    int16_t fsMult20;
+    int16_t fsMult120;
+    int16_t fsMultDistLen;
+    int16_t fsMultLPCAnalasysLen;
 
 #ifdef NETEQ_STEREO
     MasterSlaveInfo *msInfo = inst->msInfo;
 #endif
 
-    /* fs is WebRtc_UWord16 (to hold fs=48000) */
+    /* fs is uint16_t (to hold fs=48000) */
     fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs); /* calculate fs/8000 */
 
     /* Pre-calculate common multiplications with fs_mult */
-    fsMult4 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 4);
-    fsMult20 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 20);
-    fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 120);
-    fsMultDistLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, DISTLEN);
-    fsMultLPCAnalasysLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, LPCANALASYSLEN);
+    fsMult4 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 4);
+    fsMult20 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 20);
+    fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 120);
+    fsMultDistLen = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, DISTLEN);
+    fsMultLPCAnalasysLen = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, LPCANALASYSLEN);
 
     /*
      * Perform all the initial setup if it's the first expansion.
@@ -168,47 +168,47 @@
     {
         /* Setup more variables */
 #ifdef SCRATCH
-        WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr
+        int32_t *pw32_autoCorr = (int32_t*) (pw16_scratchPtr
             + SCRATCH_PW32_AUTO_CORR);
-        WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
-        WebRtc_Word16 *pw16_bestCorrIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR_INDEX;
-        WebRtc_Word16 *pw16_bestCorr = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR;
-        WebRtc_Word16 *pw16_bestDistIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST_INDEX;
-        WebRtc_Word16 *pw16_bestDist = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST;
-        WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_PW16_CORR_VEC;
-        WebRtc_Word32 *pw32_corr2 = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW16_CORR2);
+        int16_t *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
+        int16_t *pw16_bestCorrIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR_INDEX;
+        int16_t *pw16_bestCorr = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR;
+        int16_t *pw16_bestDistIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST_INDEX;
+        int16_t *pw16_bestDist = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST;
+        int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_PW16_CORR_VEC;
+        int32_t *pw32_corr2 = (int32_t*) (pw16_scratchPtr + SCRATCH_PW16_CORR2);
 #else
-        WebRtc_Word32 pw32_autoCorr[UNVOICED_LPC_ORDER+1];
-        WebRtc_Word16 pw16_rc[UNVOICED_LPC_ORDER];
-        WebRtc_Word16 pw16_corrVec[FSMULT*102]; /* 102 for NB */
-        WebRtc_Word16 pw16_bestCorrIndex[CHECK_NO_OF_CORRMAX];
-        WebRtc_Word16 pw16_bestCorr[CHECK_NO_OF_CORRMAX];
-        WebRtc_Word16 pw16_bestDistIndex[CHECK_NO_OF_CORRMAX];
-        WebRtc_Word16 pw16_bestDist[CHECK_NO_OF_CORRMAX];
-        WebRtc_Word32 pw32_corr2[(99*FSMULT)+1];
+        int32_t pw32_autoCorr[UNVOICED_LPC_ORDER+1];
+        int16_t pw16_rc[UNVOICED_LPC_ORDER];
+        int16_t pw16_corrVec[FSMULT*102]; /* 102 for NB */
+        int16_t pw16_bestCorrIndex[CHECK_NO_OF_CORRMAX];
+        int16_t pw16_bestCorr[CHECK_NO_OF_CORRMAX];
+        int16_t pw16_bestDistIndex[CHECK_NO_OF_CORRMAX];
+        int16_t pw16_bestDist[CHECK_NO_OF_CORRMAX];
+        int32_t pw32_corr2[(99*FSMULT)+1];
 #endif
-        WebRtc_Word32 pw32_bestDist[CHECK_NO_OF_CORRMAX];
-        WebRtc_Word16 w16_ind = 0;
-        WebRtc_Word16 w16_corrVecLen;
-        WebRtc_Word16 w16_corrScale;
-        WebRtc_Word16 w16_distScale;
-        WebRtc_Word16 w16_indMin, w16_indMax;
-        WebRtc_Word16 w16_len;
-        WebRtc_Word32 w32_en1, w32_en2, w32_cc;
-        WebRtc_Word16 w16_en1Scale, w16_en2Scale;
-        WebRtc_Word16 w16_en1, w16_en2;
-        WebRtc_Word32 w32_en1_mul_en2;
-        WebRtc_Word16 w16_sqrt_en1en2;
-        WebRtc_Word16 w16_ccShiftL;
-        WebRtc_Word16 w16_bestcorr; /* Correlation in Q14 */
-        WebRtc_Word16 *pw16_vec1, *pw16_vec2;
-        WebRtc_Word16 w16_factor;
-        WebRtc_Word16 w16_DistLag, w16_CorrLag, w16_diffLag;
-        WebRtc_Word16 w16_energyLen;
-        WebRtc_Word16 w16_slope;
-        WebRtc_Word16 w16_startInd;
-        WebRtc_Word16 w16_noOfcorr2;
-        WebRtc_Word16 w16_scale;
+        int32_t pw32_bestDist[CHECK_NO_OF_CORRMAX];
+        int16_t w16_ind = 0;
+        int16_t w16_corrVecLen;
+        int16_t w16_corrScale;
+        int16_t w16_distScale;
+        int16_t w16_indMin, w16_indMax;
+        int16_t w16_len;
+        int32_t w32_en1, w32_en2, w32_cc;
+        int16_t w16_en1Scale, w16_en2Scale;
+        int16_t w16_en1, w16_en2;
+        int32_t w32_en1_mul_en2;
+        int16_t w16_sqrt_en1en2;
+        int16_t w16_ccShiftL;
+        int16_t w16_bestcorr; /* Correlation in Q14 */
+        int16_t *pw16_vec1, *pw16_vec2;
+        int16_t w16_factor;
+        int16_t w16_DistLag, w16_CorrLag, w16_diffLag;
+        int16_t w16_energyLen;
+        int16_t w16_slope;
+        int16_t w16_startInd;
+        int16_t w16_noOfcorr2;
+        int16_t w16_scale;
 
         /* Initialize some variables */
         ExpandState->w16_lagsDirection = 1;
@@ -273,7 +273,7 @@
 
             }
 
-            /* Shift the distortion values to fit in WebRtc_Word16 */
+            /* Shift the distortion values to fit in int16_t */
             WebRtcSpl_VectorBitShiftW32ToW16(pw16_bestDist, CHECK_NO_OF_CORRMAX, pw32_bestDist,
                 w16_distScale);
 
@@ -282,15 +282,15 @@
              * Do this by a cross multiplication.
              */
 
-            w32_en1 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0],pw16_bestDist[1]);
-            w32_en2 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[0]);
+            w32_en1 = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[0],pw16_bestDist[1]);
+            w32_en2 = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[1],pw16_bestDist[0]);
             if (w32_en1 >= w32_en2)
             {
                 /* 0 wins over 1 */
                 w32_en1
-                    = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0], pw16_bestDist[2]);
+                    = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[0], pw16_bestDist[2]);
                 w32_en2
-                    = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2], pw16_bestDist[0]);
+                    = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[2], pw16_bestDist[0]);
                 if (w32_en1 >= w32_en2)
                 {
                     /* 0 wins over 2 */
@@ -306,10 +306,10 @@
             {
                 /* 1 wins over 0 */
                 w32_en1
-                    = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[2]);
+                    = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[1],pw16_bestDist[2]);
                 w32_en2
-                    = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2],pw16_bestDist[1]);
-                if ((WebRtc_Word32) w32_en1 >= (WebRtc_Word32) w32_en2)
+                    = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[2],pw16_bestDist[1]);
+                if ((int32_t) w32_en1 >= (int32_t) w32_en2)
                 {
                     /* 1 wins over 2 */
                     w16_ind = 1;
@@ -376,7 +376,7 @@
             = WebRtcSpl_MaxAbsValueW16(
                 &inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd
                     - w16_noOfcorr2],
-                (WebRtc_Word16) (w16_len + w16_startInd + w16_noOfcorr2 - 1));
+                (int16_t) (w16_len + w16_startInd + w16_noOfcorr2 - 1));
         w16_corrScale = ((31 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_tmp, w16_tmp)))
             + (31 - WebRtcSpl_NormW32(w16_len))) - 31;
         w16_corrScale = WEBRTC_SPL_MAX(0, w16_corrScale);
@@ -418,15 +418,15 @@
                 /* if sum is odd */
                 w16_en1Scale += 1;
             }
-            w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
-            w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+            w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+            w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
             w32_en1_mul_en2 = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
-            w16_sqrt_en1en2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_en1_mul_en2);
+            w16_sqrt_en1en2 = (int16_t) WebRtcSpl_SqrtFloor(w32_en1_mul_en2);
 
             /* Calculate cc/sqrt(en1*en2) in Q14 */
             w16_ccShiftL = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
             w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_ccShiftL);
-            w16_bestcorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrt_en1en2);
+            w16_bestcorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrt_en1en2);
             w16_bestcorr = WEBRTC_SPL_MIN(16384, w16_bestcorr); /* set maximum to 1.0 */
 
         }
@@ -462,10 +462,10 @@
             /* calculate w32_en1/w32_en2 in Q13 */
             w32_en1_mul_en2 = WebRtcSpl_DivW32W16(
                 WEBRTC_SPL_SHIFT_W32(w32_en1, -w16_en1Scale),
-                (WebRtc_Word16) (WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale)));
+                (int16_t) (WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale)));
 
             /* calculate factor in Q13 (sqrt of en1/en2 in Q26) */
-            w16_factor = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
+            w16_factor = (int16_t) WebRtcSpl_SqrtFloor(
                 WEBRTC_SPL_LSHIFT_W32(w32_en1_mul_en2, 13));
 
             /* Copy the two vectors and give them the same energy */
@@ -587,17 +587,17 @@
         if (w16_randLen <= RANDVEC_NO_OF_SAMPLES)
         {
             WEBRTC_SPL_MEMCPY_W16(pw16_randVec,
-                (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl, w16_randLen);
+                (int16_t*) WebRtcNetEQ_kRandnTbl, w16_randLen);
         }
         else
         { /* only applies to SWB where length could be larger than 256 */
 #if FSMULT >= 2  /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */
-            WEBRTC_SPL_MEMCPY_W16(pw16_randVec, (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl,
+            WEBRTC_SPL_MEMCPY_W16(pw16_randVec, (int16_t*) WebRtcNetEQ_kRandnTbl,
                 RANDVEC_NO_OF_SAMPLES);
             inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
             assert(w16_randLen <= FSMULT * 120 + 30);
             WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
-                (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
+                (int16_t) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
 #else
             assert(0);
 #endif
@@ -630,7 +630,7 @@
         w32_tmp = WEBRTC_SPL_SHIFT_W32(w32_tmp, w16_tmp);
         w32_tmp = WebRtcSpl_SqrtFloor(w32_tmp);
         ExpandState->w16_arGainScale = 13 + ((w16_tmp + 7 - w16_scale) >> 1);
-        ExpandState->w16_arGain = (WebRtc_Word16) w32_tmp;
+        ExpandState->w16_arGain = (int16_t) w32_tmp;
 
         /********************************************************************
          * Calculate vfraction from bestcorr                                *
@@ -647,21 +647,21 @@
         if (w16_bestcorr > 7875)
         {
             /* if x>0.480665 */
-            WebRtc_Word16 w16_x1, w16_x2, w16_x3;
+            int16_t w16_x1, w16_x2, w16_x3;
             w16_x1 = w16_bestcorr;
-            w32_tmp = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) w16_x1, w16_x1);
-            w16_x2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
+            w32_tmp = WEBRTC_SPL_MUL_16_16((int32_t) w16_x1, w16_x1);
+            w16_x2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
             w32_tmp = WEBRTC_SPL_MUL_16_16(w16_x1, w16_x2);
-            w16_x3 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
+            w16_x3 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
             w32_tmp
-                = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) WebRtcNetEQ_kMixFractionFuncTbl[0], 14);
+                = (int32_t) WEBRTC_SPL_LSHIFT_W32((int32_t) WebRtcNetEQ_kMixFractionFuncTbl[0], 14);
             w32_tmp
-                += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[1], w16_x1);
+                += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[1], w16_x1);
             w32_tmp
-                += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[2], w16_x2);
+                += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[2], w16_x2);
             w32_tmp
-                += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[3], w16_x3);
-            ExpandState->w16_vFraction = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 12);
+                += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[3], w16_x3);
+            ExpandState->w16_vFraction = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 12);
             ExpandState->w16_vFraction = WEBRTC_SPL_MIN(ExpandState->w16_vFraction, 16384);
             ExpandState->w16_vFraction = WEBRTC_SPL_MAX(ExpandState->w16_vFraction, 0);
         }
@@ -681,9 +681,9 @@
             /* Calculate (1-(1/slope))/w16_DistLag = (slope-1)/(w16_DistLag*slope) */
             w32_tmp = w16_slope - 8192;
             w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 12); /* Value in Q25 (13+12=25) */
-            w16_tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(w16_DistLag,
+            w16_tmp = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(w16_DistLag,
                 w16_slope, 8); /* Value in Q5  (13-8=5)  */
-            w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+            w16_tmp = (int16_t) WebRtcSpl_DivW32W16(w32_tmp,
                 w16_tmp); /* Res in Q20 (25-5=20) */
 
             if (w16_slope > 14746)
@@ -709,7 +709,7 @@
                 /* Calculate (1-slope)/w16_DistLag */
                 w32_tmp = 8192 - w16_slope;
                 w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
-                ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+                ExpandState->w16_muteSlope = (int16_t) WebRtcSpl_DivW32W16(w32_tmp,
                     w16_DistLag); /* Res in Q20 (20-0=20) */
             }
             ExpandState->w16_onset = 0;
@@ -723,7 +723,7 @@
             w32_tmp = 8192 - w16_slope;
             w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
             w32_tmp = WEBRTC_SPL_MAX(w32_tmp, 0);
-            ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+            ExpandState->w16_muteSlope = (int16_t) WebRtcSpl_DivW32W16(w32_tmp,
                 w16_DistLag); /* Res   in Q20    (20-0=20) */
             w16_tmp = WebRtcNetEQ_k5243div[fs_mult]; /* 0.005/fs_mult = 5243/fs_mult */
             ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(w16_tmp, ExpandState->w16_muteSlope);
@@ -772,7 +772,7 @@
             inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
             assert(w16_randLen <= FSMULT * 120 + 30);
             WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
-                (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
+                (int16_t) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
 #else
             assert(0);
 #endif
@@ -880,7 +880,7 @@
             for (i = 0; i < ExpandState->w16_overlap; i++)
             {
                 /* Do overlap add between new vector and overlap */
-                ExpandState->pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+                ExpandState->pw16_overlapVec[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
                     WEBRTC_SPL_MUL_16_16(ExpandState->pw16_overlapVec[i], w16_winMute) +
                     WEBRTC_SPL_MUL_16_16(
                         WEBRTC_SPL_MUL_16_16_RSFT(ExpandState->w16_expandMuteFactor,
@@ -913,7 +913,7 @@
             UNVOICED_LPC_ORDER);
         if (ExpandState->w16_arGainScale > 0)
         {
-            w32_tmp = ((WebRtc_Word32) 1) << (ExpandState->w16_arGainScale - 1);
+            w32_tmp = ((int32_t) 1) << (ExpandState->w16_arGainScale - 1);
         }
         else
         {
@@ -941,7 +941,7 @@
          >=64*fs_mult         => go from 1 to 0 in about 32 ms
          */
         w16_tmp = (31 - WebRtcSpl_NormW32(ExpandState->w16_maxLag)) - 5; /* getbits(w16_maxLag) -5 */
-        w16_vfractionChange = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(256, w16_tmp);
+        w16_vfractionChange = (int16_t) WEBRTC_SPL_RSHIFT_W32(256, w16_tmp);
         if (ExpandState->w16_stopMuting == 1)
         {
             w16_vfractionChange = 0;
@@ -963,7 +963,7 @@
             w16_tmp2 = 16384 - ExpandState->w16_currentVFraction;
             WebRtcSpl_ScaleAndAddVectorsWithRound(pw16_voicedVec + w16_tmp,
                 ExpandState->w16_currentVFraction, pw16_unvoicedVec + w16_tmp, w16_tmp2, 14,
-                pw16_outData + w16_tmp, (WebRtc_Word16) (w16_lag - w16_tmp));
+                pw16_outData + w16_tmp, (int16_t) (w16_lag - w16_tmp));
         }
 
         /* Select muting factor */
@@ -991,9 +991,9 @@
             {
                 WebRtcNetEQ_MuteSignal(pw16_outData, ExpandState->w16_muteSlope, w16_lag);
 
-                w16_tmp = 16384 - (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_lag,
+                w16_tmp = 16384 - (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_lag,
                     ExpandState->w16_muteSlope) + 8192) >> 6); /* 20-14 = 6 */
-                w16_tmp = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp,
+                w16_tmp = (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_tmp,
                     ExpandState->w16_expandMuteFactor) + 8192) >> 14);
 
                 /* Guard against getting stuck with very small (but sometimes audible) gain */
@@ -1025,7 +1025,7 @@
 
         if (BGNState->w16_scaleShift > 1)
         {
-            w32_tmp = ((WebRtc_Word32) 1) << (BGNState->w16_scaleShift - 1);
+            w32_tmp = ((int32_t) 1) << (BGNState->w16_scaleShift - 1);
         }
         else
         {
@@ -1051,7 +1051,7 @@
         {
             /* fade BGN to zero */
             /* calculate muting slope, approx 2^18/fsHz */
-            WebRtc_Word16 muteFactor;
+            int16_t muteFactor;
             if (fs_mult == 1)
             {
                 muteFactor = -32;
@@ -1136,7 +1136,7 @@
          * Only do this if StopMuting != 1 or if explicitly BGNonly, otherwise Expand is
          * called from Merge or Normal and special measures must be taken.
          */
-        inst->statInst.expandLength += (WebRtc_UWord32) *pw16_len;
+        inst->statInst.expandLength += (uint32_t) *pw16_len;
         if (ExpandState->w16_expandMuteFactor == 0 || BGNonly)
         {
             /* Only noise expansion */
@@ -1177,13 +1177,13 @@
 
 int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
 #ifdef SCRATCH
-                            WebRtc_Word16 *pw16_scratchPtr,
+                            int16_t *pw16_scratchPtr,
 #endif
-                            WebRtc_Word16 *pw16_outData, WebRtc_Word16 len)
+                            int16_t *pw16_outData, int16_t len)
 {
 
-    WebRtc_Word16 pos = 0;
-    WebRtc_Word16 tempLen = len;
+    int16_t pos = 0;
+    int16_t tempLen = len;
 
     while (tempLen > 0)
     {
diff --git a/modules/audio_coding/neteq/interface/webrtc_neteq.h b/modules/audio_coding/neteq/interface/webrtc_neteq.h
index 10adfa0..d6c68fd 100644
--- a/modules/audio_coding/neteq/interface/webrtc_neteq.h
+++ b/modules/audio_coding/neteq/interface/webrtc_neteq.h
@@ -111,25 +111,25 @@
  * API function calls for each codec
  */
 
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecode)(void* state, WebRtc_Word16* encoded,
-                                                WebRtc_Word16 len, WebRtc_Word16* decoded,
-                                                WebRtc_Word16* speechType);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodePLC)(void* state, WebRtc_Word16* decoded,
-                                                   WebRtc_Word16 frames);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodeInit)(void* state);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncAddLatePkt)(void* state, WebRtc_Word16* encoded,
-                                                    WebRtc_Word16 len);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetMDinfo)(void* state);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
-                                                      WebRtc_Word16* length);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
-                                                  WebRtc_Word32 packet_size,
-                                                  WebRtc_UWord16 rtp_seq_number,
-                                                  WebRtc_UWord32 send_ts,
-                                                  WebRtc_UWord32 arr_ts);
+typedef int16_t (*WebRtcNetEQ_FuncDecode)(void* state, int16_t* encoded,
+                                                int16_t len, int16_t* decoded,
+                                                int16_t* speechType);
+typedef int16_t (*WebRtcNetEQ_FuncDecodePLC)(void* state, int16_t* decoded,
+                                                   int16_t frames);
+typedef int16_t (*WebRtcNetEQ_FuncDecodeInit)(void* state);
+typedef int16_t (*WebRtcNetEQ_FuncAddLatePkt)(void* state, int16_t* encoded,
+                                                    int16_t len);
+typedef int16_t (*WebRtcNetEQ_FuncGetMDinfo)(void* state);
+typedef int16_t (*WebRtcNetEQ_FuncGetPitchInfo)(void* state, int16_t* encoded,
+                                                      int16_t* length);
+typedef int16_t (*WebRtcNetEQ_FuncUpdBWEst)(void* state, const uint16_t *encoded,
+                                                  int32_t packet_size,
+                                                  uint16_t rtp_seq_number,
+                                                  uint32_t send_ts,
+                                                  uint32_t arr_ts);
 typedef int (*WebRtcNetEQ_FuncDurationEst)(void* state, const uint8_t* payload,
                                            int payload_length_bytes);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetErrorCode)(void* state);
+typedef int16_t (*WebRtcNetEQ_FuncGetErrorCode)(void* state);
 
 /**********************************************************
  * Structures
@@ -138,7 +138,7 @@
 typedef struct
 {
     enum WebRtcNetEQDecoder codec;
-    WebRtc_Word16 payloadType;
+    int16_t payloadType;
     WebRtcNetEQ_FuncDecode funcDecode;
     WebRtcNetEQ_FuncDecode funcDecodeRCU;
     WebRtcNetEQ_FuncDecodePLC funcDecodePLC;
@@ -150,15 +150,15 @@
     WebRtcNetEQ_FuncDurationEst funcDurationEst;
     WebRtcNetEQ_FuncGetErrorCode funcGetErrorCode;
     void* codec_state;
-    WebRtc_UWord16 codec_fs;
+    uint16_t codec_fs;
 } WebRtcNetEQ_CodecDef;
 
 typedef struct
 {
-    WebRtc_UWord16 fraction_lost;
-    WebRtc_UWord32 cum_lost;
-    WebRtc_UWord32 ext_max;
-    WebRtc_UWord32 jitter;
+    uint16_t fraction_lost;
+    uint32_t cum_lost;
+    uint32_t ext_max;
+    uint32_t jitter;
 } WebRtcNetEQ_RTCPStat;
 
 /**********************************************************
@@ -184,7 +184,7 @@
 
 /* Init functions */
 
-int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs);
+int WebRtcNetEQ_Init(void *inst, uint16_t fs);
 int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon);
 int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs);
 int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode);
@@ -196,29 +196,29 @@
 int WebRtcNetEQ_CodecDbReset(void *inst);
 int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst);
 int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec);
-int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
-                                   WebRtc_Word16 *MaxEntries);
-int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
+int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, int16_t *UsedEntries,
+                                   int16_t *MaxEntries);
+int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, int16_t Entry,
                                     enum WebRtcNetEQDecoder *codec);
 
 /* Real-time functions */
 
-int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
-                      WebRtc_UWord32 uw32_timeRec);
-int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
+int WebRtcNetEQ_RecIn(void *inst, int16_t *p_w16datagramstart, int16_t w16_RTPlen,
+                      uint32_t uw32_timeRec);
+int WebRtcNetEQ_RecOut(void *inst, int16_t *pw16_outData, int16_t *pw16_len);
 int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
 int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
-int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp);
+int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, uint32_t *timestamp);
 int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType);
 
 /* VQmon related functions */
-int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
-                                      WebRtc_UWord16 *concealedVoiceDurationMs,
-                                      WebRtc_UWord8 *concealedVoiceFlags);
-int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
-                                      WebRtc_UWord8 *adaptationRate);
-int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
-                                     WebRtc_UWord16 *maxDelayMs);
+int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, uint16_t *validVoiceDurationMs,
+                                      uint16_t *concealedVoiceDurationMs,
+                                      uint8_t *concealedVoiceFlags);
+int WebRtcNetEQ_VQmonGetConfiguration(void *inst, uint16_t *absMaxDelayMs,
+                                      uint8_t *adaptationRate);
+int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, uint16_t *avgDelayMs,
+                                     uint16_t *maxDelayMs);
 
 #ifdef __cplusplus
 }
diff --git a/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h b/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
index 3509cf7..c124e26 100644
--- a/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
+++ b/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
@@ -24,11 +24,11 @@
 
 typedef struct
 {
-    WebRtc_UWord8 payloadType;
-    WebRtc_UWord16 sequenceNumber;
-    WebRtc_UWord32 timeStamp;
-    WebRtc_UWord32 SSRC;
-    WebRtc_UWord8 markerBit;
+    uint8_t payloadType;
+    uint16_t sequenceNumber;
+    uint32_t timeStamp;
+    uint32_t SSRC;
+    uint8_t markerBit;
 } WebRtcNetEQ_RTPInfo;
 
 /****************************************************************************
@@ -48,8 +48,8 @@
  *                            -1 - Error
  */
 int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
-                               const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
-                               WebRtc_UWord32 timeRec);
+                               const uint8_t *payloadPtr, int16_t payloadLenBytes,
+                               uint32_t timeRec);
 
 /****************************************************************************
  * WebRtcNetEQ_GetMasterSlaveInfoSize(...)
@@ -86,9 +86,9 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
-                                  WebRtc_Word16 *pw16_len, void *msInfo,
-                                  WebRtc_Word16 isMaster);
+int WebRtcNetEQ_RecOutMasterSlave(void *inst, int16_t *pw16_outData,
+                                  int16_t *pw16_len, void *msInfo,
+                                  int16_t isMaster);
 
 typedef struct
 {
@@ -143,7 +143,7 @@
 typedef int (*WebRtcNetEQ_VADInitFunction)(void *VAD_inst);
 typedef int (*WebRtcNetEQ_VADSetmodeFunction)(void *VAD_inst, int mode);
 typedef int (*WebRtcNetEQ_VADFunction)(void *VAD_inst, int fs,
-    WebRtc_Word16 *frame, int frameLen);
+    int16_t *frame, int frameLen);
 
 /****************************************************************************
  * WebRtcNetEQ_SetVADInstance(...)
@@ -210,8 +210,8 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
-                               WebRtc_Word16 *pw16_len);
+int WebRtcNetEQ_RecOutNoDecode(void *inst, int16_t *pw16_outData,
+                               int16_t *pw16_len);
 
 /****************************************************************************
  * WebRtcNetEQ_FlushBuffers(...)
diff --git a/modules/audio_coding/neteq/mcu.h b/modules/audio_coding/neteq/mcu.h
index 499684a..6994a36 100644
--- a/modules/audio_coding/neteq/mcu.h
+++ b/modules/audio_coding/neteq/mcu.h
@@ -43,19 +43,19 @@
 typedef struct
 {
 
-    WebRtc_Word16 current_Codec;
-    WebRtc_Word16 current_Payload;
-    WebRtc_UWord32 timeStamp; /* Next timestamp that should be played */
-    WebRtc_Word16 millisecondsPerCall;
-    WebRtc_UWord16 timestampsPerCall; /* Output chunk size */
-    WebRtc_UWord16 fs;
-    WebRtc_UWord32 ssrc; /* Current ssrc */
-    WebRtc_Word16 new_codec;
-    WebRtc_Word16 first_packet;
+    int16_t current_Codec;
+    int16_t current_Payload;
+    uint32_t timeStamp; /* Next timestamp that should be played */
+    int16_t millisecondsPerCall;
+    uint16_t timestampsPerCall; /* Output chunk size */
+    uint16_t fs;
+    uint32_t ssrc; /* Current ssrc */
+    int16_t new_codec;
+    int16_t first_packet;
 
     /* MCU/DSP Communication layer */
-    WebRtc_Word16 *pw16_readAddress;
-    WebRtc_Word16 *pw16_writeAddress;
+    int16_t *pw16_readAddress;
+    int16_t *pw16_writeAddress;
     void *main_inst;
 
     CodecDbInst_t codec_DB_inst; /* Information about all the codecs, i.e. which
@@ -71,21 +71,21 @@
     dtmf_inst_t DTMF_inst;
 #endif
     int NoOfExpandCalls;
-    WebRtc_Word16 AVT_PlayoutOn;
+    int16_t AVT_PlayoutOn;
     enum WebRtcNetEQPlayoutMode NetEqPlayoutMode;
 
-    WebRtc_Word16 one_desc; /* Number of times running on one desc */
+    int16_t one_desc; /* Number of times running on one desc */
 
-    WebRtc_UWord32 lostTS; /* Number of timestamps lost */
-    WebRtc_UWord32 lastReportTS; /* Timestamp elapsed since last report was given */
+    uint32_t lostTS; /* Number of timestamps lost */
+    uint32_t lastReportTS; /* Timestamp elapsed since last report was given */
 
     int waiting_times[kLenWaitingTimes];  /* Waiting time statistics storage. */
     int len_waiting_times;
     int next_waiting_time_index;
 
-    WebRtc_UWord32 externalTS;
-    WebRtc_UWord32 internalTS;
-    WebRtc_Word16 TSscalingInitialized;
+    uint32_t externalTS;
+    uint32_t internalTS;
+    int16_t TSscalingInitialized;
     enum TsScaling scalingFactor;
 
 #ifdef NETEQ_STEREO
@@ -187,7 +187,7 @@
  * Return value         :  0 - Ok
  *                        <0 - Error
  */
-int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs_hz);
+int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, uint16_t fs_hz);
 
 /****************************************************************************
  * WebRtcNetEQ_SignalMcu(...)
@@ -217,7 +217,7 @@
  */
 
 int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacket,
-                              WebRtc_UWord32 uw32_timeRec);
+                              uint32_t uw32_timeRec);
 
 /****************************************************************************
  * WebRtcNetEQ_RecInInternal(...)
@@ -234,7 +234,7 @@
  *                        -1 - Error
  */
 int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
-                                      SplitInfo_t *split_inst, WebRtc_Word16 *flushed);
+                                      SplitInfo_t *split_inst, int16_t *flushed);
 
 /****************************************************************************
  * WebRtcNetEQ_GetTimestampScaling(...)
@@ -264,8 +264,8 @@
  * Return value         : Internal timestamp
  */
 
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
-                                                            WebRtc_UWord32 externalTS);
+uint32_t WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
+                                                      uint32_t externalTS);
 
 /****************************************************************************
  * WebRtcNetEQ_ScaleTimestampInternalToExternal(...)
@@ -279,6 +279,6 @@
  * Return value         : External timestamp
  */
 
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
-                                                            WebRtc_UWord32 internalTS);
+uint32_t WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
+                                                      uint32_t internalTS);
 #endif
diff --git a/modules/audio_coding/neteq/mcu_address_init.c b/modules/audio_coding/neteq/mcu_address_init.c
index 0306a85..666ecc8 100644
--- a/modules/audio_coding/neteq/mcu_address_init.c
+++ b/modules/audio_coding/neteq/mcu_address_init.c
@@ -19,8 +19,8 @@
                                void * Data2DspAddress, void *main_inst)
 {
 
-    inst->pw16_readAddress = (WebRtc_Word16*) Data2McuAddress;
-    inst->pw16_writeAddress = (WebRtc_Word16*) Data2DspAddress;
+    inst->pw16_readAddress = (int16_t*) Data2McuAddress;
+    inst->pw16_writeAddress = (int16_t*) Data2DspAddress;
     inst->main_inst = main_inst;
 
     inst->millisecondsPerCall = 10;
diff --git a/modules/audio_coding/neteq/mcu_dsp_common.c b/modules/audio_coding/neteq/mcu_dsp_common.c
index 13025d4..744a131 100644
--- a/modules/audio_coding/neteq/mcu_dsp_common.c
+++ b/modules/audio_coding/neteq/mcu_dsp_common.c
@@ -29,7 +29,7 @@
 }
 
 /* The DSP side will call this function to interrupt the MCU side */
-int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem)
+int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, int16_t *pw16_shared_mem)
 {
     inst->MCUinst.pw16_readAddress = pw16_shared_mem;
     inst->MCUinst.pw16_writeAddress = pw16_shared_mem;
diff --git a/modules/audio_coding/neteq/mcu_dsp_common.h b/modules/audio_coding/neteq/mcu_dsp_common.h
index e3f4213..badffa1 100644
--- a/modules/audio_coding/neteq/mcu_dsp_common.h
+++ b/modules/audio_coding/neteq/mcu_dsp_common.h
@@ -36,26 +36,26 @@
 {
     DSPInst_t DSPinst; /* DSP part of the NetEQ instance */
     MCUInst_t MCUinst; /* MCU part of the NetEQ instance */
-    WebRtc_Word16 ErrorCode; /* Store last error code */
+    int16_t ErrorCode; /* Store last error code */
 #ifdef NETEQ_STEREO
-    WebRtc_Word16 masterSlave; /* 0 = not set, 1 = master, 2 = slave */
+    int16_t masterSlave; /* 0 = not set, 1 = master, 2 = slave */
 #endif /* NETEQ_STEREO */
 } MainInst_t;
 
 /* Struct used for communication between DSP and MCU sides of NetEQ */
 typedef struct
 {
-    WebRtc_UWord32 playedOutTS; /* Timestamp position at end of DSP data */
-    WebRtc_UWord16 samplesLeft; /* Number of samples stored */
-    WebRtc_Word16 MD; /* Multiple description codec information */
-    WebRtc_Word16 lastMode; /* Latest mode of NetEQ playout */
-    WebRtc_Word16 frameLen; /* Frame length of previously decoded packet */
+    uint32_t playedOutTS; /* Timestamp position at end of DSP data */
+    uint16_t samplesLeft; /* Number of samples stored */
+    int16_t MD; /* Multiple description codec information */
+    int16_t lastMode; /* Latest mode of NetEQ playout */
+    int16_t frameLen; /* Frame length of previously decoded packet */
 } DSP2MCU_info_t;
 
 /* Initialize instances with read and write address */
 int WebRtcNetEQ_DSPinit(MainInst_t *inst);
 
 /* The DSP side will call this function to interrupt the MCU side */
-int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem);
+int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, int16_t *pw16_shared_mem);
 
 #endif
diff --git a/modules/audio_coding/neteq/merge.c b/modules/audio_coding/neteq/merge.c
index 5f020a9..bd5239c 100644
--- a/modules/audio_coding/neteq/merge.c
+++ b/modules/audio_coding/neteq/merge.c
@@ -44,11 +44,11 @@
 /* Scratch usage:
 
  Type           Name                    size            startpos        endpos
- WebRtc_Word16  pw16_expanded           210*fs/8000     0               209*fs/8000
- WebRtc_Word16  pw16_expandedLB         100             210*fs/8000     99+210*fs/8000
- WebRtc_Word16  pw16_decodedLB          40              100+210*fs/8000 139+210*fs/8000
- WebRtc_Word32  pw32_corr               2*60            140+210*fs/8000 260+210*fs/8000
- WebRtc_Word16  pw16_corrVec            68              210*fs/8000     67+210*fs/8000
+ int16_t  pw16_expanded           210*fs/8000     0               209*fs/8000
+ int16_t  pw16_expandedLB         100             210*fs/8000     99+210*fs/8000
+ int16_t  pw16_decodedLB          40              100+210*fs/8000 139+210*fs/8000
+ int32_t  pw32_corr               2*60            140+210*fs/8000 260+210*fs/8000
+ int16_t  pw16_corrVec            68              210*fs/8000     67+210*fs/8000
 
  [gap in scratch vector]
 
@@ -86,40 +86,40 @@
 
 int WebRtcNetEQ_Merge(DSPInst_t *inst,
 #ifdef SCRATCH
-                      WebRtc_Word16 *pw16_scratchPtr,
+                      int16_t *pw16_scratchPtr,
 #endif
-                      WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
-                      WebRtc_Word16 *pw16_len)
+                      int16_t *pw16_decoded, int len, int16_t *pw16_outData,
+                      int16_t *pw16_len)
 {
 
-    WebRtc_Word16 fs_mult;
-    WebRtc_Word16 fs_shift;
-    WebRtc_Word32 w32_En_new_frame, w32_En_old_frame;
-    WebRtc_Word16 w16_expmax, w16_newmax;
-    WebRtc_Word16 w16_tmp, w16_tmp2;
-    WebRtc_Word32 w32_tmp;
+    int16_t fs_mult;
+    int16_t fs_shift;
+    int32_t w32_En_new_frame, w32_En_old_frame;
+    int16_t w16_expmax, w16_newmax;
+    int16_t w16_tmp, w16_tmp2;
+    int32_t w32_tmp;
 #ifdef SCRATCH
-    WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_pw16_expanded;
-    WebRtc_Word16 *pw16_expandedLB = pw16_scratchPtr + SCRATCH_pw16_expandedLB;
-    WebRtc_Word16 *pw16_decodedLB = pw16_scratchPtr + SCRATCH_pw16_decodedLB;
-    WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
-    WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;
+    int16_t *pw16_expanded = pw16_scratchPtr + SCRATCH_pw16_expanded;
+    int16_t *pw16_expandedLB = pw16_scratchPtr + SCRATCH_pw16_expandedLB;
+    int16_t *pw16_decodedLB = pw16_scratchPtr + SCRATCH_pw16_decodedLB;
+    int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_pw32_corr);
+    int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;
 #else
-    WebRtc_Word16 pw16_expanded[(125+80+5)*FSMULT];
-    WebRtc_Word16 pw16_expandedLB[100];
-    WebRtc_Word16 pw16_decodedLB[40];
-    WebRtc_Word32 pw32_corr[60];
-    WebRtc_Word16 pw16_corrVec[4+60+4];
+    int16_t pw16_expanded[(125+80+5)*FSMULT];
+    int16_t pw16_expandedLB[100];
+    int16_t pw16_decodedLB[40];
+    int32_t pw32_corr[60];
+    int16_t pw16_corrVec[4+60+4];
 #endif
-    WebRtc_Word16 *pw16_corr = &pw16_corrVec[4];
-    WebRtc_Word16 w16_stopPos = 0, w16_bestIndex, w16_interpLen;
-    WebRtc_Word16 w16_bestVal; /* bestVal is dummy */
-    WebRtc_Word16 w16_startfact, w16_inc;
-    WebRtc_Word16 w16_expandedLen;
-    WebRtc_Word16 w16_startPos;
-    WebRtc_Word16 w16_expLen, w16_newLen = 0;
-    WebRtc_Word16 *pw16_decodedOut;
-    WebRtc_Word16 w16_muted;
+    int16_t *pw16_corr = &pw16_corrVec[4];
+    int16_t w16_stopPos = 0, w16_bestIndex, w16_interpLen;
+    int16_t w16_bestVal; /* bestVal is dummy */
+    int16_t w16_startfact, w16_inc;
+    int16_t w16_expandedLen;
+    int16_t w16_startPos;
+    int16_t w16_expLen, w16_newLen = 0;
+    int16_t *pw16_decodedOut;
+    int16_t w16_muted;
 
     int w16_decodedLen = len;
 
@@ -209,13 +209,13 @@
 
     /* Adjust muting factor (main muting factor times expand muting factor) */
     inst->w16_muteFactor
-        = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
+        = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
             inst->ExpandInst.w16_expandMuteFactor, 14);
 
     /* Adjust muting factor if new vector is more or less of the BGN energy */
     len = WEBRTC_SPL_MIN(64*fs_mult, w16_decodedLen);
-    w16_expmax = WebRtcSpl_MaxAbsValueW16(pw16_expanded, (WebRtc_Word16) len);
-    w16_newmax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+    w16_expmax = WebRtcSpl_MaxAbsValueW16(pw16_expanded, (int16_t) len);
+    w16_newmax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len);
 
     /* Calculate energy of old data */
     w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_expmax, w16_expmax));
@@ -251,10 +251,10 @@
         w16_tmp = w16_tmp + 14;
         w32_En_old_frame = WEBRTC_SPL_SHIFT_W32(w32_En_old_frame, w16_tmp);
         w16_tmp
-            = WebRtcSpl_DivW32W16ResW16(w32_En_old_frame, (WebRtc_Word16) w32_En_new_frame);
+            = WebRtcSpl_DivW32W16ResW16(w32_En_old_frame, (int16_t) w32_En_new_frame);
         /* Calculate sqrt(w32_En_old_frame/w32_En_new_frame) in Q14 */
-        w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
-            WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,14));
+        w16_muted = (int16_t) WebRtcSpl_SqrtFloor(
+            WEBRTC_SPL_LSHIFT_W32((int32_t)w16_tmp,14));
     }
     else
     {
@@ -288,56 +288,56 @@
         /* Downsample to 4 kHz */
         if (inst->fs == 8000)
         {
-            WebRtcSpl_DownsampleFast(&pw16_expanded[2], (WebRtc_Word16) (w16_expandedLen - 2),
-                pw16_expandedLB, (WebRtc_Word16) (100),
-                (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl, (WebRtc_Word16) 3,
-                (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+            WebRtcSpl_DownsampleFast(&pw16_expanded[2], (int16_t) (w16_expandedLen - 2),
+                pw16_expandedLB, (int16_t) (100),
+                (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl, (int16_t) 3,
+                (int16_t) 2, (int16_t) 0);
             if (w16_decodedLen <= 80)
             {
                 /* Not quite long enough, so we have to cheat a bit... */
-                WebRtc_Word16 temp_len = w16_decodedLen - 2;
+                int16_t temp_len = w16_decodedLen - 2;
                 w16_tmp = temp_len / 2;
                 WebRtcSpl_DownsampleFast(&pw16_decoded[2], temp_len,
                                          pw16_decodedLB, w16_tmp,
-                                         (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
-                    (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+                                         (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl,
+                    (int16_t) 3, (int16_t) 2, (int16_t) 0);
                 WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40 - w16_tmp));
             }
             else
             {
                 WebRtcSpl_DownsampleFast(&pw16_decoded[2],
-                    (WebRtc_Word16) (w16_decodedLen - 2), pw16_decodedLB,
-                    (WebRtc_Word16) (40), (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
-                    (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+                    (int16_t) (w16_decodedLen - 2), pw16_decodedLB,
+                    (int16_t) (40), (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl,
+                    (int16_t) 3, (int16_t) 2, (int16_t) 0);
             }
 #ifdef NETEQ_WIDEBAND
         }
         else if (inst->fs==16000)
         {
             WebRtcSpl_DownsampleFast(
-                &pw16_expanded[4], (WebRtc_Word16)(w16_expandedLen-4),
-                pw16_expandedLB, (WebRtc_Word16)(100),
-                (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
-                (WebRtc_Word16)4, (WebRtc_Word16)0);
+                &pw16_expanded[4], (int16_t)(w16_expandedLen-4),
+                pw16_expandedLB, (int16_t)(100),
+                (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5,
+                (int16_t)4, (int16_t)0);
             if (w16_decodedLen<=160)
             {
                 /* Not quite long enough, so we have to cheat a bit... */
-                WebRtc_Word16 temp_len = w16_decodedLen - 4;
+                int16_t temp_len = w16_decodedLen - 4;
                 w16_tmp = temp_len / 4;
                 WebRtcSpl_DownsampleFast(
                     &pw16_decoded[4], temp_len,
                     pw16_decodedLB, w16_tmp,
-                    (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
-                    (WebRtc_Word16)4, (WebRtc_Word16)0);
+                    (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5,
+                    (int16_t)4, (int16_t)0);
                 WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
             }
             else
             {
                 WebRtcSpl_DownsampleFast(
-                    &pw16_decoded[4], (WebRtc_Word16)(w16_decodedLen-4),
-                    pw16_decodedLB, (WebRtc_Word16)(40),
-                    (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
-                    (WebRtc_Word16)4, (WebRtc_Word16)0);
+                    &pw16_decoded[4], (int16_t)(w16_decodedLen-4),
+                    pw16_decodedLB, (int16_t)(40),
+                    (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5,
+                    (int16_t)4, (int16_t)0);
             }
 #endif
 #ifdef NETEQ_32KHZ_WIDEBAND
@@ -348,29 +348,29 @@
              * TODO(hlundin) Why is the offset into pw16_expanded 6?
              */
             WebRtcSpl_DownsampleFast(
-                &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
-                pw16_expandedLB, (WebRtc_Word16)(100),
-                (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
-                (WebRtc_Word16)8, (WebRtc_Word16)0);
+                &pw16_expanded[6], (int16_t)(w16_expandedLen-6),
+                pw16_expandedLB, (int16_t)(100),
+                (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7,
+                (int16_t)8, (int16_t)0);
             if (w16_decodedLen<=320)
             {
                 /* Not quite long enough, so we have to cheat a bit... */
-                WebRtc_Word16 temp_len = w16_decodedLen - 6;
+                int16_t temp_len = w16_decodedLen - 6;
                 w16_tmp = temp_len / 8;
                 WebRtcSpl_DownsampleFast(
                       &pw16_decoded[6], temp_len,
                       pw16_decodedLB, w16_tmp,
-                      (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
-                      (WebRtc_Word16)8, (WebRtc_Word16)0);
+                      (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7,
+                      (int16_t)8, (int16_t)0);
                 WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
             }
             else
             {
                 WebRtcSpl_DownsampleFast(
-                    &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
-                    pw16_decodedLB, (WebRtc_Word16)(40),
-                    (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
-                    (WebRtc_Word16)8, (WebRtc_Word16)0);
+                    &pw16_decoded[6], (int16_t)(w16_decodedLen-6),
+                    pw16_decodedLB, (int16_t)(40),
+                    (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7,
+                    (int16_t)8, (int16_t)0);
             }
 #endif
 #ifdef NETEQ_48KHZ_WIDEBAND
@@ -381,10 +381,10 @@
              * TODO(hlundin) Why is the offset into pw16_expanded 6?
              */
             WebRtcSpl_DownsampleFast(
-                &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
-                pw16_expandedLB, (WebRtc_Word16)(100),
-                (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
-                (WebRtc_Word16)12, (WebRtc_Word16)0);
+                &pw16_expanded[6], (int16_t)(w16_expandedLen-6),
+                pw16_expandedLB, (int16_t)(100),
+                (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7,
+                (int16_t)12, (int16_t)0);
             if (w16_decodedLen<=320)
             {
                 /* Not quite long enough, so we have to cheat a bit... */
@@ -393,29 +393,29 @@
                  * but w16_tmp = temp_len / 8.
                  * (Was w16_tmp = ((w16_decodedLen-6)>>3) before re-write.)
                  */
-                WebRtc_Word16 temp_len = w16_decodedLen - 6;
+                int16_t temp_len = w16_decodedLen - 6;
                 w16_tmp = temp_len / 8;
                 WebRtcSpl_DownsampleFast(
                     &pw16_decoded[6], temp_len,
                     pw16_decodedLB, w16_tmp,
-                    (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
-                    (WebRtc_Word16)12, (WebRtc_Word16)0);
+                    (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7,
+                    (int16_t)12, (int16_t)0);
                 WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
             }
             else
             {
                 WebRtcSpl_DownsampleFast(
-                    &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
-                    pw16_decodedLB, (WebRtc_Word16)(40),
-                    (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
-                    (WebRtc_Word16)12, (WebRtc_Word16)0);
+                    &pw16_decoded[6], (int16_t)(w16_decodedLen-6),
+                    pw16_decodedLB, (int16_t)(40),
+                    (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7,
+                    (int16_t)12, (int16_t)0);
             }
 #endif
         }
 
         /* Calculate correlation without any normalization (40 samples) */
-        w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) inst->ExpandInst.w16_maxLag,
-            (WebRtc_Word16) (fs_mult * 2)) + 1;
+        w16_tmp = WebRtcSpl_DivW32W16ResW16((int32_t) inst->ExpandInst.w16_maxLag,
+            (int16_t) (fs_mult * 2)) + 1;
         w16_stopPos = WEBRTC_SPL_MIN(60, w16_tmp);
         w32_tmp = WEBRTC_SPL_MUL_16_16(w16_expmax, w16_newmax);
         if (w32_tmp > 26843546)
@@ -428,9 +428,9 @@
         }
 
         WebRtcNetEQ_CrossCorr(pw32_corr, pw16_decodedLB, pw16_expandedLB, 40,
-            (WebRtc_Word16) w16_stopPos, w16_tmp, 1);
+            (int16_t) w16_stopPos, w16_tmp, 1);
 
-        /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+        /* Normalize correlation to 14 bits and put in a int16_t vector */
         WebRtcSpl_MemSetW16(pw16_corrVec, 0, (4 + 60 + 4));
         w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_stopPos);
         w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
@@ -445,8 +445,8 @@
         w16_tmp = WEBRTC_SPL_MAX(0, WEBRTC_SPL_MAX(w16_startPos,
                 inst->timestampsPerCall+inst->ExpandInst.w16_overlap) - w16_decodedLen);
         /* Downscale starting index to 4kHz domain */
-        w16_tmp2 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) w16_tmp,
-            (WebRtc_Word16) (fs_mult << 1));
+        w16_tmp2 = WebRtcSpl_DivW32W16ResW16((int32_t) w16_tmp,
+            (int16_t) (fs_mult << 1));
 
 #ifdef NETEQ_STEREO
     } /* end if (msInfo->msMode != NETEQ_SLAVE)  */
@@ -500,10 +500,10 @@
     if (inst->w16_muteFactor < 16384)
     {
         WebRtcNetEQ_UnmuteSignal(pw16_decoded, &inst->w16_muteFactor, pw16_decoded, w16_inc,
-            (WebRtc_Word16) w16_interpLen);
+            (int16_t) w16_interpLen);
         WebRtcNetEQ_UnmuteSignal(&pw16_decoded[w16_interpLen], &inst->w16_muteFactor,
             &pw16_decodedOut[w16_interpLen], w16_inc,
-            (WebRtc_Word16) (w16_decodedLen - w16_interpLen));
+            (int16_t) (w16_decodedLen - w16_interpLen));
     }
     else
     {
@@ -514,7 +514,7 @@
     }
 
     /* Do overlap and interpolate linearly */
-    w16_inc = WebRtcSpl_DivW32W16ResW16(16384, (WebRtc_Word16) (w16_interpLen + 1)); /* Q14 */
+    w16_inc = WebRtcSpl_DivW32W16ResW16(16384, (int16_t) (w16_interpLen + 1)); /* Q14 */
     w16_startfact = (16384 - w16_inc);
     WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_expanded, w16_bestIndex);
     WebRtcNetEQ_MixVoiceUnvoice(pw16_decodedOut, &pw16_expanded[w16_bestIndex], pw16_decoded,
diff --git a/modules/audio_coding/neteq/min_distortion.c b/modules/audio_coding/neteq/min_distortion.c
index 4c9ee1c..47e2b44 100644
--- a/modules/audio_coding/neteq/min_distortion.c
+++ b/modules/audio_coding/neteq/min_distortion.c
@@ -16,17 +16,17 @@
 
 #include "signal_processing_library.h"
 
-WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
-                                        WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
-                                        WebRtc_Word16 len, WebRtc_Word32 *pw16_dist)
+int16_t WebRtcNetEQ_MinDistortion(const int16_t *pw16_data,
+                                  int16_t w16_minLag, int16_t w16_maxLag,
+                                  int16_t len, int32_t *pw16_dist)
 {
     int i, j;
-    const WebRtc_Word16 *pw16_data1;
-    const WebRtc_Word16 *pw16_data2;
-    WebRtc_Word32 w32_diff;
-    WebRtc_Word32 w32_sumdiff;
-    WebRtc_Word16 bestIndex = -1;
-    WebRtc_Word32 minDist = WEBRTC_SPL_WORD32_MAX;
+    const int16_t *pw16_data1;
+    const int16_t *pw16_data2;
+    int32_t w32_diff;
+    int32_t w32_sumdiff;
+    int16_t bestIndex = -1;
+    int32_t minDist = WEBRTC_SPL_WORD32_MAX;
 
     for (i = w16_minLag; i <= w16_maxLag; i++)
     {
diff --git a/modules/audio_coding/neteq/mix_voice_unvoice.c b/modules/audio_coding/neteq/mix_voice_unvoice.c
index 9895630..6c70d49 100644
--- a/modules/audio_coding/neteq/mix_voice_unvoice.c
+++ b/modules/audio_coding/neteq/mix_voice_unvoice.c
@@ -17,19 +17,19 @@
 
 #include "signal_processing_library.h"
 
-void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
-                                 WebRtc_Word16 *pw16_unvoicedVec,
-                                 WebRtc_Word16 *w16_current_vfraction,
-                                 WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N)
+void WebRtcNetEQ_MixVoiceUnvoice(int16_t *pw16_outData, int16_t *pw16_voicedVec,
+                                 int16_t *pw16_unvoicedVec,
+                                 int16_t *w16_current_vfraction,
+                                 int16_t w16_vfraction_change, int16_t N)
 {
     int i;
-    WebRtc_Word16 w16_tmp2;
-    WebRtc_Word16 vfraction = *w16_current_vfraction;
+    int16_t w16_tmp2;
+    int16_t vfraction = *w16_current_vfraction;
 
     w16_tmp2 = 16384 - vfraction;
     for (i = 0; i < N; i++)
     {
-        pw16_outData[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+        pw16_outData[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
             WEBRTC_SPL_MUL_16_16(vfraction, pw16_voicedVec[i]) +
             WEBRTC_SPL_MUL_16_16(w16_tmp2, pw16_unvoicedVec[i]) + 8192,
             14);
diff --git a/modules/audio_coding/neteq/mute_signal.c b/modules/audio_coding/neteq/mute_signal.c
index ee899cf..767a71d 100644
--- a/modules/audio_coding/neteq/mute_signal.c
+++ b/modules/audio_coding/neteq/mute_signal.c
@@ -16,16 +16,16 @@
 
 #include "signal_processing_library.h"
 
-void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
-                            WebRtc_Word16 N)
+void WebRtcNetEQ_MuteSignal(int16_t *pw16_inout, int16_t muteSlope,
+                            int16_t N)
 {
     int i;
-    WebRtc_Word32 w32_tmp = 1048608; /* (16384<<6 + 32) */
+    int32_t w32_tmp = 1048608; /* (16384<<6 + 32) */
 
     for (i = 0; i < N; i++)
     {
         pw16_inout[i]
-            = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16((WebRtc_Word16)(w32_tmp>>6), pw16_inout[i])
+            = (int16_t) ((WEBRTC_SPL_MUL_16_16((int16_t)(w32_tmp>>6), pw16_inout[i])
                 + 8192) >> 14);
         w32_tmp -= muteSlope;
     }
diff --git a/modules/audio_coding/neteq/neteq_defines.h b/modules/audio_coding/neteq/neteq_defines.h
index 1f092df..9390f2b 100644
--- a/modules/audio_coding/neteq/neteq_defines.h
+++ b/modules/audio_coding/neteq/neteq_defines.h
@@ -143,8 +143,8 @@
 /* Payload data will contain the SID frame if there is one*/
 
 #define DSP_INSTR_DTMF_GENERATE                  0x6000
-/* Payload data will be one WebRtc_Word16 with the current DTMF value and one
- * WebRtc_Word16 with the current volume value
+/* Payload data will be one int16_t with the current DTMF value and one
+ * int16_t with the current volume value
  */
 #define DSP_INSTR_NORMAL_ONE_DESC                0x7000
 /* No encoded frames */
diff --git a/modules/audio_coding/neteq/neteq_statistics.h b/modules/audio_coding/neteq/neteq_statistics.h
index 727b109..f355b58 100644
--- a/modules/audio_coding/neteq/neteq_statistics.h
+++ b/modules/audio_coding/neteq/neteq_statistics.h
@@ -24,15 +24,15 @@
 {
 
     /* variables for in-call statistics; queried through WebRtcNetEQ_GetNetworkStatistics */
-    WebRtc_UWord32 expandLength; /* number of samples produced through expand */
-    WebRtc_UWord32 preemptiveLength; /* number of samples produced through pre-emptive
+    uint32_t expandLength; /* number of samples produced through expand */
+    uint32_t preemptiveLength; /* number of samples produced through pre-emptive
      expand */
-    WebRtc_UWord32 accelerateLength; /* number of samples removed through accelerate */
+    uint32_t accelerateLength; /* number of samples removed through accelerate */
     int addedSamples; /* number of samples inserted in off mode */
 
     /* variables for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics */
-    WebRtc_UWord32 expandedVoiceSamples; /* number of voice samples produced through expand */
-    WebRtc_UWord32 expandedNoiseSamples; /* number of noise (background) samples produced
+    uint32_t expandedVoiceSamples; /* number of voice samples produced through expand */
+    uint32_t expandedNoiseSamples; /* number of noise (background) samples produced
      through expand */
 
 } DSPStats_t;
diff --git a/modules/audio_coding/neteq/normal.c b/modules/audio_coding/neteq/normal.c
index b33940a..8cbda52 100644
--- a/modules/audio_coding/neteq/normal.c
+++ b/modules/audio_coding/neteq/normal.c
@@ -20,7 +20,7 @@
 /* Scratch usage:
 
  Type           Name                    size            startpos        endpos
- WebRtc_Word16  pw16_expanded           125*fs/8000     0               125*fs/8000-1
+ int16_t  pw16_expanded           125*fs/8000     0               125*fs/8000-1
 
  func           WebRtcNetEQ_Expand      40+370*fs/8000  125*fs/8000     39+495*fs/8000
 
@@ -66,21 +66,21 @@
 
 int WebRtcNetEQ_Normal(DSPInst_t *inst,
 #ifdef SCRATCH
-                       WebRtc_Word16 *pw16_scratchPtr,
+                       int16_t *pw16_scratchPtr,
 #endif
-                       WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
-                       WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
+                       int16_t *pw16_decoded, int16_t len,
+                       int16_t *pw16_outData, int16_t *pw16_len)
 {
 
     int i;
-    WebRtc_Word16 fs_mult;
-    WebRtc_Word16 fs_shift;
-    WebRtc_Word32 w32_En_speech;
-    WebRtc_Word16 enLen;
-    WebRtc_Word16 w16_muted;
-    WebRtc_Word16 w16_inc, w16_frac;
-    WebRtc_Word16 w16_tmp;
-    WebRtc_Word32 w32_tmp;
+    int16_t fs_mult;
+    int16_t fs_shift;
+    int32_t w32_En_speech;
+    int16_t enLen;
+    int16_t w16_muted;
+    int16_t w16_inc, w16_frac;
+    int16_t w16_tmp;
+    int32_t w32_tmp;
 
     /* Sanity check */
     if (len < 0)
@@ -108,15 +108,15 @@
 
         /* Define memory where temporary result from Expand algorithm can be stored. */
 #ifdef SCRATCH
-        WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_PW16_EXPANDED;
+        int16_t *pw16_expanded = pw16_scratchPtr + SCRATCH_PW16_EXPANDED;
 #else
-        WebRtc_Word16 pw16_expanded[FSMULT * 125];
+        int16_t pw16_expanded[FSMULT * 125];
 #endif
-        WebRtc_Word16 expandedLen = 0;
-        WebRtc_Word16 w16_decodedMax;
+        int16_t expandedLen = 0;
+        int16_t w16_decodedMax;
 
         /* Find largest value in new data */
-        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len);
 
         /* Generate interpolation data using Expand */
         /* First, set Expand parameters to appropriate values. */
@@ -129,7 +129,7 @@
 #ifdef SCRATCH
             pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
 #endif
-            pw16_expanded, &expandedLen, (WebRtc_Word16) (inst->w16_mode == MODE_FADE_TO_BGN));
+            pw16_expanded, &expandedLen, (int16_t) (inst->w16_mode == MODE_FADE_TO_BGN));
 
         inst->ExpandInst.w16_stopMuting = 0; /* Restore value */
         inst->ExpandInst.w16_consecExp = 0; /* Last was not Expand any more */
@@ -144,7 +144,7 @@
         {
             /* w16_muteFactor * w16_expandMuteFactor */
             inst->w16_muteFactor
-                = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
+                = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
                     inst->ExpandInst.w16_expandMuteFactor, 14);
         }
 
@@ -154,7 +154,7 @@
             WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
         w16_tmp = WEBRTC_SPL_MAX(w16_tmp, 0);
         w32_En_speech = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, enLen, w16_tmp);
-        w32_En_speech = WebRtcSpl_DivW32W16(w32_En_speech, (WebRtc_Word16) (enLen >> w16_tmp));
+        w32_En_speech = WebRtcSpl_DivW32W16(w32_En_speech, (int16_t) (enLen >> w16_tmp));
 
         if ((w32_En_speech != 0) && (w32_En_speech > inst->BGNInst.w32_energy))
         {
@@ -162,10 +162,10 @@
             w16_tmp = WebRtcSpl_NormW32(w32_En_speech) - 16;
             /* we want inst->BGNInst.energy/En_speech in Q14 */
             w32_tmp = WEBRTC_SPL_SHIFT_W32(inst->BGNInst.w32_energy, (w16_tmp+14));
-            w16_tmp = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w32_En_speech, w16_tmp);
-            w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp, w16_tmp);
-            w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
-                WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) w16_tmp,
+            w16_tmp = (int16_t) WEBRTC_SPL_SHIFT_W32(w32_En_speech, w16_tmp);
+            w16_tmp = (int16_t) WebRtcSpl_DivW32W16(w32_tmp, w16_tmp);
+            w16_muted = (int16_t) WebRtcSpl_SqrtFloor(
+                WEBRTC_SPL_LSHIFT_W32((int32_t) w16_tmp,
                     14)); /* w16_muted in Q14 (sqrt(Q28)) */
         }
         else
@@ -184,7 +184,7 @@
             /* scale with mute factor */
             w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
             /* shift 14 with proper rounding */
-            pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
+            pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
             /* increase mute_factor towards 16384 */
             inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
         }
@@ -198,7 +198,7 @@
         w16_frac = w16_inc;
         for (i = 0; i < 8 * fs_mult; i++)
         {
-            pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+            pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
                 (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
                     WEBRTC_SPL_MUL_16_16((32 - w16_frac), pw16_expanded[i]) + 8),
                 5);
@@ -209,7 +209,7 @@
     }
     else if (inst->w16_mode==MODE_RFC3389CNG)
     { /* previous was RFC 3389 CNG...*/
-        WebRtc_Word16 pw16_CngInterp[32];
+        int16_t pw16_CngInterp[32];
         /* Reset mute factor and start up fresh */
         inst->w16_muteFactor = 16384;
         if (inst->CNG_Codec_inst != NULL)
@@ -238,7 +238,7 @@
         w16_frac = w16_inc;
         for (i = 0; i < 8 * fs_mult; i++)
         {
-            pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+            pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32(
                 (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
                     WEBRTC_SPL_MUL_16_16((32-w16_frac), pw16_CngInterp[i]) + 8),
                 5);
@@ -260,7 +260,7 @@
             /* scale with mute factor */
             w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
             /* shift 14 with proper rounding */
-            pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
+            pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
             /* increase mute_factor towards 16384 */
             inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
         }
diff --git a/modules/audio_coding/neteq/packet_buffer.c b/modules/audio_coding/neteq/packet_buffer.c
index c51805e..39f40ef 100644
--- a/modules/audio_coding/neteq/packet_buffer.c
+++ b/modules/audio_coding/neteq/packet_buffer.c
@@ -26,12 +26,12 @@
 #include <stdio.h>
 
 extern FILE *delay_fid2; /* file pointer to delay log file */
-extern WebRtc_UWord32 tot_received_packets;
+extern uint32_t tot_received_packets;
 #endif /* NETEQ_DELAY_LOGGING */
 
 
 int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
-                                 WebRtc_Word16 *pw16_memory, int memorySize)
+                                 int16_t *pw16_memory, int memorySize)
 {
     int i;
     int pos = 0;
@@ -45,11 +45,11 @@
     }
 
     /* Clear the buffer instance */
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) bufferInst, 0,
-        sizeof(PacketBuf_t) / sizeof(WebRtc_Word16));
+    WebRtcSpl_MemSetW16((int16_t*) bufferInst, 0,
+        sizeof(PacketBuf_t) / sizeof(int16_t));
 
     /* Clear the buffer memory */
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) pw16_memory, 0, memorySize);
+    WebRtcSpl_MemSetW16((int16_t*) pw16_memory, 0, memorySize);
 
     /* Set maximum number of packets */
     bufferInst->maxInsertPositions = maxNoOfPackets;
@@ -57,26 +57,26 @@
     /* Initialize array pointers */
     /* After each pointer has been set, the index pos is advanced to point immediately
      * after the the recently allocated vector. Note that one step for the pos index
-     * corresponds to a WebRtc_Word16.
+     * corresponds to a int16_t.
      */
 
-    bufferInst->timeStamp = (WebRtc_UWord32*) &pw16_memory[pos];
-    pos += maxNoOfPackets << 1; /* advance maxNoOfPackets * WebRtc_UWord32 */
+    bufferInst->timeStamp = (uint32_t*) &pw16_memory[pos];
+    pos += maxNoOfPackets << 1; /* advance maxNoOfPackets * uint32_t */
 
-    bufferInst->payloadLocation = (WebRtc_Word16**) &pw16_memory[pos];
-    pos += maxNoOfPackets * (sizeof(WebRtc_Word16*) / sizeof(WebRtc_Word16)); /* advance */
+    bufferInst->payloadLocation = (int16_t**) &pw16_memory[pos];
+    pos += maxNoOfPackets * (sizeof(int16_t*) / sizeof(int16_t)); /* advance */
 
-    bufferInst->seqNumber = (WebRtc_UWord16*) &pw16_memory[pos];
-    pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_UWord16 */
+    bufferInst->seqNumber = (uint16_t*) &pw16_memory[pos];
+    pos += maxNoOfPackets; /* advance maxNoOfPackets * uint16_t */
 
     bufferInst->payloadType = &pw16_memory[pos];
-    pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+    pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */
 
     bufferInst->payloadLengthBytes = &pw16_memory[pos];
-    pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+    pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */
 
     bufferInst->rcuPlCntr = &pw16_memory[pos];
-    pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+    pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */
 
     bufferInst->waitingTime = (int*) (&pw16_memory[pos]);
     /* Advance maxNoOfPackets * sizeof(waitingTime element). */
@@ -140,7 +140,7 @@
 
 
 int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
-                                   WebRtc_Word16 *flushed)
+                                   int16_t *flushed)
 {
     int nextPos;
     int i;
@@ -161,7 +161,7 @@
     }
 
     /* Sanity check for payload length
-     (payloadLen in bytes and memory size in WebRtc_Word16) */
+     (payloadLen in bytes and memory size in int16_t) */
     if ((RTPpacket->payloadLen > (bufferInst->memorySizeW16 << 1)) || (RTPpacket->payloadLen
         <= 0))
     {
@@ -184,7 +184,7 @@
         if (bufferInst->currentMemoryPos + ((RTPpacket->payloadLen + 1) >> 1)
             >= &bufferInst->startPayloadMemory[bufferInst->memorySizeW16])
         {
-            WebRtc_Word16 *tempMemAddress;
+            int16_t *tempMemAddress;
 
             /*
              * Payload does not fit at the end of the memory, put it in the beginning
@@ -323,13 +323,13 @@
     temp_var = NETEQ_DELAY_LOGGING_SIGNAL_RECIN;
     if ((fwrite(&temp_var, sizeof(int),
                 1, delay_fid2) != 1) ||
-        (fwrite(&RTPpacket->timeStamp, sizeof(WebRtc_UWord32),
+        (fwrite(&RTPpacket->timeStamp, sizeof(uint32_t),
                 1, delay_fid2) != 1) ||
-        (fwrite(&RTPpacket->seqNumber, sizeof(WebRtc_UWord16),
+        (fwrite(&RTPpacket->seqNumber, sizeof(uint16_t),
                 1, delay_fid2) != 1) ||
         (fwrite(&RTPpacket->payloadType, sizeof(int),
                 1, delay_fid2) != 1) ||
-        (fwrite(&RTPpacket->payloadLen, sizeof(WebRtc_Word16),
+        (fwrite(&RTPpacket->payloadLen, sizeof(int16_t),
                 1, delay_fid2) != 1)) {
       return -1;
     }
@@ -369,9 +369,9 @@
 
     /* Copy the actual data payload to RTP packet struct */
 
-    WEBRTC_SPL_MEMCPY_W16((WebRtc_Word16*) RTPpacket->payload,
+    WEBRTC_SPL_MEMCPY_W16((int16_t*) RTPpacket->payload,
         bufferInst->payloadLocation[bufferPosition],
-        (bufferInst->payloadLengthBytes[bufferPosition] + 1) >> 1); /*length in WebRtc_Word16*/
+        (bufferInst->payloadLengthBytes[bufferPosition] + 1) >> 1); /*length in int16_t*/
 
     /* Copy payload parameters */
     RTPpacket->payloadLen = bufferInst->payloadLengthBytes[bufferPosition];
@@ -503,14 +503,13 @@
     buffer_inst->payloadLengthBytes[buffer_pos]);
 }
 
-WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst,
-                                              const CodecDbInst_t*
-                                              codec_database) {
+int32_t WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst,
+                                        const CodecDbInst_t* codec_database) {
   int i, count;
   int last_duration;
   int last_codec_pos;
   int last_payload_type;
-  WebRtc_Word32 size_samples;
+  int32_t size_samples;
 
   count = 0;
   last_duration = buffer_inst->packSizeSamples;
@@ -583,9 +582,9 @@
 {
     int i;
     int ok = 0;
-    WebRtc_Word16 w16_tmp;
-    WebRtc_Word16 codecBytes;
-    WebRtc_Word16 codecBuffers;
+    int16_t w16_tmp;
+    int16_t codecBytes;
+    int16_t codecBuffers;
 
     /* Initialize return variables to zero */
     *maxBytes = 0;
@@ -786,12 +785,12 @@
      * Add size needed by the additional pointers for each slot inside struct,
      * as indicated on each line below.
      */
-    w16_tmp = (sizeof(WebRtc_UWord32) /* timeStamp */
-    + sizeof(WebRtc_Word16*) /* payloadLocation */
-    + sizeof(WebRtc_UWord16) /* seqNumber */
-    + sizeof(WebRtc_Word16)  /* payloadType */
-    + sizeof(WebRtc_Word16)  /* payloadLengthBytes */
-    + sizeof(WebRtc_Word16)  /* rcuPlCntr   */
+    w16_tmp = (sizeof(uint32_t) /* timeStamp */
+    + sizeof(int16_t*) /* payloadLocation */
+    + sizeof(uint16_t) /* seqNumber */
+    + sizeof(int16_t)  /* payloadType */
+    + sizeof(int16_t)  /* payloadLengthBytes */
+    + sizeof(int16_t)  /* rcuPlCntr   */
     + sizeof(int));          /* waitingTime */
     /* Add the extra size per slot to the memory count */
     *maxBytes += w16_tmp * (*maxSlots);
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index 1aa435f..afd74db 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -31,10 +31,10 @@
 {
 
     /* Variables common to the entire buffer */
-    WebRtc_UWord16 packSizeSamples; /* packet size in samples of last decoded packet */
-    WebRtc_Word16 *startPayloadMemory; /* pointer to the payload memory */
-    int memorySizeW16; /* the size (in WebRtc_Word16) of the payload memory */
-    WebRtc_Word16 *currentMemoryPos; /* The memory position to insert next payload */
+    uint16_t packSizeSamples; /* packet size in samples of last decoded packet */
+    int16_t *startPayloadMemory; /* pointer to the payload memory */
+    int memorySizeW16; /* the size (in int16_t) of the payload memory */
+    int16_t *currentMemoryPos; /* The memory position to insert next payload */
     int numPacketsInBuffer; /* The number of packets in the buffer */
     int insertPosition; /* The position to insert next packet */
     int maxInsertPositions; /* Maximum number of packets allowed */
@@ -42,18 +42,18 @@
     /* Arrays with one entry per packet slot */
     /* NOTE: If these are changed, the changes must be accounted for at the end of
      the function WebRtcNetEQ_GetDefaultCodecSettings(). */
-    WebRtc_UWord32 *timeStamp; /* Timestamp in slot n */
-    WebRtc_Word16 **payloadLocation; /* Memory location of payload in slot n */
-    WebRtc_UWord16 *seqNumber; /* Sequence number in slot n */
-    WebRtc_Word16 *payloadType; /* Payload type of packet in slot n */
-    WebRtc_Word16 *payloadLengthBytes; /* Payload length of packet in slot n */
-    WebRtc_Word16 *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload
+    uint32_t *timeStamp; /* Timestamp in slot n */
+    int16_t **payloadLocation; /* Memory location of payload in slot n */
+    uint16_t *seqNumber; /* Sequence number in slot n */
+    int16_t *payloadType; /* Payload type of packet in slot n */
+    int16_t *payloadLengthBytes; /* Payload length of packet in slot n */
+    int16_t *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload
      2 for redundant payload */
     int *waitingTime;
 
 
     /* Statistics counter */
-    WebRtc_UWord16 discardedPackets; /* Number of discarded packets */
+    uint16_t discardedPackets; /* Number of discarded packets */
 
 } PacketBuf_t;
 
@@ -70,7 +70,7 @@
  *		- bufferInst	: Buffer instance to be initialized
  *		- noOfPackets	: Maximum number of packets that buffer should hold
  *		- memory		: Pointer to the storage memory for the payloads
- *		- memorySize	: The size of the payload memory (in WebRtc_Word16)
+ *		- memorySize	: The size of the payload memory (in int16_t)
  *
  * Output:
  *      - bufferInst    : Updated buffer instance
@@ -80,7 +80,7 @@
  */
 
 int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
-                                 WebRtc_Word16 *pw16_memory, int memorySize);
+                                 int16_t *pw16_memory, int memorySize);
 
 /****************************************************************************
  * WebRtcNetEQ_PacketBufferFlush(...)
@@ -117,7 +117,7 @@
  */
 
 int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
-                                   WebRtc_Word16 *flushed);
+                                   int16_t *flushed);
 
 /****************************************************************************
  * WebRtcNetEQ_PacketBufferExtract(...)
@@ -208,9 +208,8 @@
  * Return value         : The buffer size in samples
  */
 
-WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst,
-                                              const CodecDbInst_t*
-                                              codec_database);
+int32_t WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst,
+                                        const CodecDbInst_t* codec_database);
 
 /****************************************************************************
  * WebRtcNetEQ_IncrementWaitingTimes(...)
diff --git a/modules/audio_coding/neteq/peak_detection.c b/modules/audio_coding/neteq/peak_detection.c
index 678c7f9..8c85d2a 100644
--- a/modules/audio_coding/neteq/peak_detection.c
+++ b/modules/audio_coding/neteq/peak_detection.c
@@ -17,7 +17,7 @@
 #include "signal_processing_library.h"
 
 /* Table of constants used in parabolic fit function WebRtcNetEQ_PrblFit */
-const WebRtc_Word16 WebRtcNetEQ_kPrblCf[17][3] = { { 120, 32, 64 }, { 140, 44, 75 },
+const int16_t WebRtcNetEQ_kPrblCf[17][3] = { { 120, 32, 64 }, { 140, 44, 75 },
                                                     { 150, 50, 80 }, { 160, 57, 85 },
                                                     { 180, 72, 96 }, { 200, 89, 107 },
                                                     { 210, 98, 112 }, { 220, 108, 117 },
@@ -27,17 +27,17 @@
                                                     { 330, 242, 176 }, { 340, 257, 181 },
                                                     { 360, 288, 192 } };
 
-WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
-                                        WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
-                                        WebRtc_Word16 *pw16_winIndex,
-                                        WebRtc_Word16 *pw16_winValue)
+int16_t WebRtcNetEQ_PeakDetection(int16_t *pw16_data, int16_t w16_dataLen,
+                                  int16_t w16_nmbPeaks, int16_t fs_mult,
+                                  int16_t *pw16_winIndex,
+                                  int16_t *pw16_winValue)
 {
     /* Local variables */
     int i;
-    WebRtc_Word16 w16_tmp;
-    WebRtc_Word16 w16_tmp2;
-    WebRtc_Word16 indMin = 0;
-    WebRtc_Word16 indMax = 0;
+    int16_t w16_tmp;
+    int16_t w16_tmp2;
+    int16_t indMin = 0;
+    int16_t indMax = 0;
 
     /* Peak detection */
 
@@ -53,7 +53,7 @@
             w16_dataLen++;
         }
 
-        pw16_winIndex[i] = WebRtcSpl_MaxIndexW16(pw16_data, (WebRtc_Word16) (w16_dataLen - 1));
+        pw16_winIndex[i] = WebRtcSpl_MaxIndexW16(pw16_data, (int16_t) (w16_dataLen - 1));
 
         if (i != w16_nmbPeaks - 1)
         {
@@ -103,14 +103,14 @@
     return 0;
 }
 
-WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
-                                  WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult)
+int16_t WebRtcNetEQ_PrblFit(int16_t *pw16_3pts, int16_t *pw16_Ind,
+                            int16_t *pw16_outVal, int16_t fs_mult)
 {
     /* Variables */
-    WebRtc_Word32 Num, Den;
-    WebRtc_Word32 temp;
-    WebRtc_Word16 flag, stp, strt, lmt;
-    WebRtc_UWord16 PFind[13];
+    int32_t Num, Den;
+    int32_t temp;
+    int16_t flag, stp, strt, lmt;
+    uint16_t PFind[13];
 
     if (fs_mult == 1)
     {
@@ -162,24 +162,24 @@
 
     Den = pw16_3pts[0] + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],-2) + pw16_3pts[2];
 
-    temp = (WebRtc_Word32) WEBRTC_SPL_MUL(Num, (WebRtc_Word32)120); /* need 32_16 really */
+    temp = (int32_t) WEBRTC_SPL_MUL(Num, (int32_t)120); /* need 32_16 really */
     flag = 1;
     stp = WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0] - WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0];
     strt = (WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0]
         + WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0]) >> 1;
 
-    if (temp < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)strt))
+    if (temp < (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)strt))
     {
         lmt = strt - stp;
         while (flag)
         {
             if ((flag == fs_mult) || (temp
-                > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
+                > (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)lmt)))
             {
                 *pw16_outVal
-                    = (WebRtc_Word16)
-                    (((WebRtc_Word32) ((WebRtc_Word32) WEBRTC_SPL_MUL(Den,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][1])
-                        + (WebRtc_Word32) WEBRTC_SPL_MUL(Num,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][2])
+                    = (int16_t)
+                    (((int32_t) ((int32_t) WEBRTC_SPL_MUL(Den,(int32_t)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][1])
+                        + (int32_t) WEBRTC_SPL_MUL(Num,(int32_t)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][2])
                         + WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256))) >> 8);
                 *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) - flag;
                 flag = 0;
@@ -191,24 +191,24 @@
             }
         }
     }
-    else if (temp > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)(strt+stp)))
+    else if (temp > (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)(strt+stp)))
     {
         lmt = strt + (stp << 1);
         while (flag)
         {
             if ((flag == fs_mult) || (temp
-                < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
+                < (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)lmt)))
             {
-                WebRtc_Word32 temp_term_1, temp_term_2, temp_term_3;
+                int32_t temp_term_1, temp_term_2, temp_term_3;
 
                 temp_term_1 = WEBRTC_SPL_MUL(Den,
-                    (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][1]);
+                    (int32_t) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][1]);
                 temp_term_2 = WEBRTC_SPL_MUL(Num,
-                    (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][2]);
+                    (int32_t) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][2]);
                 temp_term_3 = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256);
 
                 *pw16_outVal
-                    = (WebRtc_Word16) ((temp_term_1 + temp_term_2 + temp_term_3) >> 8);
+                    = (int16_t) ((temp_term_1 + temp_term_2 + temp_term_3) >> 8);
 
                 *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) + flag;
                 flag = 0;
diff --git a/modules/audio_coding/neteq/preemptive_expand.c b/modules/audio_coding/neteq/preemptive_expand.c
index 167bc3a..172a171 100644
--- a/modules/audio_coding/neteq/preemptive_expand.c
+++ b/modules/audio_coding/neteq/preemptive_expand.c
@@ -28,9 +28,9 @@
 /* Scratch usage:
 
  Type             Name                 size            startpos         endpos
- WebRtc_Word16    pw16_downSampSpeech  110             0                109
- WebRtc_Word32    pw32_corr            2*50            110              209
- WebRtc_Word16    pw16_corr            50              0                49
+ int16_t    pw16_downSampSpeech  110             0                109
+ int32_t    pw32_corr            2*50            110              209
+ int16_t    pw16_corr            50              0                49
 
  Total: 110+2*50
  */
@@ -72,45 +72,45 @@
 
 int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
 #ifdef SCRATCH
-                                 WebRtc_Word16 *pw16_scratchPtr,
+                                 int16_t *pw16_scratchPtr,
 #endif
-                                 const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
-                                 WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
-                                 WebRtc_Word16 BGNonly)
+                                 const int16_t *pw16_decoded, int len, int oldDataLen,
+                                 int16_t *pw16_outData, int16_t *pw16_len,
+                                 int16_t BGNonly)
 {
 
 #ifdef SCRATCH
     /* Use scratch memory for internal temporary vectors */
-    WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
-    WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
-    WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
+    int16_t *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
+    int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
+    int16_t *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
 #else
     /* Allocate memory for temporary vectors */
-    WebRtc_Word16 pw16_downSampSpeech[PREEMPTIVE_DOWNSAMPLED_LEN];
-    WebRtc_Word32 pw32_corr[PREEMPTIVE_CORR_LEN];
-    WebRtc_Word16 pw16_corr[PREEMPTIVE_CORR_LEN];
+    int16_t pw16_downSampSpeech[PREEMPTIVE_DOWNSAMPLED_LEN];
+    int32_t pw32_corr[PREEMPTIVE_CORR_LEN];
+    int16_t pw16_corr[PREEMPTIVE_CORR_LEN];
 #endif
-    WebRtc_Word16 w16_decodedMax = 0;
-    WebRtc_Word16 w16_tmp = 0;
-    WebRtc_Word16 w16_tmp2;
-    WebRtc_Word32 w32_tmp;
-    WebRtc_Word32 w32_tmp2;
+    int16_t w16_decodedMax = 0;
+    int16_t w16_tmp = 0;
+    int16_t w16_tmp2;
+    int32_t w32_tmp;
+    int32_t w32_tmp2;
 
-    const WebRtc_Word16 w16_startLag = PREEMPTIVE_MIN_LAG;
-    const WebRtc_Word16 w16_endLag = PREEMPTIVE_MAX_LAG;
-    const WebRtc_Word16 w16_corrLen = PREEMPTIVE_CORR_LEN;
-    const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
-    WebRtc_Word16 *pw16_vectmp;
-    WebRtc_Word16 w16_inc, w16_startfact;
-    WebRtc_Word16 w16_bestIndex, w16_bestVal;
-    WebRtc_Word16 w16_VAD = 1;
-    WebRtc_Word16 fsMult;
-    WebRtc_Word16 fsMult120;
-    WebRtc_Word32 w32_en1, w32_en2, w32_cc;
-    WebRtc_Word16 w16_en1, w16_en2;
-    WebRtc_Word16 w16_en1Scale, w16_en2Scale;
-    WebRtc_Word16 w16_sqrtEn1En2;
-    WebRtc_Word16 w16_bestCorr = 0;
+    const int16_t w16_startLag = PREEMPTIVE_MIN_LAG;
+    const int16_t w16_endLag = PREEMPTIVE_MAX_LAG;
+    const int16_t w16_corrLen = PREEMPTIVE_CORR_LEN;
+    const int16_t *pw16_vec1, *pw16_vec2;
+    int16_t *pw16_vectmp;
+    int16_t w16_inc, w16_startfact;
+    int16_t w16_bestIndex, w16_bestVal;
+    int16_t w16_VAD = 1;
+    int16_t fsMult;
+    int16_t fsMult120;
+    int32_t w32_en1, w32_en2, w32_cc;
+    int16_t w16_en1, w16_en2;
+    int16_t w16_en1Scale, w16_en2Scale;
+    int16_t w16_sqrtEn1En2;
+    int16_t w16_bestCorr = 0;
     int ok;
 
 #ifdef NETEQ_STEREO
@@ -120,7 +120,7 @@
     fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
 
     /* Pre-calculate common multiplication with fsMult */
-    fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
+    fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
 
     inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
 
@@ -128,7 +128,7 @@
      * Sanity check for len variable; must be (almost) 30 ms (120*fsMult + max(bestIndex)).
      * Also, the new part must be at least .625 ms (w16_overlap).
      */
-    if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult) || oldDataLen >= len
+    if (len < (int16_t) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult) || oldDataLen >= len
         - inst->ExpandInst.w16_overlap)
     {
         /* Length of decoded data too short */
@@ -138,7 +138,7 @@
         
         /* simply move all data from decoded to outData */
 
-        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
         return NETEQ_OTHER_ERROR;
     }
@@ -162,7 +162,7 @@
 
             /* simply move all data from decoded to outData */
 
-            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
             return NETEQ_OTHER_ERROR;
         }
@@ -207,7 +207,7 @@
         /****************************************************************/
 
         /* find maximum absolute value */
-        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+        w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len);
 
         /* downsample the decoded speech to 4 kHz */
         ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
@@ -221,7 +221,7 @@
 
             /* simply move all data from decoded to outData */
 
-            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+            WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
             return NETEQ_OTHER_ERROR;
         }
@@ -236,9 +236,9 @@
         /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */WebRtcNetEQ_CrossCorr(
             pw32_corr, &pw16_downSampSpeech[w16_endLag],
             &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
-            (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
+            (int16_t) (w16_endLag - w16_startLag), w16_tmp, -1);
 
-        /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+        /* Normalize correlation to 14 bits and put in a int16_t vector */
         w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
         w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
         w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
@@ -247,8 +247,8 @@
 
         /* Find limits for peak finding, in order to avoid overful NetEQ algorithm buffer. */
         /* Calculate difference between MAX_OUTPUT_SIZE and len in 4 kHz domain. */
-        w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) (NETEQ_MAX_OUTPUT_SIZE - len),
-            (WebRtc_Word16) (fsMult << 1)) - w16_startLag;
+        w16_tmp = WebRtcSpl_DivW32W16ResW16((int32_t) (NETEQ_MAX_OUTPUT_SIZE - len),
+            (int16_t) (fsMult << 1)) - w16_startLag;
         w16_tmp = WEBRTC_SPL_MIN(w16_corrLen, w16_tmp); /* no more than corrLen = 50 */
 
 #ifdef NETEQ_STEREO
@@ -326,13 +326,13 @@
         pw16_vec2 = &pw16_decoded[fsMult120];
 
         /* Calculate energies for vec1 and vec2 */
-        w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
-            (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
-        w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
-            (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
+        w32_en1 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1,
+            (int16_t*) pw16_vec1, w16_bestIndex, w16_tmp);
+        w32_en2 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec2,
+            (int16_t*) pw16_vec2, w16_bestIndex, w16_tmp);
 
         /* Calculate cross-correlation at the found lag */
-        w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
+        w32_cc = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, (int16_t*) pw16_vec2,
             w16_bestIndex, w16_tmp);
 
         /* Check VAD constraint 
@@ -350,7 +350,7 @@
         w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
         w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
         w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
-        w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
+        w16_tmp2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
         w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
 
         /* Scale w32_tmp properly before comparing with w32_tmp2 */
@@ -358,7 +358,7 @@
         if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
         {
             /* Cannot scale only w32_tmp, must scale w32_temp2 too */
-            WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
+            int16_t tempshift = WebRtcSpl_NormW32(w32_tmp);
             w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
             w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
                 WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
@@ -398,21 +398,21 @@
                 w16_en1Scale += 1;
             }
 
-            /* Convert energies to WebRtc_Word16 */
-            w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
-            w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+            /* Convert energies to int16_t */
+            w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+            w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
 
             /* Calculate energy product */
             w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
 
             /* Calculate square-root of energy product */
-            w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
+            w16_sqrtEn1En2 = (int16_t) WebRtcSpl_SqrtFloor(w32_tmp);
 
             /* Calculate cc/sqrt(en1*en2) in Q14 */
             w16_tmp = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
             w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
             w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
-            w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
+            w16_bestCorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
             w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
         }
 
@@ -440,14 +440,14 @@
         /* Do expand operation by overlap add */
 
         /* Set length of the first part, not to be modified */
-        WebRtc_Word16 w16_startIndex = WEBRTC_SPL_MAX(oldDataLen, fsMult120);
+        int16_t w16_startIndex = WEBRTC_SPL_MAX(oldDataLen, fsMult120);
 
         /*
          * Calculate cross-fading slope so that the fading factor goes from
          * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
          */
-        w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
-            (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
+        w16_inc = (int16_t) WebRtcSpl_DivW32W16((int32_t) 16384,
+            (int16_t) (w16_bestIndex + 1)); /* in Q14 */
 
         /* Initiate fading factor */
         w16_startfact = 16384 - w16_inc;
@@ -465,14 +465,14 @@
         /* Generate interpolated part of length bestIndex (1 pitch period) */
         pw16_vectmp = pw16_outData + w16_startIndex;
         /* Reuse mixing function from Expand */
-        WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec2,
-            (WebRtc_Word16*) pw16_vec1, &w16_startfact, w16_inc, w16_bestIndex);
+        WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (int16_t*) pw16_vec2,
+            (int16_t*) pw16_vec1, &w16_startfact, w16_inc, w16_bestIndex);
 
         /* Move the last part (also unmodified) */
         /* Take from decoded at 15 ms */
         pw16_vec2 = &pw16_decoded[w16_startIndex];
         WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[w16_startIndex + w16_bestIndex], pw16_vec2,
-            (WebRtc_Word16) (len - w16_startIndex));
+            (int16_t) (len - w16_startIndex));
 
         /* Set the mode flag */
         if (w16_VAD)
@@ -513,7 +513,7 @@
 
         /* Simply move all data from decoded to outData */
 
-        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+        WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len);
 
         return 0;
     }
diff --git a/modules/audio_coding/neteq/random_vector.c b/modules/audio_coding/neteq/random_vector.c
index 217bacd..c168ab5 100644
--- a/modules/audio_coding/neteq/random_vector.c
+++ b/modules/audio_coding/neteq/random_vector.c
@@ -18,7 +18,7 @@
  * Values are normalized so that
  * sqrt(dot(pw16_NETEQFIX_RANDN_TBL,pw16_NETEQFIX_RANDN_TBL)/256)=2^13
  */
-const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[RANDVEC_NO_OF_SAMPLES] =
+const int16_t WebRtcNetEQ_kRandnTbl[RANDVEC_NO_OF_SAMPLES] =
 {
 	2680, 5532, 441, 5520, 16170, -5146, -1024, -8733, 3115, 9598, -10380, -4959, -1280, -21716, 7133, -1522, 
 	13458, -3902, 2789, -675, 3441, 5016, -13599, -4003, -2739, 3922, -7209, 13352, -11617, -7241, 12905, -2314, 
@@ -39,15 +39,15 @@
 };
 
 
-void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
-                           WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval)
+void WebRtcNetEQ_RandomVec(uint32_t *w32_seed, int16_t *pw16_randVec,
+                           int16_t w16_len, int16_t w16_incval)
 {
     int i;
-    WebRtc_Word16 w16_pos;
+    int16_t w16_pos;
     for (i = 0; i < w16_len; i++)
     {
         *w32_seed = (*w32_seed) + w16_incval;
-        w16_pos = (WebRtc_Word16) ((*w32_seed) & (RANDVEC_NO_OF_SAMPLES - 1));
+        w16_pos = (int16_t) ((*w32_seed) & (RANDVEC_NO_OF_SAMPLES - 1));
         pw16_randVec[i] = WebRtcNetEQ_kRandnTbl[w16_pos];
     }
 }
diff --git a/modules/audio_coding/neteq/recin.c b/modules/audio_coding/neteq/recin.c
index c2f0d2d..75733b2 100644
--- a/modules/audio_coding/neteq/recin.c
+++ b/modules/audio_coding/neteq/recin.c
@@ -26,16 +26,16 @@
 
 
 int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
-                              WebRtc_UWord32 uw32_timeRec)
+                              uint32_t uw32_timeRec)
 {
     RTPPacket_t RTPpacket[2];
     int i_k;
     int i_ok = 0, i_No_Of_Payloads = 1;
-    WebRtc_Word16 flushed = 0;
-    WebRtc_Word16 codecPos;
+    int16_t flushed = 0;
+    int16_t codecPos;
     int curr_Codec;
-    WebRtc_Word16 isREDPayload = 0;
-    WebRtc_Word32 temp_bufsize;
+    int16_t isREDPayload = 0;
+    int32_t temp_bufsize;
 #ifdef NETEQ_RED_CODEC
     RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */
     RTPpacketPtr[0] = &RTPpacket[0];
@@ -158,7 +158,7 @@
         /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */
         if (MCU_inst->TSscalingInitialized == 1)
         {
-            WebRtc_UWord32 newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst,
+            uint32_t newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst,
                 RTPpacket[i_k].timeStamp);
 
             /* save the incoming timestamp for next time */
@@ -202,7 +202,7 @@
             /* Is this a CNG packet? how should we handle this?*/
 #ifdef NETEQ_CNG_CODEC
             /* Get CNG sample rate */
-            WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
+            uint16_t fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
                 RTPpacket[i_k].payloadType);
 
             /* Force sampling frequency to 32000 Hz CNG 48000 Hz. */
@@ -301,9 +301,9 @@
 
             MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos](
                 MCU_inst->codec_DB_inst.codec_state[codecPos],
-                (G_CONST WebRtc_UWord16 *) RTPpacket[0].payload,
-                (WebRtc_Word32) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber,
-                (WebRtc_UWord32) RTPpacket[0].timeStamp, (WebRtc_UWord32) uw32_timeRec);
+                (G_CONST uint16_t *) RTPpacket[0].payload,
+                (int32_t) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber,
+                (uint32_t) RTPpacket[0].timeStamp, (uint32_t) uw32_timeRec);
         }
     }
 
@@ -320,11 +320,11 @@
         {
             /* Change the auto-mode parameters if packet length has changed */
             WebRtcNetEQ_SetPacketSpeechLen(&(MCU_inst->BufferStat_inst.Automode_inst),
-                (WebRtc_Word16) temp_bufsize, MCU_inst->fs);
+                (int16_t) temp_bufsize, MCU_inst->fs);
         }
 
         /* update statistics */
-        if ((WebRtc_Word32) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0
+        if ((int32_t) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0
             && !MCU_inst->new_codec)
         {
             /*
@@ -395,7 +395,7 @@
         {
             /* TODO(tlegrand): remove scaling once ACM has full 48 kHz
              * support. */
-            WebRtc_UWord16 sample_freq =
+            uint16_t sample_freq =
                 WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
                                             rtpPayloadType);
             if (sample_freq == 48000) {
@@ -416,11 +416,11 @@
     return 0;
 }
 
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
-                                                            WebRtc_UWord32 externalTS)
+uint32_t WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
+                                                      uint32_t externalTS)
 {
-    WebRtc_Word32 timestampDiff;
-    WebRtc_UWord32 internalTS;
+    int32_t timestampDiff;
+    uint32_t internalTS;
 
     /* difference between this and last incoming timestamp */
     timestampDiff = externalTS - MCU_inst->externalTS;
@@ -459,14 +459,14 @@
     return internalTS;
 }
 
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
-                                                            WebRtc_UWord32 internalTS)
+uint32_t WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
+                                                      uint32_t internalTS)
 {
-    WebRtc_Word32 timestampDiff;
-    WebRtc_UWord32 externalTS;
+    int32_t timestampDiff;
+    uint32_t externalTS;
 
     /* difference between this and last incoming timestamp */
-    timestampDiff = (WebRtc_Word32) internalTS - MCU_inst->internalTS;
+    timestampDiff = (int32_t) internalTS - MCU_inst->internalTS;
 
     switch (MCU_inst->scalingFactor)
     {
diff --git a/modules/audio_coding/neteq/recout.c b/modules/audio_coding/neteq/recout.c
index 8eb49e3..63abbd1 100644
--- a/modules/audio_coding/neteq/recout.c
+++ b/modules/audio_coding/neteq/recout.c
@@ -41,7 +41,7 @@
 /* Scratch usage:
 
  Type           Name                            size             startpos      endpos
- WebRtc_Word16  pw16_NetEqAlgorithm_buffer      1080*fs/8000     0             1080*fs/8000-1
+ int16_t  pw16_NetEqAlgorithm_buffer      1080*fs/8000     0             1080*fs/8000-1
  struct         dspInfo                         6                1080*fs/8000  1085*fs/8000
 
  func           WebRtcNetEQ_Normal              40+495*fs/8000   0             39+495*fs/8000
@@ -91,49 +91,49 @@
 
 #ifdef NETEQ_DELAY_LOGGING
 extern FILE *delay_fid2; /* file pointer to delay log file */
-extern WebRtc_UWord32 tot_received_packets;
+extern uint32_t tot_received_packets;
 #endif
 
 
-int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
-                               WebRtc_Word16 *pw16_len, WebRtc_Word16 BGNonly)
+int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, int16_t *pw16_outData,
+                               int16_t *pw16_len, int16_t BGNonly)
 {
 
-    WebRtc_Word16 blockLen, payloadLen, len = 0, pos;
-    WebRtc_Word16 w16_tmp1, w16_tmp2, w16_tmp3, DataEnough;
-    WebRtc_Word16 *blockPtr;
-    WebRtc_Word16 MD = 0;
+    int16_t blockLen, payloadLen, len = 0, pos;
+    int16_t w16_tmp1, w16_tmp2, w16_tmp3, DataEnough;
+    int16_t *blockPtr;
+    int16_t MD = 0;
 
-    WebRtc_Word16 speechType = TYPE_SPEECH;
-    WebRtc_UWord16 instr;
-    WebRtc_UWord16 uw16_tmp;
+    int16_t speechType = TYPE_SPEECH;
+    uint16_t instr;
+    uint16_t uw16_tmp;
 #ifdef SCRATCH
     char pw8_ScratchBuffer[((SIZE_SCRATCH_BUFFER + 1) * 2)];
-    WebRtc_Word16 *pw16_scratchPtr = (WebRtc_Word16*) pw8_ScratchBuffer;
+    int16_t *pw16_scratchPtr = (int16_t*) pw8_ScratchBuffer;
     /* pad with 240*fs_mult to match the overflow guard below */
-    WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
-    WebRtc_Word16 *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr
+    int16_t pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
+    int16_t *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr
         + SCRATCH_ALGORITHM_BUFFER;
     DSP2MCU_info_t *dspInfo = (DSP2MCU_info_t*) (pw16_scratchPtr + SCRATCH_DSP_INFO);
 #else
     /* pad with 240*fs_mult to match the overflow guard below */
-    WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
-    WebRtc_Word16 pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE+240*6];
+    int16_t pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
+    int16_t pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE+240*6];
     DSP2MCU_info_t dspInfoStruct;
     DSP2MCU_info_t *dspInfo = &dspInfoStruct;
 #endif
-    WebRtc_Word16 fs_mult;
+    int16_t fs_mult;
     int borrowedSamples;
     int oldBorrowedSamples;
     int return_value = 0;
-    WebRtc_Word16 lastModeBGNonly = (inst->w16_mode & MODE_BGN_ONLY) != 0; /* check BGN flag */
+    int16_t lastModeBGNonly = (inst->w16_mode & MODE_BGN_ONLY) != 0; /* check BGN flag */
     void *mainInstBackup = inst->main_inst;
 
 #ifdef NETEQ_DELAY_LOGGING
     int temp_var;
 #endif
-    WebRtc_Word16 dtmfValue = -1;
-    WebRtc_Word16 dtmfVolume = -1;
+    int16_t dtmfValue = -1;
+    int16_t dtmfVolume = -1;
     int playDtmf = 0;
 #ifdef NETEQ_ATEVENT_DECODE
     int dtmfSwitch = 0;
@@ -141,7 +141,7 @@
 #ifdef NETEQ_STEREO
     MasterSlaveInfo *msInfo = inst->msInfo;
 #endif
-    WebRtc_Word16 *sharedMem = pw16_NetEqAlgorithm_buffer; /* Reuse memory SHARED_MEM_SIZE size */
+    int16_t *sharedMem = pw16_NetEqAlgorithm_buffer; /* Reuse memory SHARED_MEM_SIZE size */
     inst->pw16_readAddress = sharedMem;
     inst->pw16_writeAddress = sharedMem;
 
@@ -164,8 +164,8 @@
          */
 
         /* Get the information from master to correct synchronization */
-        WebRtc_UWord32 currentMasterTimestamp;
-        WebRtc_UWord32 currentSlaveTimestamp;
+        uint32_t currentMasterTimestamp;
+        uint32_t currentSlaveTimestamp;
 
         currentMasterTimestamp = msInfo->endTimestamp - msInfo->samplesLeftWithOverlap;
         currentSlaveTimestamp = inst->endTimestamp - (inst->endPosition - inst->curPosition);
@@ -244,8 +244,8 @@
     if (msInfo->msMode == NETEQ_MASTER)
     {
         /* clear info to slave */
-        WebRtcSpl_MemSetW16((WebRtc_Word16 *) msInfo, 0,
-            sizeof(MasterSlaveInfo) / sizeof(WebRtc_Word16));
+        WebRtcSpl_MemSetW16((int16_t *) msInfo, 0,
+            sizeof(MasterSlaveInfo) / sizeof(int16_t));
         /* re-set mode */
         msInfo->msMode = NETEQ_MASTER;
 
@@ -263,7 +263,7 @@
     return_value = WebRtcNetEQ_DSP2MCUinterrupt((MainInst_t *) inst->main_inst, sharedMem);
 
     /* Read MCU data and instructions */
-    instr = (WebRtc_UWord16) (inst->pw16_readAddress[0] & 0xf000);
+    instr = (uint16_t) (inst->pw16_readAddress[0] & 0xf000);
 
 #ifdef NETEQ_STEREO
     if (msInfo->msMode == NETEQ_MASTER)
@@ -303,7 +303,7 @@
 #endif
     }
 
-    blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of WebRtc_Word16 */
+    blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of int16_t */
     payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
     blockPtr++;
 
@@ -322,7 +322,7 @@
             temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS;
             if ((fwrite(&temp_var, sizeof(int),
                         1, delay_fid2) != 1) ||
-                (fwrite(&inst->fs, sizeof(WebRtc_UWord16),
+                (fwrite(&inst->fs, sizeof(uint16_t),
                         1, delay_fid2) != 1)) {
               return -1;
             }
@@ -410,7 +410,7 @@
         {
             if (inst->codec_ptr_inst.funcDecode != NULL)
             {
-                WebRtc_Word16 dec_Len;
+                int16_t dec_Len;
                 if (!BGNonly)
                 {
                     /* Do decoding as normal
@@ -517,11 +517,11 @@
             if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
               return -1;
             }
-            if (fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32),
+            if (fwrite(&inst->endTimestamp, sizeof(uint32_t),
                        1, delay_fid2) != 1) {
               return -1;
             }
-            if (fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16),
+            if (fwrite(&dspInfo->samplesLeft, sizeof(uint16_t),
                        1, delay_fid2) != 1) {
               return -1;
             }
@@ -625,7 +625,7 @@
                     /* call VAD with new decoded data */
                     inst->VADInst.VADDecision |= inst->VADInst.VADFunction(
                         inst->VADInst.VADState, (int) inst->fs,
-                        (WebRtc_Word16 *) &pw16_decoded_buffer[VADSamplePtr],
+                        (int16_t *) &pw16_decoded_buffer[VADSamplePtr],
                         (VADframeSize * fs_mult * 8));
 
                     VADSamplePtr += VADframeSize * fs_mult * 8; /* increment sample counter */
@@ -642,9 +642,9 @@
 #endif /* NETEQ_VAD */
 
     /* Adjust timestamp if needed */
-    uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[1];
-    inst->endTimestamp += (((WebRtc_UWord32) uw16_tmp) << 16);
-    uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[2];
+    uw16_tmp = (uint16_t) inst->pw16_readAddress[1];
+    inst->endTimestamp += (((uint32_t) uw16_tmp) << 16);
+    uw16_tmp = (uint16_t) inst->pw16_readAddress[2];
     inst->endTimestamp += uw16_tmp;
 
     if (BGNonly && len > 0)
@@ -907,7 +907,7 @@
 #ifdef NETEQ_CNG_CODEC
             if (blockLen > 0)
             {
-                if (WebRtcCng_UpdateSid(inst->CNG_Codec_inst, (WebRtc_UWord8*) blockPtr,
+                if (WebRtcCng_UpdateSid(inst->CNG_Codec_inst, (uint8_t*) blockPtr,
                     payloadLen) < 0)
                 {
                     /* error returned from CNG function */
@@ -1022,7 +1022,7 @@
                  * Generate extra DTMF data to fill the space between
                  * curPosition and endPosition
                  */
-                WebRtc_Word16 tempLen;
+                int16_t tempLen;
 
                 tempLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
                     &pw16_NetEqAlgorithm_buffer[len], inst->fs,
@@ -1269,7 +1269,7 @@
             {
                 inst->speechBuffer[inst->curPosition + pos]
                     =
-                    (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+                    (int16_t) WEBRTC_SPL_RSHIFT_W32(
                         WEBRTC_SPL_MUL_16_16( inst->speechBuffer[inst->endPosition - w16_tmp3 + pos],
                             16384-w16_tmp1 ) +
                         WEBRTC_SPL_MUL_16_16( pw16_NetEqAlgorithm_buffer[pos], w16_tmp1 ),
@@ -1336,9 +1336,9 @@
     if (playDtmf != 0)
     {
 #ifdef NETEQ_ATEVENT_DECODE
-        WebRtc_Word16 outDataIndex = 0;
-        WebRtc_Word16 overdubLen = -1; /* default len */
-        WebRtc_Word16 dtmfLen;
+        int16_t outDataIndex = 0;
+        int16_t overdubLen = -1; /* default len */
+        int16_t dtmfLen;
 
         /*
          * Overdub the output with DTMF. Note that this is not executed if the
@@ -1455,9 +1455,9 @@
      */
     if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_RFC3389CNG))
     {
-        WebRtc_UWord32 uw32_tmpTS;
+        uint32_t uw32_tmpTS;
         uw32_tmpTS = inst->endTimestamp - (inst->endPosition - inst->curPosition);
-        if ((WebRtc_Word32) (uw32_tmpTS - inst->videoSyncTimestamp) > 0)
+        if ((int32_t) (uw32_tmpTS - inst->videoSyncTimestamp) > 0)
         {
             inst->videoSyncTimestamp = uw32_tmpTS;
         }
diff --git a/modules/audio_coding/neteq/rtcp.c b/modules/audio_coding/neteq/rtcp.c
index 35f73da..d1ce934 100644
--- a/modules/audio_coding/neteq/rtcp.c
+++ b/modules/audio_coding/neteq/rtcp.c
@@ -18,24 +18,24 @@
 
 #include "signal_processing_library.h"
 
-int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo)
+int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo)
 {
     /*
      * Initialize everything to zero and then set the start values for the RTP packet stream.
      */
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) RTCP_inst, 0,
-        sizeof(WebRtcNetEQ_RTCP_t) / sizeof(WebRtc_Word16));
+    WebRtcSpl_MemSetW16((int16_t*) RTCP_inst, 0,
+        sizeof(WebRtcNetEQ_RTCP_t) / sizeof(int16_t));
     RTCP_inst->base_seq = uw16_seqNo;
     RTCP_inst->max_seq = uw16_seqNo;
     return 0;
 }
 
-int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
-                           WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime)
+int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo,
+                           uint32_t uw32_timeStamp, uint32_t uw32_recTime)
 {
-    WebRtc_Word16 w16_SeqDiff;
-    WebRtc_Word32 w32_TimeDiff;
-    WebRtc_Word32 w32_JitterDiff;
+    int16_t w16_SeqDiff;
+    int32_t w32_TimeDiff;
+    int32_t w32_JitterDiff;
 
     /*
      * Update number of received packets, and largest packet number received.
@@ -66,16 +66,16 @@
 }
 
 int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
-                             WebRtc_UWord16 *puw16_fraction_lost,
-                             WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
-                             WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset)
+                             uint16_t *puw16_fraction_lost,
+                             uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max,
+                             uint32_t *puw32_jitter, int16_t doNotReset)
 {
-    WebRtc_UWord32 uw32_exp_nr, uw32_exp_interval, uw32_rec_interval;
-    WebRtc_Word32 w32_lost;
+    uint32_t uw32_exp_nr, uw32_exp_interval, uw32_rec_interval;
+    int32_t w32_lost;
 
     /* Extended highest sequence number received */
     *puw32_ext_max
-        = (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)RTCP_inst->cycles, 16)
+        = (uint32_t) WEBRTC_SPL_LSHIFT_W32((uint32_t)RTCP_inst->cycles, 16)
             + RTCP_inst->max_seq;
 
     /*
@@ -91,7 +91,7 @@
     else if (uw32_exp_nr > RTCP_inst->received)
     {
         *puw32_cum_lost = uw32_exp_nr - RTCP_inst->received;
-        if (*puw32_cum_lost > (WebRtc_UWord32) 0xFFFFFF)
+        if (*puw32_cum_lost > (uint32_t) 0xFFFFFF)
         {
             *puw32_cum_lost = 0xFFFFFF;
         }
@@ -112,14 +112,14 @@
     {
         RTCP_inst->rec_prior = RTCP_inst->received;
     }
-    w32_lost = (WebRtc_Word32) (uw32_exp_interval - uw32_rec_interval);
+    w32_lost = (int32_t) (uw32_exp_interval - uw32_rec_interval);
     if (uw32_exp_interval == 0 || w32_lost <= 0 || RTCP_inst->received == 0)
     {
         *puw16_fraction_lost = 0;
     }
     else
     {
-        *puw16_fraction_lost = (WebRtc_UWord16) (WEBRTC_SPL_LSHIFT_W32(w32_lost, 8)
+        *puw16_fraction_lost = (uint16_t) (WEBRTC_SPL_LSHIFT_W32(w32_lost, 8)
             / uw32_exp_interval);
     }
     if (*puw16_fraction_lost > 0xFF)
diff --git a/modules/audio_coding/neteq/rtcp.h b/modules/audio_coding/neteq/rtcp.h
index 009e019..5e066eb 100644
--- a/modules/audio_coding/neteq/rtcp.h
+++ b/modules/audio_coding/neteq/rtcp.h
@@ -19,16 +19,16 @@
 
 typedef struct
 {
-    WebRtc_UWord16 cycles; /* The number of wrap-arounds for the sequence number */
-    WebRtc_UWord16 max_seq; /* The maximum sequence number received
+    uint16_t cycles; /* The number of wrap-arounds for the sequence number */
+    uint16_t max_seq; /* The maximum sequence number received
      (starts from 0 again after wrap around) */
-    WebRtc_UWord16 base_seq; /* The sequence number of the first packet that arrived */
-    WebRtc_UWord32 received; /* The number of packets that has been received */
-    WebRtc_UWord32 rec_prior; /* Number of packets received when last report was generated */
-    WebRtc_UWord32 exp_prior; /* Number of packets that should have been received if no
+    uint16_t base_seq; /* The sequence number of the first packet that arrived */
+    uint32_t received; /* The number of packets that has been received */
+    uint32_t rec_prior; /* Number of packets received when last report was generated */
+    uint32_t exp_prior; /* Number of packets that should have been received if no
      packets were lost. Stored value from last report. */
-    WebRtc_UWord32 jitter; /* Jitter statistics at this instance (calculated according to RFC) */
-    WebRtc_Word32 transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */
+    uint32_t jitter; /* Jitter statistics at this instance (calculated according to RFC) */
+    int32_t transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */
 } WebRtcNetEQ_RTCP_t;
 
 /****************************************************************************
@@ -46,7 +46,7 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo);
+int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo);
 
 /****************************************************************************
  * WebRtcNetEQ_RTCPUpdate(...)
@@ -65,8 +65,8 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
-                           WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime);
+int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo,
+                           uint32_t uw32_timeStamp, uint32_t uw32_recTime);
 
 /****************************************************************************
  * WebRtcNetEQ_RTCPGetStats(...)
@@ -95,8 +95,8 @@
  */
 
 int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
-                             WebRtc_UWord16 *puw16_fraction_lost,
-                             WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
-                             WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset);
+                             uint16_t *puw16_fraction_lost,
+                             uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max,
+                             uint32_t *puw32_jitter, int16_t doNotReset);
 
 #endif
diff --git a/modules/audio_coding/neteq/rtp.c b/modules/audio_coding/neteq/rtp.c
index 63cdf65..f23f351 100644
--- a/modules/audio_coding/neteq/rtp.c
+++ b/modules/audio_coding/neteq/rtp.c
@@ -18,7 +18,7 @@
 
 #include "neteq_error_codes.h"
 
-int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
+int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen,
                                RTPPacket_t* RTPheader)
 {
     int i_P, i_X, i_CC, i_startPosition;
@@ -32,20 +32,20 @@
     }
 
 #ifdef WEBRTC_BIG_ENDIAN
-    i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */
-    i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */
-    i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */
-    i_CC = ((WebRtc_UWord16) (pw16_Datagram[0] >> 8) & 0xF); /* Get the CC number */
+    i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */
+    i_P = (((uint16_t) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */
+    i_X = (((uint16_t) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */
+    i_CC = ((uint16_t) (pw16_Datagram[0] >> 8) & 0xF); /* Get the CC number */
     RTPheader->payloadType = pw16_Datagram[0] & 0x7F; /* Get the coder type	*/
     RTPheader->seqNumber = pw16_Datagram[1]; /* Get the sequence number	*/
-    RTPheader->timeStamp = ((((WebRtc_UWord32) ((WebRtc_UWord16) pw16_Datagram[2])) << 16)
-        | (WebRtc_UWord16) (pw16_Datagram[3])); /* Get timestamp */
-    RTPheader->ssrc = (((WebRtc_UWord32) pw16_Datagram[4]) << 16)
-        + (((WebRtc_UWord32) pw16_Datagram[5])); /* Get the SSRC */
+    RTPheader->timeStamp = ((((uint32_t) ((uint16_t) pw16_Datagram[2])) << 16)
+        | (uint16_t) (pw16_Datagram[3])); /* Get timestamp */
+    RTPheader->ssrc = (((uint32_t) pw16_Datagram[4]) << 16)
+        + (((uint32_t) pw16_Datagram[5])); /* Get the SSRC */
 
     if (i_X == 1)
     {
-        /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
+        /* Extension header exists. Find out how many int32_t it consists of. */
         i_extlength = pw16_Datagram[7 + 2 * i_CC];
     }
     if (i_P == 1)
@@ -54,7 +54,7 @@
         if (i_DatagramLen & 0x1)
         {
             /* odd number of bytes => last byte in higher byte */
-            i_padlength = (((WebRtc_UWord16) pw16_Datagram[i_DatagramLen >> 1]) >> 8);
+            i_padlength = (((uint16_t) pw16_Datagram[i_DatagramLen >> 1]) >> 8);
         }
         else
         {
@@ -63,27 +63,27 @@
         }
     }
 #else /* WEBRTC_LITTLE_ENDIAN */
-    i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */
-    i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */
-    i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */
-    i_CC = (WebRtc_UWord16) (pw16_Datagram[0] & 0xF); /* Get the CC number */
+    i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */
+    i_P = (((uint16_t) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */
+    i_X = (((uint16_t) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */
+    i_CC = (uint16_t) (pw16_Datagram[0] & 0xF); /* Get the CC number */
     RTPheader->payloadType = (pw16_Datagram[0] >> 8) & 0x7F; /* Get the coder type */
-    RTPheader->seqNumber = (((((WebRtc_UWord16) pw16_Datagram[1]) >> 8) & 0xFF)
-        | (((WebRtc_UWord16) (pw16_Datagram[1] & 0xFF)) << 8)); /* Get the packet number */
-    RTPheader->timeStamp = ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF) << 24)
-        | ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF00) << 8)
-        | ((((WebRtc_UWord16) pw16_Datagram[3]) >> 8) & 0xFF)
-        | ((((WebRtc_UWord16) pw16_Datagram[3]) & 0xFF) << 8); /* Get timestamp */
-    RTPheader->ssrc = ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF) << 24)
-        | ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF00) << 8)
-        | ((((WebRtc_UWord16) pw16_Datagram[5]) >> 8) & 0xFF)
-        | ((((WebRtc_UWord16) pw16_Datagram[5]) & 0xFF) << 8); /* Get the SSRC */
+    RTPheader->seqNumber = (((((uint16_t) pw16_Datagram[1]) >> 8) & 0xFF)
+        | (((uint16_t) (pw16_Datagram[1] & 0xFF)) << 8)); /* Get the packet number */
+    RTPheader->timeStamp = ((((uint16_t) pw16_Datagram[2]) & 0xFF) << 24)
+        | ((((uint16_t) pw16_Datagram[2]) & 0xFF00) << 8)
+        | ((((uint16_t) pw16_Datagram[3]) >> 8) & 0xFF)
+        | ((((uint16_t) pw16_Datagram[3]) & 0xFF) << 8); /* Get timestamp */
+    RTPheader->ssrc = ((((uint16_t) pw16_Datagram[4]) & 0xFF) << 24)
+        | ((((uint16_t) pw16_Datagram[4]) & 0xFF00) << 8)
+        | ((((uint16_t) pw16_Datagram[5]) >> 8) & 0xFF)
+        | ((((uint16_t) pw16_Datagram[5]) & 0xFF) << 8); /* Get the SSRC */
 
     if (i_X == 1)
     {
-        /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
-        i_extlength = (((((WebRtc_UWord16) pw16_Datagram[7 + 2 * i_CC]) >> 8) & 0xFF)
-            | (((WebRtc_UWord16) (pw16_Datagram[7 + 2 * i_CC] & 0xFF)) << 8));
+        /* Extension header exists. Find out how many int32_t it consists of. */
+        i_extlength = (((((uint16_t) pw16_Datagram[7 + 2 * i_CC]) >> 8) & 0xFF)
+            | (((uint16_t) (pw16_Datagram[7 + 2 * i_CC] & 0xFF)) << 8));
     }
     if (i_P == 1)
     {
@@ -96,7 +96,7 @@
         else
         {
             /* even number of bytes => last byte in lower byte */
-            i_padlength = (((WebRtc_UWord16) pw16_Datagram[(i_DatagramLen >> 1) - 1]) >> 8);
+            i_padlength = (((uint16_t) pw16_Datagram[(i_DatagramLen >> 1) - 1]) >> 8);
         }
     }
 #endif
@@ -120,8 +120,8 @@
 int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
                                 int *i_No_Of_Payloads)
 {
-    const WebRtc_Word16 *pw16_data = RTPheader[0]->payload; /* Pointer to the data */
-    WebRtc_UWord16 uw16_offsetTimeStamp = 65535, uw16_secondPayload = 65535;
+    const int16_t *pw16_data = RTPheader[0]->payload; /* Pointer to the data */
+    uint16_t uw16_offsetTimeStamp = 65535, uw16_secondPayload = 65535;
     int i_blockLength, i_k;
     int i_discardedBlockLength = 0;
     int singlePayload = 0;
@@ -133,7 +133,7 @@
         singlePayload = 1;
         /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
         i_blockLength = -4;
-        RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
+        RTPheader[0]->payloadType = ((((uint16_t)pw16_data[0]) & 0x7F00) >> 8);
     }
     else
     {
@@ -141,7 +141,7 @@
         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));
+            i_discardedBlockLength += (4+(((uint16_t)pw16_data[1]) & 0x3FF));
             pw16_data+=2;
         }
         if (pw16_data>=(RTPheader[0]->payload+((RTPheader[0]->payloadLen+1)>>1)))
@@ -149,11 +149,11 @@
             return RED_SPLIT_ERROR2; /* Error, we are outside the packet */
         }
         singlePayload = 0; /* the packet contains more than one payload */
-        uw16_secondPayload = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
-        RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[2]) & 0x7F00) >> 8);
-        uw16_offsetTimeStamp = ((((WebRtc_UWord16)pw16_data[0]) & 0xFF) << 6) +
-        ((((WebRtc_UWord16)pw16_data[1]) & 0xFC00) >> 10);
-        i_blockLength = (((WebRtc_UWord16)pw16_data[1]) & 0x3FF);
+        uw16_secondPayload = ((((uint16_t)pw16_data[0]) & 0x7F00) >> 8);
+        RTPheader[0]->payloadType = ((((uint16_t)pw16_data[2]) & 0x7F00) >> 8);
+        uw16_offsetTimeStamp = ((((uint16_t)pw16_data[0]) & 0xFF) << 6) +
+        ((((uint16_t)pw16_data[1]) & 0xFC00) >> 10);
+        i_blockLength = (((uint16_t)pw16_data[1]) & 0x3FF);
     }
 #else /* WEBRTC_LITTLE_ENDIAN */
     if ((pw16_data[0] & 0x80) == 0)
@@ -162,7 +162,7 @@
         singlePayload = 1;
         /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
         i_blockLength = -4;
-        RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
+        RTPheader[0]->payloadType = (((uint16_t) pw16_data[0]) & 0x7F);
     }
     else
     {
@@ -170,8 +170,8 @@
         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)
-                + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8));
+            i_discardedBlockLength += (4 + ((((uint16_t) pw16_data[1]) & 0x3) << 8)
+                + ((((uint16_t) pw16_data[1]) & 0xFF00) >> 8));
             pw16_data += 2;
         }
         if (pw16_data >= (RTPheader[0]->payload + ((RTPheader[0]->payloadLen + 1) >> 1)))
@@ -179,12 +179,12 @@
             return RED_SPLIT_ERROR2; /* Error, we are outside the packet */;
         }
         singlePayload = 0; /* the packet contains more than one payload */
-        uw16_secondPayload = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
-        RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[2]) & 0x7F);
-        uw16_offsetTimeStamp = ((((WebRtc_UWord16) pw16_data[0]) & 0xFF00) >> 2)
-            + ((((WebRtc_UWord16) pw16_data[1]) & 0xFC) >> 2);
-        i_blockLength = ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
-            + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8);
+        uw16_secondPayload = (((uint16_t) pw16_data[0]) & 0x7F);
+        RTPheader[0]->payloadType = (((uint16_t) pw16_data[2]) & 0x7F);
+        uw16_offsetTimeStamp = ((((uint16_t) pw16_data[0]) & 0xFF00) >> 2)
+            + ((((uint16_t) pw16_data[1]) & 0xFC) >> 2);
+        i_blockLength = ((((uint16_t) pw16_data[1]) & 0x3) << 8)
+            + ((((uint16_t) pw16_data[1]) & 0xFF00) >> 8);
     }
 #endif
 
diff --git a/modules/audio_coding/neteq/rtp.h b/modules/audio_coding/neteq/rtp.h
index 8490d62..4642eae 100644
--- a/modules/audio_coding/neteq/rtp.h
+++ b/modules/audio_coding/neteq/rtp.h
@@ -21,14 +21,14 @@
 
 typedef struct
 {
-    WebRtc_UWord16 seqNumber;
-    WebRtc_UWord32 timeStamp;
-    WebRtc_UWord32 ssrc;
+    uint16_t seqNumber;
+    uint32_t timeStamp;
+    uint32_t ssrc;
     int payloadType;
-    const WebRtc_Word16 *payload;
-    WebRtc_Word16 payloadLen;
-    WebRtc_Word16 starts_byte1;
-    WebRtc_Word16 rcuPlCntr;
+    const int16_t *payload;
+    int16_t payloadLen;
+    int16_t starts_byte1;
+    int16_t rcuPlCntr;
 } RTPPacket_t;
 
 /****************************************************************************
@@ -47,7 +47,7 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
+int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen,
                                RTPPacket_t* RTPheader);
 
 /****************************************************************************
diff --git a/modules/audio_coding/neteq/set_fs.c b/modules/audio_coding/neteq/set_fs.c
index b2ad5ca..ac97454 100644
--- a/modules/audio_coding/neteq/set_fs.c
+++ b/modules/audio_coding/neteq/set_fs.c
@@ -17,9 +17,9 @@
 #include "dtmf_buffer.h"
 #include "neteq_error_codes.h"
 
-int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs)
+int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, uint16_t fs)
 {
-    WebRtc_Word16 ok = 0;
+    int16_t ok = 0;
 
     switch (fs)
     {
diff --git a/modules/audio_coding/neteq/signal_mcu.c b/modules/audio_coding/neteq/signal_mcu.c
index e51d5f2..ebe035d 100644
--- a/modules/audio_coding/neteq/signal_mcu.c
+++ b/modules/audio_coding/neteq/signal_mcu.c
@@ -59,21 +59,21 @@
 {
 
     int i_bufferpos, i_res;
-    WebRtc_UWord16 uw16_instr;
+    uint16_t uw16_instr;
     DSP2MCU_info_t dspInfo;
-    WebRtc_Word16 *blockPtr, blockLen;
-    WebRtc_UWord32 uw32_availableTS;
+    int16_t *blockPtr, blockLen;
+    uint32_t uw32_availableTS;
     RTPPacket_t temp_pkt;
-    WebRtc_Word32 w32_bufsize, w32_tmp;
-    WebRtc_Word16 payloadType = -1;
-    WebRtc_Word16 wantedNoOfTimeStamps;
-    WebRtc_Word32 totalTS;
-    WebRtc_Word16 oldPT, latePacketExist = 0;
-    WebRtc_UWord32 oldTS, prevTS, uw32_tmp;
-    WebRtc_UWord16 prevSeqNo;
-    WebRtc_Word16 nextSeqNoAvail;
-    WebRtc_Word16 fs_mult, w16_tmp;
-    WebRtc_Word16 lastModeBGNonly = 0;
+    int32_t w32_bufsize, w32_tmp;
+    int16_t payloadType = -1;
+    int16_t wantedNoOfTimeStamps;
+    int32_t totalTS;
+    int16_t oldPT, latePacketExist = 0;
+    uint32_t oldTS, prevTS, uw32_tmp;
+    uint16_t prevSeqNo;
+    int16_t nextSeqNoAvail;
+    int16_t fs_mult, w16_tmp;
+    int16_t lastModeBGNonly = 0;
 #ifdef NETEQ_DELAY_LOGGING
     int temp_var;
 #endif
@@ -94,7 +94,7 @@
     /* Set blockPtr to first payload block */
     blockPtr = &inst->pw16_writeAddress[3];
 
-    /* Clear instruction word and number of lost samples (2*WebRtc_Word16) */
+    /* Clear instruction word and number of lost samples (2*int16_t) */
     inst->pw16_writeAddress[0] = 0;
     inst->pw16_writeAddress[1] = 0;
     inst->pw16_writeAddress[2] = 0;
@@ -342,7 +342,7 @@
             if (WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType))
             {
                 /* The currently extracted packet is CNG; get CNG fs */
-                WebRtc_UWord16 tempFs;
+                uint16_t tempFs;
 
                 tempFs = WebRtcNetEQ_DbGetSampleRate(&inst->codec_DB_inst, payloadType);
                 /* TODO(tlegrand): Remove this limitation once ACM has full
@@ -356,12 +356,12 @@
                     inst->fs = tempFs;
                 }
             }
-            WebRtcSpl_MemSetW16((WebRtc_Word16*) &cinst, 0,
-                                sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
+            WebRtcSpl_MemSetW16((int16_t*) &cinst, 0,
+                                sizeof(CodecFuncInst_t) / sizeof(int16_t));
             cinst.codec_fs = inst->fs;
         }
         cinst.timeStamp = inst->timeStamp;
-        blockLen = (sizeof(CodecFuncInst_t)) >> (sizeof(WebRtc_Word16) - 1); /* in Word16 */
+        blockLen = (sizeof(CodecFuncInst_t)) >> (sizeof(int16_t) - 1); /* in Word16 */
         *blockPtr = blockLen * 2;
         blockPtr++;
         WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst,sizeof(CodecFuncInst_t));
@@ -395,7 +395,7 @@
         else
         {
             /* CNG exists */
-            blockLen = (sizeof(cinst.codec_state)) >> (sizeof(WebRtc_Word16) - 1);
+            blockLen = (sizeof(cinst.codec_state)) >> (sizeof(int16_t) - 1);
             *blockPtr = blockLen * 2;
             blockPtr++;
             WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst.codec_state,sizeof(cinst.codec_state));
@@ -500,15 +500,15 @@
     /* Do DTMF without extracting any new packets from buffer */
     if (uw16_instr == BUFSTATS_DO_DTMF_ONLY)
     {
-        WebRtc_UWord32 timeStampJump = 0;
+        uint32_t timeStampJump = 0;
 
         /* Update timestamp */
         if ((inst->BufferStat_inst.uw32_CNGplayedTS > 0) && (dspInfo.lastMode != MODE_DTMF))
         {
             /* Jump in timestamps if needed */
             timeStampJump = inst->BufferStat_inst.uw32_CNGplayedTS;
-            inst->pw16_writeAddress[1] = (WebRtc_UWord16) (timeStampJump >> 16);
-            inst->pw16_writeAddress[2] = (WebRtc_UWord16) (timeStampJump & 0xFFFF);
+            inst->pw16_writeAddress[1] = (uint16_t) (timeStampJump >> 16);
+            inst->pw16_writeAddress[2] = (uint16_t) (timeStampJump & 0xFFFF);
         }
 
         inst->timeStamp = dspInfo.playedOutTS + timeStampJump;
@@ -532,7 +532,7 @@
                     | DSP_INSTR_ACCELERATE;
             *blockPtr = 0;
             inst->BufferStat_inst.Automode_inst.sampleMemory
-            = (WebRtc_Word32) dspInfo.samplesLeft;
+            = (int32_t) dspInfo.samplesLeft;
             inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
             return 0;
         }
@@ -576,7 +576,7 @@
                     | DSP_INSTR_PREEMPTIVE_EXPAND;
             *blockPtr = 0;
             inst->BufferStat_inst.Automode_inst.sampleMemory
-            = (WebRtc_Word32) dspInfo.samplesLeft;
+            = (int32_t) dspInfo.samplesLeft;
             inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
             return 0;
         }
@@ -591,7 +591,7 @@
                     | DSP_INSTR_PREEMPTIVE_EXPAND;
             *blockPtr = 0;
             inst->BufferStat_inst.Automode_inst.sampleMemory
-            = (WebRtc_Word32) dspInfo.samplesLeft;
+            = (int32_t) dspInfo.samplesLeft;
             inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
             return 0;
         }
@@ -631,8 +631,8 @@
         && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION_INC_TS))
     {
         uw32_tmp = (uw32_availableTS - dspInfo.playedOutTS);
-        inst->pw16_writeAddress[1] = (WebRtc_UWord16) (uw32_tmp >> 16);
-        inst->pw16_writeAddress[2] = (WebRtc_UWord16) (uw32_tmp & 0xFFFF);
+        inst->pw16_writeAddress[1] = (uint16_t) (uw32_tmp >> 16);
+        inst->pw16_writeAddress[2] = (uint16_t) (uw32_tmp & 0xFFFF);
         if (inst->BufferStat_inst.w16_cngOn == CNG_OFF)
         {
             /*
@@ -682,9 +682,9 @@
             temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE;
             if ((fwrite(&temp_var, sizeof(int),
                         1, delay_fid2) != 1) ||
-                (fwrite(&temp_pkt.timeStamp, sizeof(WebRtc_UWord32),
+                (fwrite(&temp_pkt.timeStamp, sizeof(uint32_t),
                         1, delay_fid2) != 1) ||
-                (fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16),
+                (fwrite(&dspInfo.samplesLeft, sizeof(uint16_t),
                         1, delay_fid2) != 1)) {
               return -1;
             }
@@ -744,7 +744,7 @@
         else
         {
             inst->BufferStat_inst.Automode_inst.sampleMemory
-            = (WebRtc_Word32) dspInfo.samplesLeft + totalTS;
+            = (int32_t) dspInfo.samplesLeft + totalTS;
             inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
         }
     }
diff --git a/modules/audio_coding/neteq/split_and_insert.c b/modules/audio_coding/neteq/split_and_insert.c
index 03c1569..ce2e821 100644
--- a/modules/audio_coding/neteq/split_and_insert.c
+++ b/modules/audio_coding/neteq/split_and_insert.c
@@ -21,15 +21,15 @@
 #include "neteq_error_codes.h"
 
 int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
-                                      SplitInfo_t *split_inst, WebRtc_Word16 *flushed)
+                                      SplitInfo_t *split_inst, int16_t *flushed)
 {
 
     int i_ok;
     int len;
     int i;
     RTPPacket_t temp_packet;
-    WebRtc_Word16 localFlushed = 0;
-    const WebRtc_Word16 *pw16_startPayload;
+    int16_t localFlushed = 0;
+    const int16_t *pw16_startPayload;
     *flushed = 0;
 
     len = packet->payloadLen;
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc b/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
index 25f10b0..838f3d9 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
+++ b/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc_neteq_help_macros.h"
 
-NETEQTEST_Decoder::NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt)
+NETEQTEST_Decoder::NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, uint16_t fs, const char * name, uint8_t pt)
 :
 _decoder(NULL),
 _decoderType(type),
@@ -42,11 +42,11 @@
 #ifdef CODEC_ISAC
 #include "isac.h"
 
-decoder_iSAC::decoder_iSAC(WebRtc_UWord8 pt) 
+decoder_iSAC::decoder_iSAC(uint8_t pt) 
 :
 NETEQTEST_Decoder(kDecoderISAC, 16000, "iSAC", pt)
 {
-    WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
+    int16_t err = WebRtcIsac_Create((ISACStruct **) &_decoder);
     if (err)
     {
         exit(EXIT_FAILURE);
@@ -79,11 +79,11 @@
 #endif
 
 #ifdef CODEC_ISAC_SWB
-decoder_iSACSWB::decoder_iSACSWB(WebRtc_UWord8 pt) 
+decoder_iSACSWB::decoder_iSACSWB(uint8_t pt) 
 :
 NETEQTEST_Decoder(kDecoderISACswb, 32000, "iSAC swb", pt)
 {
-    WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
+    int16_t err = WebRtcIsac_Create((ISACStruct **) &_decoder);
     if (err)
     {
         exit(EXIT_FAILURE);
@@ -114,9 +114,9 @@
 #endif
 
 #ifdef CODEC_ISAC_FB
-decoder_iSACFB::decoder_iSACFB(WebRtc_UWord8 pt)
+decoder_iSACFB::decoder_iSACFB(uint8_t pt)
     : NETEQTEST_Decoder(kDecoderISACfb, 32000, "iSAC fb", pt) {
-  WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
+  int16_t err = WebRtcIsac_Create((ISACStruct **) &_decoder);
   if (err) {
     exit(EXIT_FAILURE);
   }
@@ -143,7 +143,7 @@
 #ifdef CODEC_G711
 #include "g711_interface.h"
 
-decoder_PCMU::decoder_PCMU(WebRtc_UWord8 pt) 
+decoder_PCMU::decoder_PCMU(uint8_t pt) 
 :
 NETEQTEST_Decoder(kDecoderPCMu, 8000, "G.711-u", pt)
 {
@@ -160,7 +160,7 @@
 
 }
 
-decoder_PCMA::decoder_PCMA(WebRtc_UWord8 pt) 
+decoder_PCMA::decoder_PCMA(uint8_t pt) 
 :
 NETEQTEST_Decoder(kDecoderPCMa, 8000, "G.711-A", pt)
 {
@@ -229,11 +229,11 @@
 
 #ifdef CODEC_ILBC
 #include "ilbc.h"
-decoder_ILBC::decoder_ILBC(WebRtc_UWord8 pt)
+decoder_ILBC::decoder_ILBC(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderILBC, 8000, "iLBC", pt)
 {
-    WebRtc_Word16 err = WebRtcIlbcfix_DecoderCreate((iLBC_decinst_t **) &_decoder);
+    int16_t err = WebRtcIlbcfix_DecoderCreate((iLBC_decinst_t **) &_decoder);
     if (err)
     {
         exit(EXIT_FAILURE);
@@ -257,11 +257,11 @@
 
 #ifdef CODEC_G729
 #include "G729Interface.h"
-decoder_G729::decoder_G729(WebRtc_UWord8 pt)
+decoder_G729::decoder_G729(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG729, 8000, "G.729", pt)
 {
-    WebRtc_Word16 err = WebRtcG729_CreateDec((G729_decinst_t **) &_decoder);
+    int16_t err = WebRtcG729_CreateDec((G729_decinst_t **) &_decoder);
     if (err)
     {
         exit(EXIT_FAILURE);
@@ -285,11 +285,11 @@
 
 #ifdef CODEC_G729_1
 #include "G729_1Interface.h"
-decoder_G729_1::decoder_G729_1(WebRtc_UWord8 pt)
+decoder_G729_1::decoder_G729_1(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG729_1, 16000, "G.729.1", pt)
 {
-    WebRtc_Word16 err = WebRtcG7291_Create((G729_1_inst_t **) &_decoder);
+    int16_t err = WebRtcG7291_Create((G729_1_inst_t **) &_decoder);
     if (err)
     {
         exit(EXIT_FAILURE);
@@ -313,11 +313,11 @@
 
 #ifdef CODEC_G722
 #include "g722_interface.h"
-decoder_G722::decoder_G722(WebRtc_UWord8 pt)
+decoder_G722::decoder_G722(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722, 16000, "G.722", pt)
 {
-     WebRtc_Word16 err = WebRtcG722_CreateDecoder((G722DecInst **) &_decoder);
+     int16_t err = WebRtcG722_CreateDecoder((G722DecInst **) &_decoder);
      if (err)
      {
          exit(EXIT_FAILURE);
@@ -346,7 +346,7 @@
 #endif
 
 #ifdef CODEC_G722_1_16
-decoder_G722_1_16::decoder_G722_1_16(WebRtc_UWord8 pt)
+decoder_G722_1_16::decoder_G722_1_16(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1_16, 16000, "G.722.1 (16 kbps)", pt)
 {
@@ -372,7 +372,7 @@
 #endif
 
 #ifdef CODEC_G722_1_24
-decoder_G722_1_24::decoder_G722_1_24(WebRtc_UWord8 pt)
+decoder_G722_1_24::decoder_G722_1_24(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1_24, 16000, "G.722.1 (24 kbps)", pt)
 {
@@ -398,7 +398,7 @@
 #endif
 
 #ifdef CODEC_G722_1_32
-decoder_G722_1_32::decoder_G722_1_32(WebRtc_UWord8 pt)
+decoder_G722_1_32::decoder_G722_1_32(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1_32, 16000, "G.722.1 (32 kbps)", pt)
 {
@@ -424,7 +424,7 @@
 #endif
 
 #ifdef CODEC_G722_1C_24
-decoder_G722_1C_24::decoder_G722_1C_24(WebRtc_UWord8 pt)
+decoder_G722_1C_24::decoder_G722_1C_24(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1C_24, 32000, "G.722.1C (24 kbps)", pt)
 {
@@ -448,7 +448,7 @@
 #endif
 
 #ifdef CODEC_G722_1C_32
-decoder_G722_1C_32::decoder_G722_1C_32(WebRtc_UWord8 pt)
+decoder_G722_1C_32::decoder_G722_1C_32(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1C_32, 32000, "G.722.1C (32 kbps)", pt)
 {
@@ -472,7 +472,7 @@
 #endif
 
 #ifdef CODEC_G722_1C_48
-decoder_G722_1C_48::decoder_G722_1C_48(WebRtc_UWord8 pt)
+decoder_G722_1C_48::decoder_G722_1C_48(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderG722_1C_48, 32000, "G.722.1C (48 kbps)", pt)
 {
@@ -498,7 +498,7 @@
 #ifdef CODEC_AMR
 #include "AMRInterface.h"
 #include "AMRCreation.h"
-decoder_AMR::decoder_AMR(WebRtc_UWord8 pt)
+decoder_AMR::decoder_AMR(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderAMR, 8000, "AMR", pt)
 {
@@ -526,7 +526,7 @@
 #ifdef CODEC_AMRWB
 #include "AMRWBInterface.h"
 #include "AMRWBCreation.h"
-decoder_AMRWB::decoder_AMRWB(WebRtc_UWord8 pt)
+decoder_AMRWB::decoder_AMRWB(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderAMRWB, 16000, "AMR wb", pt)
 {
@@ -554,7 +554,7 @@
 #ifdef CODEC_GSMFR
 #include "GSMFRInterface.h"
 #include "GSMFRCreation.h"
-decoder_GSMFR::decoder_GSMFR(WebRtc_UWord8 pt)
+decoder_GSMFR::decoder_GSMFR(uint8_t pt)
 :
 NETEQTEST_Decoder(kDecoderGSMFR, 8000, "GSM-FR", pt)
 {
@@ -579,7 +579,7 @@
 
 #if (defined(CODEC_SPEEX_8) || defined (CODEC_SPEEX_16))
 #include "SpeexInterface.h"
-decoder_SPEEX::decoder_SPEEX(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+decoder_SPEEX::decoder_SPEEX(uint8_t pt, uint16_t fs)
 :
 NETEQTEST_Decoder(fs == 8000 ? kDecoderSPEEX_8 : kDecoderSPEEX_16, 
                   fs, "SPEEX", pt)
@@ -608,7 +608,7 @@
 
 #ifdef CODEC_CELT_32
 #include "celt_interface.h"
-decoder_CELT::decoder_CELT(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+decoder_CELT::decoder_CELT(uint8_t pt, uint16_t fs)
 :
 NETEQTEST_Decoder(kDecoderCELT_32, fs, "CELT", pt)
 {
@@ -630,7 +630,7 @@
     return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
 }
 
-decoder_CELTslave::decoder_CELTslave(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+decoder_CELTslave::decoder_CELTslave(uint8_t pt, uint16_t fs)
 :
 NETEQTEST_Decoder(kDecoderCELT_32, fs, "CELT", pt)
 {
@@ -677,7 +677,7 @@
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
 #include "webrtc_cng.h"
-decoder_CNG::decoder_CNG(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+decoder_CNG::decoder_CNG(uint8_t pt, uint16_t fs)
 :
 NETEQTEST_Decoder(kDecoderCNG, fs, "CNG", pt)
 {
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h b/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
index 43f16a5..a58bf0d 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
+++ b/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
@@ -21,25 +21,25 @@
 class NETEQTEST_Decoder
 {
 public:
-    NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt = 0);
+    NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, uint16_t fs, const char * name, uint8_t pt = 0);
     virtual ~NETEQTEST_Decoder() {};
 
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
 
     int getName(char * name, int maxLen) const { strncpy( name, _name.c_str(), maxLen ); return 0;};
 
-    void setPT(WebRtc_UWord8 pt) { _pt = pt; };
-    WebRtc_UWord16 getFs() const { return (_fs); };
+    void setPT(uint8_t pt) { _pt = pt; };
+    uint16_t getFs() const { return (_fs); };
     enum WebRtcNetEQDecoder getType() const { return (_decoderType); };
-    WebRtc_UWord8 getPT() const { return (_pt); };
+    uint8_t getPT() const { return (_pt); };
 
 protected:
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq, WebRtcNetEQ_CodecDef & codecInst);
 
     void * _decoder;
     enum WebRtcNetEQDecoder _decoderType;
-    WebRtc_UWord8 _pt;
-    WebRtc_UWord16 _fs;
+    uint8_t _pt;
+    uint16_t _fs;
     std::string _name;
 
 private:
@@ -49,7 +49,7 @@
 class decoder_iSAC : public NETEQTEST_Decoder
 {
 public:
-    decoder_iSAC(WebRtc_UWord8 pt = 0);
+    decoder_iSAC(uint8_t pt = 0);
     virtual ~decoder_iSAC();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -58,7 +58,7 @@
 class decoder_iSACSWB : public NETEQTEST_Decoder
 {
 public:
-    decoder_iSACSWB(WebRtc_UWord8 pt = 0);
+    decoder_iSACSWB(uint8_t pt = 0);
     virtual ~decoder_iSACSWB();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -66,7 +66,7 @@
 
 class decoder_iSACFB : public NETEQTEST_Decoder {
  public:
-  decoder_iSACFB(WebRtc_UWord8 pt = 0);
+  decoder_iSACFB(uint8_t pt = 0);
   virtual ~decoder_iSACFB();
   int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -75,7 +75,7 @@
 class decoder_PCMU : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCMU(WebRtc_UWord8 pt = 0);
+    decoder_PCMU(uint8_t pt = 0);
     virtual ~decoder_PCMU() {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -84,7 +84,7 @@
 class decoder_PCMA : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCMA(WebRtc_UWord8 pt = 0);
+    decoder_PCMA(uint8_t pt = 0);
     virtual ~decoder_PCMA() {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -92,28 +92,28 @@
 class decoder_PCM16B_NB : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCM16B_NB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16B, 8000, "PCM16 nb", pt) {};
+    decoder_PCM16B_NB(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderPCM16B, 8000, "PCM16 nb", pt) {};
     virtual ~decoder_PCM16B_NB() {};
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 class decoder_PCM16B_WB : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCM16B_WB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bwb, 16000, "PCM16 wb", pt) {};
+    decoder_PCM16B_WB(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bwb, 16000, "PCM16 wb", pt) {};
     virtual ~decoder_PCM16B_WB() {};
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 class decoder_PCM16B_SWB32 : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCM16B_SWB32(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb32kHz, 32000, "PCM16 swb32", pt) {};
+    decoder_PCM16B_SWB32(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb32kHz, 32000, "PCM16 swb32", pt) {};
     virtual ~decoder_PCM16B_SWB32() {};
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 class decoder_PCM16B_SWB48 : public NETEQTEST_Decoder
 {
 public:
-    decoder_PCM16B_SWB48(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb48kHz, 48000, "PCM16 swb48", pt) {};
+    decoder_PCM16B_SWB48(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb48kHz, 48000, "PCM16 swb48", pt) {};
     virtual ~decoder_PCM16B_SWB48() {};
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -122,7 +122,7 @@
 class decoder_ILBC : public NETEQTEST_Decoder
 {
 public:
-    decoder_ILBC(WebRtc_UWord8 pt = 0);
+    decoder_ILBC(uint8_t pt = 0);
     virtual ~decoder_ILBC();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -131,7 +131,7 @@
 class decoder_G729 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G729(WebRtc_UWord8 pt = 0);
+    decoder_G729(uint8_t pt = 0);
     virtual ~decoder_G729();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -139,7 +139,7 @@
 class decoder_G729_1 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G729_1(WebRtc_UWord8 pt = 0);
+    decoder_G729_1(uint8_t pt = 0);
     virtual ~decoder_G729_1();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -148,7 +148,7 @@
 class decoder_G722 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722(WebRtc_UWord8 pt = 0);
+    decoder_G722(uint8_t pt = 0);
     virtual ~decoder_G722();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -157,7 +157,7 @@
 class decoder_G722_1_16 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1_16(WebRtc_UWord8 pt = 0);
+    decoder_G722_1_16(uint8_t pt = 0);
     virtual ~decoder_G722_1_16();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -165,7 +165,7 @@
 class decoder_G722_1_24 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1_24(WebRtc_UWord8 pt = 0);
+    decoder_G722_1_24(uint8_t pt = 0);
     virtual ~decoder_G722_1_24();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -173,7 +173,7 @@
 class decoder_G722_1_32 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1_32(WebRtc_UWord8 pt = 0);
+    decoder_G722_1_32(uint8_t pt = 0);
     virtual ~decoder_G722_1_32();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -182,7 +182,7 @@
 class decoder_G722_1C_24 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1C_24(WebRtc_UWord8 pt = 0);
+    decoder_G722_1C_24(uint8_t pt = 0);
     virtual ~decoder_G722_1C_24();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -190,7 +190,7 @@
 class decoder_G722_1C_32 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1C_32(WebRtc_UWord8 pt = 0);
+    decoder_G722_1C_32(uint8_t pt = 0);
     virtual ~decoder_G722_1C_32();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -198,7 +198,7 @@
 class decoder_G722_1C_48 : public NETEQTEST_Decoder
 {
 public:
-    decoder_G722_1C_48(WebRtc_UWord8 pt = 0);
+    decoder_G722_1C_48(uint8_t pt = 0);
     virtual ~decoder_G722_1C_48();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -207,7 +207,7 @@
 class decoder_AMR : public NETEQTEST_Decoder
 {
 public:
-    decoder_AMR(WebRtc_UWord8 pt = 0);
+    decoder_AMR(uint8_t pt = 0);
     virtual ~decoder_AMR();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -215,7 +215,7 @@
 class decoder_AMRWB : public NETEQTEST_Decoder
 {
 public:
-    decoder_AMRWB(WebRtc_UWord8 pt = 0);
+    decoder_AMRWB(uint8_t pt = 0);
     virtual ~decoder_AMRWB();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -223,7 +223,7 @@
 class decoder_GSMFR : public NETEQTEST_Decoder
 {
 public:
-    decoder_GSMFR(WebRtc_UWord8 pt = 0);
+    decoder_GSMFR(uint8_t pt = 0);
     virtual ~decoder_GSMFR();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -231,8 +231,8 @@
 class decoder_G726 : public NETEQTEST_Decoder
 {
 public:
-    //virtual decoder_G726(WebRtc_UWord8 pt = 0) = 0;
-    decoder_G726(enum WebRtcNetEQDecoder type, const char * name, WebRtc_UWord8 pt = 0);
+    //virtual decoder_G726(uint8_t pt = 0) = 0;
+    decoder_G726(enum WebRtcNetEQDecoder type, const char * name, uint8_t pt = 0);
     virtual ~decoder_G726();
     virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
 };
@@ -240,35 +240,35 @@
 class decoder_G726_16 : public decoder_G726
 {
 public:
-    decoder_G726_16(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_16, "G.726 (16 kbps)", pt) {};
+    decoder_G726_16(uint8_t pt = 0) : decoder_G726(kDecoderG726_16, "G.726 (16 kbps)", pt) {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 
 class decoder_G726_24 : public decoder_G726
 {
 public:
-    decoder_G726_24(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_24, "G.726 (24 kbps)", pt) {};
+    decoder_G726_24(uint8_t pt = 0) : decoder_G726(kDecoderG726_24, "G.726 (24 kbps)", pt) {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 
 class decoder_G726_32 : public decoder_G726
 {
 public:
-    decoder_G726_32(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_32, "G.726 (32 kbps)", pt) {};
+    decoder_G726_32(uint8_t pt = 0) : decoder_G726(kDecoderG726_32, "G.726 (32 kbps)", pt) {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 
 class decoder_G726_40 : public decoder_G726
 {
 public:
-    decoder_G726_40(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_40, "G.726 (40 kbps)", pt) {};
+    decoder_G726_40(uint8_t pt = 0) : decoder_G726(kDecoderG726_40, "G.726 (40 kbps)", pt) {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 
 class decoder_SPEEX : public NETEQTEST_Decoder
 {
 public:
-    decoder_SPEEX(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
+    decoder_SPEEX(uint8_t pt = 0, uint16_t fs = 8000);
     virtual ~decoder_SPEEX();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -276,14 +276,14 @@
 class decoder_CELT : public NETEQTEST_Decoder
 {
 public:
-    decoder_CELT(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 32000);
+    decoder_CELT(uint8_t pt = 0, uint16_t fs = 32000);
     virtual ~decoder_CELT();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 class decoder_CELTslave : public NETEQTEST_Decoder
 {
 public:
-    decoder_CELTslave(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 32000);
+    decoder_CELTslave(uint8_t pt = 0, uint16_t fs = 32000);
     virtual ~decoder_CELTslave();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -291,7 +291,7 @@
 class decoder_RED : public NETEQTEST_Decoder
 {
 public:
-    decoder_RED(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderRED, 8000, "RED", pt) {};
+    decoder_RED(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderRED, 8000, "RED", pt) {};
     virtual ~decoder_RED() {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -299,7 +299,7 @@
 class decoder_AVT : public NETEQTEST_Decoder
 {
 public:
-    decoder_AVT(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderAVT, 8000, "AVT", pt) {};
+    decoder_AVT(uint8_t pt = 0) : NETEQTEST_Decoder(kDecoderAVT, 8000, "AVT", pt) {};
     virtual ~decoder_AVT() {};
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
@@ -308,7 +308,7 @@
 class decoder_CNG : public NETEQTEST_Decoder
 {
 public:
-    decoder_CNG(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
+    decoder_CNG(uint8_t pt = 0, uint16_t fs = 8000);
     virtual ~decoder_CNG();
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc b/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
index d37cd64..5c28d49 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
+++ b/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
@@ -29,8 +29,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     if (fread(&length, 2, 1, fp) == 0)
     {
@@ -52,10 +52,10 @@
         return -1;
     }
     // Store in local variable until we have passed the reset below.
-    WebRtc_UWord32 receiveTime = ntohl(offset);
+    uint32_t receiveTime = ntohl(offset);
 
     // Use length here because a plen of 0 specifies rtcp.
-    length = (WebRtc_UWord16) (length - _kRDHeaderLen);
+    length = (uint16_t) (length - _kRDHeaderLen);
 
     // check buffer size
     if (_datagram && _memSize < length + 1)
@@ -66,7 +66,7 @@
     if (!_datagram)
     {
         // Add one extra byte, to be able to fake a dummy payload of one byte.
-        _datagram = new WebRtc_UWord8[length + 1];
+        _datagram = new uint8_t[length + 1];
         _memSize = length + 1;
     }
     memset(_datagram, 0, length + 1);
@@ -154,8 +154,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     // length including RTPplay header
     length = htons(_datagramLen + _kRDHeaderLen);
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc b/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
index d175c76..b77c305 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
+++ b/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
@@ -30,7 +30,7 @@
 }
 
 NETEQTEST_NetEQClass::NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
-        WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
+        uint16_t fs, WebRtcNetEQNetworkType nwType)
     :
     _inst(NULL),
     _instMem(NULL),
@@ -84,7 +84,7 @@
         _instMem = NULL;
     }
 
-    _instMem = new WebRtc_Word8[memSize];
+    _instMem = new int8_t[memSize];
 
     int ret = WebRtcNetEQ_Assign(&_inst, _instMem);
 
@@ -97,7 +97,7 @@
 }
 
 
-int NETEQTEST_NetEQClass::init(WebRtc_UWord16 fs)
+int NETEQTEST_NetEQClass::init(uint16_t fs)
 {
     int ret;
 
@@ -165,7 +165,7 @@
         _bufferMem = NULL;
     }
 
-    _bufferMem = new WebRtc_Word8[memSize];
+    _bufferMem = new int8_t[memSize];
 
     memset(_bufferMem, -1, memSize);
 
@@ -253,7 +253,7 @@
         QueryPerformanceCounter(&countA); // get start count for processor
 #endif
 
-        err = WebRtcNetEQ_RecIn(_inst, (WebRtc_Word16 *) rtp.datagram(), rtp.dataLen(), rtp.time() * _fsmult * 8);
+        err = WebRtcNetEQ_RecIn(_inst, (int16_t *) rtp.datagram(), rtp.dataLen(), rtp.time() * _fsmult * 8);
 
 #ifdef WINDOWS_TIMING
         QueryPerformanceCounter(&countB); // get stop count for processor
@@ -272,10 +272,10 @@
 }
 
 
-WebRtc_Word16 NETEQTEST_NetEQClass::recOut(WebRtc_Word16 *outData, void *msInfo, enum WebRtcNetEQOutputType *outputType)
+int16_t NETEQTEST_NetEQClass::recOut(int16_t *outData, void *msInfo, enum WebRtcNetEQOutputType *outputType)
 {
     int err;
-    WebRtc_Word16 outLen = 0;
+    int16_t outLen = 0;
 #ifdef WINDOWS_TIMING
     LARGE_INTEGER countA, countB;
 #endif
@@ -299,7 +299,7 @@
     else
     {
         // master/slave mode
-        err = WebRtcNetEQ_RecOutMasterSlave(_inst, outData, &outLen, msInfo, static_cast<WebRtc_Word16>(_isMaster));
+        err = WebRtcNetEQ_RecOutMasterSlave(_inst, outData, &outLen, msInfo, static_cast<int16_t>(_isMaster));
     }
 
 #ifdef WINDOWS_TIMING
@@ -338,10 +338,10 @@
 }
 
 
-WebRtc_UWord32 NETEQTEST_NetEQClass::getSpeechTimeStamp()
+uint32_t NETEQTEST_NetEQClass::getSpeechTimeStamp()
 {
 
-    WebRtc_UWord32 ts = 0;
+    uint32_t ts = 0;
     int err;
 
     err = WebRtcNetEQ_GetSpeechTimeStamp(_inst, &ts);
@@ -379,7 +379,7 @@
 //}
 //
 //NETEQTEST_NetEQVector::NETEQTEST_NetEQVector(int numChannels, enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
-//                      WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
+//                      uint16_t fs, WebRtcNetEQNetworkType nwType)
 //                      :
 //channels(numChannels, new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, fs, nwType))
 //{
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h b/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
index 3e43125..8e987b8 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
+++ b/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
@@ -30,16 +30,16 @@
 public:
     NETEQTEST_NetEQClass();
     NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, 
-        WebRtc_UWord16 fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
+        uint16_t fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
     ~NETEQTEST_NetEQClass();
 
     int assign();
-    int init(WebRtc_UWord16 fs = 8000);
+    int init(uint16_t fs = 8000);
     int assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
     int loadCodec(WebRtcNetEQ_CodecDef & codecInst);
     int recIn(NETEQTEST_RTPpacket & rtp);
-    WebRtc_Word16 recOut(WebRtc_Word16 *outData, void *msInfo = NULL, enum WebRtcNetEQOutputType *outputType = NULL);
-    WebRtc_UWord32 getSpeechTimeStamp();
+    int16_t recOut(int16_t *outData, void *msInfo = NULL, enum WebRtcNetEQOutputType *outputType = NULL);
+    uint32_t getSpeechTimeStamp();
     WebRtcNetEQOutputType getOutputType();
 
     void * instance() { return (_inst); };
@@ -66,8 +66,8 @@
 
 private:
     void *          _inst;
-    WebRtc_Word8 *    _instMem;
-    WebRtc_Word8 *    _bufferMem;
+    int8_t *    _instMem;
+    int8_t *    _bufferMem;
     bool            _preparseRTP;
     int             _fsmult;
     bool            _isMaster;
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
index ecca018..49ce02d 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
+++ b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
@@ -105,8 +105,8 @@
         return(-1);
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     if (fread(&length,2,1,fp)==0)
     {
@@ -127,10 +127,10 @@
         reset();
         return(-1);
     }
-    WebRtc_UWord32 receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
+    uint32_t receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
 
     // Use length here because a plen of 0 specifies rtcp
-    length = (WebRtc_UWord16) (length - _kRDHeaderLen);
+    length = (uint16_t) (length - _kRDHeaderLen);
 
     // check buffer size
     if (_datagram && _memSize < length)
@@ -140,7 +140,7 @@
 
     if (!_datagram)
     {
-        _datagram = new WebRtc_UWord8[length];
+        _datagram = new uint8_t[length];
         _memSize = length;
     }
 
@@ -185,7 +185,7 @@
 
     if (!_datagram)
     {
-        _datagram = new WebRtc_UWord8[length];
+        _datagram = new uint8_t[length];
         _memSize = length;
     }
 
@@ -216,8 +216,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     // length including RTPplay header
     length = htons(_datagramLen + _kRDHeaderLen);
@@ -253,7 +253,7 @@
 }
 
 
-void NETEQTEST_RTPpacket::blockPT(WebRtc_UWord8 pt)
+void NETEQTEST_RTPpacket::blockPT(uint8_t pt)
 {
     _blockList[pt] = true;
 }
@@ -309,7 +309,7 @@
     }
 }
 
-WebRtc_UWord8 * NETEQTEST_RTPpacket::datagram() const
+uint8_t * NETEQTEST_RTPpacket::datagram() const
 {
     if (_datagramLen > 0)
     {
@@ -321,7 +321,7 @@
     }
 }
 
-WebRtc_UWord8 * NETEQTEST_RTPpacket::payload() const
+uint8_t * NETEQTEST_RTPpacket::payload() const
 {
     if (_payloadLen > 0)
     {
@@ -333,13 +333,13 @@
     }
 }
 
-WebRtc_Word16 NETEQTEST_RTPpacket::payloadLen()
+int16_t NETEQTEST_RTPpacket::payloadLen()
 {
     parseHeader();
     return _payloadLen;
 }
 
-WebRtc_Word16 NETEQTEST_RTPpacket::dataLen() const
+int16_t NETEQTEST_RTPpacket::dataLen() const
 {
     return _datagramLen;
 }
@@ -354,7 +354,7 @@
     return _lost;
 }
 
-WebRtc_UWord8  NETEQTEST_RTPpacket::payloadType() const
+uint8_t  NETEQTEST_RTPpacket::payloadType() const
 {
     WebRtcNetEQ_RTPInfo tempRTPinfo;
 
@@ -370,7 +370,7 @@
     return tempRTPinfo.payloadType;
 }
 
-WebRtc_UWord16 NETEQTEST_RTPpacket::sequenceNumber() const
+uint16_t NETEQTEST_RTPpacket::sequenceNumber() const
 {
     WebRtcNetEQ_RTPInfo tempRTPinfo;
 
@@ -386,7 +386,7 @@
     return tempRTPinfo.sequenceNumber;
 }
 
-WebRtc_UWord32 NETEQTEST_RTPpacket::timeStamp() const
+uint32_t NETEQTEST_RTPpacket::timeStamp() const
 {
     WebRtcNetEQ_RTPInfo tempRTPinfo;
 
@@ -402,7 +402,7 @@
     return tempRTPinfo.timeStamp;
 }
 
-WebRtc_UWord32 NETEQTEST_RTPpacket::SSRC() const
+uint32_t NETEQTEST_RTPpacket::SSRC() const
 {
     WebRtcNetEQ_RTPInfo tempRTPinfo;
 
@@ -418,7 +418,7 @@
     return tempRTPinfo.SSRC;
 }
 
-WebRtc_UWord8  NETEQTEST_RTPpacket::markerBit() const
+uint8_t  NETEQTEST_RTPpacket::markerBit() const
 {
     WebRtcNetEQ_RTPInfo tempRTPinfo;
 
@@ -436,7 +436,7 @@
 
 
 
-int NETEQTEST_RTPpacket::setPayloadType(WebRtc_UWord8 pt)
+int NETEQTEST_RTPpacket::setPayloadType(uint8_t pt)
 {
 
     if (_datagramLen < 12)
@@ -455,7 +455,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setSequenceNumber(WebRtc_UWord16 sn)
+int NETEQTEST_RTPpacket::setSequenceNumber(uint16_t sn)
 {
 
     if (_datagramLen < 12)
@@ -475,7 +475,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setTimeStamp(WebRtc_UWord32 ts)
+int NETEQTEST_RTPpacket::setTimeStamp(uint32_t ts)
 {
 
     if (_datagramLen < 12)
@@ -497,7 +497,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setSSRC(WebRtc_UWord32 ssrc)
+int NETEQTEST_RTPpacket::setSSRC(uint32_t ssrc)
 {
 
     if (_datagramLen < 12)
@@ -519,7 +519,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setMarkerBit(WebRtc_UWord8 mb)
+int NETEQTEST_RTPpacket::setMarkerBit(uint8_t mb)
 {
 
     if (_datagramLen < 12)
@@ -628,7 +628,7 @@
 }
 
 
-void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const
+void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqNo, uint32_t timestamp, uint32_t ssrc, uint8_t markerBit) const
 {
     rtp_data[0]=(unsigned char)0x80;
     if (markerBit)
@@ -655,11 +655,11 @@
     rtp_data[11]=(unsigned char)(ssrc & 0xFF);
 }
 
-WebRtc_UWord16
+uint16_t
     NETEQTEST_RTPpacket::parseRTPheader(WebRtcNetEQ_RTPInfo *RTPinfo,
-                                        WebRtc_UWord8 **payloadPtr) const
+                                        uint8_t **payloadPtr) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     int i_P, i_X, i_CC;
 
     assert(_datagramLen >= 12);
@@ -671,59 +671,59 @@
 
     if (payloadPtr)
     {
-        *payloadPtr = (WebRtc_UWord8*) &rtp_data[i_startPosition >> 1];
+        *payloadPtr = (uint8_t*) &rtp_data[i_startPosition >> 1];
     }
 
-    return (WebRtc_UWord16) (_datagramLen - i_startPosition - i_padlength);
+    return (uint16_t) (_datagramLen - i_startPosition - i_padlength);
 }
 
 
 void NETEQTEST_RTPpacket::parseBasicHeader(WebRtcNetEQ_RTPInfo *RTPinfo,
                                            int *i_P, int *i_X, int *i_CC) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     if (_datagramLen < 12)
     {
         assert(false);
         return;
     }
 
-    *i_P=(((WebRtc_UWord16)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
-    *i_X=(((WebRtc_UWord16)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
-    *i_CC=(WebRtc_UWord16)(rtp_data[0] & 0xF); /* Get the CC number  */
+    *i_P=(((uint16_t)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
+    *i_X=(((uint16_t)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
+    *i_CC=(uint16_t)(rtp_data[0] & 0xF); /* Get the CC number  */
     /* Get the marker bit */
-    RTPinfo->markerBit = (WebRtc_UWord8) ((rtp_data[0] >> 15) & 0x01);
+    RTPinfo->markerBit = (uint8_t) ((rtp_data[0] >> 15) & 0x01);
     /* Get the coder type */
-    RTPinfo->payloadType = (WebRtc_UWord8) ((rtp_data[0] >> 8) & 0x7F);
+    RTPinfo->payloadType = (uint8_t) ((rtp_data[0] >> 8) & 0x7F);
     /* Get the packet number */
-    RTPinfo->sequenceNumber = ((( ((WebRtc_UWord16)rtp_data[1]) >> 8) & 0xFF) |
-        ( ((WebRtc_UWord16)(rtp_data[1] & 0xFF)) << 8));
+    RTPinfo->sequenceNumber = ((( ((uint16_t)rtp_data[1]) >> 8) & 0xFF) |
+        ( ((uint16_t)(rtp_data[1] & 0xFF)) << 8));
     /* Get timestamp */
-    RTPinfo->timeStamp = ((((WebRtc_UWord16)rtp_data[2]) & 0xFF) << 24) |
-        ((((WebRtc_UWord16)rtp_data[2]) & 0xFF00) << 8) |
-        ((((WebRtc_UWord16)rtp_data[3]) >> 8) & 0xFF) |
-        ((((WebRtc_UWord16)rtp_data[3]) & 0xFF) << 8);
+    RTPinfo->timeStamp = ((((uint16_t)rtp_data[2]) & 0xFF) << 24) |
+        ((((uint16_t)rtp_data[2]) & 0xFF00) << 8) |
+        ((((uint16_t)rtp_data[3]) >> 8) & 0xFF) |
+        ((((uint16_t)rtp_data[3]) & 0xFF) << 8);
     /* Get the SSRC */
-    RTPinfo->SSRC=((((WebRtc_UWord16)rtp_data[4]) & 0xFF) << 24) |
-        ((((WebRtc_UWord16)rtp_data[4]) & 0xFF00) << 8) |
-        ((((WebRtc_UWord16)rtp_data[5]) >> 8) & 0xFF) |
-        ((((WebRtc_UWord16)rtp_data[5]) & 0xFF) << 8);
+    RTPinfo->SSRC=((((uint16_t)rtp_data[4]) & 0xFF) << 24) |
+        ((((uint16_t)rtp_data[4]) & 0xFF00) << 8) |
+        ((((uint16_t)rtp_data[5]) >> 8) & 0xFF) |
+        ((((uint16_t)rtp_data[5]) & 0xFF) << 8);
 }
 
 int NETEQTEST_RTPpacket::calcHeaderLength(int i_X, int i_CC) const
 {
     int i_extlength = 0;
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
 
     if (i_X == 1)
     {
         // Extension header exists.
-        // Find out how many WebRtc_Word32 it consists of.
+        // Find out how many int32_t it consists of.
         assert(_datagramLen > 2 * (7 + 2 * i_CC));
         if (_datagramLen > 2 * (7 + 2 * i_CC))
         {
-            i_extlength = (((((WebRtc_UWord16) rtp_data[7 + 2 * i_CC]) >> 8)
-                & 0xFF) | (((WebRtc_UWord16) (rtp_data[7 + 2 * i_CC] & 0xFF))
+            i_extlength = (((((uint16_t) rtp_data[7 + 2 * i_CC]) >> 8)
+                & 0xFF) | (((uint16_t) (rtp_data[7 + 2 * i_CC] & 0xFF))
                 << 8)) + 1;
         }
     }
@@ -733,7 +733,7 @@
 
 int NETEQTEST_RTPpacket::calcPadLength(int i_P) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     if (i_P == 1)
     {
         /* Padding exists. Find out how many bytes the padding consists of. */
@@ -745,7 +745,7 @@
         else
         {
             /* even number of bytes => last byte in lower byte */
-            return ((WebRtc_UWord16) rtp_data[(_datagramLen >> 1) - 1]) >> 8;
+            return ((uint16_t) rtp_data[(_datagramLen >> 1) - 1]) >> 8;
         }
     }
     return 0;
@@ -760,9 +760,9 @@
         return;
     }
 
-    WebRtc_UWord8 *readDataPtr = _payloadPtr;
-    WebRtc_UWord8 *writeDataPtr = _payloadPtr;
-    WebRtc_UWord8 *slaveData = slaveRtp->_payloadPtr;
+    uint8_t *readDataPtr = _payloadPtr;
+    uint8_t *writeDataPtr = _payloadPtr;
+    uint8_t *slaveData = slaveRtp->_payloadPtr;
 
     while (readDataPtr - _payloadPtr < _payloadLen)
     {
@@ -829,8 +829,8 @@
 
     parseHeader();
 
-    WebRtc_UWord8* ptr = payload();
-    WebRtc_UWord8* payloadEndPtr = ptr + payloadLen();
+    uint8_t* ptr = payload();
+    uint8_t* payloadEndPtr = ptr + payloadLen();
     int num_encodings = 0;
     int total_len = 0;
 
@@ -841,7 +841,7 @@
         {
             // Header found.
             red.payloadType = ptr[0] & 0x7F;
-            WebRtc_UWord32 offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
+            uint32_t offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
             red.sequenceNumber = sequenceNumber();
             red.timeStamp = timeStamp() - offset;
             red.markerBit = markerBit();
@@ -873,6 +873,6 @@
 
     for (int i = 0; i < _payloadLen; ++i)
     {
-        _payloadPtr[i] = static_cast<WebRtc_UWord8>(rand());
+        _payloadPtr[i] = static_cast<uint8_t>(rand());
     }
 }
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 8ce04c5..fda7b95 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -35,32 +35,32 @@
     virtual int readFromFile(FILE *fp);
     int readFixedFromFile(FILE *fp, size_t len);
     virtual int writeToFile(FILE *fp);
-    void blockPT(WebRtc_UWord8 pt);
+    void blockPT(uint8_t pt);
     void selectSSRC(uint32_t ssrc);
-    //WebRtc_Word16 payloadType();
+    //int16_t payloadType();
     void parseHeader();
     void parseHeader(WebRtcNetEQ_RTPInfo & rtpInfo);
     WebRtcNetEQ_RTPInfo const * RTPinfo() const;
-    WebRtc_UWord8 * datagram() const;
-    WebRtc_UWord8 * payload() const;
-    WebRtc_Word16 payloadLen();
-    WebRtc_Word16 dataLen() const;
+    uint8_t * datagram() const;
+    uint8_t * payload() const;
+    int16_t payloadLen();
+    int16_t dataLen() const;
     bool isParsed() const;
     bool isLost() const;
-    WebRtc_UWord32 time() const { return _receiveTime; };
+    uint32_t time() const { return _receiveTime; };
 
-    WebRtc_UWord8  payloadType() const;
-    WebRtc_UWord16 sequenceNumber() const;
-    WebRtc_UWord32 timeStamp() const;
-    WebRtc_UWord32 SSRC() const;
-    WebRtc_UWord8  markerBit() const;
+    uint8_t  payloadType() const;
+    uint16_t sequenceNumber() const;
+    uint32_t timeStamp() const;
+    uint32_t SSRC() const;
+    uint8_t  markerBit() const;
 
-    int setPayloadType(WebRtc_UWord8 pt);
-    int setSequenceNumber(WebRtc_UWord16 sn);
-    int setTimeStamp(WebRtc_UWord32 ts);
-    int setSSRC(WebRtc_UWord32 ssrc);
-    int setMarkerBit(WebRtc_UWord8 mb);
-    void setTime(WebRtc_UWord32 receiveTime) { _receiveTime = receiveTime; };
+    int setPayloadType(uint8_t pt);
+    int setSequenceNumber(uint16_t sn);
+    int setTimeStamp(uint32_t ts);
+    int setSSRC(uint32_t ssrc);
+    int setMarkerBit(uint8_t mb);
+    void setTime(uint32_t receiveTime) { _receiveTime = receiveTime; };
 
     int setRTPheader(const WebRtcNetEQ_RTPInfo *RTPinfo);
 
@@ -70,16 +70,16 @@
 
     void scramblePayload(void);
 
-    WebRtc_UWord8 *       _datagram;
-    WebRtc_UWord8 *       _payloadPtr;
+    uint8_t *       _datagram;
+    uint8_t *       _payloadPtr;
     int                 _memSize;
-    WebRtc_Word16         _datagramLen;
-    WebRtc_Word16         _payloadLen;
+    int16_t         _datagramLen;
+    int16_t         _payloadLen;
     WebRtcNetEQ_RTPInfo  _rtpInfo;
     bool                _rtpParsed;
-    WebRtc_UWord32        _receiveTime;
+    uint32_t        _receiveTime;
     bool                _lost;
-    std::map<WebRtc_UWord8, bool> _blockList;
+    std::map<uint8_t, bool> _blockList;
     uint32_t            _selectSSRC;
     bool                _filterSSRC;
 
@@ -92,12 +92,12 @@
     int calcHeaderLength(int i_X, int i_CC) const;
 
 private:
-    void makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType,
-                       WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp,
-                       WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const;
-    WebRtc_UWord16 parseRTPheader(WebRtcNetEQ_RTPInfo *RTPinfo,
-                                  WebRtc_UWord8 **payloadPtr = NULL) const;
-    WebRtc_UWord16 parseRTPheader(WebRtc_UWord8 **payloadPtr = NULL)
+    void makeRTPheader(unsigned char* rtp_data, uint8_t payloadType,
+                       uint16_t seqNo, uint32_t timestamp,
+                       uint32_t ssrc, uint8_t markerBit) const;
+    uint16_t parseRTPheader(WebRtcNetEQ_RTPInfo *RTPinfo,
+                            uint8_t **payloadPtr = NULL) const;
+    uint16_t parseRTPheader(uint8_t **payloadPtr = NULL)
         { return parseRTPheader(&_rtpInfo, payloadPtr);};
     int calcPadLength(int i_P) const;
     void splitStereoSample(NETEQTEST_RTPpacket* slaveRtp, int stride);
diff --git a/modules/audio_coding/neteq/test/NetEqRTPplay.cc b/modules/audio_coding/neteq/test/NetEqRTPplay.cc
index c8b742d..3ad8e05 100644
--- a/modules/audio_coding/neteq/test/NetEqRTPplay.cc
+++ b/modules/audio_coding/neteq/test/NetEqRTPplay.cc
@@ -113,19 +113,19 @@
 /* Function declarations */
 /*************************/
 
-void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen);
-int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime);
-void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d);
+void stereoInterleave(int16_t *data, int16_t totalLen);
+int getNextRecoutTime(FILE *fp, uint32_t *nextTime);
+void getNextExtraDelay(FILE *fp, uint32_t *t, int *d);
 bool splitStereo(NETEQTEST_RTPpacket* rtp, NETEQTEST_RTPpacket* rtpSlave,
-                 const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
-                 const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
+                 const int16_t *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
+                 const int16_t *cngPtype, int noOfCngCodecs,
                  bool *isStereo);
-void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders);
-int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec);
-void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber);
-void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders);
+void parsePtypeFile(FILE *ptypeFile, std::map<uint8_t, decoderStruct>* decoders);
+int populateUsedCodec(std::map<uint8_t, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec);
+void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<uint8_t, decoderStruct>* decoders, int channelNumber);
+void free_coders(std::map<uint8_t, decoderStruct> & decoders);
 int doAPItest();
-bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode);
+bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<uint8_t, decoderStruct> & decoders, enum stereoModes *stereoMode);
 
 
 
@@ -133,13 +133,13 @@
 /* Global variables */
 /********************/
 
-WebRtc_Word16 NetEqPacketBuffer[MAX_NETEQ_BUFFERSIZE>>1];
-WebRtc_Word16 NetEqPacketBufferSlave[MAX_NETEQ_BUFFERSIZE>>1];
+int16_t NetEqPacketBuffer[MAX_NETEQ_BUFFERSIZE>>1];
+int16_t NetEqPacketBufferSlave[MAX_NETEQ_BUFFERSIZE>>1];
 
 #ifdef NETEQ_DELAY_LOGGING
 extern "C" {
     FILE *delay_fid2;   /* file pointer */
-    WebRtc_UWord32 tot_received_packets=0;
+    uint32_t tot_received_packets=0;
 }
 #endif
 
@@ -147,8 +147,8 @@
 extern char BUILD_DATE;
 #endif
 
-WebRtc_UWord32 writtenSamples = 0;
-WebRtc_UWord32 simClock=0;
+uint32_t writtenSamples = 0;
+uint32_t simClock=0;
 
 int main(int argc, char* argv[])
 {
@@ -157,8 +157,8 @@
     enum WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd-1];
     int noOfCodecs;
     int ok;
-    WebRtc_Word16 out_data[640*2];
-    WebRtc_Word16 outLen, writeLen;
+    int16_t out_data[640*2];
+    int16_t outLen, writeLen;
     int fs = 8000;
     WebRtcNetEQ_RTCPStat RTCPstat;
 #ifdef WIN32
@@ -182,7 +182,7 @@
     bool rtpOnly = false;
     int packetLen = 0;
     int packetCount = 0;
-    std::map<WebRtc_UWord8, decoderStruct> decoders;
+    std::map<uint8_t, decoderStruct> decoders;
     bool dummyRtp = false;
     bool noDecode = false;
     bool filterSSRC = false;
@@ -489,7 +489,7 @@
     for (int i = 0; i < numInst; i++)
     {
         // create memory, allocate, initialize, and allocate packet buffer memory
-        NetEQvector.push_back (new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, static_cast<WebRtc_UWord16>(fs), kTCPLargeJitter));
+        NetEQvector.push_back (new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, static_cast<uint16_t>(fs), kTCPLargeJitter));
 
         createAndInsertDecoders (NetEQvector[i], &decoders, i /* channel */);
 
@@ -519,10 +519,10 @@
 
 
 #ifdef ZERO_TS_START
-    WebRtc_UWord32 firstTS = rtp->timeStamp();
+    uint32_t firstTS = rtp->timeStamp();
     rtp->setTimeStamp(0);
 #else
-    WebRtc_UWord32 firstTS = 0;
+    uint32_t firstTS = 0;
 #endif
 
     // check stereo mode
@@ -535,8 +535,8 @@
     }
 
 #ifdef PLAY_CLEAN
-    WebRtc_UWord32 prevTS = rtp->timeStamp();
-    WebRtc_UWord32 currTS, prev_time;
+    uint32_t prevTS = rtp->timeStamp();
+    uint32_t currTS, prev_time;
 #endif
 
 #ifdef JUNK_DATA
@@ -552,7 +552,7 @@
     }
 #endif
 
-    WebRtc_UWord32 nextRecoutTime;
+    uint32_t nextRecoutTime;
     int lastRecout = getNextRecoutTime(recoutTimes, &nextRecoutTime); // does nothing if recoutTimes == NULL
 
     if (recoutTimes)
@@ -560,9 +560,9 @@
     else
         simClock = rtp->time(); // start immediately with first packet
 
-    WebRtc_UWord32 start_clock = simClock;
+    uint32_t start_clock = simClock;
 
-    WebRtc_UWord32 nextExtraDelayTime;
+    uint32_t nextExtraDelayTime;
     int extraDelay = -1;
     getNextExtraDelay(extraDelays, &nextExtraDelayTime, &extraDelay);
 
@@ -660,7 +660,7 @@
             if (stereoMode > stereoModeMono)
             {
                 // stereo
-                WebRtc_Word16 tempLen;
+                int16_t tempLen;
                 tempLen = NetEQvector[0]->recOut( out_data, msInfo ); // master
                 outLen = NetEQvector[1]->recOut( &out_data[tempLen], msInfo ); // slave
 
@@ -703,7 +703,7 @@
     if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
       return -1;
     }
-    if (fwrite(&tot_received_packets, sizeof(WebRtc_UWord32),
+    if (fwrite(&tot_received_packets, sizeof(uint32_t),
                1, delay_fid2) != 1) {
       return -1;
     }
@@ -767,8 +767,8 @@
 /****************/
 
 bool splitStereo(NETEQTEST_RTPpacket* rtp, NETEQTEST_RTPpacket* rtpSlave,
-                 const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
-                 const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
+                 const int16_t *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
+                 const int16_t *cngPtype, int noOfCngCodecs,
                  bool *isStereo)
 {
 
@@ -784,7 +784,7 @@
     }
 
     // check payload type
-    WebRtc_Word16 ptype = rtp->payloadType();
+    int16_t ptype = rtp->payloadType();
 
     // is this a cng payload?
     for (int k = 0; k < noOfCngCodecs; k++) {
@@ -824,19 +824,19 @@
 
 }
 
-void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen)
+void stereoInterleave(int16_t *data, int16_t totalLen)
 {
     int k;
 
     for(k = totalLen/2; k < totalLen; k++) {
-        WebRtc_Word16 temp = data[k];
-        memmove(&data[2*k - totalLen + 2], &data[2*k - totalLen + 1], (totalLen - k -1) *  sizeof(WebRtc_Word16));
+        int16_t temp = data[k];
+        memmove(&data[2*k - totalLen + 2], &data[2*k - totalLen + 1], (totalLen - k -1) *  sizeof(int16_t));
         data[2*k - totalLen + 1] = temp;
     }
 }
 
 
-int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime) {
+int getNextRecoutTime(FILE *fp, uint32_t *nextTime) {
 
     float tempTime;
 
@@ -846,7 +846,7 @@
 
     if (fread(&tempTime, sizeof(float), 1, fp) != 0) {
         // not end of file
-        *nextTime = (WebRtc_UWord32) tempTime;
+        *nextTime = (uint32_t) tempTime;
         return 0;
     }
 
@@ -856,7 +856,7 @@
     return 1;
 }
 
-void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d) {
+void getNextExtraDelay(FILE *fp, uint32_t *t, int *d) {
 
     float temp[2];
 
@@ -867,7 +867,7 @@
 
     if (fread(&temp, sizeof(float), 2, fp) != 0) {
         // not end of file
-        *t = (WebRtc_UWord32) temp[0];
+        *t = (uint32_t) temp[0];
         *d = (int) temp[1];
         return;
     }
@@ -879,7 +879,7 @@
 }
 
 
-void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders)
+void parsePtypeFile(FILE *ptypeFile, std::map<uint8_t, decoderStruct>* decoders)
 {
     int n, pt;
     char codec[100];
@@ -1264,7 +1264,7 @@
             }
 
             // insert into codecs map
-            (*decoders)[static_cast<WebRtc_UWord8>(pt)] = tempDecoder;
+            (*decoders)[static_cast<uint8_t>(pt)] = tempDecoder;
 
         }
 
@@ -1274,7 +1274,7 @@
 }
 
 
-bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode)
+bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<uint8_t, decoderStruct> & decoders, enum stereoModes *stereoMode)
 {
         if (decoders.count(rtp.payloadType()) > 0
             && decoders[rtp.payloadType()].codec != kDecoderRED
@@ -1292,11 +1292,11 @@
 }
 
 
-int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec)
+int populateUsedCodec(std::map<uint8_t, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec)
 {
     int numCodecs = 0;
 
-    std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+    std::map<uint8_t, decoderStruct>::iterator it;
 
     it = decoders->begin();
 
@@ -1310,9 +1310,9 @@
 }
 
 
-void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber)
+void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<uint8_t, decoderStruct>* decoders, int channelNumber)
 {
-    std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+    std::map<uint8_t, decoderStruct>::iterator it;
 
     for (it = decoders->begin(); it != decoders->end();  it++)
     {
@@ -1320,7 +1320,7 @@
             ((*it).second.stereo > stereoModeMono ))
         {
             // create decoder instance
-            WebRtc_UWord8 pt = static_cast<WebRtc_UWord8>( (*it).first );
+            uint8_t pt = static_cast<uint8_t>( (*it).first );
             NETEQTEST_Decoder **dec = &((*it).second.decoder[channelNumber]);
             enum WebRtcNetEQDecoder type = (*it).second.codec;
 
@@ -1524,7 +1524,7 @@
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
             case kDecoderCNG:
-                *dec = new decoder_CNG( pt, static_cast<WebRtc_UWord16>((*it).second.fs) );
+                *dec = new decoder_CNG( pt, static_cast<uint16_t>((*it).second.fs) );
                 break;
 #endif
 #ifdef CODEC_ISACLC
@@ -1577,9 +1577,9 @@
 }
 
 
-void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders)
+void free_coders(std::map<uint8_t, decoderStruct> & decoders)
 {
-    std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+    std::map<uint8_t, decoderStruct>::iterator it;
 
     for (it = decoders.begin(); it != decoders.end();  it++)
     {
@@ -1608,7 +1608,7 @@
     int NetEqBufferMaxPackets, BufferSizeInBytes;
     WebRtcNetEQ_CodecDef codecInst;
     WebRtcNetEQ_RTCPStat RTCPstat;
-    WebRtc_UWord32 timestamp;
+    uint32_t timestamp;
     int memorySize;
     int ok;
     int overhead_bytes;
@@ -1635,7 +1635,7 @@
     CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbReset(inst))
     CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
     CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbRemove(inst, usedCodec))
-    WebRtc_Word16 temp1, temp2;
+    int16_t temp1, temp2;
     CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetSizeInfo(inst, &temp1, &temp2))
     CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetCodecInfo(inst, 0, &usedCodec))
 
@@ -1646,8 +1646,8 @@
     WebRtcNetEQOutputType temptype;
     CHECK_MINUS_ONE(WebRtcNetEQ_GetSpeechOutputType(inst, &temptype))
 
-    WebRtc_UWord8 tempFlags;
-    WebRtc_UWord16 utemp1, utemp2;
+    uint8_t tempFlags;
+    uint16_t utemp1, utemp2;
     CHECK_MINUS_ONE(WebRtcNetEQ_VQmonRecOutStatistics(inst, &utemp1, &utemp2, &tempFlags))
     CHECK_MINUS_ONE(WebRtcNetEQ_VQmonGetRxStatistics(inst, &utemp1, &utemp2))
 
diff --git a/modules/audio_coding/neteq/test/RTPencode.cc b/modules/audio_coding/neteq/test/RTPencode.cc
index 32b0bcc..45313be 100644
--- a/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/modules/audio_coding/neteq/test/RTPencode.cc
@@ -73,12 +73,12 @@
 int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
 void defineCodecs(enum WebRtcNetEQDecoder *usedCodec, int *noOfCodecs );
 int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels);
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc);
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
-                        int seqNo, WebRtc_UWord32 ssrc);
+int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc);
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
+                        int seqNo, uint32_t ssrc);
 int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples);
+void stereoDeInterleave(int16_t* audioSamples, int numSamples);
 void stereoInterleave(unsigned char* data, int dataLen, int stride);
 
 /*********************/
@@ -199,11 +199,11 @@
 #endif
 #ifdef CODEC_AMR
 	AMR_encinst_t *AMRenc_inst[2];
-	WebRtc_Word16		  AMR_bitrate;
+	int16_t		  AMR_bitrate;
 #endif
 #ifdef CODEC_AMRWB
 	AMRWB_encinst_t *AMRWBenc_inst[2];
-	WebRtc_Word16		  AMRWB_bitrate;
+	int16_t		  AMRWB_bitrate;
 #endif
 #ifdef CODEC_ILBC
 	iLBC_encinst_t *iLBCenc_inst[2];
@@ -250,21 +250,21 @@
 	int useVAD, vad;
     int useRed=0;
 	int len, enc_len;
-	WebRtc_Word16 org_data[4000];
+	int16_t org_data[4000];
 	unsigned char rtp_data[8000];
-	WebRtc_Word16 seqNo=0xFFF;
-	WebRtc_UWord32 ssrc=1235412312;
-	WebRtc_UWord32 timestamp=0xAC1245;
-        WebRtc_UWord16 length, plen;
-	WebRtc_UWord32 offset;
+	int16_t seqNo=0xFFF;
+	uint32_t ssrc=1235412312;
+	uint32_t timestamp=0xAC1245;
+        uint16_t length, plen;
+	uint32_t offset;
 	double sendtime = 0;
     int red_PT[2] = {0};
-    WebRtc_UWord32 red_TS[2] = {0};
-    WebRtc_UWord16 red_len[2] = {0};
+    uint32_t red_TS[2] = {0};
+    uint16_t red_len[2] = {0};
     int RTPheaderLen=12;
 	unsigned char red_data[8000];
 #ifdef INSERT_OLD_PACKETS
-	WebRtc_UWord16 old_length, old_plen;
+	uint16_t old_length, old_plen;
 	int old_enc_len;
 	int first_old_packet=1;
 	unsigned char old_rtp_data[8000];
@@ -273,7 +273,7 @@
 #ifdef INSERT_DTMF_PACKETS
 	int NTone = 1;
 	int DTMFfirst = 1;
-	WebRtc_UWord32 DTMFtimestamp;
+	uint32_t DTMFtimestamp;
     bool dtmfSent = false;
 #endif
     bool usingStereo = false;
@@ -553,7 +553,7 @@
 	/* write file header */
 	//fprintf(out_file, "#!RTPencode%s\n", "1.0");
 	fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
-	WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
+	uint32_t dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
         if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
           return -1;
         }
@@ -615,7 +615,7 @@
             /* write RTP packet to file */
             length = htons(12 + enc_len + 8);
             plen = htons(12 + enc_len);
-            offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
+            offset = (uint32_t) sendtime; //(timestamp/(fs/1000));
             offset = htonl(offset);
             if (fwrite(&length, 2, 1, out_file) != 1) {
               return -1;
@@ -710,7 +710,7 @@
 			/* write RTP packet to file */
                           length = htons(12 + enc_len + 8);
                           plen = htons(12 + enc_len);
-                          offset = (WebRtc_UWord32) sendtime;
+                          offset = (uint32_t) sendtime;
                           //(timestamp/(fs/1000));
                           offset = htonl(offset);
                           if (fwrite(&length, 2, 1, out_file) != 1) {
@@ -778,7 +778,7 @@
                 if(usedCodec==kDecoderISAC)
                 {
                     assert(!usingStereo); // Cannot handle stereo yet
-                    red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (WebRtc_Word16*)red_data);
+                    red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (int16_t*)red_data);
                 }
                 else
                 {
@@ -1726,13 +1726,13 @@
 
 
 
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , 
+int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , 
 						  int * vad, int useVAD, int bitrate, int numChannels){
 
 	short cdlen = 0;
-	WebRtc_Word16 *tempdata;
+	int16_t *tempdata;
 	static int first_cng=1;
-	WebRtc_Word16 tempLen;
+	int16_t tempLen;
 
 	*vad =1;
 
@@ -1797,91 +1797,91 @@
         /* Encode with the selected coder type */
         if (coder==kDecoderPCMu) { /*g711 u-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+            cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (int16_t*) encoded);
 #endif
         }  
         else if (coder==kDecoderPCMa) { /*g711 A-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+            cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (int16_t*) encoded);
         }
 #endif
 #ifdef CODEC_PCM16B
         else if ((coder==kDecoderPCM16B)||(coder==kDecoderPCM16Bwb)||
             (coder==kDecoderPCM16Bswb32kHz)||(coder==kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
-                cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (WebRtc_Word16*) encoded);
+                cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (int16_t*) encoded);
             }
 #endif
 #ifdef CODEC_G722
         else if (coder==kDecoderG722) { /*g722 */
-            cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (int16_t*)encoded);
             cdlen=frameLen>>1;
         }
 #endif
 #ifdef CODEC_G722_1_16
         else if (coder==kDecoderG722_1_16) { /* g722.1 16kbit/s mode */
-            cdlen=WebRtcG7221_Encode16((G722_1_16_encinst_t*)G722_1_16enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221_Encode16((G722_1_16_encinst_t*)G722_1_16enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G722_1_24
         else if (coder==kDecoderG722_1_24) { /* g722.1 24kbit/s mode*/
-            cdlen=WebRtcG7221_Encode24((G722_1_24_encinst_t*)G722_1_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221_Encode24((G722_1_24_encinst_t*)G722_1_24enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G722_1_32
         else if (coder==kDecoderG722_1_32) { /* g722.1 32kbit/s mode */
-            cdlen=WebRtcG7221_Encode32((G722_1_32_encinst_t*)G722_1_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221_Encode32((G722_1_32_encinst_t*)G722_1_32enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G722_1C_24
         else if (coder==kDecoderG722_1C_24) { /* g722.1 32 kHz 24kbit/s mode*/
-            cdlen=WebRtcG7221C_Encode24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221C_Encode24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G722_1C_32
         else if (coder==kDecoderG722_1C_32) { /* g722.1 32 kHz 32kbit/s mode */
-            cdlen=WebRtcG7221C_Encode32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221C_Encode32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G722_1C_48
         else if (coder==kDecoderG722_1C_48) { /* g722.1 32 kHz 48kbit/s mode */
-            cdlen=WebRtcG7221C_Encode48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG7221C_Encode48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_G729
         else if (coder==kDecoderG729) { /*g729 */
-            WebRtc_Word16 dataPos=0;
-            WebRtc_Word16 len=0;
+            int16_t dataPos=0;
+            int16_t len=0;
             cdlen = 0;
             for (dataPos=0;dataPos<frameLen;dataPos+=80) {
-                len=WebRtcG729_Encode(G729enc_inst[k], &indata[dataPos], 80, (WebRtc_Word16*)(&encoded[cdlen]));
+                len=WebRtcG729_Encode(G729enc_inst[k], &indata[dataPos], 80, (int16_t*)(&encoded[cdlen]));
                 cdlen += len;
             }
         }
 #endif
 #ifdef CODEC_G729_1
         else if (coder==kDecoderG729_1) { /*g729.1 */
-            WebRtc_Word16 dataPos=0;
-            WebRtc_Word16 len=0;
+            int16_t dataPos=0;
+            int16_t len=0;
             cdlen = 0;
             for (dataPos=0;dataPos<frameLen;dataPos+=160) {
-                len=WebRtcG7291_Encode(G729_1_inst[k], &indata[dataPos], (WebRtc_Word16*)(&encoded[cdlen]), bitrate, frameLen/320 /* num 20ms frames*/);
+                len=WebRtcG7291_Encode(G729_1_inst[k], &indata[dataPos], (int16_t*)(&encoded[cdlen]), bitrate, frameLen/320 /* num 20ms frames*/);
                 cdlen += len;
             }
         }
 #endif
 #ifdef CODEC_AMR
         else if (coder==kDecoderAMR) { /*AMR */
-            cdlen=WebRtcAmr_Encode(AMRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMR_bitrate);
+            cdlen=WebRtcAmr_Encode(AMRenc_inst[k], indata, frameLen, (int16_t*)encoded, AMR_bitrate);
         }
 #endif
 #ifdef CODEC_AMRWB
         else if (coder==kDecoderAMRWB) { /*AMR-wb */
-            cdlen=WebRtcAmrWb_Encode(AMRWBenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMRWB_bitrate);
+            cdlen=WebRtcAmrWb_Encode(AMRWBenc_inst[k], indata, frameLen, (int16_t*)encoded, AMRWB_bitrate);
         }
 #endif
 #ifdef CODEC_ILBC
         else if (coder==kDecoderILBC) { /*iLBC */
-            cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(WebRtc_Word16*)encoded);
+            cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(int16_t*)encoded);
         }
 #endif
 #if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
@@ -1890,9 +1890,9 @@
             cdlen=0;
             while (cdlen<=0) {
 #ifdef CODEC_ISAC /* floating point */
-                cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
 #else /* fixed point */
-                cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
 #endif
                 noOfCalls++;
             }
@@ -1903,7 +1903,7 @@
             int noOfCalls=0;
             cdlen=0;
             while (cdlen<=0) {
-                cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(int16_t*)encoded);
                 noOfCalls++;
             }
         }
@@ -1915,14 +1915,14 @@
             while (cdlen <= 0) {
                 cdlen = WebRtcIsac_Encode(ISACFB_inst[k],
                                           &indata[noOfCalls * 480],
-                                          (WebRtc_Word16*)encoded);
+                                          (int16_t*)encoded);
                 noOfCalls++;
             }
         }
 #endif
 #ifdef CODEC_GSMFR
         else if (coder==kDecoderGSMFR) { /* GSM FR */
-            cdlen=WebRtcGSMFR_Encode(GSMFRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcGSMFR_Encode(GSMFRenc_inst[k], indata, frameLen, (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_SPEEX_8
@@ -1937,7 +1937,7 @@
                 printf("Error encoding speex frame!\n");
                 exit(0);
             }
-            cdlen=WebRtcSpeex_GetBitstream(SPEEX8enc_inst[k], (WebRtc_Word16*)encoded);
+            cdlen=WebRtcSpeex_GetBitstream(SPEEX8enc_inst[k], (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_SPEEX_16
@@ -1952,7 +1952,7 @@
                 printf("Error encoding speex frame!\n");
                 exit(0);
             }
-            cdlen=WebRtcSpeex_GetBitstream(SPEEX16enc_inst[k], (WebRtc_Word16*)encoded);
+            cdlen=WebRtcSpeex_GetBitstream(SPEEX16enc_inst[k], (int16_t*)encoded);
         }
 #endif
 #ifdef CODEC_CELT_32
@@ -1982,7 +1982,7 @@
 
 
 
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc){
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc){
 			
 			rtp_data[0]=(unsigned char)0x80;
 			rtp_data[1]=(unsigned char)(payloadType & 0xFF);
@@ -2002,13 +2002,13 @@
 }
 
 
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
-                        int seqNo, WebRtc_UWord32 ssrc)
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
+                        int seqNo, uint32_t ssrc)
 {
 
     int i;
     unsigned char *rtpPointer;
-    WebRtc_UWord16 offset;
+    uint16_t offset;
 
     /* first create "standard" RTP header */
     makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
@@ -2018,7 +2018,7 @@
     /* add one sub-header for each redundant payload (not the primary) */
     for(i=0; i<numPayloads-1; i++) {                                            /* |0 1 2 3 4 5 6 7| */
         if(blockLen[i] > 0) {
-            offset = (WebRtc_UWord16) (timestamp[numPayloads-1] - timestamp[i]);
+            offset = (uint16_t) (timestamp[numPayloads-1] - timestamp[i]);
 
             rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F|   block PT  | */
             rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF);             /* |  timestamp-   | */
@@ -2056,22 +2056,22 @@
 	return(4);
 }
 
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples)
+void stereoDeInterleave(int16_t* audioSamples, int numSamples)
 {
 
-    WebRtc_Word16 *tempVec;
-    WebRtc_Word16 *readPtr, *writeL, *writeR;
+    int16_t *tempVec;
+    int16_t *readPtr, *writeL, *writeR;
 
     if (numSamples <= 0)
         return;
 
-    tempVec = (WebRtc_Word16 *) malloc(sizeof(WebRtc_Word16) * numSamples);
+    tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples);
     if (tempVec == NULL) {
         printf("Error allocating memory\n");
         exit(0);
     }
 
-    memcpy(tempVec, audioSamples, numSamples*sizeof(WebRtc_Word16));
+    memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t));
 
     writeL = audioSamples;
     writeR = &audioSamples[numSamples/2];
diff --git a/modules/audio_coding/neteq/test/RTPjitter.cc b/modules/audio_coding/neteq/test/RTPjitter.cc
index e3270be..3f94350 100644
--- a/modules/audio_coding/neteq/test/RTPjitter.cc
+++ b/modules/audio_coding/neteq/test/RTPjitter.cc
@@ -35,7 +35,7 @@
 
 struct arr_time {
 	float time;
-	WebRtc_UWord32 ix;
+	uint32_t ix;
 };
 
 int filelen(FILE *fid)
@@ -64,8 +64,8 @@
 	char			firstline[FIRSTLINELEN];
 	unsigned char	*rtp_vec = NULL, **packet_ptr, *temp_packet;
 	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
-	WebRtc_UWord16			len;
-	WebRtc_UWord32			*offset;
+	uint16_t			len;
+	uint32_t			*offset;
 
 /* check number of parameters */
 	if (argc != 4) {
@@ -131,9 +131,9 @@
 	// read all RTP packets into vector
 	rtp_len=0;
 	Npack=0;
-	len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
+	len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
 	while(len==2) {
-		len = ntohs(*((WebRtc_UWord16 *)(rtp_vec + rtp_len)));
+		len = ntohs(*((uint16_t *)(rtp_vec + rtp_len)));
 		rtp_len += 2;
 		if(fread(&rtp_vec[rtp_len], sizeof(unsigned char), len-2, in_file)!=(unsigned) (len-2)) {
 			fprintf(stderr,"Error: currupt packet length\n");
@@ -141,7 +141,7 @@
 		}
 		rtp_len += len-2;
 		Npack++;
-		len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
+		len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
 	}
 
 	packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
@@ -149,7 +149,7 @@
 	packet_ptr[0]=rtp_vec;
 	k=1;
 	while(k<Npack) {
-		len = ntohs(*((WebRtc_UWord16 *) packet_ptr[k-1]));
+		len = ntohs(*((uint16_t *) packet_ptr[k-1]));
 		packet_ptr[k]=packet_ptr[k-1]+len;
 		k++;
 	}
@@ -157,20 +157,20 @@
 	for(k=0; k<dat_len && k<Npack; k++) {
 		if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){ 
 			temp_packet = packet_ptr[time_vec[k].ix];
-			offset = (WebRtc_UWord32 *) (temp_packet+4);
+			offset = (uint32_t *) (temp_packet+4);
 			if ( time_vec[k].time >= 0 ) {
-				*offset = htonl((WebRtc_UWord32) time_vec[k].time);
+				*offset = htonl((uint32_t) time_vec[k].time);
 			}
 			else {
-				*offset = htonl((WebRtc_UWord32) 0);
+				*offset = htonl((uint32_t) 0);
 				fprintf(stderr, "Warning: negative receive time in dat file transformed to 0.\n");
 			}
 
 			// write packet to file
                         if (fwrite(temp_packet, sizeof(unsigned char),
-                                   ntohs(*((WebRtc_UWord16*) temp_packet)),
+                                   ntohs(*((uint16_t*) temp_packet)),
                                    out_file) !=
-                            ntohs(*((WebRtc_UWord16*) temp_packet))) {
+                            ntohs(*((uint16_t*) temp_packet))) {
                           return -1;
                         }
 		}
diff --git a/modules/audio_coding/neteq/test/RTPtimeshift.cc b/modules/audio_coding/neteq/test/RTPtimeshift.cc
index 66a00bd..1a0fc67 100644
--- a/modules/audio_coding/neteq/test/RTPtimeshift.cc
+++ b/modules/audio_coding/neteq/test/RTPtimeshift.cc
@@ -60,9 +60,9 @@
   }
 
   // get new start TS and start SeqNo from arguments
-  WebRtc_UWord32 TSdiff = atoi(argv[3]) - packet.timeStamp();
-  WebRtc_UWord16 SNdiff = 0;
-  WebRtc_UWord32 ATdiff = 0;
+  uint32_t TSdiff = atoi(argv[3]) - packet.timeStamp();
+  uint16_t SNdiff = 0;
+  uint32_t ATdiff = 0;
   if (argc > 4) {
     SNdiff = atoi(argv[4]) - packet.sequenceNumber();
     if (argc > 5) {
diff --git a/modules/audio_coding/neteq/unmute_signal.c b/modules/audio_coding/neteq/unmute_signal.c
index ee9daa8..3128f21 100644
--- a/modules/audio_coding/neteq/unmute_signal.c
+++ b/modules/audio_coding/neteq/unmute_signal.c
@@ -17,25 +17,25 @@
 #include "signal_processing_library.h"
 
 
-void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
-                              WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
-                              WebRtc_Word16 N)
+void WebRtcNetEQ_UnmuteSignal(int16_t *pw16_inVec, int16_t *startMuteFact,
+                              int16_t *pw16_outVec, int16_t unmuteFact,
+                              int16_t N)
 {
     int i;
-    WebRtc_UWord16 w16_tmp;
-    WebRtc_Word32 w32_tmp;
+    uint16_t w16_tmp;
+    int32_t w32_tmp;
 
-    w16_tmp = (WebRtc_UWord16) *startMuteFact;
-    w32_tmp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,6) + 32;
+    w16_tmp = (uint16_t) *startMuteFact;
+    w32_tmp = WEBRTC_SPL_LSHIFT_W32((int32_t)w16_tmp,6) + 32;
     for (i = 0; i < N; i++)
     {
         pw16_outVec[i]
-            = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp, pw16_inVec[i]) + 8192) >> 14);
+            = (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_tmp, pw16_inVec[i]) + 8192) >> 14);
         w32_tmp += unmuteFact;
         w32_tmp = WEBRTC_SPL_MAX(0, w32_tmp);
-        w16_tmp = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 6); /* 20 - 14 = 6 */
+        w16_tmp = (uint16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 6); /* 20 - 14 = 6 */
         w16_tmp = WEBRTC_SPL_MIN(16384, w16_tmp);
     }
-    *startMuteFact = (WebRtc_Word16) w16_tmp;
+    *startMuteFact = (int16_t) w16_tmp;
 }
 
diff --git a/modules/audio_coding/neteq/webrtc_neteq.c b/modules/audio_coding/neteq/webrtc_neteq.c
index 477b0d0..38b3ce4 100644
--- a/modules/audio_coding/neteq/webrtc_neteq.c
+++ b/modules/audio_coding/neteq/webrtc_neteq.c
@@ -30,7 +30,7 @@
     if ((macroExpr) == -1) { \
     (macroInstPtr)->ErrorCode = - (NETEQ_OTHER_ERROR); \
     } else { \
-    (macroInstPtr)->ErrorCode = -((WebRtc_Word16) (macroExpr)); \
+    (macroInstPtr)->ErrorCode = -((int16_t) (macroExpr)); \
     } \
     return(-1); \
     } }
@@ -280,7 +280,7 @@
 /* Assign functions (create not allowed in order to avoid malloc in lib) */
 int WebRtcNetEQ_AssignSize(int *sizeinbytes)
 {
-    *sizeinbytes = (sizeof(MainInst_t) * 2) / sizeof(WebRtc_Word16);
+    *sizeinbytes = (sizeof(MainInst_t) * 2) / sizeof(int16_t);
     return (0);
 }
 
@@ -294,8 +294,8 @@
     WebRtcSpl_Init();
 
     /* Clear memory */
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) NetEqMainInst, 0,
-        (sizeof(MainInst_t) / sizeof(WebRtc_Word16)));
+    WebRtcSpl_MemSetW16((int16_t*) NetEqMainInst, 0,
+        (sizeof(MainInst_t) / sizeof(int16_t)));
     ok = WebRtcNetEQ_McuReset(&NetEqMainInst->MCUinst);
     if (ok != 0)
     {
@@ -361,7 +361,7 @@
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
     if (NetEqMainInst == NULL) return (-1);
     ok = WebRtcNetEQ_PacketBufferInit(&NetEqMainInst->MCUinst.PacketBuffer_inst,
-        MaxNoOfPackets, (WebRtc_Word16*) NETEQ_Buffer_Addr, (sizeinbytes >> 1));
+        MaxNoOfPackets, (int16_t*) NETEQ_Buffer_Addr, (sizeinbytes >> 1));
     if (ok != 0)
     {
         NetEqMainInst->ErrorCode = -ok;
@@ -390,7 +390,7 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs)
+int WebRtcNetEQ_Init(void *inst, uint16_t fs)
 {
     int ok = 0;
 
@@ -450,8 +450,8 @@
     WebRtcNetEQ_RTCPInit(&(NetEqMainInst->MCUinst.RTCP_inst), 0);
 
     /* set BufferStat struct to zero */
-    WebRtcSpl_MemSetW16((WebRtc_Word16*) &(NetEqMainInst->MCUinst.BufferStat_inst), 0,
-        sizeof(BufstatsInst_t) / sizeof(WebRtc_Word16));
+    WebRtcSpl_MemSetW16((int16_t*) &(NetEqMainInst->MCUinst.BufferStat_inst), 0,
+        sizeof(BufstatsInst_t) / sizeof(int16_t));
 
     /* reset automode */
     WebRtcNetEQ_ResetAutomode(&(NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst),
@@ -607,8 +607,8 @@
     return (0);
 }
 
-int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
-                                   WebRtc_Word16 *MaxEntries)
+int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, int16_t *UsedEntries,
+                                   int16_t *MaxEntries)
 {
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
     if (NetEqMainInst == NULL) return (-1);
@@ -617,7 +617,7 @@
     return (0);
 }
 
-int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
+int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, int16_t Entry,
                                     enum WebRtcNetEQDecoder *codec)
 {
     int i;
@@ -668,7 +668,7 @@
     if (NetEqMainInst == NULL) return (-1);
 
     /* check if currently used codec is being removed */
-    if (NetEqMainInst->MCUinst.current_Codec == (WebRtc_Word16) codec)
+    if (NetEqMainInst->MCUinst.current_Codec == (int16_t) codec)
     {
         /* set function pointers to NULL to prevent RecOut from using the codec */
         NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
@@ -695,8 +695,8 @@
  * Real-time functions
  */
 
-int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
-                      WebRtc_UWord32 uw32_timeRec)
+int WebRtcNetEQ_RecIn(void *inst, int16_t *p_w16datagramstart, int16_t w16_RTPlen,
+                      uint32_t uw32_timeRec)
 {
     int ok = 0;
     RTPPacket_t RTPpacket;
@@ -745,8 +745,8 @@
  *                            -1 - Error
  */
 int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
-                               const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
-                               WebRtc_UWord32 uw32_timeRec)
+                               const uint8_t *payloadPtr, int16_t payloadLenBytes,
+                               uint32_t uw32_timeRec)
 {
     int ok = 0;
     RTPPacket_t RTPpacket;
@@ -769,7 +769,7 @@
     RTPpacket.seqNumber = rtpInfo->sequenceNumber;
     RTPpacket.timeStamp = rtpInfo->timeStamp;
     RTPpacket.ssrc = rtpInfo->SSRC;
-    RTPpacket.payload = (const WebRtc_Word16*) payloadPtr;
+    RTPpacket.payload = (const int16_t*) payloadPtr;
     RTPpacket.payloadLen = payloadLenBytes;
     RTPpacket.starts_byte1 = 0;
 
@@ -782,7 +782,7 @@
     return (ok);
 }
 
-int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
+int WebRtcNetEQ_RecOut(void *inst, int16_t *pw16_outData, int16_t *pw16_len)
 {
     int ok = 0;
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
@@ -836,9 +836,9 @@
  *						  -1 - Error
  */
 
-int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
-                                  WebRtc_Word16 *pw16_len, void *msInfo,
-                                  WebRtc_Word16 isMaster)
+int WebRtcNetEQ_RecOutMasterSlave(void *inst, int16_t *pw16_outData,
+                                  int16_t *pw16_len, void *msInfo,
+                                  int16_t isMaster)
 {
 #ifndef NETEQ_STEREO
     /* Stereo not supported */
@@ -914,8 +914,8 @@
 }
 
 /* Special RecOut that does not do any decoding. */
-int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
-                               WebRtc_Word16 *pw16_len)
+int WebRtcNetEQ_RecOutNoDecode(void *inst, int16_t *pw16_outData,
+                               int16_t *pw16_len)
 {
     int ok = 0;
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
@@ -999,7 +999,7 @@
     return (ok);
 }
 
-int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp)
+int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, uint32_t *timestamp)
 {
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
     if (NetEqMainInst == NULL) return (-1);
@@ -1101,18 +1101,18 @@
 #define WEBRTC_NETEQ_CONCEALMENTFLAG_SUPRESS    0x04
 #define WEBRTC_NETEQ_CONCEALMENTFLAG_CNGACTIVE  0x80
 
-int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
-                                      WebRtc_UWord16 *concealedVoiceDurationMs,
-                                      WebRtc_UWord8 *concealedVoiceFlags)
+int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, uint16_t *validVoiceDurationMs,
+                                      uint16_t *concealedVoiceDurationMs,
+                                      uint8_t *concealedVoiceFlags)
 {
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-    WebRtc_Word16 fs_mult;
-    WebRtc_Word16 ms_lost;
+    int16_t fs_mult;
+    int16_t ms_lost;
     if (NetEqMainInst == NULL) return (-1);
     fs_mult = WebRtcSpl_DivW32W16ResW16(NetEqMainInst->MCUinst.fs, 8000);
 
     ms_lost = WebRtcSpl_DivW32W16ResW16(
-        (WebRtc_Word32) NetEqMainInst->DSPinst.w16_concealedTS, (WebRtc_Word16) (8 * fs_mult));
+        (int32_t) NetEqMainInst->DSPinst.w16_concealedTS, (int16_t) (8 * fs_mult));
     if (ms_lost > NetEqMainInst->DSPinst.millisecondsPerCall) ms_lost
         = NetEqMainInst->DSPinst.millisecondsPerCall;
 
@@ -1131,8 +1131,8 @@
     return (0);
 }
 
-int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
-                                      WebRtc_UWord8 *adaptationRate)
+int WebRtcNetEQ_VQmonGetConfiguration(void *inst, uint16_t *absMaxDelayMs,
+                                      uint8_t *adaptationRate)
 {
     /* Dummy check the inst, just to avoid compiler warnings. */
     if (inst == NULL)
@@ -1146,13 +1146,13 @@
     return (0);
 }
 
-int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
-                                     WebRtc_UWord16 *maxDelayMs)
+int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, uint16_t *avgDelayMs,
+                                     uint16_t *maxDelayMs)
 {
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
     if (NetEqMainInst == NULL) return (-1);
-    *avgDelayMs = (WebRtc_UWord16) (NetEqMainInst->MCUinst.BufferStat_inst.avgDelayMsQ8 >> 8);
-    *maxDelayMs = (WebRtc_UWord16) NetEqMainInst->MCUinst.BufferStat_inst.maxDelayMs;
+    *avgDelayMs = (uint16_t) (NetEqMainInst->MCUinst.BufferStat_inst.avgDelayMsQ8 >> 8);
+    *maxDelayMs = (uint16_t) NetEqMainInst->MCUinst.BufferStat_inst.maxDelayMs;
     return (0);
 }
 
@@ -1166,8 +1166,8 @@
 
 {
 
-    WebRtc_UWord16 tempU16;
-    WebRtc_UWord32 tempU32, tempU32_2;
+    uint16_t tempU16;
+    uint32_t tempU32, tempU32_2;
     int numShift;
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
 
@@ -1182,7 +1182,7 @@
 
     if (NetEqMainInst->MCUinst.fs != 0)
     {
-        WebRtc_Word32 temp32;
+        int32_t temp32;
         /* Query packet buffer for number of samples. */
         temp32 = WebRtcNetEQ_PacketBufferGetSize(
             &NetEqMainInst->MCUinst.PacketBuffer_inst,
@@ -1190,13 +1190,13 @@
 
         /* Divide by sample rate.
          * Calculate temp32 * 1000 / fs to get result in ms. */
-        stats->currentBufferSize = (WebRtc_UWord16)
+        stats->currentBufferSize = (uint16_t)
             WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
 
         /* Add number of samples yet to play in sync buffer. */
-        temp32 = (WebRtc_Word32) (NetEqMainInst->DSPinst.endPosition -
+        temp32 = (int32_t) (NetEqMainInst->DSPinst.endPosition -
             NetEqMainInst->DSPinst.curPosition);
-        stats->currentBufferSize += (WebRtc_UWord16)
+        stats->currentBufferSize += (uint16_t)
             WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
     }
     else
@@ -1213,11 +1213,11 @@
     {
         /* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
         stats->preferredBufferSize
-            = (WebRtc_UWord16) WEBRTC_SPL_MUL_16_16(
-                (WebRtc_Word16) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */
+            = (uint16_t) WEBRTC_SPL_MUL_16_16(
+                (int16_t) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */
                 WebRtcSpl_DivW32W16ResW16(
-                    (WebRtc_Word32) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */
-                    WebRtcSpl_DivW32W16ResW16( (WebRtc_Word32) NetEqMainInst->MCUinst.fs, (WebRtc_Word16) 1000 ) /* samples per ms */
+                    (int32_t) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */
+                    WebRtcSpl_DivW32W16ResW16( (int32_t) NetEqMainInst->MCUinst.fs, (int16_t) 1000 ) /* samples per ms */
                 ) );
 
         /* add extra delay */
@@ -1281,13 +1281,13 @@
                 tempU32 >>= 1; /* right-shift 1 step */
                 numShift--; /* compensate in numerator */
             }
-            tempU16 = (WebRtc_UWord16) tempU32;
+            tempU16 = (uint16_t) tempU32;
 
             /* do the shift of numerator */
             tempU32
-                = WEBRTC_SPL_SHIFT_W32( (WebRtc_UWord32) NetEqMainInst->MCUinst.lostTS, numShift);
+                = WEBRTC_SPL_SHIFT_W32( (uint32_t) NetEqMainInst->MCUinst.lostTS, numShift);
 
-            stats->currentPacketLossRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+            stats->currentPacketLossRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32,
                 tempU16);
         }
     }
@@ -1307,7 +1307,7 @@
 
     /* number of discarded samples */
     tempU32_2
-        = WEBRTC_SPL_MUL_16_U16( (WebRtc_Word16) NetEqMainInst->MCUinst.PacketBuffer_inst.packSizeSamples,
+        = WEBRTC_SPL_MUL_16_U16( (int16_t) NetEqMainInst->MCUinst.PacketBuffer_inst.packSizeSamples,
             NetEqMainInst->MCUinst.PacketBuffer_inst.discardedPackets);
 
     if (tempU32_2 == 0)
@@ -1344,12 +1344,12 @@
                 tempU32 >>= 1; /* right-shift 1 step */
                 numShift--; /* compensate in numerator */
             }
-            tempU16 = (WebRtc_UWord16) tempU32;
+            tempU16 = (uint16_t) tempU32;
 
             /* do the shift of numerator */
             tempU32 = WEBRTC_SPL_SHIFT_W32( tempU32_2, numShift);
 
-            stats->currentDiscardRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
+            stats->currentDiscardRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, tempU16);
         }
     }
     else
@@ -1400,13 +1400,13 @@
                 tempU32 >>= 1; /* right-shift 1 step */
                 numShift--; /* compensate in numerator */
             }
-            tempU16 = (WebRtc_UWord16) tempU32;
+            tempU16 = (uint16_t) tempU32;
 
             /* do the shift of numerator */
             tempU32
                 = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.accelerateLength, numShift);
 
-            stats->currentAccelerateRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+            stats->currentAccelerateRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32,
                 tempU16);
         }
     }
@@ -1454,13 +1454,13 @@
                 tempU32 >>= 1; /* right-shift 1 step */
                 numShift--; /* compensate in numerator */
             }
-            tempU16 = (WebRtc_UWord16) tempU32;
+            tempU16 = (uint16_t) tempU32;
 
             /* do the shift of numerator */
             tempU32
                 = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.expandLength, numShift);
 
-            stats->currentExpandRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
+            stats->currentExpandRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, tempU16);
         }
     }
     else
@@ -1507,13 +1507,13 @@
                 tempU32 >>= 1; /* right-shift 1 step */
                 numShift--; /* compensate in numerator */
             }
-            tempU16 = (WebRtc_UWord16) tempU32;
+            tempU16 = (uint16_t) tempU32;
 
             /* do the shift of numerator */
             tempU32
                 = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.preemptiveLength, numShift);
 
-            stats->currentPreemptiveRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+            stats->currentPreemptiveRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32,
                 tempU16);
         }
     }
diff --git a/modules/audio_coding/neteq4/neteq_impl.cc b/modules/audio_coding/neteq4/neteq_impl.cc
index 08a8272..7c485c6 100644
--- a/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/modules/audio_coding/neteq4/neteq_impl.cc
@@ -548,7 +548,7 @@
     }
 
     // Update statistics.
-    if ((WebRtc_Word32) (main_header.timestamp - timestamp_) >= 0 &&
+    if ((int32_t) (main_header.timestamp - timestamp_) >= 0 &&
         !new_codec_) {
       // Only update statistics if incoming packet is not older than last played
       // out packet, and if new codec flag is not set.
diff --git a/modules/audio_coding/neteq4/preemptive_expand.cc b/modules/audio_coding/neteq4/preemptive_expand.cc
index 9929078..8522368 100644
--- a/modules/audio_coding/neteq4/preemptive_expand.cc
+++ b/modules/audio_coding/neteq4/preemptive_expand.cc
@@ -53,7 +53,7 @@
 }
 
 PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
-    const WebRtc_Word16 *input, int input_length, size_t peak_index,
+    const int16_t *input, int input_length, size_t peak_index,
     int16_t best_correlation, bool active_speech,
     AudioMultiVector<int16_t>* output) const {
   // Pre-calculate common multiplication with |fs_mult_|.
diff --git a/modules/audio_coding/neteq4/preemptive_expand.h b/modules/audio_coding/neteq4/preemptive_expand.h
index 96a8511..9e22f47 100644
--- a/modules/audio_coding/neteq4/preemptive_expand.h
+++ b/modules/audio_coding/neteq4/preemptive_expand.h
@@ -43,7 +43,7 @@
   // |output|. The number of samples added through time-stretching is
   // is provided in the output |length_change_samples|. The method returns
   // the outcome of the operation as an enumerator value.
-  ReturnCodes Process(const WebRtc_Word16 *pw16_decoded,
+  ReturnCodes Process(const int16_t *pw16_decoded,
                       int len,
                       int oldDataLen,
                       AudioMultiVector<int16_t>* output,
@@ -59,7 +59,7 @@
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching.
   virtual ReturnCodes CheckCriteriaAndStretch(
-      const WebRtc_Word16 *pw16_decoded, int len, size_t w16_bestIndex,
+      const int16_t *pw16_decoded, int len, size_t w16_bestIndex,
       int16_t w16_bestCorr, bool w16_VAD,
       AudioMultiVector<int16_t>* output) const;
 
diff --git a/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.cc b/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.cc
index b215bd3..5c47f2c 100644
--- a/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.cc
+++ b/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.cc
@@ -27,8 +27,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     if (fread(&length, 2, 1, fp) == 0)
     {
@@ -50,10 +50,10 @@
         return -1;
     }
     // Store in local variable until we have passed the reset below.
-    WebRtc_UWord32 receiveTime = ntohl(offset);
+    uint32_t receiveTime = ntohl(offset);
 
     // Use length here because a plen of 0 specifies rtcp.
-    length = (WebRtc_UWord16) (length - _kRDHeaderLen);
+    length = (uint16_t) (length - _kRDHeaderLen);
 
     // check buffer size
     if (_datagram && _memSize < length)
@@ -63,7 +63,7 @@
 
     if (!_datagram)
     {
-        _datagram = new WebRtc_UWord8[length];
+        _datagram = new uint8_t[length];
         _memSize = length;
     }
     memset(_datagram, 0, length);
@@ -142,8 +142,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     // length including RTPplay header
     length = htons(_datagramLen + _kRDHeaderLen);
diff --git a/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.cc b/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.cc
index 49c8bc9..c6d3270 100644
--- a/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.cc
+++ b/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.cc
@@ -103,8 +103,8 @@
         return(-1);
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     if (fread(&length,2,1,fp)==0)
     {
@@ -125,10 +125,10 @@
         reset();
         return(-1);
     }
-    WebRtc_UWord32 receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
+    uint32_t receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
 
     // Use length here because a plen of 0 specifies rtcp
-    length = (WebRtc_UWord16) (length - _kRDHeaderLen);
+    length = (uint16_t) (length - _kRDHeaderLen);
 
     // check buffer size
     if (_datagram && _memSize < length)
@@ -138,7 +138,7 @@
 
     if (!_datagram)
     {
-        _datagram = new WebRtc_UWord8[length];
+        _datagram = new uint8_t[length];
         _memSize = length;
     }
 
@@ -178,7 +178,7 @@
 
     if (!_datagram)
     {
-        _datagram = new WebRtc_UWord8[length];
+        _datagram = new uint8_t[length];
         _memSize = length;
     }
 
@@ -210,8 +210,8 @@
         return -1;
     }
 
-    WebRtc_UWord16 length, plen;
-    WebRtc_UWord32 offset;
+    uint16_t length, plen;
+    uint32_t offset;
 
     // length including RTPplay header
     length = htons(_datagramLen + _kRDHeaderLen);
@@ -247,7 +247,7 @@
 }
 
 
-void NETEQTEST_RTPpacket::blockPT(WebRtc_UWord8 pt)
+void NETEQTEST_RTPpacket::blockPT(uint8_t pt)
 {
     _blockList[pt] = true;
 }
@@ -300,7 +300,7 @@
     }
 }
 
-WebRtc_UWord8 * NETEQTEST_RTPpacket::datagram() const
+uint8_t * NETEQTEST_RTPpacket::datagram() const
 {
     if (_datagramLen > 0)
     {
@@ -312,7 +312,7 @@
     }
 }
 
-WebRtc_UWord8 * NETEQTEST_RTPpacket::payload() const
+uint8_t * NETEQTEST_RTPpacket::payload() const
 {
     if (_payloadLen > 0)
     {
@@ -324,13 +324,13 @@
     }
 }
 
-WebRtc_Word16 NETEQTEST_RTPpacket::payloadLen()
+int16_t NETEQTEST_RTPpacket::payloadLen()
 {
     parseHeader();
     return _payloadLen;
 }
 
-WebRtc_Word16 NETEQTEST_RTPpacket::dataLen() const
+int16_t NETEQTEST_RTPpacket::dataLen() const
 {
     return _datagramLen;
 }
@@ -345,7 +345,7 @@
     return _lost;
 }
 
-WebRtc_UWord8  NETEQTEST_RTPpacket::payloadType() const
+uint8_t  NETEQTEST_RTPpacket::payloadType() const
 {
     webrtc::WebRtcRTPHeader tempRTPinfo;
 
@@ -361,7 +361,7 @@
     return tempRTPinfo.header.payloadType;
 }
 
-WebRtc_UWord16 NETEQTEST_RTPpacket::sequenceNumber() const
+uint16_t NETEQTEST_RTPpacket::sequenceNumber() const
 {
     webrtc::WebRtcRTPHeader tempRTPinfo;
 
@@ -377,7 +377,7 @@
     return tempRTPinfo.header.sequenceNumber;
 }
 
-WebRtc_UWord32 NETEQTEST_RTPpacket::timeStamp() const
+uint32_t NETEQTEST_RTPpacket::timeStamp() const
 {
     webrtc::WebRtcRTPHeader tempRTPinfo;
 
@@ -393,7 +393,7 @@
     return tempRTPinfo.header.timestamp;
 }
 
-WebRtc_UWord32 NETEQTEST_RTPpacket::SSRC() const
+uint32_t NETEQTEST_RTPpacket::SSRC() const
 {
     webrtc::WebRtcRTPHeader tempRTPinfo;
 
@@ -409,7 +409,7 @@
     return tempRTPinfo.header.ssrc;
 }
 
-WebRtc_UWord8  NETEQTEST_RTPpacket::markerBit() const
+uint8_t  NETEQTEST_RTPpacket::markerBit() const
 {
     webrtc::WebRtcRTPHeader tempRTPinfo;
 
@@ -427,7 +427,7 @@
 
 
 
-int NETEQTEST_RTPpacket::setPayloadType(WebRtc_UWord8 pt)
+int NETEQTEST_RTPpacket::setPayloadType(uint8_t pt)
 {
 
     if (_datagramLen < 12)
@@ -446,7 +446,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setSequenceNumber(WebRtc_UWord16 sn)
+int NETEQTEST_RTPpacket::setSequenceNumber(uint16_t sn)
 {
 
     if (_datagramLen < 12)
@@ -466,7 +466,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setTimeStamp(WebRtc_UWord32 ts)
+int NETEQTEST_RTPpacket::setTimeStamp(uint32_t ts)
 {
 
     if (_datagramLen < 12)
@@ -488,7 +488,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setSSRC(WebRtc_UWord32 ssrc)
+int NETEQTEST_RTPpacket::setSSRC(uint32_t ssrc)
 {
 
     if (_datagramLen < 12)
@@ -510,7 +510,7 @@
 
 }
 
-int NETEQTEST_RTPpacket::setMarkerBit(WebRtc_UWord8 mb)
+int NETEQTEST_RTPpacket::setMarkerBit(uint8_t mb)
 {
 
     if (_datagramLen < 12)
@@ -619,7 +619,7 @@
 }
 
 
-void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const
+void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqNo, uint32_t timestamp, uint32_t ssrc, uint8_t markerBit) const
 {
     rtp_data[0]=(unsigned char)0x80;
     if (markerBit)
@@ -646,11 +646,11 @@
     rtp_data[11]=(unsigned char)(ssrc & 0xFF);
 }
 
-WebRtc_UWord16
+uint16_t
     NETEQTEST_RTPpacket::parseRTPheader(webrtc::WebRtcRTPHeader* RTPinfo,
-                                        WebRtc_UWord8 **payloadPtr) const
+                                        uint8_t **payloadPtr) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     int i_P, i_X, i_CC;
 
     assert(_datagramLen >= 12);
@@ -662,60 +662,60 @@
 
     if (payloadPtr)
     {
-        *payloadPtr = (WebRtc_UWord8*) &rtp_data[i_startPosition >> 1];
+        *payloadPtr = (uint8_t*) &rtp_data[i_startPosition >> 1];
     }
 
-    return (WebRtc_UWord16) (_datagramLen - i_startPosition - i_padlength);
+    return (uint16_t) (_datagramLen - i_startPosition - i_padlength);
 }
 
 
 void NETEQTEST_RTPpacket::parseBasicHeader(webrtc::WebRtcRTPHeader* RTPinfo,
                                            int *i_P, int *i_X, int *i_CC) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     if (_datagramLen < 12)
     {
         assert(false);
         return;
     }
 
-    *i_P=(((WebRtc_UWord16)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
-    *i_X=(((WebRtc_UWord16)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
-    *i_CC=(WebRtc_UWord16)(rtp_data[0] & 0xF); /* Get the CC number  */
+    *i_P=(((uint16_t)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
+    *i_X=(((uint16_t)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
+    *i_CC=(uint16_t)(rtp_data[0] & 0xF); /* Get the CC number  */
     /* Get the marker bit */
-    RTPinfo->header.markerBit = (WebRtc_UWord8) ((rtp_data[0] >> 15) & 0x01);
+    RTPinfo->header.markerBit = (uint8_t) ((rtp_data[0] >> 15) & 0x01);
     /* Get the coder type */
-    RTPinfo->header.payloadType = (WebRtc_UWord8) ((rtp_data[0] >> 8) & 0x7F);
+    RTPinfo->header.payloadType = (uint8_t) ((rtp_data[0] >> 8) & 0x7F);
     /* Get the packet number */
     RTPinfo->header.sequenceNumber =
-        ((( ((WebRtc_UWord16)rtp_data[1]) >> 8) & 0xFF) |
-        ( ((WebRtc_UWord16)(rtp_data[1] & 0xFF)) << 8));
+        ((( ((uint16_t)rtp_data[1]) >> 8) & 0xFF) |
+        ( ((uint16_t)(rtp_data[1] & 0xFF)) << 8));
     /* Get timestamp */
-    RTPinfo->header.timestamp = ((((WebRtc_UWord16)rtp_data[2]) & 0xFF) << 24) |
-        ((((WebRtc_UWord16)rtp_data[2]) & 0xFF00) << 8) |
-        ((((WebRtc_UWord16)rtp_data[3]) >> 8) & 0xFF) |
-        ((((WebRtc_UWord16)rtp_data[3]) & 0xFF) << 8);
+    RTPinfo->header.timestamp = ((((uint16_t)rtp_data[2]) & 0xFF) << 24) |
+        ((((uint16_t)rtp_data[2]) & 0xFF00) << 8) |
+        ((((uint16_t)rtp_data[3]) >> 8) & 0xFF) |
+        ((((uint16_t)rtp_data[3]) & 0xFF) << 8);
     /* Get the SSRC */
-    RTPinfo->header.ssrc = ((((WebRtc_UWord16)rtp_data[4]) & 0xFF) << 24) |
-        ((((WebRtc_UWord16)rtp_data[4]) & 0xFF00) << 8) |
-        ((((WebRtc_UWord16)rtp_data[5]) >> 8) & 0xFF) |
-        ((((WebRtc_UWord16)rtp_data[5]) & 0xFF) << 8);
+    RTPinfo->header.ssrc = ((((uint16_t)rtp_data[4]) & 0xFF) << 24) |
+        ((((uint16_t)rtp_data[4]) & 0xFF00) << 8) |
+        ((((uint16_t)rtp_data[5]) >> 8) & 0xFF) |
+        ((((uint16_t)rtp_data[5]) & 0xFF) << 8);
 }
 
 int NETEQTEST_RTPpacket::calcHeaderLength(int i_X, int i_CC) const
 {
     int i_extlength = 0;
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
 
     if (i_X == 1)
     {
         // Extension header exists.
-        // Find out how many WebRtc_Word32 it consists of.
+        // Find out how many int32_t it consists of.
         assert(_datagramLen > 2 * (7 + 2 * i_CC));
         if (_datagramLen > 2 * (7 + 2 * i_CC))
         {
-            i_extlength = (((((WebRtc_UWord16) rtp_data[7 + 2 * i_CC]) >> 8)
-                & 0xFF) | (((WebRtc_UWord16) (rtp_data[7 + 2 * i_CC] & 0xFF))
+            i_extlength = (((((uint16_t) rtp_data[7 + 2 * i_CC]) >> 8)
+                & 0xFF) | (((uint16_t) (rtp_data[7 + 2 * i_CC] & 0xFF))
                 << 8)) + 1;
         }
     }
@@ -725,7 +725,7 @@
 
 int NETEQTEST_RTPpacket::calcPadLength(int i_P) const
 {
-    WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) _datagram;
+    int16_t *rtp_data = (int16_t *) _datagram;
     if (i_P == 1)
     {
         /* Padding exists. Find out how many bytes the padding consists of. */
@@ -737,7 +737,7 @@
         else
         {
             /* even number of bytes => last byte in lower byte */
-            return ((WebRtc_UWord16) rtp_data[(_datagramLen >> 1) - 1]) >> 8;
+            return ((uint16_t) rtp_data[(_datagramLen >> 1) - 1]) >> 8;
         }
     }
     return 0;
@@ -752,9 +752,9 @@
         return;
     }
 
-    WebRtc_UWord8 *readDataPtr = _payloadPtr;
-    WebRtc_UWord8 *writeDataPtr = _payloadPtr;
-    WebRtc_UWord8 *slaveData = slaveRtp->_payloadPtr;
+    uint8_t *readDataPtr = _payloadPtr;
+    uint8_t *writeDataPtr = _payloadPtr;
+    uint8_t *slaveData = slaveRtp->_payloadPtr;
 
     while (readDataPtr - _payloadPtr < _payloadLen)
     {
@@ -821,8 +821,8 @@
 
     parseHeader();
 
-    WebRtc_UWord8* ptr = payload();
-    WebRtc_UWord8* payloadEndPtr = ptr + payloadLen();
+    uint8_t* ptr = payload();
+    uint8_t* payloadEndPtr = ptr + payloadLen();
     int num_encodings = 0;
     int total_len = 0;
 
@@ -833,7 +833,7 @@
         {
             // Header found.
             red.header.payloadType = ptr[0] & 0x7F;
-            WebRtc_UWord32 offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
+            uint32_t offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
             red.header.sequenceNumber = sequenceNumber();
             red.header.timestamp = timeStamp() - offset;
             red.header.markerBit = markerBit();
@@ -865,6 +865,6 @@
 
     for (int i = 0; i < _payloadLen; ++i)
     {
-        _payloadPtr[i] = static_cast<WebRtc_UWord8>(rand());
+        _payloadPtr[i] = static_cast<uint8_t>(rand());
     }
 }
diff --git a/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h b/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h
index 1199d97..684a160 100644
--- a/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h
+++ b/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h
@@ -35,31 +35,31 @@
     virtual int readFromFile(FILE *fp);
     int readFixedFromFile(FILE *fp, size_t len);
     virtual int writeToFile(FILE *fp);
-    void blockPT(WebRtc_UWord8 pt);
-    //WebRtc_Word16 payloadType();
+    void blockPT(uint8_t pt);
+    //int16_t payloadType();
     void parseHeader();
     void parseHeader(webrtc::WebRtcRTPHeader* rtp_header);
     const webrtc::WebRtcRTPHeader* RTPinfo() const;
-    WebRtc_UWord8 * datagram() const;
-    WebRtc_UWord8 * payload() const;
-    WebRtc_Word16 payloadLen();
-    WebRtc_Word16 dataLen() const;
+    uint8_t * datagram() const;
+    uint8_t * payload() const;
+    int16_t payloadLen();
+    int16_t dataLen() const;
     bool isParsed() const;
     bool isLost() const;
-    WebRtc_UWord32 time() const { return _receiveTime; };
+    uint32_t time() const { return _receiveTime; };
 
-    WebRtc_UWord8  payloadType() const;
-    WebRtc_UWord16 sequenceNumber() const;
-    WebRtc_UWord32 timeStamp() const;
-    WebRtc_UWord32 SSRC() const;
-    WebRtc_UWord8  markerBit() const;
+    uint8_t  payloadType() const;
+    uint16_t sequenceNumber() const;
+    uint32_t timeStamp() const;
+    uint32_t SSRC() const;
+    uint8_t  markerBit() const;
 
-    int setPayloadType(WebRtc_UWord8 pt);
-    int setSequenceNumber(WebRtc_UWord16 sn);
-    int setTimeStamp(WebRtc_UWord32 ts);
-    int setSSRC(WebRtc_UWord32 ssrc);
-    int setMarkerBit(WebRtc_UWord8 mb);
-    void setTime(WebRtc_UWord32 receiveTime) { _receiveTime = receiveTime; };
+    int setPayloadType(uint8_t pt);
+    int setSequenceNumber(uint16_t sn);
+    int setTimeStamp(uint32_t ts);
+    int setSSRC(uint32_t ssrc);
+    int setMarkerBit(uint8_t mb);
+    void setTime(uint32_t receiveTime) { _receiveTime = receiveTime; };
 
     int setRTPheader(const webrtc::WebRtcRTPHeader* RTPinfo);
 
@@ -69,16 +69,16 @@
 
     void scramblePayload(void);
 
-    WebRtc_UWord8 *       _datagram;
-    WebRtc_UWord8 *       _payloadPtr;
+    uint8_t *       _datagram;
+    uint8_t *       _payloadPtr;
     int                 _memSize;
-    WebRtc_Word16         _datagramLen;
-    WebRtc_Word16         _payloadLen;
+    int16_t         _datagramLen;
+    int16_t         _payloadLen;
     webrtc::WebRtcRTPHeader _rtpInfo;
     bool                _rtpParsed;
-    WebRtc_UWord32        _receiveTime;
+    uint32_t        _receiveTime;
     bool                _lost;
-    std::map<WebRtc_UWord8, bool> _blockList;
+    std::map<uint8_t, bool> _blockList;
 
 protected:
     static const int _kRDHeaderLen;
@@ -89,12 +89,12 @@
     int calcHeaderLength(int i_X, int i_CC) const;
 
 private:
-    void makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType,
-                       WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp,
-                       WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const;
-    WebRtc_UWord16 parseRTPheader(webrtc::WebRtcRTPHeader* RTPinfo,
-                                  WebRtc_UWord8 **payloadPtr = NULL) const;
-    WebRtc_UWord16 parseRTPheader(WebRtc_UWord8 **payloadPtr = NULL)
+    void makeRTPheader(unsigned char* rtp_data, uint8_t payloadType,
+                       uint16_t seqNo, uint32_t timestamp,
+                       uint32_t ssrc, uint8_t markerBit) const;
+    uint16_t parseRTPheader(webrtc::WebRtcRTPHeader* RTPinfo,
+                            uint8_t **payloadPtr = NULL) const;
+    uint16_t parseRTPheader(uint8_t **payloadPtr = NULL)
         { return parseRTPheader(&_rtpInfo, payloadPtr);};
     int calcPadLength(int i_P) const;
     void splitStereoSample(NETEQTEST_RTPpacket* slaveRtp, int stride);
diff --git a/modules/audio_coding/neteq4/test/RTPencode.cc b/modules/audio_coding/neteq4/test/RTPencode.cc
index c79d5db..2c38556 100644
--- a/modules/audio_coding/neteq4/test/RTPencode.cc
+++ b/modules/audio_coding/neteq4/test/RTPencode.cc
@@ -75,12 +75,12 @@
 int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
 void defineCodecs(webrtc::NetEqDecoder *usedCodec, int *noOfCodecs );
 int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels);
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc);
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
-                        int seqNo, WebRtc_UWord32 ssrc);
+int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc);
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
+                        int seqNo, uint32_t ssrc);
 int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples);
+void stereoDeInterleave(int16_t* audioSamples, int numSamples);
 void stereoInterleave(unsigned char* data, int dataLen, int stride);
 
 /*********************/
@@ -201,11 +201,11 @@
 #endif
 #ifdef CODEC_AMR
 	AMR_encinst_t *AMRenc_inst[2];
-	WebRtc_Word16		  AMR_bitrate;
+	int16_t		  AMR_bitrate;
 #endif
 #ifdef CODEC_AMRWB
 	AMRWB_encinst_t *AMRWBenc_inst[2];
-	WebRtc_Word16		  AMRWB_bitrate;
+	int16_t		  AMRWB_bitrate;
 #endif
 #ifdef CODEC_ILBC
 	iLBC_encinst_t *iLBCenc_inst[2];
@@ -249,21 +249,21 @@
 	int useVAD, vad;
     int useRed=0;
 	int len, enc_len;
-	WebRtc_Word16 org_data[4000];
+	int16_t org_data[4000];
 	unsigned char rtp_data[8000];
-	WebRtc_Word16 seqNo=0xFFF;
-	WebRtc_UWord32 ssrc=1235412312;
-	WebRtc_UWord32 timestamp=0xAC1245;
-        WebRtc_UWord16 length, plen;
-	WebRtc_UWord32 offset;
+	int16_t seqNo=0xFFF;
+	uint32_t ssrc=1235412312;
+	uint32_t timestamp=0xAC1245;
+        uint16_t length, plen;
+	uint32_t offset;
 	double sendtime = 0;
     int red_PT[2] = {0};
-    WebRtc_UWord32 red_TS[2] = {0};
-    WebRtc_UWord16 red_len[2] = {0};
+    uint32_t red_TS[2] = {0};
+    uint16_t red_len[2] = {0};
     int RTPheaderLen=12;
 	unsigned char red_data[8000];
 #ifdef INSERT_OLD_PACKETS
-	WebRtc_UWord16 old_length, old_plen;
+	uint16_t old_length, old_plen;
 	int old_enc_len;
 	int first_old_packet=1;
 	unsigned char old_rtp_data[8000];
@@ -272,7 +272,7 @@
 #ifdef INSERT_DTMF_PACKETS
 	int NTone = 1;
 	int DTMFfirst = 1;
-	WebRtc_UWord32 DTMFtimestamp;
+	uint32_t DTMFtimestamp;
     bool dtmfSent = false;
 #endif
     bool usingStereo = false;
@@ -530,7 +530,7 @@
 	/* write file header */
 	//fprintf(out_file, "#!RTPencode%s\n", "1.0");
 	fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
-	WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
+	uint32_t dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
         if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
           return -1;
         }
@@ -592,7 +592,7 @@
             /* write RTP packet to file */
             length = htons(12 + enc_len + 8);
             plen = htons(12 + enc_len);
-            offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
+            offset = (uint32_t) sendtime; //(timestamp/(fs/1000));
             offset = htonl(offset);
             if (fwrite(&length, 2, 1, out_file) != 1) {
               return -1;
@@ -687,7 +687,7 @@
 			/* write RTP packet to file */
                           length = htons(12 + enc_len + 8);
                           plen = htons(12 + enc_len);
-                          offset = (WebRtc_UWord32) sendtime;
+                          offset = (uint32_t) sendtime;
                           //(timestamp/(fs/1000));
                           offset = htonl(offset);
                           if (fwrite(&length, 2, 1, out_file) != 1) {
@@ -755,7 +755,7 @@
                 if(usedCodec==webrtc::kDecoderISAC)
                 {
                     assert(!usingStereo); // Cannot handle stereo yet
-                    red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (WebRtc_Word16*)red_data);
+                    red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (int16_t*)red_data);
                 }
                 else
                 {
@@ -1532,13 +1532,13 @@
 
 
 
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , 
+int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , 
 						  int * vad, int useVAD, int bitrate, int numChannels){
 
 	short cdlen = 0;
-	WebRtc_Word16 *tempdata;
+	int16_t *tempdata;
 	static int first_cng=1;
-	WebRtc_Word16 tempLen;
+	int16_t tempLen;
 
 	*vad =1;
 
@@ -1601,29 +1601,29 @@
         /* Encode with the selected coder type */
         if (coder==webrtc::kDecoderPCMu) { /*g711 u-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+            cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (int16_t*) encoded);
 #endif
         }  
         else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+            cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (int16_t*) encoded);
         }
 #endif
 #ifdef CODEC_PCM16B
         else if ((coder==webrtc::kDecoderPCM16B)||(coder==webrtc::kDecoderPCM16Bwb)||
             (coder==webrtc::kDecoderPCM16Bswb32kHz)||(coder==webrtc::kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
-                cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (WebRtc_Word16*) encoded);
+                cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (int16_t*) encoded);
             }
 #endif
 #ifdef CODEC_G722
         else if (coder==webrtc::kDecoderG722) { /*g722 */
-            cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (WebRtc_Word16*)encoded);
+            cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (int16_t*)encoded);
             cdlen=frameLen>>1;
         }
 #endif
 #ifdef CODEC_ILBC
         else if (coder==webrtc::kDecoderILBC) { /*iLBC */
-            cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(WebRtc_Word16*)encoded);
+            cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(int16_t*)encoded);
         }
 #endif
 #if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
@@ -1632,9 +1632,9 @@
             cdlen=0;
             while (cdlen<=0) {
 #ifdef CODEC_ISAC /* floating point */
-                cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
 #else /* fixed point */
-                cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
 #endif
                 noOfCalls++;
             }
@@ -1645,7 +1645,7 @@
             int noOfCalls=0;
             cdlen=0;
             while (cdlen<=0) {
-                cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(WebRtc_Word16*)encoded);
+                cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(int16_t*)encoded);
                 noOfCalls++;
             }
         }
@@ -1677,7 +1677,7 @@
 
 
 
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc){
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc){
 			
 			rtp_data[0]=(unsigned char)0x80;
 			rtp_data[1]=(unsigned char)(payloadType & 0xFF);
@@ -1697,13 +1697,13 @@
 }
 
 
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
-                        int seqNo, WebRtc_UWord32 ssrc)
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
+                        int seqNo, uint32_t ssrc)
 {
 
     int i;
     unsigned char *rtpPointer;
-    WebRtc_UWord16 offset;
+    uint16_t offset;
 
     /* first create "standard" RTP header */
     makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
@@ -1713,7 +1713,7 @@
     /* add one sub-header for each redundant payload (not the primary) */
     for(i=0; i<numPayloads-1; i++) {                                            /* |0 1 2 3 4 5 6 7| */
         if(blockLen[i] > 0) {
-            offset = (WebRtc_UWord16) (timestamp[numPayloads-1] - timestamp[i]);
+            offset = (uint16_t) (timestamp[numPayloads-1] - timestamp[i]);
 
             rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F|   block PT  | */
             rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF);             /* |  timestamp-   | */
@@ -1751,22 +1751,22 @@
 	return(4);
 }
 
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples)
+void stereoDeInterleave(int16_t* audioSamples, int numSamples)
 {
 
-    WebRtc_Word16 *tempVec;
-    WebRtc_Word16 *readPtr, *writeL, *writeR;
+    int16_t *tempVec;
+    int16_t *readPtr, *writeL, *writeR;
 
     if (numSamples <= 0)
         return;
 
-    tempVec = (WebRtc_Word16 *) malloc(sizeof(WebRtc_Word16) * numSamples);
+    tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples);
     if (tempVec == NULL) {
         printf("Error allocating memory\n");
         exit(0);
     }
 
-    memcpy(tempVec, audioSamples, numSamples*sizeof(WebRtc_Word16));
+    memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t));
 
     writeL = audioSamples;
     writeR = &audioSamples[numSamples/2];
diff --git a/modules/audio_coding/neteq4/test/RTPjitter.cc b/modules/audio_coding/neteq4/test/RTPjitter.cc
index 77b29ec..301ceb6 100644
--- a/modules/audio_coding/neteq4/test/RTPjitter.cc
+++ b/modules/audio_coding/neteq4/test/RTPjitter.cc
@@ -37,7 +37,7 @@
 
 struct arr_time {
 	float time;
-	WebRtc_UWord32 ix;
+	uint32_t ix;
 };
 
 int filelen(FILE *fid)
@@ -66,8 +66,8 @@
 	char			firstline[FIRSTLINELEN];
 	unsigned char	*rtp_vec = NULL, **packet_ptr, *temp_packet;
 	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
-	WebRtc_UWord16			len;
-	WebRtc_UWord32			*offset;
+	uint16_t			len;
+	uint32_t			*offset;
 
 /* check number of parameters */
 	if (argc != 4) {
@@ -133,9 +133,9 @@
 	// read all RTP packets into vector
 	rtp_len=0;
 	Npack=0;
-	len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
+	len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
 	while(len==2) {
-		len = ntohs(*((WebRtc_UWord16 *)(rtp_vec + rtp_len)));
+		len = ntohs(*((uint16_t *)(rtp_vec + rtp_len)));
 		rtp_len += 2;
 		if(fread(&rtp_vec[rtp_len], sizeof(unsigned char), len-2, in_file)!=(unsigned) (len-2)) {
 			fprintf(stderr,"Error: currupt packet length\n");
@@ -143,7 +143,7 @@
 		}
 		rtp_len += len-2;
 		Npack++;
-		len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
+		len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
 	}
 
 	packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
@@ -151,7 +151,7 @@
 	packet_ptr[0]=rtp_vec;
 	k=1;
 	while(k<Npack) {
-		len = ntohs(*((WebRtc_UWord16 *) packet_ptr[k-1]));
+		len = ntohs(*((uint16_t *) packet_ptr[k-1]));
 		packet_ptr[k]=packet_ptr[k-1]+len;
 		k++;
 	}
@@ -159,20 +159,20 @@
 	for(k=0; k<dat_len && k<Npack; k++) {
 		if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){ 
 			temp_packet = packet_ptr[time_vec[k].ix];
-			offset = (WebRtc_UWord32 *) (temp_packet+4);
+			offset = (uint32_t *) (temp_packet+4);
 			if ( time_vec[k].time >= 0 ) {
-				*offset = htonl((WebRtc_UWord32) time_vec[k].time);
+				*offset = htonl((uint32_t) time_vec[k].time);
 			}
 			else {
-				*offset = htonl((WebRtc_UWord32) 0);
+				*offset = htonl((uint32_t) 0);
 				fprintf(stderr, "Warning: negative receive time in dat file transformed to 0.\n");
 			}
 
 			// write packet to file
                         if (fwrite(temp_packet, sizeof(unsigned char),
-                                   ntohs(*((WebRtc_UWord16*) temp_packet)),
+                                   ntohs(*((uint16_t*) temp_packet)),
                                    out_file) !=
-                            ntohs(*((WebRtc_UWord16*) temp_packet))) {
+                            ntohs(*((uint16_t*) temp_packet))) {
                           return -1;
                         }
 		}
diff --git a/modules/audio_coding/neteq4/test/RTPtimeshift.cc b/modules/audio_coding/neteq4/test/RTPtimeshift.cc
index dc7ff9f..ba3a08e 100644
--- a/modules/audio_coding/neteq4/test/RTPtimeshift.cc
+++ b/modules/audio_coding/neteq4/test/RTPtimeshift.cc
@@ -63,9 +63,9 @@
 	}
 
     // get new start TS and start SeqNo from arguments
-	WebRtc_UWord32 TSdiff = atoi(argv[3]) - packet.timeStamp();
-	WebRtc_UWord16 SNdiff = 0;
-	WebRtc_UWord32 ATdiff = 0;
+	uint32_t TSdiff = atoi(argv[3]) - packet.timeStamp();
+	uint16_t SNdiff = 0;
+	uint32_t ATdiff = 0;
     if (argc > 4)
     {
         if (argv[4] >= 0)