diff --git a/webrtc/common_audio/signal_processing/cross_correlation.c b/webrtc/common_audio/signal_processing/cross_correlation.c
index 42000d608..8f39d1bce 100644
--- a/webrtc/common_audio/signal_processing/cross_correlation.c
+++ b/webrtc/common_audio/signal_processing/cross_correlation.c
@@ -16,8 +16,8 @@ void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
                                  const int16_t* seq2,
                                  int16_t dim_seq,
                                  int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2) {
+                                 int right_shifts,
+                                 int step_seq2) {
   int i = 0, j = 0;
 
   for (i = 0; i < dim_cross_correlation; i++) {
diff --git a/webrtc/common_audio/signal_processing/cross_correlation_mips.c b/webrtc/common_audio/signal_processing/cross_correlation_mips.c
index 7d9a6c644..ac3df6d0a 100644
--- a/webrtc/common_audio/signal_processing/cross_correlation_mips.c
+++ b/webrtc/common_audio/signal_processing/cross_correlation_mips.c
@@ -15,8 +15,8 @@ void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
                                      const int16_t* seq2,
                                      int16_t dim_seq,
                                      int16_t dim_cross_correlation,
-                                     int16_t right_shifts,
-                                     int16_t step_seq2) {
+                                     int right_shifts,
+                                     int step_seq2) {
 
   int32_t t0 = 0, t1 = 0, t2 = 0, t3 = 0, sum = 0;
   int16_t *pseq2 = NULL;
diff --git a/webrtc/common_audio/signal_processing/cross_correlation_neon.c b/webrtc/common_audio/signal_processing/cross_correlation_neon.c
index c358c701a..9bf16cfb6 100644
--- a/webrtc/common_audio/signal_processing/cross_correlation_neon.c
+++ b/webrtc/common_audio/signal_processing/cross_correlation_neon.c
@@ -74,8 +74,8 @@ void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
                                     const int16_t* seq2,
                                     int16_t dim_seq,
                                     int16_t dim_cross_correlation,
-                                    int16_t right_shifts,
-                                    int16_t step_seq2) {
+                                    int right_shifts,
+                                    int step_seq2) {
   int i = 0;
 
   for (i = 0; i < dim_cross_correlation; i++) {
diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index 745715dce..fdfcba024 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -542,24 +542,24 @@ typedef void (*CrossCorrelation)(int32_t* cross_correlation,
                                  const int16_t* seq2,
                                  int16_t dim_seq,
                                  int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2);
+                                 int right_shifts,
+                                 int step_seq2);
 extern CrossCorrelation WebRtcSpl_CrossCorrelation;
 void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
                                  const int16_t* seq1,
                                  const int16_t* seq2,
                                  int16_t dim_seq,
                                  int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2);
+                                 int right_shifts,
+                                 int step_seq2);
 #if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
                                     const int16_t* seq1,
                                     const int16_t* seq2,
                                     int16_t dim_seq,
                                     int16_t dim_cross_correlation,
-                                    int16_t right_shifts,
-                                    int16_t step_seq2);
+                                    int right_shifts,
+                                    int step_seq2);
 #endif
 #if defined(MIPS32_LE)
 void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
@@ -567,8 +567,8 @@ void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
                                      const int16_t* seq2,
                                      int16_t dim_seq,
                                      int16_t dim_cross_correlation,
-                                     int16_t right_shifts,
-                                     int16_t step_seq2);
+                                     int right_shifts,
+                                     int step_seq2);
 #endif
 
 // Creates (the first half of) a Hanning window. Size must be at least 1 and
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
index b016f4017..1ec5d67d1 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
@@ -68,8 +68,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
  *                      -1 - Error
  */
 
-int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
-                          int16_t quality);
+int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
+                      int16_t quality);
 int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
 
 /****************************************************************************
@@ -103,9 +103,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                         int16_t nrOfSamples, uint8_t* SIDdata,
-                         int16_t* bytesOut, int16_t forceSID);
+int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
+                     int16_t nrOfSamples, uint8_t* SIDdata,
+                     int16_t* bytesOut, int16_t forceSID);
 
 /****************************************************************************
  * WebRtcCng_UpdateSid(...)
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 9862f1253..32e28591d 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -36,7 +36,7 @@ typedef struct WebRtcCngDecoder_ {
 
 typedef struct WebRtcCngEncoder_ {
   int16_t enc_nrOfCoefs;
-  uint16_t enc_sampfreq;
+  int enc_sampfreq;
   int16_t enc_interval;
   int16_t enc_msSinceSID;
   int32_t enc_Energy;
@@ -142,8 +142,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
-                          int16_t quality) {
+int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
+                      int16_t quality) {
   int i;
   WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
   memset(inst, 0, sizeof(WebRtcCngEncoder));
@@ -227,9 +227,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst) {
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                         int16_t nrOfSamples, uint8_t* SIDdata,
-                         int16_t* bytesOut, int16_t forceSID) {
+int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
+                     int16_t nrOfSamples, uint8_t* SIDdata,
+                     int16_t* bytesOut, int16_t forceSID) {
   WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
 
   int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@@ -388,10 +388,12 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
     inst->enc_msSinceSID = 0;
     *bytesOut = inst->enc_nrOfCoefs + 1;
 
-    inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
+    inst->enc_msSinceSID +=
+        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
     return inst->enc_nrOfCoefs + 1;
   } else {
-    inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
+    inst->enc_msSinceSID +=
+        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
     *bytesOut = 0;
     return 0;
   }
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index d06c588d0..6a669e2c6 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -39,7 +39,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
     }
 }
 
-int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
 {
     // Free encoder memory
     return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
@@ -79,7 +79,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
     }
 }
 
-int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
 {
     // Free encoder memory
     return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
index 7fe11a7eb..a5ecbe70f 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -73,7 +73,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
 
 
 
@@ -142,7 +142,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
  *                              -1 - Error
  */
 
-int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
 
 
 /****************************************************************************
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
index d8f8c93a8..c24b4a6ac 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -31,7 +31,7 @@ void WebRtcIlbcfix_AugmentedCbCorr(
     int16_t low,    /* (i) Lag to start from (typically
                              20) */
     int16_t high,   /* (i) Lag to end at (typically 39) */
-    int16_t scale)   /* (i) Scale factor to use for
+    int scale)   /* (i) Scale factor to use for
                               the crossDot */
 {
   int lagcount;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
index 533d0a49e..a0435c434 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -36,7 +36,6 @@ void WebRtcIlbcfix_AugmentedCbCorr(
     int16_t low,    /* (i) Lag to start from (typically
                                                    20) */
     int16_t high,   /* (i) Lag to end at (typically 39 */
-    int16_t scale);   /* (i) Scale factor to use for
-                                                   the crossDot */
+    int scale);   /* (i) Scale factor to use for the crossDot */
 
 #endif
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
index f8a09334f..2b7e08206 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -34,7 +34,7 @@ void WebRtcIlbcfix_CbMemEnergy(
     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 */
+    int scale,   /* (i) The scaling of all energy values */
     int16_t base_size  /* (i) Index to where the energy values should be stored */
                                ) {
   int16_t *ppi, *ppo, *pp;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 1b50c0b4e..68dd7dac5 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -27,7 +27,7 @@ void WebRtcIlbcfix_CbMemEnergy(
     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 */
+    int scale,   /* (i) The scaling of all energy values */
     int16_t base_size  /* (i) Index to where the energy values should be stored */
                                );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
index 7e6daf921..39f18c239 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -22,7 +22,7 @@
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
     int16_t *interpSamples, /* (i) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t scale,   /* (i) The scaling of all energy values */
+    int 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 */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
index 6c181bd0e..e73d414d4 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -22,7 +22,7 @@
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
     int16_t *interpSamples, /* (i) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t scale,   /* (i) The scaling of all energy values */
+    int 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 */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
index e5d14246f..b6754418a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -28,7 +28,7 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
     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 */
+    int scale,   /* (i) The scaling of all energy values */
     int16_t base_size  /* (i) Index to where the energy values should be stored */
                                    )
 {
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
index c7e1e5417..c7bf92945 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -26,7 +26,7 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
     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 */
+    int scale,   /* (i) The scaling of all energy values */
     int16_t base_size  /* (i) Index to where the energy values should be stored */
                                    );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
index 4c6196bdf..2a77f4f66 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -46,7 +46,9 @@ void WebRtcIlbcfix_CbSearch(
     int16_t block  /* (i) the subblock number */
                             ) {
   int16_t i, j, stage, range;
-  int16_t *pp, scale, tmp;
+  int16_t *pp;
+  int16_t tmp;
+  int scale;
   int16_t bits, temp1, temp2;
   int16_t base_size;
   int32_t codedEner, targetEner;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index fde541455..db4b8d405 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -121,8 +121,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     shifts = WEBRTC_SPL_MAX(0, shifts);
 
     /* compute cross correlation */
-    WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               ENH_BLOCKL_HALF, 50, (int16_t)shifts, -1);
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
+                               shifts, -1);
 
     /* Find 3 highest correlations that should be compared for the
        highest (corr*corr)/ener */
@@ -207,8 +207,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
       shifts=0;
 
     /* compute cross correlation */
-    WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               plc_blockl, 3, (int16_t)shifts, 1);
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
+                               1);
 
     /* find lag */
     lag=WebRtcSpl_MaxIndexW32(corr32, 3);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index 88ad33b5a..e41c095f8 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -88,10 +88,10 @@ int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
   }
 }
 
-int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
-                             const int16_t* speechIn,
-                             int16_t len,
-                             uint8_t* encoded) {
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         int16_t len,
+                         uint8_t* encoded) {
   int16_t pos = 0;
   int16_t encpos = 0;
 
@@ -141,11 +141,11 @@ int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
 }
 
 
-int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
-                             const uint8_t* encoded,
-                             int16_t len,
-                             int16_t* decoded,
-                             int16_t* speechType)
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         int16_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
 {
   int i=0;
   /* Allow for automatic switching between the frame sizes
@@ -194,11 +194,11 @@ int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
   return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                 const uint8_t* encoded,
-                                 int16_t len,
-                                 int16_t* decoded,
-                                 int16_t* speechType)
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             int16_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
 {
   int i=0;
   if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
@@ -222,11 +222,11 @@ int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
   return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                 const uint8_t* encoded,
-                                 int16_t len,
-                                 int16_t* decoded,
-                                 int16_t* speechType)
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             int16_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
 {
   int i=0;
   if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
index d903ac7e8..0659e5005 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -23,7 +23,7 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
     int16_t mode,  /* (i) frame size mode */
     int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
index 4871b5c1a..cdd219207 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -25,7 +25,7 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
     int16_t mode,     /* (i) frame size mode */
     int use_enhancer           /* (i) 1 to use enhancer
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
index 1a2fa0892..9c562dbfc 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -23,7 +23,7 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     IlbcEncoder *iLBCenc_inst,  /* (i/o) Encoder instance */
     int16_t mode) {  /* (i) frame size mode */
   iLBCenc_inst->mode = mode;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
index 2eea27c8e..7154661fb 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -25,7 +25,7 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
     int16_t mode     /* (i) frame size mode */
                                          );
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
index b7e1735e5..493496848 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
@@ -135,10 +135,10 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
-                               const int16_t *speechIn,
-                               int16_t len,
-                               uint8_t* encoded);
+  int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
+                           const int16_t *speechIn,
+                           int16_t len,
+                           uint8_t* encoded);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecoderInit(...)
@@ -180,21 +180,21 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+  int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                           const uint8_t* encoded,
+                           int16_t len,
+                           int16_t* decoded,
+                           int16_t* speechType);
+  int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                               const uint8_t* encoded,
+                               int16_t len,
+                               int16_t* decoded,
+                               int16_t* speechType);
+  int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
                                const uint8_t* encoded,
                                int16_t len,
                                int16_t* decoded,
                                int16_t* speechType);
-  int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                   const uint8_t* encoded,
-                                   int16_t len,
-                                   int16_t* decoded,
-                                   int16_t* speechType);
-  int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                   const uint8_t* encoded,
-                                   int16_t len,
-                                   int16_t* decoded,
-                                   int16_t* speechType);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecodePlc(...)
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
index 048745a3a..0da6d5498 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -29,7 +29,8 @@ void WebRtcIlbcfix_MyCorr(
     const int16_t *seq2, /* (i) second sequence */
     int16_t dim2   /* (i) dimension seq2 */
                           ){
-  int16_t max, scale, loops;
+  int16_t max, loops;
+  int scale;
 
   /* Calculate correlation between the two sequences. Scale the
      result of the multiplcication to maximum 26 bits in order
@@ -37,7 +38,7 @@ void WebRtcIlbcfix_MyCorr(
   max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
   scale=WebRtcSpl_GetSizeInBits(max);
 
-  scale = (int16_t)(2 * scale - 26);
+  scale = 2 * scale - 26;
   if (scale<0) {
     scale=0;
   }
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index 370bf9d8a..1e966a7af 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -41,7 +41,8 @@ int main(int argc, char* argv[])
 {
   FILE *ifileid,*efileid,*ofileid, *chfileid;
   short encoded_data[55], data[240], speechType;
-  short len, mode, pli;
+  int len;
+  short mode, pli;
   int blockcount = 0;
 
   IlbcEncoderInstance *Enc_Inst;
@@ -173,7 +174,8 @@ int main(int argc, char* argv[])
       /* decoding */
       fprintf(stderr, "--- Decoding block %i --- ",blockcount);
       if (pli==1) {
-        len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, &speechType);
+        len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, (int16_t)len, data,
+                                 &speechType);
       } else {
         len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
       }
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 65c5b9030..befb3558b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -184,7 +184,7 @@ int AudioEncoderDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
     decoder_sample_rate_hz_ = sample_rate_hz;
   }
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret =
+  int ret =
       T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len),
                         decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
index bf9f875d7..a1eb2714f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
@@ -25,12 +25,12 @@ struct IsacFix {
   static const uint16_t kFixSampleRate = 16000;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
-                                int16_t framesize) {
+                                int framesize) {
     return WebRtcIsacfix_Control(inst, rate, framesize);
   }
   static inline int16_t ControlBwe(instance_type* inst,
                                    int32_t rate_bps,
-                                   int16_t frame_size_ms,
+                                   int frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
                                     enforce_frame_size);
@@ -38,11 +38,11 @@ struct IsacFix {
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsacfix_Create(inst);
   }
-  static inline int16_t DecodeInternal(instance_type* inst,
-                                       const uint8_t* encoded,
-                                       int16_t len,
-                                       int16_t* decoded,
-                                       int16_t* speech_type) {
+  static inline int DecodeInternal(instance_type* inst,
+                                   const uint8_t* encoded,
+                                   int16_t len,
+                                   int16_t* decoded,
+                                   int16_t* speech_type) {
     return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
   }
   static inline int16_t DecodePlc(instance_type* inst,
@@ -53,9 +53,9 @@ struct IsacFix {
   static inline int16_t DecoderInit(instance_type* inst) {
     return WebRtcIsacfix_DecoderInit(inst);
   }
-  static inline int16_t Encode(instance_type* inst,
-                               const int16_t* speech_in,
-                               uint8_t* encoded) {
+  static inline int Encode(instance_type* inst,
+                           const int16_t* speech_in,
+                           uint8_t* encoded) {
     return WebRtcIsacfix_Encode(inst, speech_in, encoded);
   }
   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
index 961fd3fad..92dcf51ce 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -128,9 +128,9 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                               const int16_t *speechIn,
-                               uint8_t* encoded);
+  int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                           const int16_t *speechIn,
+                           uint8_t* encoded);
 
 
 
@@ -251,11 +251,11 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                               const uint8_t* encoded,
-                               int16_t len,
-                               int16_t *decoded,
-                               int16_t *speechType);
+  int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+                           const uint8_t* encoded,
+                           int16_t len,
+                           int16_t *decoded,
+                           int16_t *speechType);
 
 
   /****************************************************************************
@@ -280,11 +280,11 @@ extern "C" {
    */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                 const uint16_t *encoded,
-                                 int16_t len,
-                                 int16_t *decoded,
-                                 int16_t *speechType);
+  int 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
 
 
@@ -378,8 +378,8 @@ extern "C" {
    */
 
   int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                                int16_t          rate,
-                                int16_t          framesize);
+                                int16_t rate,
+                                int framesize);
 
 
 
@@ -407,7 +407,7 @@ extern "C" {
 
   int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
                                    int16_t rateBPS,
-                                   int16_t frameSizeMs,
+                                   int frameSizeMs,
                                    int16_t enforceFrameSize);
 
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index 23048a5c3..808aeb7fd 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -226,10 +226,10 @@ int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
  * Return value             : number of bytes in the stream so far
  *                            -1 if error detected
  */
-int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
-                                        Bitstr_dec *streamData,
-                                        const int32_t *envQ8,
-                                        const int16_t lenData)
+int WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
+                                    Bitstr_dec *streamData,
+                                    const int32_t *envQ8,
+                                    const int16_t lenData)
 {
   uint32_t    W_lower;
   uint32_t    W_upper;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
index 584bc471f..40bbb4cda 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
@@ -74,7 +74,7 @@ int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
  * Return value             : number of bytes in the stream so far
  *                            <0 if error detected
  */
-int16_t WebRtcIsacfix_DecLogisticMulti2(
+int WebRtcIsacfix_DecLogisticMulti2(
     int16_t *data,
     Bitstr_dec *streamData,
     const int32_t *env,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
index 8ecbd14bc..3d003e4ed 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -32,9 +32,9 @@ int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr* bwest_str,
                                     uint32_t send_ts,
                                     uint32_t arr_ts);
 
-int16_t WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
-                                       IsacFixDecoderInstance* ISACdec_obj,
-                                       int16_t* current_framesamples);
+int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
+                             IsacFixDecoderInstance* ISACdec_obj,
+                             int16_t* current_framesamples);
 
 int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
                                           IsacFixDecoderInstance* ISACdec_obj,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index 5e095da68..f0ae07e13 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -27,14 +27,14 @@
 
 
 
-int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
-                                 IsacFixDecoderInstance *ISACdec_obj,
-                                 int16_t       *current_framesamples)
+int WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
+                             IsacFixDecoderInstance *ISACdec_obj,
+                             int16_t       *current_framesamples)
 {
   int k;
   int err;
   int16_t BWno;
-  int16_t len = 0;
+  int len = 0;
 
   int16_t model;
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 39653789e..aab8f43d8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -450,10 +450,10 @@ static void GenerateDitherQ7(int16_t *bufQ7,
  * function to decode the complex spectrum from the bitstream
  * returns the total number of bytes in the stream
  */
-int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
-                                 int16_t *frQ7,
-                                 int16_t *fiQ7,
-                                 int16_t AvgPitchGain_Q12)
+int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
+                             int16_t *frQ7,
+                             int16_t *fiQ7,
+                             int16_t AvgPitchGain_Q12)
 {
   int16_t  data[FRAMESAMPLES];
   int32_t  invARSpec2_Q16[FRAMESAMPLES/4];
@@ -461,7 +461,7 @@ int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
   int16_t  RCQ15[AR_ORDER];
   int16_t  gainQ10;
   int32_t  gain2_Q10;
-  int16_t  len;
+  int len;
   int          k;
 
   /* create dither signal */
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index ec20c71f7..e4489df33 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -22,10 +22,10 @@
 #include "structs.h"
 
 /* decode complex spectrum (return number of bytes in stream) */
-int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
-                                 int16_t *frQ7,
-                                 int16_t *fiQ7,
-                                 int16_t AvgPitchGain_Q12);
+int WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
+                             int16_t *frQ7,
+                             int16_t *fiQ7,
+                             int16_t AvgPitchGain_Q12);
 
 /* encode complex spectrum */
 int WebRtcIsacfix_EncodeSpec(const int16_t *fr,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index f8abc8a0b..f1e5cd0cf 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -399,12 +399,12 @@ static void write_be16(const uint16_t* src, size_t nbytes, uint8_t* dest) {
  *                          : -1 - Error
  */
 
-int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                             const int16_t    *speechIn,
-                             uint8_t* encoded)
+int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                         const int16_t    *speechIn,
+                         uint8_t* encoded)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  int16_t stream_len;
+  int stream_len;
 
   /* typecast pointer to rela structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -421,7 +421,7 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
                                         &ISAC_inst->bwestimator_obj,
                                         ISAC_inst->CodingMode);
   if (stream_len<0) {
-    ISAC_inst->errorcode = - stream_len;
+    ISAC_inst->errorcode = -(int16_t)stream_len;
     return -1;
   }
 
@@ -766,17 +766,17 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 
-int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                             const uint8_t* encoded,
-                             int16_t          len,
-                             int16_t          *decoded,
-                             int16_t     *speechType)
+int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+                         const uint8_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) */
   int16_t     number_of_samples;
-  int16_t declen = 0;
+  int declen = 0;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -809,7 +809,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
 
   if (declen < 0) {
     /* Some error inside the decoder */
-    ISAC_inst->errorcode = -declen;
+    ISAC_inst->errorcode = -(int16_t)declen;
     memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES);
     return -1;
   }
@@ -859,17 +859,17 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                               const uint16_t   *encoded,
-                               int16_t          len,
-                               int16_t          *decoded,
-                               int16_t    *speechType)
+int 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) */
   int16_t     number_of_samples;
-  int16_t declen = 0;
+  int declen = 0;
   int16_t dummy[FRAMESAMPLES/2];
 
 
@@ -903,7 +903,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
 
   if (declen < 0) {
     /* Some error inside the decoder */
-    ISAC_inst->errorcode = -declen;
+    ISAC_inst->errorcode = -(int16_t)declen;
     memset(decoded, 0, sizeof(int16_t) * FRAMESAMPLES);
     return -1;
   }
@@ -1076,8 +1076,8 @@ int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                              int16_t          rate,
-                              int16_t          framesize)
+                              int16_t rate,
+                              int framesize)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* typecast pointer to real structure */
@@ -1101,7 +1101,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
 
 
   if (framesize  == 30 || framesize == 60)
-    ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * framesize;
+    ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * framesize);
   else {
     ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
     return -1;
@@ -1136,7 +1136,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
 
 int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
                                  int16_t rateBPS,
-                                 int16_t frameSizeMs,
+                                 int frameSizeMs,
                                  int16_t enforceFrameSize)
 {
   ISACFIX_SubStruct *ISAC_inst;
@@ -1170,7 +1170,7 @@ int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
 
   /* Set initial framesize. If enforceFrameSize is set the frame size will not change */
   if ((frameSizeMs  == 30) || (frameSizeMs == 60)) {
-    ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * frameSizeMs;
+    ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * frameSizeMs);
   } else {
     ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
     return -1;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index ba50b0c67..bc1a194ab 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -101,14 +101,15 @@ int main(int argc, char* argv[])
   int i, errtype, h = 0, k, packetLossPercent = 0;
   int16_t CodingMode;
   int16_t bottleneck;
-  int16_t framesize = 30;           /* ms */
+  int framesize = 30;           /* ms */
   int cur_framesmpls, err = 0, lostPackets = 0;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
   int16_t stream_len = 0;
-  int16_t framecnt, declen = 0;
+  int16_t framecnt;
+  int declen = 0;
   int16_t shortdata[FRAMESAMPLES_10ms];
   int16_t decoded[MAX_FRAMESAMPLES];
   uint16_t streamdata[500];
@@ -766,7 +767,7 @@ int main(int argc, char* argv[])
 #else
           declen = -1;
 #endif
-          prevFrameSize = declen/240;
+          prevFrameSize = static_cast<int16_t>(declen / 240);
         }
       }
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 9aabd04a9..218b97bb0 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -88,8 +88,8 @@ int main(int argc, char* argv[]) {
   int16_t CodingMode;
   int16_t bottleneck;
 
-  int16_t framesize = 30; /* ms */
-  // int16_t framesize = 60; /* To invoke cisco complexity case at frame 2252 */
+  int framesize = 30; /* ms */
+  // int framesize = 60; /* To invoke cisco complexity case at frame 2252 */
 
   int cur_framesmpls, err;
 
@@ -99,7 +99,7 @@ int main(int argc, char* argv[]) {
   double length_file;
 
   int16_t stream_len = 0;
-  int16_t declen;
+  int declen;
 
   int16_t shortdata[FRAMESAMPLES_10ms];
   int16_t decoded[MAX_FRAMESAMPLES];
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
index 5a7580746..8c7053359 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
@@ -24,12 +24,12 @@ struct IsacFloat {
   static const bool has_swb = true;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
-                                int16_t framesize) {
+                                int framesize) {
     return WebRtcIsac_Control(inst, rate, framesize);
   }
   static inline int16_t ControlBwe(instance_type* inst,
                                    int32_t rate_bps,
-                                   int16_t frame_size_ms,
+                                   int frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
                                  enforce_frame_size);
@@ -37,11 +37,11 @@ struct IsacFloat {
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsac_Create(inst);
   }
-  static inline int16_t DecodeInternal(instance_type* inst,
-                                       const uint8_t* encoded,
-                                       int16_t len,
-                                       int16_t* decoded,
-                                       int16_t* speech_type) {
+  static inline int DecodeInternal(instance_type* inst,
+                                   const uint8_t* encoded,
+                                   int16_t len,
+                                   int16_t* decoded,
+                                   int16_t* speech_type) {
     return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
   }
   static inline int16_t DecodePlc(instance_type* inst,
@@ -53,9 +53,9 @@ struct IsacFloat {
   static inline int16_t DecoderInit(instance_type* inst) {
     return WebRtcIsac_DecoderInit(inst);
   }
-  static inline int16_t Encode(instance_type* inst,
-                               const int16_t* speech_in,
-                               uint8_t* encoded) {
+  static inline int Encode(instance_type* inst,
+                           const int16_t* speech_in,
+                           uint8_t* encoded) {
     return WebRtcIsac_Encode(inst, speech_in, encoded);
   }
   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
index 6d0c32deb..1a83d722b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -144,7 +144,7 @@ extern "C" {
    *                            : -1 - Error
    */
 
-  int16_t WebRtcIsac_Encode(
+  int WebRtcIsac_Encode(
       ISACStruct*        ISAC_main_inst,
       const int16_t* speechIn,
       uint8_t* encoded);
@@ -214,7 +214,7 @@ extern "C" {
    *                              -1 - Error.
    */
 
-  int16_t WebRtcIsac_Decode(
+  int WebRtcIsac_Decode(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
       int16_t         len,
@@ -269,7 +269,7 @@ extern "C" {
   int16_t WebRtcIsac_Control(
       ISACStruct*   ISAC_main_inst,
       int32_t rate,
-      int16_t framesize);
+      int framesize);
 
 
   /******************************************************************************
@@ -300,7 +300,7 @@ extern "C" {
   int16_t WebRtcIsac_ControlBwe(
       ISACStruct* ISAC_main_inst,
       int32_t rateBPS,
-      int16_t frameSizeMs,
+      int frameSizeMs,
       int16_t enforceFrameSize);
 
 
@@ -701,7 +701,7 @@ extern "C" {
    * Return value              : >0 - number of samples in decoded vector
    *                             -1 - Error
    */
-  int16_t WebRtcIsac_DecodeRcu(
+  int WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
       int16_t         len,
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
index 06c15cb39..2419e2403 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
@@ -80,9 +80,9 @@ static const uint32_t kCrcTable[256] = {
  *                             -1 - Error
  */
 
-int16_t WebRtcIsac_GetCrc(const int16_t* bitstream,
-                          int16_t        len_bitstream_in_bytes,
-                          uint32_t*      crc)
+int WebRtcIsac_GetCrc(const int16_t* bitstream,
+                      int len_bitstream_in_bytes,
+                      uint32_t* crc)
 {
   uint8_t* bitstream_ptr_uw8;
   uint32_t crc_state;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
index 19d1bf31d..09583dfc5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
@@ -36,10 +36,10 @@
  *                   -1 - Error
  */
 
-int16_t WebRtcIsac_GetCrc(
+int WebRtcIsac_GetCrc(
     const int16_t* encoded,
-    int16_t        no_of_word8s,
-    uint32_t*      crc);
+    int no_of_word8s,
+    uint32_t* crc);
 
 
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index db78e6de2..3ed776b5d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -494,15 +494,15 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
  *                                 samples.
  *                            : -1 - Error
  */
-int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
-                          const int16_t* speechIn,
-                          uint8_t* encoded) {
+int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
+                      const int16_t* speechIn,
+                      uint8_t* encoded) {
   float inFrame[FRAMESAMPLES_10ms];
   int16_t speechInLB[FRAMESAMPLES_10ms];
   int16_t speechInUB[FRAMESAMPLES_10ms];
-  int16_t streamLenLB = 0;
-  int16_t streamLenUB = 0;
-  int16_t streamLen = 0;
+  int streamLenLB = 0;
+  int streamLenUB = 0;
+  int streamLen = 0;
   int16_t k = 0;
   int garbageLen = 0;
   int32_t bottleneck = 0;
@@ -601,8 +601,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
 
     /* Tell to upper-band the number of bytes used so far.
      * This is for payload limitation. */
-    instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
-                                         LEN_CHECK_SUM_WORD8;
+    instUB->ISACencUB_obj.numBytesUsed =
+        (int16_t)(streamLenLB + 1 + LEN_CHECK_SUM_WORD8);
     /* Encode upper-band. */
     switch (instISAC->bandwidthKHz) {
       case isac12kHz: {
@@ -1045,12 +1045,12 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
   return 0;
 }
 
-static int16_t Decode(ISACStruct* ISAC_main_inst,
-                      const uint8_t* encoded,
-                      int16_t lenEncodedBytes,
-                      int16_t* decoded,
-                      int16_t* speechType,
-                      int16_t isRCUPayload) {
+static int Decode(ISACStruct* ISAC_main_inst,
+                  const uint8_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). */
@@ -1060,8 +1060,8 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
   float outFrame[MAX_FRAMESAMPLES];
   int16_t outFrameLB[MAX_FRAMESAMPLES];
   int16_t outFrameUB[MAX_FRAMESAMPLES];
-  int16_t numDecodedBytesLB;
-  int16_t numDecodedBytesUB;
+  int numDecodedBytesLB;
+  int numDecodedBytesUB;
   int16_t lenEncodedLBBytes;
   int16_t validChecksum = 1;
   int16_t k;
@@ -1350,11 +1350,11 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
  *                              -1 - Error
  */
 
-int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
-                          const uint8_t* encoded,
-                          int16_t lenEncodedBytes,
-                          int16_t* decoded,
-                          int16_t* speechType) {
+int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
+                      const uint8_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);
@@ -1382,11 +1382,11 @@ int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
 
 
 
-int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
-                             const uint8_t* encoded,
-                             int16_t lenEncodedBytes,
-                             int16_t* decoded,
-                             int16_t* speechType) {
+int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
+                         const uint8_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);
@@ -1485,7 +1485,7 @@ static int16_t ControlUb(ISACUBStruct* instISAC, double rate) {
 
 int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
                            int32_t bottleneckBPS,
-                           int16_t frameSize) {
+                           int frameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   int16_t status;
   double rateLB;
@@ -1526,7 +1526,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
     return -1;
   }
 
-  status = ControlLb(&instISAC->instLB, rateLB, frameSize);
+  status = ControlLb(&instISAC->instLB, rateLB, (int16_t)frameSize);
   if (status < 0) {
     instISAC->errorCode = -status;
     return -1;
@@ -1594,7 +1594,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
  */
 int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
                               int32_t bottleneckBPS,
-                              int16_t frameSizeMs,
+                              int frameSizeMs,
                               int16_t enforceFrameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum ISACBandwidth bandwidth;
@@ -1641,8 +1641,8 @@ int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
    *  will not change */
   if (frameSizeMs != 0) {
     if ((frameSizeMs  == 30) || (frameSizeMs == 60)) {
-      instISAC->instLB.ISACencLB_obj.new_framelength = (FS / 1000) *
-          frameSizeMs;
+      instISAC->instLB.ISACencLB_obj.new_framelength =
+          (int16_t)((FS / 1000) * frameSizeMs);
     } else {
       instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
       return -1;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 8b93e65ba..a751c2479 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -79,7 +79,7 @@ TEST_F(IsacTest, IsacUpdateBWE) {
   WebRtcIsac_EncoderInit(isac_codec_, 0);
   WebRtcIsac_DecoderInit(isac_codec_);
 
-  int16_t encoded_bytes;
+  int encoded_bytes;
 
   // Test with call with a small packet (sync packet).
   EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1,
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 717da09b6..1a20ca587 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -47,14 +47,15 @@ int main(int argc, char* argv[]) {
   int i, errtype, VADusage = 0, packetLossPercent = 0;
   int16_t CodingMode;
   int32_t bottleneck = 0;
-  int16_t framesize = 30; /* ms */
+  int framesize = 30; /* ms */
   int cur_framesmpls, err;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
   int16_t stream_len = 0;
-  int16_t declen = 0, lostFrame = 0, declenTC = 0;
+  int declen = 0, declenTC = 0;
+  int16_t lostFrame = 0;
 
   int16_t shortdata[SWBFRAMESAMPLES_10ms];
   int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 6ec818ee7..a11e408b3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -191,7 +191,7 @@ int main(int argc, char* argv[])
 
   short streamLen;
   short numSamplesRead;
-  short lenDecodedAudio;
+  int lenDecodedAudio;
   short senderIdx;
   short receiverIdx;
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 959be7d4c..b1d8a2418 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -62,7 +62,7 @@ int main(int argc, char* argv[]) {
   unsigned long totalsmpls = 0;
 
   int32_t bottleneck = 39;
-  int16_t frameSize = 30; /* ms */
+  int frameSize = 30; /* ms */
   int16_t codingMode = 1;
   int16_t shortdata[FRAMESAMPLES_SWB_10ms];
   int16_t decoded[MAX_FRAMESAMPLES_SWB];
@@ -73,9 +73,9 @@ int main(int argc, char* argv[]) {
   ISACStruct* ISAC_main_inst;
 
   int16_t stream_len = 0;
-  int16_t declen = 0;
+  int declen = 0;
   int16_t err;
-  int16_t cur_framesmpls;
+  int cur_framesmpls;
   int endfile;
 #ifdef WIN32
   double length_file;
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index c05d773c0..1eeb5caa9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -198,7 +198,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
   CHECK_EQ(input_buffer_.size(),
            static_cast<size_t>(num_10ms_frames_per_packet_) *
            samples_per_10ms_frame_);
-  int16_t status = WebRtcOpus_Encode(
+  int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
                            static_cast<int16_t>(num_channels_)),
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
index dccc7ca71..925cd85df 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
@@ -64,11 +64,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
  * Return value                 : >=0 - Length (in bytes) of coded data
  *                                -1 - Error
  */
-int16_t WebRtcOpus_Encode(OpusEncInst* inst,
-                          const int16_t* audio_in,
-                          int16_t samples,
-                          int16_t length_encoded_buffer,
-                          uint8_t* encoded);
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      int16_t samples,
+                      int16_t length_encoded_buffer,
+                      uint8_t* encoded);
 
 /****************************************************************************
  * WebRtcOpus_SetBitRate(...)
@@ -236,9 +236,9 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
  * Return value              : >0 - Samples per channel in decoded vector
  *                             -1 - Error
  */
-int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                          int16_t encoded_bytes, int16_t* decoded,
-                          int16_t* audio_type);
+int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
+                      int16_t encoded_bytes, int16_t* decoded,
+                      int16_t* audio_type);
 
 /****************************************************************************
  * WebRtcOpus_DecodePlc(...)
@@ -254,8 +254,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
  * Return value                   : >0 - number of samples in decoded PLC vector
  *                                  -1 - Error
  */
-int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
-                             int16_t number_of_lost_frames);
+int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                         int number_of_lost_frames);
 
 /****************************************************************************
  * WebRtcOpus_DecodeFec(...)
@@ -275,9 +275,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
  *                              0 - No FEC data in the packet
  *                             -1 - Error
  */
-int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                             int16_t encoded_bytes, int16_t* decoded,
-                             int16_t* audio_type);
+int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
+                         int16_t encoded_bytes, int16_t* decoded,
+                         int16_t* audio_type);
 
 /****************************************************************************
  * WebRtcOpus_DurationEst(...)
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index a30b1cb90..328fc4837 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -131,10 +131,10 @@ OpusFecTest::OpusFecTest()
 }
 
 void OpusFecTest::EncodeABlock() {
-  int16_t value = WebRtcOpus_Encode(opus_encoder_,
-                                    &in_data_[data_pointer_],
-                                    block_length_sample_,
-                                    max_bytes_, &bit_stream_[0]);
+  int value = WebRtcOpus_Encode(opus_encoder_,
+                                &in_data_[data_pointer_],
+                                block_length_sample_,
+                                max_bytes_, &bit_stream_[0]);
   EXPECT_GT(value, 0);
 
   encoded_bytes_ = value;
@@ -142,7 +142,7 @@ void OpusFecTest::EncodeABlock() {
 
 void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
   int16_t audio_type;
-  int16_t value_1 = 0, value_2 = 0;
+  int value_1 = 0, value_2 = 0;
 
   if (lost_previous) {
     // Decode previous frame.
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 527de1013..e2506166a 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -78,11 +78,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   }
 }
 
-int16_t WebRtcOpus_Encode(OpusEncInst* inst,
-                          const int16_t* audio_in,
-                          int16_t samples,
-                          int16_t length_encoded_buffer,
-                          uint8_t* encoded) {
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      int16_t samples,
+                      int16_t length_encoded_buffer,
+                      uint8_t* encoded) {
   int res;
 
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
@@ -291,9 +291,9 @@ static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
   return res;
 }
 
-int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                          int16_t encoded_bytes, int16_t* decoded,
-                          int16_t* audio_type) {
+int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
+                      int16_t encoded_bytes, int16_t* decoded,
+                      int16_t* audio_type) {
   int decoded_samples;
 
   if (encoded_bytes == 0) {
@@ -318,8 +318,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
   return decoded_samples;
 }
 
-int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
-                             int16_t number_of_lost_frames) {
+int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                         int number_of_lost_frames) {
   int16_t audio_type = 0;
   int decoded_samples;
   int plc_samples;
@@ -339,9 +339,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
   return decoded_samples;
 }
 
-int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                             int16_t encoded_bytes, int16_t* decoded,
-                             int16_t* audio_type) {
+int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
+                         int16_t encoded_bytes, int16_t* decoded,
+                         int16_t* audio_type) {
   int decoded_samples;
   int fec_samples;
 
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index 09301df51..ad7e2f9be 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -273,17 +273,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
       int16_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
       for (int i = 0; i < loop_encode; i++) {
-        if (channels == 1) {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_mono_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GT(bitstream_len_byte, -1);
-        } else {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_stereo_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GT(bitstream_len_byte, -1);
-        }
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GT(bitstream_len_byte_int, -1);
+        bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
 
         // Simulate packet loss by setting |packet_loss_| to "true" in
         // |percent_loss| percent of the loops.
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index c3f1dbbc8..99ff95a2e 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -163,9 +163,9 @@ int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
                                      SpeechType* speech_type) {
   DCHECK_EQ(sample_rate_hz, 8000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
+                                 static_cast<int16_t>(encoded_len), decoded,
+                                 &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
@@ -330,11 +330,11 @@ int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
                                      SpeechType* speech_type) {
   DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
-                                  static_cast<int16_t>(encoded_len), decoded,
-                                  &temp_type);
+  int ret = WebRtcOpus_Decode(dec_state_, encoded,
+                              static_cast<int16_t>(encoded_len), decoded,
+                              &temp_type);
   if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
@@ -352,11 +352,11 @@ int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
 
   DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
+                                 static_cast<int16_t>(encoded_len), decoded,
+                                 &temp_type);
   if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 7451ae26f..289e66d17 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -272,7 +272,7 @@ void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
 }
 
 void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
-                             int16_t* factor, int16_t increment,
+                             int16_t* factor, int increment,
                              int16_t* output) {
   uint16_t factor_16b = *factor;
   int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
@@ -284,7 +284,7 @@ void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
   *factor = factor_16b;
 }
 
-void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
+void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
   int32_t factor = (16384 << 6) + 32;
   for (size_t i = 0; i < length; i++) {
     signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.h b/webrtc/modules/audio_coding/neteq/dsp_helper.h
index af4f4d6c8..f9032562f 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.h
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -110,11 +110,11 @@ class DspHelper {
   // sample and increases the gain by |increment| (Q20) for each sample. The
   // result is written to |output|. |length| samples are processed.
   static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
-                           int16_t increment, int16_t* output);
+                           int increment, int16_t* output);
 
   // Starts at unity gain and gradually fades out |signal|. For each sample,
   // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
-  static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
+  static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
 
   // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
   // has |input_length| samples, and the method will write |output_length|
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index 15d7a84ae..4bcb7a88e 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -239,14 +239,12 @@ int Expand::Process(AudioMultiVector* output) {
     if (consecutive_expands_ == 3) {
       // Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
       // mute_slope = 0.0010 / fs_mult in Q20.
-      parameters.mute_slope = std::max(parameters.mute_slope,
-                                       static_cast<int16_t>(1049 / fs_mult));
+      parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
     }
     if (consecutive_expands_ == 7) {
       // Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
       // mute_slope = 0.0020 / fs_mult in Q20.
-      parameters.mute_slope = std::max(parameters.mute_slope,
-                                       static_cast<int16_t>(2097 / fs_mult));
+      parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
     }
 
     // Mute segment according to slope value.
@@ -368,7 +366,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   InitializeForAnExpandPeriod();
 
   // Calculate correlation in downsampled domain (4 kHz sample rate).
-  int16_t correlation_scale;
+  int correlation_scale;
   int correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
   // If it is decided to break bit-exactness |correlation_length| should be
   // initialized to the return value of Correlation().
@@ -446,7 +444,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
                        correlation_length + start_index + correlation_lags - 1);
     correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
         + (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
-    correlation_scale = std::max(static_cast<int16_t>(0), correlation_scale);
+    correlation_scale = std::max(0, correlation_scale);
 
     // Calculate the correlation, store in |correlation_vector2|.
     WebRtcSpl_CrossCorrelation(
@@ -473,7 +471,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
     // Calculate the correlation coefficient between the two portions of the
     // signal.
-    int16_t corr_coefficient;
+    int32_t corr_coefficient;
     if ((energy1 > 0) && (energy2 > 0)) {
       int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
       int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
@@ -482,17 +480,17 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         // If sum is odd, add 1 to make it even.
         energy1_scale += 1;
       }
-      int16_t scaled_energy1 = energy1 >> energy1_scale;
-      int16_t scaled_energy2 = energy2 >> energy2_scale;
-      int16_t sqrt_energy_product = WebRtcSpl_SqrtFloor(
-          scaled_energy1 * scaled_energy2);
+      int32_t scaled_energy1 = energy1 >> energy1_scale;
+      int32_t scaled_energy2 = energy2 >> energy2_scale;
+      int16_t sqrt_energy_product = static_cast<int16_t>(
+          WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
       // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
       int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
       max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
       corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
                                              sqrt_energy_product);
-      corr_coefficient = std::min(static_cast<int16_t>(16384),
-                                  corr_coefficient);  // Cap at 1.0 in Q14.
+      // Cap at 1.0 in Q14.
+      corr_coefficient = std::min(16384, corr_coefficient);
     } else {
       corr_coefficient = 0;
     }
@@ -513,8 +511,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
       // Energy constraint fulfilled. Use both vectors and scale them
       // accordingly.
-      int16_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
-      int16_t scaled_energy1 = scaled_energy2 - 13;
+      int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      int32_t scaled_energy1 = scaled_energy2 - 13;
       // Calculate scaled_energy1 / scaled_energy2 in Q13.
       int32_t energy_ratio = WebRtcSpl_DivW32W16(
           WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
@@ -684,7 +682,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     //   voice_mix_factor = 0;
     if (corr_coefficient > 7875) {
       int16_t x1, x2, x3;
-      x1 = corr_coefficient;  // |corr_coefficient| is in Q14.
+      // |corr_coefficient| is in Q14.
+      x1 = static_cast<int16_t>(corr_coefficient);
       x2 = (x1 * x1) >> 14;   // Shift 14 to keep result in Q14.
       x3 = (x1 * x2) >> 14;
       static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
@@ -712,8 +711,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       // the division.
       // Shift the denominator from Q13 to Q5 before the division. The result of
       // the division will then be in Q20.
-      int16_t temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
-                                               (distortion_lag * slope) >> 8);
+      int temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
+                                           (distortion_lag * slope) >> 8);
       if (slope > 14746) {
         // slope > 1.8.
         // Divide by 2, with proper rounding.
@@ -732,8 +731,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         // Make sure the mute factor decreases from 1.0 to 0.9 in no more than
         // 6.25 ms.
         // mute_slope >= 0.005 / fs_mult in Q20.
-        parameters.mute_slope = std::max(static_cast<int16_t>(5243 / fs_mult),
-                                         parameters.mute_slope);
+        parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
       } else if (slope > 8028) {
         parameters.mute_slope = 0;
       }
@@ -755,7 +753,7 @@ Expand::ChannelParameters::ChannelParameters()
 }
 
 int16_t Expand::Correlation(const int16_t* input, size_t input_length,
-                            int16_t* output, int16_t* output_scale) const {
+                            int16_t* output, int* output_scale) const {
   // Set parameters depending on sample rate.
   const int16_t* filter_coefficients;
   int16_t num_coefficients;
@@ -844,7 +842,7 @@ Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
 // TODO(turajs): This can be moved to BackgroundNoise class.
 void Expand::GenerateBackgroundNoise(int16_t* random_vector,
                                      size_t channel,
-                                     int16_t mute_slope,
+                                     int mute_slope,
                                      bool too_many_expands,
                                      size_t num_noise_samples,
                                      int16_t* buffer) {
@@ -887,7 +885,7 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
         bgn_mute_factor > 0) {
       // Fade BGN to zero.
       // Calculate muting slope, approximately -2^18 / fs_hz.
-      int16_t mute_slope;
+      int mute_slope;
       if (fs_hz_ == 8000) {
         mute_slope = -32;
       } else if (fs_hz_ == 16000) {
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 5679ec1cc..672ea399a 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -72,7 +72,7 @@ class Expand {
 
   void GenerateBackgroundNoise(int16_t* random_vector,
                                size_t channel,
-                               int16_t mute_slope,
+                               int mute_slope,
                                bool too_many_expands,
                                size_t num_noise_samples,
                                int16_t* buffer);
@@ -113,7 +113,7 @@ class Expand {
     AudioVector expand_vector0;
     AudioVector expand_vector1;
     bool onset;
-    int16_t mute_slope; /* Q20 */
+    int mute_slope; /* Q20 */
   };
 
   // Calculate the auto-correlation of |input|, with length |input_length|
@@ -121,7 +121,7 @@ class Expand {
   // |input|, and is written to |output|. The scale factor is written to
   // |output_scale|. Returns the length of the correlation vector.
   int16_t Correlation(const int16_t* input, size_t input_length,
-                      int16_t* output, int16_t* output_scale) const;
+                      int16_t* output, int* output_scale) const;
 
   void UpdateLagIndex();
 
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index 0de2439a6..fa033cf63 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -311,7 +311,7 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   const int max_corr_length = kMaxCorrelationLength;
   int stop_position_downsamp = std::min(
       max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
-  int16_t correlation_shift = 0;
+  int correlation_shift = 0;
   if (expanded_max * input_max > 26843546) {
     correlation_shift = 3;
   }
@@ -330,7 +330,7 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   int16_t* correlation_ptr = &correlation16[pad_length];
   int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
                                                      stop_position_downsamp);
-  int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+  int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
   WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
                                    correlation, norm_shift);
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 1351e6663..8cd9aebf6 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -1271,7 +1271,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
            *operation == kPreemptiveExpand);
     packet_list->pop_front();
     size_t payload_length = packet->payload_length;
-    int16_t decode_length;
+    int decode_length;
     if (packet->sync_packet) {
       // Decode to silence with the same frame size as the last decode.
       LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index fd4c46a55..ce774d749 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -108,7 +108,7 @@ int Normal::Process(const int16_t* input,
       }
 
       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-      int16_t increment = 64 / fs_mult;
+      int increment = static_cast<int>(64 / fs_mult);
       for (size_t i = 0; i < length_per_channel; i++) {
         // Scale with mute factor.
         assert(channel_ix < output->Channels());
@@ -174,7 +174,7 @@ int Normal::Process(const int16_t* input,
     // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
     // still ramping up from previous muting.
     // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-    int16_t increment = 64 / fs_mult;
+    int increment = static_cast<int>(64 / fs_mult);
     size_t length_per_channel = length / output->Channels();
     for (size_t i = 0; i < length_per_channel; i++) {
       for (size_t channel_ix = 0; channel_ix < output->Channels();
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 427601e7a..87f9494f0 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -1605,7 +1605,7 @@ int NetEQTest_encode(int coder,
                      int useVAD,
                      int bitrate,
                      int numChannels) {
-  short cdlen = 0;
+  int cdlen = 0;
   int16_t* tempdata;
   static int first_cng = 1;
   int16_t tempLen;