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 cb7aa4567..1f6974a45 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 25d75ee6e..1edf58dc1 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 711b991d9..46ff3b0f0 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/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 17fa5b24b..328698380 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 aaaced16d..f0ef70a3b 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 a407fc5d3..c61d25ad1 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_GE(bitstream_len_byte, 0);
-        } else {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_stereo_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GE(bitstream_len_byte, 0);
-        }
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GE(bitstream_len_byte_int, 0);
+        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.