Reland "Upconvert various types to int.", misc. codecs portion.

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 527de101320a03e76ec17cd45a70cd751a4de6be..e2506166a923a07b3599c5d3728f0ae40328facd 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;