Revert "Upconvert various types to int."

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

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 60 matching lines @@
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
     opus_encoder_destroy(inst->encoder);
     free(inst);
     return 0;
   } else {
     return -1;
   }
 }
 
-int WebRtcOpus_Encode(OpusEncInst* inst,
-                      const int16_t* audio_in,
-                      int16_t samples,
-                      int16_t length_encoded_buffer,
-                      uint8_t* encoded) {
+int16_t 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) {
     return -1;
   }
 
   res = opus_encode(inst->encoder,
                     (const opus_int16*)audio_in,
                     samples,
                     encoded,
@@ skipping 188 matching lines @@
                         (opus_int16*)decoded, frame_size, decode_fec);
 
   if (res <= 0)
     return -1;
 
   *audio_type = DetermineAudioType(inst, encoded_bytes);
 
   return res;
 }
 
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                      int16_t encoded_bytes, int16_t* decoded,
-                      int16_t* audio_type) {
+int16_t 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) {
     *audio_type = DetermineAudioType(inst, encoded_bytes);
     decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1);
   } else {
     decoded_samples = DecodeNative(inst,
                                    encoded,
                                    encoded_bytes,
                                    kWebRtcOpusMaxFrameSizePerChannel,
                                    decoded,
                                    audio_type,
                                    0);
   }
   if (decoded_samples < 0) {
     return -1;
   }
 
   /* Update decoded sample memory, to be used by the PLC in case of losses. */
   inst->prev_decoded_samples = decoded_samples;
 
   return decoded_samples;
 }
 
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
-                         int number_of_lost_frames) {
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                             int16_t number_of_lost_frames) {
   int16_t audio_type = 0;
   int decoded_samples;
   int plc_samples;
 
   /* The number of samples we ask for is |number_of_lost_frames| times
    * |prev_decoded_samples_|. Limit the number of samples to maximum
    * |kWebRtcOpusMaxFrameSizePerChannel|. */
   plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
   plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
       plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
   decoded_samples = DecodeNative(inst, NULL, 0, plc_samples,
                                  decoded, &audio_type, 0);
   if (decoded_samples < 0) {
     return -1;
   }
 
   return decoded_samples;
 }
 
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                         int16_t encoded_bytes, int16_t* decoded,
-                         int16_t* audio_type) {
+int16_t 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;
 
   if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) {
     return 0;
   }
 
   fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
 
   decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
@@ skipping 86 matching lines @@
     return 0;
   }
 
   for (n = 0; n < channels; n++) {
     if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))
       return 1;
   }
 
   return 0;
 }