Revert "Upconvert various types to int."

webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h

 /*
  *  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 46 matching lines @@
  *      - audio_in              : Input speech data buffer
  *      - samples               : Samples per channel in audio_in
  *      - length_encoded_buffer : Output buffer size
  *
  * Output:
  *      - encoded               : Output compressed data buffer
  *
  * Return value                 : >=0 - Length (in bytes) of coded data
  *                                -1 - Error
  */
-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);
 
 /****************************************************************************
  * WebRtcOpus_SetBitRate(...)
  *
  * This function adjusts the target bitrate of the encoder.
  *
  * Input:
  *      - inst               : Encoder context
  *      - rate               : New target bitrate
  *
@@ skipping 147 matching lines @@
  *
  * Output:
  *      - decoded            : The decoded vector
  *      - audio_type         : 1 normal, 2 CNG (for Opus it should
  *                             always return 1 since we're not using Opus's
  *                             built-in DTX/CNG scheme)
  *
  * Return value              : >0 - Samples per channel in decoded vector
  *                             -1 - Error
  */
-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);
 
 /****************************************************************************
  * WebRtcOpus_DecodePlc(...)
  *
  * This function processes PLC for opus frame(s).
  * Input:
  *        - inst                  : Decoder context
  *        - number_of_lost_frames : Number of PLC frames to produce
  *
  * Output:
  *        - decoded               : The decoded vector
  *
  * Return value                   : >0 - number of samples in decoded PLC vector
  *                                  -1 - Error
  */
-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);
 
 /****************************************************************************
  * WebRtcOpus_DecodeFec(...)
  *
  * This function decodes the FEC data from an Opus packet into one or more audio
  * frames at the ACM interface's sampling rate (32 kHz).
  *
  * Input:
  *      - inst               : Decoder context
  *      - encoded            : Encoded data
  *      - encoded_bytes      : Bytes in encoded vector
  *
  * Output:
  *      - decoded            : The decoded vector (previous frame)
  *
  * Return value              : >0 - Samples per channel in decoded vector
  *                              0 - No FEC data in the packet
  *                             -1 - Error
  */
-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);
 
 /****************************************************************************
  * WebRtcOpus_DurationEst(...)
  *
  * This function calculates the duration of an opus packet.
  * Input:
  *        - inst                 : Decoder context
  *        - payload              : Encoded data pointer
  *        - payload_length_bytes : Bytes of encoded data
  *
@@ skipping 36 matching lines @@
  *                                 1 - the packet contains FEC.
  */
 int WebRtcOpus_PacketHasFec(const uint8_t* payload,
                             int payload_length_bytes);
 
 #ifdef __cplusplus
 }  // extern "C"
 #endif
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_