Revert "Upconvert various types to int."

webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h

 /*
  *  Copyright (c) 2011 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 126 matching lines @@
    *        - encoded           : the encoded data vector
    *
    * Return value:
    *                            : >0 - Length (in bytes) of coded data
    *                            :  0 - The buffer didn't reach the chosen
    *                               frame-size so it keeps buffering speech
    *                               samples.
    *                            : -1 - Error
    */
 
-  int WebRtcIsac_Encode(
+  int16_t WebRtcIsac_Encode(
       ISACStruct*        ISAC_main_inst,
       const int16_t* speechIn,
       uint8_t* encoded);
 
 
   /******************************************************************************
    * WebRtcIsac_DecoderInit(...)
    *
    * This function initializes an ISAC instance prior to the decoder calls.
    *
@@ skipping 49 matching lines @@
    *        - encoded           : encoded ISAC frame(s).
    *        - len               : bytes in encoded vector.
    *
    * Output:
    *        - decoded           : The decoded vector.
    *
    * Return value               : >0 - number of samples in decoded vector.
    *                              -1 - Error.
    */
 
-  int WebRtcIsac_Decode(
+  int16_t WebRtcIsac_Decode(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
       int16_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
 
   /******************************************************************************
    * WebRtcIsac_DecodePlc(...)
    *
@@ skipping 34 matching lines @@
    *                              in bits/second.
    *        - framesize         : frame-size in millisecond.
    *
    * Return value               : 0  - ok
    *                             -1 - Error
    */
 
   int16_t WebRtcIsac_Control(
       ISACStruct*   ISAC_main_inst,
       int32_t rate,
-      int framesize);
+      int16_t framesize);
 
 
   /******************************************************************************
    * WebRtcIsac_ControlBwe(...)
    *
    * This function sets the initial values of bottleneck and frame-size if
    * iSAC is used in channel-adaptive mode. Therefore, this API is not
    * applicable if the codec is created to operate in super-wideband mode.
    *
    * Through this API, users can enforce a frame-size for all values of
@@ skipping 10 matching lines @@
    *                              out the adaptation process, 0 to let iSAC
    *                              change the frame-size if required.
    *
    * Return value               : 0  - ok
    *                             -1 - Error
    */
 
   int16_t WebRtcIsac_ControlBwe(
       ISACStruct* ISAC_main_inst,
       int32_t rateBPS,
-      int frameSizeMs,
+      int16_t frameSizeMs,
       int16_t enforceFrameSize);
 
 
   /******************************************************************************
    * WebRtcIsac_ReadFrameLen(...)
    *
    * This function returns the length of the frame represented in the packet.
    *
    * Input:
    *        - encoded           : Encoded bit-stream
@@ skipping 380 matching lines @@
    *      - ISAC_main_inst     : ISAC instance.
    *      - encoded            : encoded ISAC RCU frame(s)
    *      - len                : bytes in encoded vector
    *
    * Output:
    *      - decoded            : The decoded vector
    *
    * Return value              : >0 - number of samples in decoded vector
    *                             -1 - Error
    */
-  int WebRtcIsac_DecodeRcu(
+  int16_t WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
       int16_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */