Update audio code to use size_t more correctly,

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
 
+#include <stddef.h>
+
 #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/typedefs.h"
 
 typedef struct WebRtcISACStruct    ISACStruct;
 
 #if defined(__cplusplus)
 extern "C" {
 #endif
 
   /******************************************************************************
@@ skipping 155 matching lines @@
    *        - arr_ts            : the arrival time of the packet (from NetEq)
    *                              in samples.
    *
    * Return value               : 0 - Ok
    *                             -1 - Error
    */
 
   int16_t WebRtcIsac_UpdateBwEstimate(
       ISACStruct*         ISAC_main_inst,
       const uint8_t* encoded,
-      int32_t         packet_size,
+      size_t         packet_size,
       uint16_t        rtp_seq_number,
       uint32_t        send_ts,
       uint32_t        arr_ts);
 
 
   /******************************************************************************
    * WebRtcIsac_Decode(...)
    *
    * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
    * of the output audio could be either 480 or 960 samples, equivalent to
    * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
    * output audio is 960 samples, which is 30 ms.
    *
    * Input:
    *        - ISAC_main_inst    : ISAC instance.
    *        - 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(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
-      int16_t         len,
+      size_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
 
   /******************************************************************************
    * WebRtcIsac_DecodePlc(...)
    *
    * This function conducts PLC for ISAC frame(s). Output speech length
    * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
    * the output is multiple of 480 samples if operating at 16 kHz and multiple
    * of 960 if operating at 32 kHz.
    *
    * Input:
    *        - ISAC_main_inst    : ISAC instance.
    *        - noOfLostFrames    : Number of PLC frames to produce.
    *
    * Output:
    *        - decoded           : The decoded vector.
    *
-   * Return value               : >0 - number of samples in decoded PLC vector
-   *                              -1 - Error
+   * Return value               : Number of samples in decoded PLC vector
    */
 
-  int16_t WebRtcIsac_DecodePlc(
+  size_t WebRtcIsac_DecodePlc(
       ISACStruct*  ISAC_main_inst,
       int16_t* decoded,
-      int16_t  noOfLostFrames);
+      size_t  noOfLostFrames);
 
 
   /******************************************************************************
    * WebRtcIsac_Control(...)
    *
    * This function sets the limit on the short-term average bit-rate and the
    * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
    * rate, an average bit-rate between 10000 to 32000 bps is valid and a
    * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
    * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
@@ skipping 441 matching lines @@
    *
    * Output:
    *      - decoded            : The decoded vector
    *
    * Return value              : >0 - number of samples in decoded vector
    *                             -1 - Error
    */
   int WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
-      int16_t         len,
+      size_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
   /* Fills in an IsacBandwidthInfo struct. |inst| should be a decoder. */
   void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
 
   /* Uses the values from an IsacBandwidthInfo struct. |inst| should be an
      encoder. */
   void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
                                    const IsacBandwidthInfo* bwinfo);
 
   /* If |inst| is a decoder but not an encoder: tell it what sample rate the
      encoder is using, for bandwidth estimation purposes. */
   void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */