Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/agc/legacy/gain_control.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 48 matching lines @@
  *      - agcInst           : AGC instance.
  *      - inFar             : Far-end input speech vector
  *      - samples           : Number of samples in input vector
  *
  * Return value:
  *                          :  0 - Normal operation.
  *                          : -1 - Error
  */
 int WebRtcAgc_AddFarend(void* agcInst,
                         const int16_t* inFar,
-                        int16_t samples);
+                        size_t samples);
 
 /*
  * This function processes a 10 ms frame of microphone speech to determine
  * if there is active speech. The length of the input speech vector must be
  * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
  * FS=48000). For very low input levels, the input signal is increased in level
  * by multiplying and overwriting the samples in inMic[].
  *
  * This function should be called before any further processing of the
  * near-end microphone signal.
  *
  * Input:
  *      - agcInst           : AGC instance.
  *      - inMic             : Microphone input speech vector for each band
  *      - num_bands         : Number of bands in input vector
  *      - samples           : Number of samples in input vector
  *
  * Return value:
  *                          :  0 - Normal operation.
  *                          : -1 - Error
  */
 int WebRtcAgc_AddMic(void* agcInst,
                      int16_t* const* inMic,
-                     int16_t num_bands,
-                     int16_t samples);
+                     size_t num_bands,
+                     size_t samples);
 
 /*
  * This function replaces the analog microphone with a virtual one.
  * It is a digital gain applied to the input signal and is used in the
  * agcAdaptiveDigital mode where no microphone level is adjustable. The length
  * of the input speech vector must be given in samples (80 when FS=8000, and 160
  * when FS=16000, FS=32000 or FS=48000).
  *
  * Input:
  *      - agcInst           : AGC instance.
  *      - inMic             : Microphone input speech vector for each band
  *      - num_bands         : Number of bands in input vector
  *      - samples           : Number of samples in input vector
  *      - micLevelIn        : Input level of microphone (static)
  *
  * Output:
  *      - inMic             : Microphone output after processing (L band)
  *      - inMic_H           : Microphone output after processing (H band)
  *      - micLevelOut       : Adjusted microphone level after processing
  *
  * Return value:
  *                          :  0 - Normal operation.
  *                          : -1 - Error
  */
 int WebRtcAgc_VirtualMic(void* agcInst,
                          int16_t* const* inMic,
-                         int16_t num_bands,
-                         int16_t samples,
+                         size_t num_bands,
+                         size_t samples,
                          int32_t micLevelIn,
                          int32_t* micLevelOut);
 
 /*
  * This function processes a 10 ms frame and adjusts (normalizes) the gain both
  * analog and digitally. The gain adjustments are done only during active
  * periods of speech. The length of the speech vectors must be given in samples
  * (80 when FS=8000, and 160 when FS=16000, FS=32000 or FS=48000). The echo
  * parameter can be used to ensure the AGC will not adjust upward in the
  * presence of echo.
@@ skipping 19 matching lines @@
  *      - saturationWarning : A returned value of 1 indicates a saturation event
  *                            has occurred and the volume cannot be further
  *                            reduced. Otherwise will be set to 0.
  *
  * Return value:
  *                          :  0 - Normal operation.
  *                          : -1 - Error
  */
 int WebRtcAgc_Process(void* agcInst,
                       const int16_t* const* inNear,
-                      int16_t num_bands,
-                      int16_t samples,
+                      size_t num_bands,
+                      size_t samples,
                       int16_t* const* out,
                       int32_t inMicLevel,
                       int32_t* outMicLevel,
                       int16_t echo,
                       uint8_t* saturationWarning);
 
 /*
  * This function sets the config parameters (targetLevelDbfs,
  * compressionGaindB and limiterEnable).
  *
@@ skipping 59 matching lines @@
                    int32_t minLevel,
                    int32_t maxLevel,
                    int16_t agcMode,
                    uint32_t fs);
 
 #if defined(__cplusplus)
 }
 #endif
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_