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

webrtc/modules/audio_processing/agc/legacy/analog_agc.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 23 matching lines @@
 
 /* The slope of in Q13*/
 static const int16_t kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
 
 /* The offset in Q14 */
 static const int16_t kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670,
         17286};
 
 static const int16_t kMuteGuardTimeMs = 8000;
 static const int16_t kInitCheck = 42;
-static const int16_t kNumSubframes = 10;
+static const size_t kNumSubframes = 10;
 
 /* Default settings if config is not used */
 #define AGC_DEFAULT_TARGET_LEVEL 3
 #define AGC_DEFAULT_COMP_GAIN 9
 /* This is the target level for the analog part in ENV scale. To convert to RMS scale you
  * have to add OFFSET_ENV_TO_RMS.
  */
 #define ANALOG_TARGET_LEVEL 11
 #define ANALOG_TARGET_LEVEL_2 5 // ANALOG_TARGET_LEVEL / 2
 /* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
@@ skipping 50 matching lines @@
  * targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n', round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
 
 static const int32_t kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67264106,
         53429779, 42440782, 33711911, 26778323, 21270778, 16895980, 13420954, 10660642,
         8468049, 6726411, 5342978, 4244078, 3371191, 2677832, 2127078, 1689598, 1342095,
         1066064, 846805, 672641, 534298, 424408, 337119, 267783, 212708, 168960, 134210,
         106606, 84680, 67264, 53430, 42441, 33712, 26778, 21271, 16896, 13421, 10661, 8468,
         6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
         213, 169, 134, 107, 85, 67};
 
-int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, int16_t num_bands,
-                     int16_t samples)
+int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, size_t num_bands,
+                     size_t samples)
 {
     int32_t nrg, max_nrg, sample, tmp32;
     int32_t *ptr;
     uint16_t targetGainIdx, gain;
-    int16_t i, n, L, tmp16, tmp_speech[16];
+    size_t i;
+    int16_t n, L, tmp16, tmp_speech[16];
     LegacyAgc* stt;
     stt = (LegacyAgc*)state;
 
     if (stt->fs == 8000) {
         L = 8;
         if (samples != 80) {
             return -1;
         }
     } else {
         L = 16;
@@ skipping 25 matching lines @@
         } else if (stt->gainTableIdx > targetGainIdx)
         {
             stt->gainTableIdx--;
         }
 
         /* Q12 */
         gain = kGainTableAnalog[stt->gainTableIdx];
 
         for (i = 0; i < samples; i++)
         {
-            int j;
+            size_t j;
             for (j = 0; j < num_bands; ++j)
             {
                 sample = (in_mic[j][i] * gain) >> 12;
                 if (sample > 32767)
                 {
                     in_mic[j][i] = 32767;
                 } else if (sample < -32768)
                 {
                     in_mic[j][i] = -32768;
                 } else
@@ skipping 64 matching lines @@
     {
         stt->inQueue = 2;
     }
 
     /* call VAD (use low band only) */
     WebRtcAgc_ProcessVad(&stt->vadMic, in_mic[0], samples);
 
     return 0;
 }
 
-int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
+int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, size_t samples)
 {
   LegacyAgc* stt;
   stt = (LegacyAgc*)state;
 
     if (stt == NULL)
     {
         return -1;
     }
 
     if (stt->fs == 8000)
@@ skipping 10 matching lines @@
         }
     } else
     {
         return -1;
     }
 
     return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
 }
 
 int WebRtcAgc_VirtualMic(void *agcInst, int16_t* const* in_near,
-                         int16_t num_bands, int16_t samples, int32_t micLevelIn,
+                         size_t num_bands, size_t samples, int32_t micLevelIn,
                          int32_t *micLevelOut)
 {
     int32_t tmpFlt, micLevelTmp, gainIdx;
     uint16_t gain;
-    int16_t ii, j;
+    size_t ii, j;
     LegacyAgc* stt;
 
     uint32_t nrg;
-    int16_t sampleCntr;
+    size_t sampleCntr;
     uint32_t frameNrg = 0;
     uint32_t frameNrgLimit = 5500;
     int16_t numZeroCrossing = 0;
     const int16_t kZeroCrossingLowLim = 15;
     const int16_t kZeroCrossingHighLim = 20;
 
     stt = (LegacyAgc*)agcInst;
 
     /*
      *  Before applying gain decide if this is a low-level signal.
@@ skipping 822 matching lines @@
     {
         stt->micVol = stt->minOutput;
     }
 
     *outMicLevel = WEBRTC_SPL_MIN(stt->micVol, stt->maxAnalog) >> stt->scale;
 
     return 0;
 }
 
 int WebRtcAgc_Process(void *agcInst, const int16_t* const* in_near,
-                      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)
 {
   LegacyAgc* stt;
 
   stt = (LegacyAgc*)agcInst;
 
     //
     if (stt == NULL)
@@ skipping 363 matching lines @@
 #endif
         return -1;
     } else
     {
 #ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt, "\n");
 #endif
         return 0;
     }
 }