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.
 */

/* analog_agc.c
 *
 * Using a feedback system, determines an appropriate analog volume level
 * given an input signal and current volume level. Targets a conservative
 * signal level and is intended for use with a digital AGC to apply
 * additional gain.
 *
 */

#include "modules/audio_processing/agc/legacy/analog_agc.h"

#include <stdlib.h>
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
#endif

#include "rtc_base/checks.h"

/* The slope of in Q13*/
static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129,
                                   2372,  1362,  472,  78};

/* The offset in Q14 */
static const int16_t kOffset1[8] = {25395, 23911, 22206, 20737,
                                    19612, 18805, 17951, 17367};

/* 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 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
 * varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future
 * replace it with
 * a table.
 */
#define OFFSET_ENV_TO_RMS 9
/* The reference input level at which the digital part gives an output of
 * targetLevelDbfs
 * (desired level) if we have no compression gain. This level should be set high
 * enough not
 * to compress the peaks due to the dynamics.
 */
#define DIGITAL_REF_AT_0_COMP_GAIN 4
/* Speed of reference level decrease.
 */
#define DIFF_REF_TO_ANALOG 5

#ifdef MIC_LEVEL_FEEDBACK
#define NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET 7
#endif
/* Size of analog gain table */
#define GAIN_TBL_LEN 32
/* Matlab code:
 * fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
 */
/* Q12 */
static const uint16_t kGainTableAnalog[GAIN_TBL_LEN] = {
    4096, 4251, 4412, 4579,  4752,  4932,  5118,  5312,  5513,  5722, 5938,
    6163, 6396, 6638, 6889,  7150,  7420,  7701,  7992,  8295,  8609, 8934,
    9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};

/* Gain/Suppression tables for virtual Mic (in Q10) */
static const uint16_t kGainTableVirtualMic[128] = {
    1052,  1081,  1110,  1141,  1172,  1204,  1237,  1271,  1305,  1341,  1378,
    1416,  1454,  1494,  1535,  1577,  1620,  1664,  1710,  1757,  1805,  1854,
    1905,  1957,  2010,  2065,  2122,  2180,  2239,  2301,  2364,  2428,  2495,
    2563,  2633,  2705,  2779,  2855,  2933,  3013,  3096,  3180,  3267,  3357,
    3449,  3543,  3640,  3739,  3842,  3947,  4055,  4166,  4280,  4397,  4517,
    4640,  4767,  4898,  5032,  5169,  5311,  5456,  5605,  5758,  5916,  6078,
    6244,  6415,  6590,  6770,  6956,  7146,  7341,  7542,  7748,  7960,  8178,
    8402,  8631,  8867,  9110,  9359,  9615,  9878,  10148, 10426, 10711, 11004,
    11305, 11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807,
    15212, 15628, 16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923,
    20468, 21028, 21603, 22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808,
    27541, 28295, 29069, 29864, 30681, 31520, 32382};
static const uint16_t kSuppressionTableVirtualMic[128] = {
    1024, 1006, 988, 970, 952, 935, 918, 902, 886, 870, 854, 839, 824, 809, 794,
    780,  766,  752, 739, 726, 713, 700, 687, 675, 663, 651, 639, 628, 616, 605,
    594,  584,  573, 563, 553, 543, 533, 524, 514, 505, 496, 487, 478, 470, 461,
    453,  445,  437, 429, 421, 414, 406, 399, 392, 385, 378, 371, 364, 358, 351,
    345,  339,  333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278, 273, 268,
    263,  258,  254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
    200,  197,  193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155,
    153,  150,  147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118,
    116,  114,  112, 110, 108, 106, 104, 102};

/* Table for target energy levels. Values in Q(-7)
 * Matlab code
 * 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,
                     size_t num_bands,
                     size_t samples) {
  int32_t nrg, max_nrg, sample, tmp32;
  int32_t* ptr;
  uint16_t targetGainIdx, gain;
  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;
    if (samples != 160) {
      return -1;
    }
  }

  /* apply slowly varying digital gain */
  if (stt->micVol > stt->maxAnalog) {
    /* |maxLevel| is strictly >= |micVol|, so this condition should be
     * satisfied here, ensuring there is no divide-by-zero. */
    RTC_DCHECK_GT(stt->maxLevel, stt->maxAnalog);

    /* Q1 */
    tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
    tmp32 = (GAIN_TBL_LEN - 1) * tmp16;
    tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
    targetGainIdx = tmp32 / tmp16;
    RTC_DCHECK_LT(targetGainIdx, GAIN_TBL_LEN);

    /* Increment through the table towards the target gain.
     * If micVol drops below maxAnalog, we allow the gain
     * to be dropped immediately. */
    if (stt->gainTableIdx < targetGainIdx) {
      stt->gainTableIdx++;
    } else if (stt->gainTableIdx > targetGainIdx) {
      stt->gainTableIdx--;
    }

    /* Q12 */
    gain = kGainTableAnalog[stt->gainTableIdx];

    for (i = 0; i < samples; i++) {
      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 {
          in_mic[j][i] = (int16_t)sample;
        }
      }
    }
  } else {
    stt->gainTableIdx = 0;
  }

  /* compute envelope */
  if (stt->inQueue > 0) {
    ptr = stt->env[1];
  } else {
    ptr = stt->env[0];
  }

  for (i = 0; i < kNumSubframes; i++) {
    /* iterate over samples */
    max_nrg = 0;
    for (n = 0; n < L; n++) {
      nrg = in_mic[0][i * L + n] * in_mic[0][i * L + n];
      if (nrg > max_nrg) {
        max_nrg = nrg;
      }
    }
    ptr[i] = max_nrg;
  }

  /* compute energy */
  if (stt->inQueue > 0) {
    ptr = stt->Rxx16w32_array[1];
  } else {
    ptr = stt->Rxx16w32_array[0];
  }

  for (i = 0; i < kNumSubframes / 2; i++) {
    if (stt->fs == 16000) {
      WebRtcSpl_DownsampleBy2(&in_mic[0][i * 32], 32, tmp_speech,
                              stt->filterState);
    } else {
      memcpy(tmp_speech, &in_mic[0][i * 16], 16 * sizeof(short));
    }
    /* Compute energy in blocks of 16 samples */
    ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
  }

  /* update queue information */
  if (stt->inQueue == 0) {
    stt->inQueue = 1;
  } else {
    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, size_t samples) {
  LegacyAgc* stt = (LegacyAgc*)state;

  int err = WebRtcAgc_GetAddFarendError(state, samples);

  if (err != 0)
    return err;

  return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
}

int WebRtcAgc_GetAddFarendError(void* state, size_t samples) {
  LegacyAgc* stt;
  stt = (LegacyAgc*)state;

  if (stt == NULL)
    return -1;

  if (stt->fs == 8000) {
    if (samples != 80)
      return -1;
  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
    if (samples != 160)
      return -1;
  } else {
    return -1;
  }

  return 0;
}

int WebRtcAgc_VirtualMic(void* agcInst,
                         int16_t* const* in_near,
                         size_t num_bands,
                         size_t samples,
                         int32_t micLevelIn,
                         int32_t* micLevelOut) {
  int32_t tmpFlt, micLevelTmp, gainIdx;
  uint16_t gain;
  size_t ii, j;
  LegacyAgc* stt;

  uint32_t nrg;
  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.
   *  The idea is that digital AGC will not adapt to low-level
   *  signals.
   */
  if (stt->fs != 8000) {
    frameNrgLimit = frameNrgLimit << 1;
  }

  frameNrg = (uint32_t)(in_near[0][0] * in_near[0][0]);
  for (sampleCntr = 1; sampleCntr < samples; sampleCntr++) {
    // increment frame energy if it is less than the limit
    // the correct value of the energy is not important
    if (frameNrg < frameNrgLimit) {
      nrg = (uint32_t)(in_near[0][sampleCntr] * in_near[0][sampleCntr]);
      frameNrg += nrg;
    }

    // Count the zero crossings
    numZeroCrossing +=
        ((in_near[0][sampleCntr] ^ in_near[0][sampleCntr - 1]) < 0);
  }

  if ((frameNrg < 500) || (numZeroCrossing <= 5)) {
    stt->lowLevelSignal = 1;
  } else if (numZeroCrossing <= kZeroCrossingLowLim) {
    stt->lowLevelSignal = 0;
  } else if (frameNrg <= frameNrgLimit) {
    stt->lowLevelSignal = 1;
  } else if (numZeroCrossing >= kZeroCrossingHighLim) {
    stt->lowLevelSignal = 1;
  } else {
    stt->lowLevelSignal = 0;
  }

  micLevelTmp = micLevelIn << stt->scale;
  /* Set desired level */
  gainIdx = stt->micVol;
  if (stt->micVol > stt->maxAnalog) {
    gainIdx = stt->maxAnalog;
  }
  if (micLevelTmp != stt->micRef) {
    /* Something has happened with the physical level, restart. */
    stt->micRef = micLevelTmp;
    stt->micVol = 127;
    *micLevelOut = 127;
    stt->micGainIdx = 127;
    gainIdx = 127;
  }
  /* Pre-process the signal to emulate the microphone level. */
  /* Take one step at a time in the gain table. */
  if (gainIdx > 127) {
    gain = kGainTableVirtualMic[gainIdx - 128];
  } else {
    gain = kSuppressionTableVirtualMic[127 - gainIdx];
  }
  for (ii = 0; ii < samples; ii++) {
    tmpFlt = (in_near[0][ii] * gain) >> 10;
    if (tmpFlt > 32767) {
      tmpFlt = 32767;
      gainIdx--;
      if (gainIdx >= 127) {
        gain = kGainTableVirtualMic[gainIdx - 127];
      } else {
        gain = kSuppressionTableVirtualMic[127 - gainIdx];
      }
    }
    if (tmpFlt < -32768) {
      tmpFlt = -32768;
      gainIdx--;
      if (gainIdx >= 127) {
        gain = kGainTableVirtualMic[gainIdx - 127];
      } else {
        gain = kSuppressionTableVirtualMic[127 - gainIdx];
      }
    }
    in_near[0][ii] = (int16_t)tmpFlt;
    for (j = 1; j < num_bands; ++j) {
      tmpFlt = (in_near[j][ii] * gain) >> 10;
      if (tmpFlt > 32767) {
        tmpFlt = 32767;
      }
      if (tmpFlt < -32768) {
        tmpFlt = -32768;
      }
      in_near[j][ii] = (int16_t)tmpFlt;
    }
  }
  /* Set the level we (finally) used */
  stt->micGainIdx = gainIdx;
  //    *micLevelOut = stt->micGainIdx;
  *micLevelOut = stt->micGainIdx >> stt->scale;
  /* Add to Mic as if it was the output from a true microphone */
  if (WebRtcAgc_AddMic(agcInst, in_near, num_bands, samples) != 0) {
    return -1;
  }
  return 0;
}

void WebRtcAgc_UpdateAgcThresholds(LegacyAgc* stt) {
  int16_t tmp16;
#ifdef MIC_LEVEL_FEEDBACK
  int zeros;

  if (stt->micLvlSat) {
    /* Lower the analog target level since we have reached its maximum */
    zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
    stt->targetIdxOffset = (3 * zeros - stt->targetIdx - 2) / 4;
  }
#endif

  /* Set analog target level in envelope dBOv scale */
  tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
  tmp16 = WebRtcSpl_DivW32W16ResW16((int32_t)tmp16, ANALOG_TARGET_LEVEL);
  stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
  if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN) {
    stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
  }
  if (stt->agcMode == kAgcModeFixedDigital) {
    /* Adjust for different parameter interpretation in FixedDigital mode */
    stt->analogTarget = stt->compressionGaindB;
  }
#ifdef MIC_LEVEL_FEEDBACK
  stt->analogTarget += stt->targetIdxOffset;
#endif
  /* Since the offset between RMS and ENV is not constant, we should make this
   * into a
   * table, but for now, we'll stick with a constant, tuned for the chosen
   * analog
   * target level.
   */
  stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
#ifdef MIC_LEVEL_FEEDBACK
  stt->targetIdx += stt->targetIdxOffset;
#endif
  /* Analog adaptation limits */
  /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
  stt->analogTargetLevel =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
  stt->startUpperLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 1]; /* -19 dBov */
  stt->startLowerLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 1]; /* -21 dBov */
  stt->upperPrimaryLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 2]; /* -18 dBov */
  stt->lowerPrimaryLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 2]; /* -22 dBov */
  stt->upperSecondaryLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 5]; /* -15 dBov */
  stt->lowerSecondaryLimit =
      RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 5]; /* -25 dBov */
  stt->upperLimit = stt->startUpperLimit;
  stt->lowerLimit = stt->startLowerLimit;
}

void WebRtcAgc_SaturationCtrl(LegacyAgc* stt,
                              uint8_t* saturated,
                              int32_t* env) {
  int16_t i, tmpW16;

  /* Check if the signal is saturated */
  for (i = 0; i < 10; i++) {
    tmpW16 = (int16_t)(env[i] >> 20);
    if (tmpW16 > 875) {
      stt->envSum += tmpW16;
    }
  }

  if (stt->envSum > 25000) {
    *saturated = 1;
    stt->envSum = 0;
  }

  /* stt->envSum *= 0.99; */
  stt->envSum = (int16_t)((stt->envSum * 32440) >> 15);
}

void WebRtcAgc_ZeroCtrl(LegacyAgc* stt, int32_t* inMicLevel, int32_t* env) {
  int16_t i;
  int64_t tmp = 0;
  int32_t midVal;

  /* Is the input signal zero? */
  for (i = 0; i < 10; i++) {
    tmp += env[i];
  }

  /* Each block is allowed to have a few non-zero
   * samples.
   */
  if (tmp < 500) {
    stt->msZero += 10;
  } else {
    stt->msZero = 0;
  }

  if (stt->muteGuardMs > 0) {
    stt->muteGuardMs -= 10;
  }

  if (stt->msZero > 500) {
    stt->msZero = 0;

    /* Increase microphone level only if it's less than 50% */
    midVal = (stt->maxAnalog + stt->minLevel + 1) / 2;
    if (*inMicLevel < midVal) {
      /* *inMicLevel *= 1.1; */
      *inMicLevel = (1126 * *inMicLevel) >> 10;
      /* Reduces risk of a muted mic repeatedly triggering excessive levels due
       * to zero signal detection. */
      *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
      stt->micVol = *inMicLevel;
    }

#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt,
            "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold,"
            " micVol: %d\n",
            stt->fcount, stt->micVol);
#endif

    stt->activeSpeech = 0;
    stt->Rxx16_LPw32Max = 0;

    /* The AGC has a tendency (due to problems with the VAD parameters), to
     * vastly increase the volume after a muting event. This timer prevents
     * upwards adaptation for a short period. */
    stt->muteGuardMs = kMuteGuardTimeMs;
  }
}

void WebRtcAgc_SpeakerInactiveCtrl(LegacyAgc* stt) {
  /* Check if the near end speaker is inactive.
   * If that is the case the VAD threshold is
   * increased since the VAD speech model gets
   * more sensitive to any sound after a long
   * silence.
   */

  int32_t tmp32;
  int16_t vadThresh;

  if (stt->vadMic.stdLongTerm < 2500) {
    stt->vadThreshold = 1500;
  } else {
    vadThresh = kNormalVadThreshold;
    if (stt->vadMic.stdLongTerm < 4500) {
      /* Scale between min and max threshold */
      vadThresh += (4500 - stt->vadMic.stdLongTerm) / 2;
    }

    /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
    tmp32 = vadThresh + 31 * stt->vadThreshold;
    stt->vadThreshold = (int16_t)(tmp32 >> 5);
  }
}

void WebRtcAgc_ExpCurve(int16_t volume, int16_t* index) {
  // volume in Q14
  // index in [0-7]
  /* 8 different curves */
  if (volume > 5243) {
    if (volume > 7864) {
      if (volume > 12124) {
        *index = 7;
      } else {
        *index = 6;
      }
    } else {
      if (volume > 6554) {
        *index = 5;
      } else {
        *index = 4;
      }
    }
  } else {
    if (volume > 2621) {
      if (volume > 3932) {
        *index = 3;
      } else {
        *index = 2;
      }
    } else {
      if (volume > 1311) {
        *index = 1;
      } else {
        *index = 0;
      }
    }
  }
}

int32_t WebRtcAgc_ProcessAnalog(void* state,
                                int32_t inMicLevel,
                                int32_t* outMicLevel,
                                int16_t vadLogRatio,
                                int16_t echo,
                                uint8_t* saturationWarning) {
  uint32_t tmpU32;
  int32_t Rxx16w32, tmp32;
  int32_t inMicLevelTmp, lastMicVol;
  int16_t i;
  uint8_t saturated = 0;
  LegacyAgc* stt;

  stt = (LegacyAgc*)state;
  inMicLevelTmp = inMicLevel << stt->scale;

  if (inMicLevelTmp > stt->maxAnalog) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n",
            stt->fcount);
#endif
    return -1;
  } else if (inMicLevelTmp < stt->minLevel) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n",
            stt->fcount);
#endif
    return -1;
  }

  if (stt->firstCall == 0) {
    int32_t tmpVol;
    stt->firstCall = 1;
    tmp32 = ((stt->maxLevel - stt->minLevel) * 51) >> 9;
    tmpVol = (stt->minLevel + tmp32);

    /* If the mic level is very low at start, increase it! */
    if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog)) {
      inMicLevelTmp = tmpVol;
    }
    stt->micVol = inMicLevelTmp;
  }

  /* Set the mic level to the previous output value if there is digital input
   * gain */
  if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog)) {
    inMicLevelTmp = stt->micVol;
  }

  /* If the mic level was manually changed to a very low value raise it! */
  if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput)) {
    tmp32 = ((stt->maxLevel - stt->minLevel) * 51) >> 9;
    inMicLevelTmp = (stt->minLevel + tmp32);
    stt->micVol = inMicLevelTmp;
#ifdef MIC_LEVEL_FEEDBACK
// stt->numBlocksMicLvlSat = 0;
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt,
            "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual"
            " decrease, raise vol\n",
            stt->fcount);
#endif
  }

  if (inMicLevelTmp != stt->micVol) {
    if (inMicLevel == stt->lastInMicLevel) {
      // We requested a volume adjustment, but it didn't occur. This is
      // probably due to a coarse quantization of the volume slider.
      // Restore the requested value to prevent getting stuck.
      inMicLevelTmp = stt->micVol;
    } else {
      // As long as the value changed, update to match.
      stt->micVol = inMicLevelTmp;
    }
  }

  if (inMicLevelTmp > stt->maxLevel) {
    // Always allow the user to raise the volume above the maxLevel.
    stt->maxLevel = inMicLevelTmp;
  }

  // Store last value here, after we've taken care of manual updates etc.
  stt->lastInMicLevel = inMicLevel;
  lastMicVol = stt->micVol;

  /* Checks if the signal is saturated. Also a check if individual samples
   * are larger than 12000 is done. If they are the counter for increasing
   * the volume level is set to -100ms
   */
  WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);

  /* The AGC is always allowed to lower the level if the signal is saturated */
  if (saturated == 1) {
    /* Lower the recording level
     * Rxx160_LP is adjusted down because it is so slow it could
     * cause the AGC to make wrong decisions. */
    /* stt->Rxx160_LPw32 *= 0.875; */
    stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 8) * 7;

    stt->zeroCtrlMax = stt->micVol;

    /* stt->micVol *= 0.903; */
    tmp32 = inMicLevelTmp - stt->minLevel;
    tmpU32 = WEBRTC_SPL_UMUL(29591, (uint32_t)(tmp32));
    stt->micVol = (tmpU32 >> 15) + stt->minLevel;
    if (stt->micVol > lastMicVol - 2) {
      stt->micVol = lastMicVol - 2;
    }
    inMicLevelTmp = stt->micVol;

#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt,
            "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
            stt->fcount, stt->micVol);
#endif

    if (stt->micVol < stt->minOutput) {
      *saturationWarning = 1;
    }

    /* Reset counter for decrease of volume level to avoid
     * decreasing too much. The saturation control can still
     * lower the level if needed. */
    stt->msTooHigh = -100;

    /* Enable the control mechanism to ensure that our measure,
     * Rxx160_LP, is in the correct range. This must be done since
     * the measure is very slow. */
    stt->activeSpeech = 0;
    stt->Rxx16_LPw32Max = 0;

    /* Reset to initial values */
    stt->msecSpeechInnerChange = kMsecSpeechInner;
    stt->msecSpeechOuterChange = kMsecSpeechOuter;
    stt->changeToSlowMode = 0;

    stt->muteGuardMs = 0;

    stt->upperLimit = stt->startUpperLimit;
    stt->lowerLimit = stt->startLowerLimit;
#ifdef MIC_LEVEL_FEEDBACK
// stt->numBlocksMicLvlSat = 0;
#endif
  }

  /* Check if the input speech is zero. If so the mic volume
   * is increased. On some computers the input is zero up as high
   * level as 17% */
  WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);

  /* Check if the near end speaker is inactive.
   * If that is the case the VAD threshold is
   * increased since the VAD speech model gets
   * more sensitive to any sound after a long
   * silence.
   */
  WebRtcAgc_SpeakerInactiveCtrl(stt);

  for (i = 0; i < 5; i++) {
    /* Computed on blocks of 16 samples */

    Rxx16w32 = stt->Rxx16w32_array[0][i];

    /* Rxx160w32 in Q(-7) */
    tmp32 = (Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos]) >> 3;
    stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
    stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;

    /* Circular buffer */
    stt->Rxx16pos++;
    if (stt->Rxx16pos == RXX_BUFFER_LEN) {
      stt->Rxx16pos = 0;
    }

    /* Rxx16_LPw32 in Q(-4) */
    tmp32 = (Rxx16w32 - stt->Rxx16_LPw32) >> kAlphaShortTerm;
    stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;

    if (vadLogRatio > stt->vadThreshold) {
      /* Speech detected! */

      /* Check if Rxx160_LP is in the correct range. If
       * it is too high/low then we set it to the maximum of
       * Rxx16_LPw32 during the first 200ms of speech.
       */
      if (stt->activeSpeech < 250) {
        stt->activeSpeech += 2;

        if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max) {
          stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
        }
      } else if (stt->activeSpeech == 250) {
        stt->activeSpeech += 2;
        tmp32 = stt->Rxx16_LPw32Max >> 3;
        stt->Rxx160_LPw32 = tmp32 * RXX_BUFFER_LEN;
      }

      tmp32 = (stt->Rxx160w32 - stt->Rxx160_LPw32) >> kAlphaLongTerm;
      stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;

      if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit) {
        stt->msTooHigh += 2;
        stt->msTooLow = 0;
        stt->changeToSlowMode = 0;

        if (stt->msTooHigh > stt->msecSpeechOuterChange) {
          stt->msTooHigh = 0;

          /* Lower the recording level */
          /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
          tmp32 = stt->Rxx160_LPw32 >> 6;
          stt->Rxx160_LPw32 = tmp32 * 53;

          /* Reduce the max gain to avoid excessive oscillation
           * (but never drop below the maximum analog level).
           */
          stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
          stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);

          stt->zeroCtrlMax = stt->micVol;

          /* 0.95 in Q15 */
          tmp32 = inMicLevelTmp - stt->minLevel;
          tmpU32 = WEBRTC_SPL_UMUL(31130, (uint32_t)(tmp32));
          stt->micVol = (tmpU32 >> 15) + stt->minLevel;
          if (stt->micVol > lastMicVol - 1) {
            stt->micVol = lastMicVol - 1;
          }
          inMicLevelTmp = stt->micVol;

          /* Enable the control mechanism to ensure that our measure,
           * Rxx160_LP, is in the correct range.
           */
          stt->activeSpeech = 0;
          stt->Rxx16_LPw32Max = 0;
#ifdef MIC_LEVEL_FEEDBACK
// stt->numBlocksMicLvlSat = 0;
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
          fprintf(stt->fpt,
                  "\tAGC->ProcessAnalog, frame %d: measure >"
                  " 2ndUpperLim, micVol = %d, maxLevel = %d\n",
                  stt->fcount, stt->micVol, stt->maxLevel);
#endif
        }
      } else if (stt->Rxx160_LPw32 > stt->upperLimit) {
        stt->msTooHigh += 2;
        stt->msTooLow = 0;
        stt->changeToSlowMode = 0;

        if (stt->msTooHigh > stt->msecSpeechInnerChange) {
          /* Lower the recording level */
          stt->msTooHigh = 0;
          /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 53;

          /* Reduce the max gain to avoid excessive oscillation
           * (but never drop below the maximum analog level).
           */
          stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
          stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);

          stt->zeroCtrlMax = stt->micVol;

          /* 0.965 in Q15 */
          tmp32 = inMicLevelTmp - stt->minLevel;
          tmpU32 =
              WEBRTC_SPL_UMUL(31621, (uint32_t)(inMicLevelTmp - stt->minLevel));
          stt->micVol = (tmpU32 >> 15) + stt->minLevel;
          if (stt->micVol > lastMicVol - 1) {
            stt->micVol = lastMicVol - 1;
          }
          inMicLevelTmp = stt->micVol;

#ifdef MIC_LEVEL_FEEDBACK
// stt->numBlocksMicLvlSat = 0;
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
          fprintf(stt->fpt,
                  "\tAGC->ProcessAnalog, frame %d: measure >"
                  " UpperLim, micVol = %d, maxLevel = %d\n",
                  stt->fcount, stt->micVol, stt->maxLevel);
#endif
        }
      } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit) {
        stt->msTooHigh = 0;
        stt->changeToSlowMode = 0;
        stt->msTooLow += 2;

        if (stt->msTooLow > stt->msecSpeechOuterChange) {
          /* Raise the recording level */
          int16_t index, weightFIX;
          int16_t volNormFIX = 16384;  // =1 in Q14.

          stt->msTooLow = 0;

          /* Normalize the volume level */
          tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
          if (stt->maxInit != stt->minLevel) {
            volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
          }

          /* Find correct curve */
          WebRtcAgc_ExpCurve(volNormFIX, &index);

          /* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05
           */
          weightFIX =
              kOffset1[index] - (int16_t)((kSlope1[index] * volNormFIX) >> 13);

          /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 67;

          tmp32 = inMicLevelTmp - stt->minLevel;
          tmpU32 =
              ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
          stt->micVol = (tmpU32 >> 14) + stt->minLevel;
          if (stt->micVol < lastMicVol + 2) {
            stt->micVol = lastMicVol + 2;
          }

          inMicLevelTmp = stt->micVol;

#ifdef MIC_LEVEL_FEEDBACK
          /* Count ms in level saturation */
          // if (stt->micVol > stt->maxAnalog) {
          if (stt->micVol > 150) {
            /* mic level is saturated */
            stt->numBlocksMicLvlSat++;
            fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
          }
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
          fprintf(stt->fpt,
                  "\tAGC->ProcessAnalog, frame %d: measure <"
                  " 2ndLowerLim, micVol = %d\n",
                  stt->fcount, stt->micVol);
#endif
        }
      } else if (stt->Rxx160_LPw32 < stt->lowerLimit) {
        stt->msTooHigh = 0;
        stt->changeToSlowMode = 0;
        stt->msTooLow += 2;

        if (stt->msTooLow > stt->msecSpeechInnerChange) {
          /* Raise the recording level */
          int16_t index, weightFIX;
          int16_t volNormFIX = 16384;  // =1 in Q14.

          stt->msTooLow = 0;

          /* Normalize the volume level */
          tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
          if (stt->maxInit != stt->minLevel) {
            volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
          }

          /* Find correct curve */
          WebRtcAgc_ExpCurve(volNormFIX, &index);

          /* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1
           */
          weightFIX =
              kOffset2[index] - (int16_t)((kSlope2[index] * volNormFIX) >> 13);

          /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 67;

          tmp32 = inMicLevelTmp - stt->minLevel;
          tmpU32 =
              ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
          stt->micVol = (tmpU32 >> 14) + stt->minLevel;
          if (stt->micVol < lastMicVol + 1) {
            stt->micVol = lastMicVol + 1;
          }

          inMicLevelTmp = stt->micVol;

#ifdef MIC_LEVEL_FEEDBACK
          /* Count ms in level saturation */
          // if (stt->micVol > stt->maxAnalog) {
          if (stt->micVol > 150) {
            /* mic level is saturated */
            stt->numBlocksMicLvlSat++;
            fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
          }
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
          fprintf(stt->fpt,
                  "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol "
                  "= %d\n",
                  stt->fcount, stt->micVol);
#endif
        }
      } else {
        /* The signal is inside the desired range which is:
         * lowerLimit < Rxx160_LP/640 < upperLimit
         */
        if (stt->changeToSlowMode > 4000) {
          stt->msecSpeechInnerChange = 1000;
          stt->msecSpeechOuterChange = 500;
          stt->upperLimit = stt->upperPrimaryLimit;
          stt->lowerLimit = stt->lowerPrimaryLimit;
        } else {
          stt->changeToSlowMode += 2;  // in milliseconds
        }
        stt->msTooLow = 0;
        stt->msTooHigh = 0;

        stt->micVol = inMicLevelTmp;
      }
#ifdef MIC_LEVEL_FEEDBACK
      if (stt->numBlocksMicLvlSat > NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET) {
        stt->micLvlSat = 1;
        fprintf(stderr, "target before = %d (%d)\n", stt->analogTargetLevel,
                stt->targetIdx);
        WebRtcAgc_UpdateAgcThresholds(stt);
        WebRtcAgc_CalculateGainTable(
            &(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
            stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget);
        stt->numBlocksMicLvlSat = 0;
        stt->micLvlSat = 0;
        fprintf(stderr, "target offset = %d\n", stt->targetIdxOffset);
        fprintf(stderr, "target after  = %d (%d)\n", stt->analogTargetLevel,
                stt->targetIdx);
      }
#endif
    }
  }

  /* Ensure gain is not increased in presence of echo or after a mute event
   * (but allow the zeroCtrl() increase on the frame of a mute detection).
   */
  if (echo == 1 ||
      (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs)) {
    if (stt->micVol > lastMicVol) {
      stt->micVol = lastMicVol;
    }
  }

  /* limit the gain */
  if (stt->micVol > stt->maxLevel) {
    stt->micVol = stt->maxLevel;
  } else if (stt->micVol < stt->minOutput) {
    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,
                      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) {
    return -1;
  }
  //

  if (stt->fs == 8000) {
    if (samples != 80) {
      return -1;
    }
  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
    if (samples != 160) {
      return -1;
    }
  } else {
    return -1;
  }

  *saturationWarning = 0;
  // TODO(minyue): PUT IN RANGE CHECKING FOR INPUT LEVELS
  *outMicLevel = inMicLevel;

#ifdef WEBRTC_AGC_DEBUG_DUMP
  stt->fcount++;
#endif

  if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, in_near, num_bands, out,
                               stt->fs, stt->lowLevelSignal) == -1) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n",
            stt->fcount);
#endif
    return -1;
  }
  if (stt->agcMode < kAgcModeFixedDigital &&
      (stt->lowLevelSignal == 0 || stt->agcMode != kAgcModeAdaptiveDigital)) {
    if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevel, outMicLevel,
                                stt->vadMic.logRatio, echo,
                                saturationWarning) == -1) {
      return -1;
    }
  }
#ifdef WEBRTC_AGC_DEBUG_DUMP
  fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\t%d\n", stt->fcount, inMicLevel,
          *outMicLevel, stt->maxLevel, stt->micVol);
#endif

  /* update queue */
  if (stt->inQueue > 1) {
    memcpy(stt->env[0], stt->env[1], 10 * sizeof(int32_t));
    memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(int32_t));
  }

  if (stt->inQueue > 0) {
    stt->inQueue--;
  }

  return 0;
}

int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig agcConfig) {
  LegacyAgc* stt;
  stt = (LegacyAgc*)agcInst;

  if (stt == NULL) {
    return -1;
  }

  if (stt->initFlag != kInitCheck) {
    stt->lastError = AGC_UNINITIALIZED_ERROR;
    return -1;
  }

  if (agcConfig.limiterEnable != kAgcFalse &&
      agcConfig.limiterEnable != kAgcTrue) {
    stt->lastError = AGC_BAD_PARAMETER_ERROR;
    return -1;
  }
  stt->limiterEnable = agcConfig.limiterEnable;
  stt->compressionGaindB = agcConfig.compressionGaindB;
  if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31)) {
    stt->lastError = AGC_BAD_PARAMETER_ERROR;
    return -1;
  }
  stt->targetLevelDbfs = agcConfig.targetLevelDbfs;

  if (stt->agcMode == kAgcModeFixedDigital) {
    /* Adjust for different parameter interpretation in FixedDigital mode */
    stt->compressionGaindB += agcConfig.targetLevelDbfs;
  }

  /* Update threshold levels for analog adaptation */
  WebRtcAgc_UpdateAgcThresholds(stt);

  /* Recalculate gain table */
  if (WebRtcAgc_CalculateGainTable(
          &(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
          stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n",
            stt->fcount);
#endif
    return -1;
  }
  /* Store the config in a WebRtcAgcConfig */
  stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
  stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
  stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;

  return 0;
}

int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config) {
  LegacyAgc* stt;
  stt = (LegacyAgc*)agcInst;

  if (stt == NULL) {
    return -1;
  }

  if (config == NULL) {
    stt->lastError = AGC_NULL_POINTER_ERROR;
    return -1;
  }

  if (stt->initFlag != kInitCheck) {
    stt->lastError = AGC_UNINITIALIZED_ERROR;
    return -1;
  }

  config->limiterEnable = stt->usedConfig.limiterEnable;
  config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
  config->compressionGaindB = stt->usedConfig.compressionGaindB;

  return 0;
}

void* WebRtcAgc_Create() {
  LegacyAgc* stt = malloc(sizeof(LegacyAgc));

#ifdef WEBRTC_AGC_DEBUG_DUMP
  stt->fpt = fopen("./agc_test_log.txt", "wt");
  stt->agcLog = fopen("./agc_debug_log.txt", "wt");
  stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
#endif

  stt->initFlag = 0;
  stt->lastError = 0;

  return stt;
}

void WebRtcAgc_Free(void* state) {
  LegacyAgc* stt;

  stt = (LegacyAgc*)state;
#ifdef WEBRTC_AGC_DEBUG_DUMP
  fclose(stt->fpt);
  fclose(stt->agcLog);
  fclose(stt->digitalAgc.logFile);
#endif
  free(stt);
}

/* minLevel     - Minimum volume level
 * maxLevel     - Maximum volume level
 */
int WebRtcAgc_Init(void* agcInst,
                   int32_t minLevel,
                   int32_t maxLevel,
                   int16_t agcMode,
                   uint32_t fs) {
  int32_t max_add, tmp32;
  int16_t i;
  int tmpNorm;
  LegacyAgc* stt;

  /* typecast state pointer */
  stt = (LegacyAgc*)agcInst;

  if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0) {
    stt->lastError = AGC_UNINITIALIZED_ERROR;
    return -1;
  }

  /* Analog AGC variables */
  stt->envSum = 0;

/* mode     = 0 - Only saturation protection
 *            1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3
 * dBOv)]
 *            2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3
 * dBOv)]
 *            3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
 */
#ifdef WEBRTC_AGC_DEBUG_DUMP
  stt->fcount = 0;
  fprintf(stt->fpt, "AGC->Init\n");
#endif
  if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
#endif
    return -1;
  }
  stt->agcMode = agcMode;
  stt->fs = fs;

  /* initialize input VAD */
  WebRtcAgc_InitVad(&stt->vadMic);

  /* If the volume range is smaller than 0-256 then
   * the levels are shifted up to Q8-domain */
  tmpNorm = WebRtcSpl_NormU32((uint32_t)maxLevel);
  stt->scale = tmpNorm - 23;
  if (stt->scale < 0) {
    stt->scale = 0;
  }
  // TODO(bjornv): Investigate if we really need to scale up a small range now
  // when we have
  // a guard against zero-increments. For now, we do not support scale up (scale
  // = 0).
  stt->scale = 0;
  maxLevel <<= stt->scale;
  minLevel <<= stt->scale;

  /* Make minLevel and maxLevel static in AdaptiveDigital */
  if (stt->agcMode == kAgcModeAdaptiveDigital) {
    minLevel = 0;
    maxLevel = 255;
    stt->scale = 0;
  }
  /* The maximum supplemental volume range is based on a vague idea
   * of how much lower the gain will be than the real analog gain. */
  max_add = (maxLevel - minLevel) / 4;

  /* Minimum/maximum volume level that can be set */
  stt->minLevel = minLevel;
  stt->maxAnalog = maxLevel;
  stt->maxLevel = maxLevel + max_add;
  stt->maxInit = stt->maxLevel;

  stt->zeroCtrlMax = stt->maxAnalog;
  stt->lastInMicLevel = 0;

  /* Initialize micVol parameter */
  stt->micVol = stt->maxAnalog;
  if (stt->agcMode == kAgcModeAdaptiveDigital) {
    stt->micVol = 127; /* Mid-point of mic level */
  }
  stt->micRef = stt->micVol;
  stt->micGainIdx = 127;
#ifdef MIC_LEVEL_FEEDBACK
  stt->numBlocksMicLvlSat = 0;
  stt->micLvlSat = 0;
#endif
#ifdef WEBRTC_AGC_DEBUG_DUMP
  fprintf(stt->fpt, "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
          stt->minLevel, stt->maxAnalog, stt->maxLevel);
#endif

  /* Minimum output volume is 4% higher than the available lowest volume level
   */
  tmp32 = ((stt->maxLevel - stt->minLevel) * 10) >> 8;
  stt->minOutput = (stt->minLevel + tmp32);

  stt->msTooLow = 0;
  stt->msTooHigh = 0;
  stt->changeToSlowMode = 0;
  stt->firstCall = 0;
  stt->msZero = 0;
  stt->muteGuardMs = 0;
  stt->gainTableIdx = 0;

  stt->msecSpeechInnerChange = kMsecSpeechInner;
  stt->msecSpeechOuterChange = kMsecSpeechOuter;

  stt->activeSpeech = 0;
  stt->Rxx16_LPw32Max = 0;

  stt->vadThreshold = kNormalVadThreshold;
  stt->inActive = 0;

  for (i = 0; i < RXX_BUFFER_LEN; i++) {
    stt->Rxx16_vectorw32[i] = (int32_t)1000; /* -54dBm0 */
  }
  stt->Rxx160w32 =
      125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */

  stt->Rxx16pos = 0;
  stt->Rxx16_LPw32 = (int32_t)16284; /* Q(-4) */

  for (i = 0; i < 5; i++) {
    stt->Rxx16w32_array[0][i] = 0;
  }
  for (i = 0; i < 10; i++) {
    stt->env[0][i] = 0;
    stt->env[1][i] = 0;
  }
  stt->inQueue = 0;

#ifdef MIC_LEVEL_FEEDBACK
  stt->targetIdxOffset = 0;
#endif

  WebRtcSpl_MemSetW32(stt->filterState, 0, 8);

  stt->initFlag = kInitCheck;
  // Default config settings.
  stt->defaultConfig.limiterEnable = kAgcTrue;
  stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
  stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;

  if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1) {
    stt->lastError = AGC_UNSPECIFIED_ERROR;
    return -1;
  }
  stt->Rxx160_LPw32 = stt->analogTargetLevel;  // Initialize rms value

  stt->lowLevelSignal = 0;

  /* Only positive values are allowed that are not too large */
  if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000)) {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
#endif
    return -1;
  } else {
#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt, "\n");
#endif
    return 0;
  }
}