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webrtc / src / webrtc / 816e15e48525a3d9e91000ccfd73b88cd84a1bb4 / . / modules / audio_processing / agc / legacy / analog_agc.c
blob: d2155648d63d2015eb72f90bf491791418045e64 [file] [log] [blame]
/*
* 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 "webrtc/modules/audio_processing/agc/legacy/analog_agc.h"
#include <stdlib.h>
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
#endif
#include "webrtc/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;
}
}