summaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_processing/agc/legacy/analog_agc.cc
diff options
context:
space:
mode:
Diffstat (limited to 'webrtc/modules/audio_processing/agc/legacy/analog_agc.cc')
-rw-r--r--webrtc/modules/audio_processing/agc/legacy/analog_agc.cc1238
1 files changed, 1238 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/agc/legacy/analog_agc.cc b/webrtc/modules/audio_processing/agc/legacy/analog_agc.cc
new file mode 100644
index 0000000..b53e3f9
--- /dev/null
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.cc
@@ -0,0 +1,1238 @@
+/*
+ * 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.
+ */
+
+/*
+ *
+ * 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>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Errors
+#define AGC_UNSPECIFIED_ERROR 18000
+#define AGC_UNINITIALIZED_ERROR 18002
+#define AGC_NULL_POINTER_ERROR 18003
+#define AGC_BAD_PARAMETER_ERROR 18004
+
+/* 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
+
+/* 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};
+
+} // namespace
+
+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 = reinterpret_cast<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(int16_t));
+ }
+ /* 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 = reinterpret_cast<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 = reinterpret_cast<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 = reinterpret_cast<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;
+
+ /* 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;
+ }
+ /* 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;
+ /* Analog adaptation limits */
+ /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
+ stt->analogTargetLevel =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
+ stt->startUpperLimit =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 1]; /* -19 dBov */
+ stt->startLowerLimit =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx + 1]; /* -21 dBov */
+ stt->upperPrimaryLimit =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 2]; /* -18 dBov */
+ stt->lowerPrimaryLimit =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx + 2]; /* -22 dBov */
+ stt->upperSecondaryLimit =
+ kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 5]; /* -15 dBov */
+ stt->lowerSecondaryLimit =
+ kRxxBufferLen * 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;
+ }
+
+ 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 = reinterpret_cast<LegacyAgc*>(state);
+ inMicLevelTmp = inMicLevel << stt->scale;
+
+ if (inMicLevelTmp > stt->maxAnalog) {
+ return -1;
+ } else if (inMicLevelTmp < stt->minLevel) {
+ 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;
+ }
+
+ 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;
+
+ 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;
+ }
+
+ /* 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 == kRxxBufferLen) {
+ 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 * kRxxBufferLen;
+ }
+
+ 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;
+ }
+ } 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;
+ }
+ } 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;
+ }
+ } 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;
+ }
+ } 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;
+ }
+ }
+ }
+
+ /* 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_Analyze(void* agcInst,
+ const int16_t* const* in_near,
+ size_t num_bands,
+ size_t samples,
+ int32_t inMicLevel,
+ int32_t* outMicLevel,
+ int16_t echo,
+ uint8_t* saturationWarning,
+ int32_t gains[11]) {
+ LegacyAgc* stt = reinterpret_cast<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;
+
+ int32_t error =
+ WebRtcAgc_ComputeDigitalGains(&stt->digitalAgc, in_near, num_bands,
+ stt->fs, stt->lowLevelSignal, gains);
+ if (error == -1) {
+ 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;
+ }
+ }
+
+ /* 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_Process(const void* agcInst,
+ const int32_t gains[11],
+ const int16_t* const* in_near,
+ size_t num_bands,
+ int16_t* const* out) {
+ const LegacyAgc* stt = (const LegacyAgc*)agcInst;
+ return WebRtcAgc_ApplyDigitalGains(gains, num_bands, stt->fs, in_near, out);
+}
+
+int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig agcConfig) {
+ LegacyAgc* stt;
+ stt = reinterpret_cast<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) {
+ 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 = reinterpret_cast<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 = static_cast<LegacyAgc*>(malloc(sizeof(LegacyAgc)));
+
+ stt->initFlag = 0;
+ stt->lastError = 0;
+
+ return stt;
+}
+
+void WebRtcAgc_Free(void* state) {
+ LegacyAgc* stt;
+
+ stt = reinterpret_cast<LegacyAgc*>(state);
+ 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 = reinterpret_cast<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)]
+ */
+ if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital) {
+ 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;
+
+ /* 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 < kRxxBufferLen; i++) {
+ stt->Rxx16_vectorw32[i] = (int32_t)1000; /* -54dBm0 */
+ }
+ stt->Rxx160w32 = 125 * kRxxBufferLen; /* (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;
+
+ 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)) {
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+} // namespace webrtc