/* * 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 #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(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(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(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(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(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(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(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(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(malloc(sizeof(LegacyAgc))); stt->initFlag = 0; stt->lastError = 0; return stt; } void WebRtcAgc_Free(void* state) { LegacyAgc* stt; stt = reinterpret_cast(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(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