/* * 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 #include #ifdef AGC_DEBUG //test log #include #endif #include "webrtc/modules/audio_processing/agc/analog_agc.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; /* 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 *in_mic, int16_t *in_mic_H, int16_t samples) { int32_t nrg, max_nrg, sample, tmp32; int32_t *ptr; uint16_t targetGainIdx, gain; int16_t i, n, L, M, subFrames, tmp16, tmp_speech[16]; Agc_t *stt; stt = (Agc_t *)state; //default/initial values corresponding to 10ms for wb and swb M = 10; L = 16; subFrames = 160; if (stt->fs == 8000) { if (samples == 80) { subFrames = 80; M = 10; L = 8; } else if (samples == 160) { subFrames = 80; M = 20; L = 8; } else { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid number of samples\n\n", (stt->fcount + 1)); #endif return -1; } } else if (stt->fs == 16000) { if (samples == 160) { subFrames = 160; M = 10; L = 16; } else if (samples == 320) { subFrames = 160; M = 20; L = 16; } else { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid number of samples\n\n", (stt->fcount + 1)); #endif return -1; } } else if (stt->fs == 32000) { /* SWB is processed as 160 sample for L and H bands */ if (samples == 160) { subFrames = 160; M = 10; L = 16; } else { #ifdef AGC_DEBUG fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid sample rate\n\n", (stt->fcount + 1)); #endif return -1; } } /* Check for valid pointers based on sampling rate */ if ((stt->fs == 32000) && (in_mic_H == NULL)) { return -1; } /* Check for valid pointer for low band */ if (in_mic == NULL) { 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. */ assert(stt->maxLevel > stt->maxAnalog); /* Q1 */ tmp16 = (int16_t)(stt->micVol - stt->maxAnalog); tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16); tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog); targetGainIdx = (uint16_t)WEBRTC_SPL_DIV(tmp32, tmp16); assert(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++) { // For lower band tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic[i], gain); sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12); if (sample > 32767) { in_mic[i] = 32767; } else if (sample < -32768) { in_mic[i] = -32768; } else { in_mic[i] = (int16_t)sample; } // For higher band if (stt->fs == 32000) { tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic_H[i], gain); sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12); if (sample > 32767) { in_mic_H[i] = 32767; } else if (sample < -32768) { in_mic_H[i] = -32768; } else { in_mic_H[i] = (int16_t)sample; } } } } else { stt->gainTableIdx = 0; } /* compute envelope */ if ((M == 10) && (stt->inQueue > 0)) { ptr = stt->env[1]; } else { ptr = stt->env[0]; } for (i = 0; i < M; i++) { /* iterate over samples */ max_nrg = 0; for (n = 0; n < L; n++) { nrg = WEBRTC_SPL_MUL_16_16(in_mic[i * L + n], in_mic[i * L + n]); if (nrg > max_nrg) { max_nrg = nrg; } } ptr[i] = max_nrg; } /* compute energy */ if ((M == 10) && (stt->inQueue > 0)) { ptr = stt->Rxx16w32_array[1]; } else { ptr = stt->Rxx16w32_array[0]; } for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++) { if (stt->fs == 16000) { WebRtcSpl_DownsampleBy2(&in_mic[i * 32], 32, tmp_speech, stt->filterState); } else { memcpy(tmp_speech, &in_mic[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) && (M == 10)) { stt->inQueue = 1; } else { stt->inQueue = 2; } /* call VAD (use low band only) */ for (i = 0; i < samples; i += subFrames) { WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames); } return 0; } int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples) { int32_t errHandle = 0; int16_t i, subFrames; Agc_t *stt; stt = (Agc_t *)state; if (stt == NULL) { return -1; } if (stt->fs == 8000) { if ((samples != 80) && (samples != 160)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 80; } else if (stt->fs == 16000) { if ((samples != 160) && (samples != 320)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 160; } else if (stt->fs == 32000) { if ((samples != 160) && (samples != 320)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 160; } else { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid sample rate\n\n", stt->fcount + 1); #endif return -1; } for (i = 0; i < samples; i += subFrames) { errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames); } return errHandle; } int WebRtcAgc_VirtualMic(void *agcInst, int16_t *in_near, int16_t *in_near_H, int16_t samples, int32_t micLevelIn, int32_t *micLevelOut) { int32_t tmpFlt, micLevelTmp, gainIdx; uint16_t gain; int16_t ii; Agc_t *stt; uint32_t nrg; int16_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 = (Agc_t *)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 = WEBRTC_SPL_MUL_16_16(in_near[0], in_near[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 = WEBRTC_SPL_MUL_16_16(in_near[sampleCntr], in_near[sampleCntr]); frameNrg += nrg; } // Count the zero crossings numZeroCrossing += ((in_near[sampleCntr] ^ in_near[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 = WEBRTC_SPL_LSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_U16(in_near[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[ii] = (int16_t)tmpFlt; if (stt->fs == 32000) { tmpFlt = WEBRTC_SPL_MUL_16_U16(in_near_H[ii], gain); tmpFlt = WEBRTC_SPL_RSHIFT_W32(tmpFlt, 10); if (tmpFlt > 32767) { tmpFlt = 32767; } if (tmpFlt < -32768) { tmpFlt = -32768; } in_near_H[ii] = (int16_t)tmpFlt; } } /* Set the level we (finally) used */ stt->micGainIdx = gainIdx; // *micLevelOut = stt->micGainIdx; *micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale); /* Add to Mic as if it was the output from a true microphone */ if (WebRtcAgc_AddMic(agcInst, in_near, in_near_H, samples) != 0) { return -1; } return 0; } void WebRtcAgc_UpdateAgcThresholds(Agc_t *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 = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2); } #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(Agc_t *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)WEBRTC_SPL_RSHIFT_W32(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)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum, (int16_t)32440, 15); } void WebRtcAgc_ZeroCtrl(Agc_t *stt, int32_t *inMicLevel, int32_t *env) { int16_t i; int32_t tmp32 = 0; int32_t midVal; /* Is the input signal zero? */ for (i = 0; i < 10; i++) { tmp32 += env[i]; } /* Each block is allowed to have a few non-zero * samples. */ if (tmp32 < 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 = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog + stt->minLevel + 1, 1); if (*inMicLevel < midVal) { /* *inMicLevel *= 1.1; */ tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel); *inMicLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 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 AGC_DEBUG //test log fprintf(stt->fpt, "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\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(Agc_t *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 += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1); } /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */ tmp32 = (int32_t)vadThresh; tmp32 += WEBRTC_SPL_MUL_16_16((int16_t)31, stt->vadThreshold); stt->vadThreshold = (int16_t)WEBRTC_SPL_RSHIFT_W32(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; Agc_t *stt; stt = (Agc_t *)state; inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale); if (inMicLevelTmp > stt->maxAnalog) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount); #endif return -1; } else if (inMicLevelTmp < stt->minLevel) { #ifdef AGC_DEBUG //test log 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 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)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 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)51, 9); inMicLevelTmp = (stt->minLevel + tmp32); stt->micVol = inMicLevelTmp; #ifdef MIC_LEVEL_FEEDBACK //stt->numBlocksMicLvlSat = 0; #endif #ifdef AGC_DEBUG //test log 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 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7); stt->zeroCtrlMax = stt->micVol; /* stt->micVol *= 0.903; */ tmp32 = inMicLevelTmp - stt->minLevel; tmpU32 = WEBRTC_SPL_UMUL(29591, (uint32_t)(tmp32)); stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel; if (stt->micVol > lastMicVol - 2) { stt->micVol = lastMicVol - 2; } inMicLevelTmp = stt->micVol; #ifdef AGC_DEBUG //test log 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 = WEBRTC_SPL_RSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx16_LPw32Max, 3); stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN); } tmp32 = WEBRTC_SPL_RSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6); stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(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; */ tmp32 = (15 * stt->maxLevel) + stt->micVol; stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4); 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 = (int32_t)WEBRTC_SPL_RSHIFT_U32(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 AGC_DEBUG //test log 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 */ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6); stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(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; */ tmp32 = (15 * stt->maxLevel) + stt->micVol; stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4); 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 = (int32_t)WEBRTC_SPL_RSHIFT_U32(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 AGC_DEBUG //test log 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 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14); if (stt->maxInit != stt->minLevel) { volNormFIX = (int16_t)WEBRTC_SPL_DIV(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)WEBRTC_SPL_MUL_16_16_RSFT(kSlope1[index], volNormFIX, 13); /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6); stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67); tmp32 = inMicLevelTmp - stt->minLevel; tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel)); stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(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 AGC_DEBUG //test log 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 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14); if (stt->maxInit != stt->minLevel) { volNormFIX = (int16_t)WEBRTC_SPL_DIV(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)WEBRTC_SPL_MUL_16_16_RSFT(kSlope2[index], volNormFIX, 13); /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6); stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67); tmp32 = inMicLevelTmp - stt->minLevel; tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel)); stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(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 AGC_DEBUG //test log 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_RSHIFT_W32(stt->micVol, stt->scale); if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale)) { *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale); } return 0; } int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, const int16_t *in_near_H, int16_t samples, int16_t *out, int16_t *out_H, int32_t inMicLevel, int32_t *outMicLevel, int16_t echo, uint8_t *saturationWarning) { Agc_t *stt; int32_t inMicLevelTmp; int16_t subFrames, i; uint8_t satWarningTmp = 0; stt = (Agc_t *)agcInst; // if (stt == NULL) { return -1; } // if (stt->fs == 8000) { if ((samples != 80) && (samples != 160)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 80; } else if (stt->fs == 16000) { if ((samples != 160) && (samples != 320)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 160; } else if (stt->fs == 32000) { if ((samples != 160) && (samples != 320)) { #ifdef AGC_DEBUG //test log fprintf(stt->fpt, "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); #endif return -1; } subFrames = 160; } else { #ifdef AGC_DEBUG// test log fprintf(stt->fpt, "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount); #endif return -1; } /* Check for valid pointers based on sampling rate */ if (stt->fs == 32000 && in_near_H == NULL) { return -1; } /* Check for valid pointers for low band */ if (in_near == NULL) { return -1; } *saturationWarning = 0; //TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS *outMicLevel = inMicLevel; inMicLevelTmp = inMicLevel; // TODO(andrew): clearly we don't need input and output pointers... // Change the interface to take a shared input/output. if (in_near != out) { // Only needed if they don't already point to the same place. memcpy(out, in_near, samples * sizeof(int16_t)); } if (stt->fs == 32000) { if (in_near_H != out_H) { memcpy(out_H, in_near_H, samples * sizeof(int16_t)); } } #ifdef AGC_DEBUG//test log stt->fcount++; #endif for (i = 0; i < samples; i += subFrames) { if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i], stt->fs, stt->lowLevelSignal) == -1) { #ifdef AGC_DEBUG//test log 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, inMicLevelTmp, outMicLevel, stt->vadMic.logRatio, echo, saturationWarning) == -1) { return -1; } } #ifdef AGC_DEBUG//test log fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *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--; } /* If 20ms frames are used the input mic level must be updated so that * the analog AGC does not think that there has been a manual volume * change. */ inMicLevelTmp = *outMicLevel; /* Store a positive saturation warning. */ if (*saturationWarning == 1) { satWarningTmp = 1; } } /* Trigger the saturation warning if displayed by any of the frames. */ *saturationWarning = satWarningTmp; return 0; } int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig) { Agc_t *stt; stt = (Agc_t *)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 AGC_DEBUG//test log fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount); #endif return -1; } /* Store the config in a WebRtcAgc_config_t */ stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB; stt->usedConfig.limiterEnable = agcConfig.limiterEnable; stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs; return 0; } int WebRtcAgc_get_config(void *agcInst, WebRtcAgc_config_t *config) { Agc_t *stt; stt = (Agc_t *)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; } int WebRtcAgc_Create(void **agcInst) { Agc_t *stt; if (agcInst == NULL) { return -1; } stt = (Agc_t *)malloc(sizeof(Agc_t)); *agcInst = stt; if (stt == NULL) { return -1; } #ifdef AGC_DEBUG 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 0; } int WebRtcAgc_Free(void *state) { Agc_t *stt; stt = (Agc_t *)state; #ifdef AGC_DEBUG fclose(stt->fpt); fclose(stt->agcLog); fclose(stt->digitalAgc.logFile); #endif free(stt); return 0; } /* 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; Agc_t *stt; /* typecast state pointer */ stt = (Agc_t *)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 AGC_DEBUG//test log stt->fcount = 0; fprintf(stt->fpt, "AGC->Init\n"); #endif if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital) { #ifdef AGC_DEBUG//test log 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 = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale); minLevel = WEBRTC_SPL_LSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(maxLevel - minLevel, 2); /* 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 AGC_DEBUG//test log 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 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)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 AGC_DEBUG//test log fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n"); #endif return -1; } else { #ifdef AGC_DEBUG//test log fprintf(stt->fpt, "\n"); #endif return 0; } }