| /* |
| * 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 <assert.h> |
| #include <stdlib.h> |
| #ifdef AGC_DEBUG //test log |
| #include <stdio.h> |
| #endif |
| #include "analog_agc.h" |
| |
| /* The slope of in Q13*/ |
| static const WebRtc_Word16 kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78}; |
| |
| /* The offset in Q14 */ |
| static const WebRtc_Word16 kOffset1[8] = {25395, 23911, 22206, 20737, 19612, 18805, 17951, |
| 17367}; |
| |
| /* The slope of in Q13*/ |
| static const WebRtc_Word16 kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337}; |
| |
| /* The offset in Q14 */ |
| static const WebRtc_Word16 kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670, |
| 17286}; |
| |
| static const WebRtc_Word16 kMuteGuardTimeMs = 8000; |
| static const WebRtc_Word16 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 WebRtc_UWord16 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 WebRtc_UWord16 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 WebRtc_UWord16 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 WebRtc_Word32 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, WebRtc_Word16 *in_mic, WebRtc_Word16 *in_mic_H, |
| WebRtc_Word16 samples) |
| { |
| WebRtc_Word32 nrg, max_nrg, sample, tmp32; |
| WebRtc_Word32 *ptr; |
| WebRtc_UWord16 targetGainIdx, gain; |
| WebRtc_Word16 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 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog); |
| tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16); |
| tmp16 = (WebRtc_Word16)(stt->maxLevel - stt->maxAnalog); |
| targetGainIdx = (WebRtc_UWord16)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] = (WebRtc_Word16)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] = (WebRtc_Word16)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 WebRtc_Word16 *in_far, WebRtc_Word16 samples) |
| { |
| WebRtc_Word32 errHandle = 0; |
| WebRtc_Word16 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, WebRtc_Word16 *in_near, WebRtc_Word16 *in_near_H, |
| WebRtc_Word16 samples, WebRtc_Word32 micLevelIn, |
| WebRtc_Word32 *micLevelOut) |
| { |
| WebRtc_Word32 tmpFlt, micLevelTmp, gainIdx; |
| WebRtc_UWord16 gain; |
| WebRtc_Word16 ii; |
| Agc_t *stt; |
| |
| WebRtc_UWord32 nrg; |
| WebRtc_Word16 sampleCntr; |
| WebRtc_UWord32 frameNrg = 0; |
| WebRtc_UWord32 frameNrgLimit = 5500; |
| WebRtc_Word16 numZeroCrossing = 0; |
| const WebRtc_Word16 kZeroCrossingLowLim = 15; |
| const WebRtc_Word16 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] = (WebRtc_Word16)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] = (WebRtc_Word16)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) |
| { |
| |
| WebRtc_Word16 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((WebRtc_Word32)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, WebRtc_UWord8 *saturated, WebRtc_Word32 *env) |
| { |
| WebRtc_Word16 i, tmpW16; |
| |
| /* Check if the signal is saturated */ |
| for (i = 0; i < 10; i++) |
| { |
| tmpW16 = (WebRtc_Word16)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 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum, |
| (WebRtc_Word16)32440, 15); |
| } |
| |
| void WebRtcAgc_ZeroCtrl(Agc_t *stt, WebRtc_Word32 *inMicLevel, WebRtc_Word32 *env) |
| { |
| WebRtc_Word16 i; |
| WebRtc_Word32 tmp32 = 0; |
| WebRtc_Word32 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. |
| */ |
| |
| WebRtc_Word32 tmp32; |
| WebRtc_Word16 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 = (WebRtc_Word32)vadThresh; |
| tmp32 += WEBRTC_SPL_MUL_16_16((WebRtc_Word16)31, stt->vadThreshold); |
| stt->vadThreshold = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 5); |
| } |
| } |
| |
| void WebRtcAgc_ExpCurve(WebRtc_Word16 volume, WebRtc_Word16 *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; |
| } |
| } |
| } |
| } |
| |
| WebRtc_Word32 WebRtcAgc_ProcessAnalog(void *state, WebRtc_Word32 inMicLevel, |
| WebRtc_Word32 *outMicLevel, |
| WebRtc_Word16 vadLogRatio, |
| WebRtc_Word16 echo, WebRtc_UWord8 *saturationWarning) |
| { |
| WebRtc_UWord32 tmpU32; |
| WebRtc_Word32 Rxx16w32, tmp32; |
| WebRtc_Word32 inMicLevelTmp, lastMicVol; |
| WebRtc_Word16 i; |
| WebRtc_UWord8 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) |
| { |
| WebRtc_Word32 tmpVol; |
| stt->firstCall = 1; |
| tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)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) * (WebRtc_Word32)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) |
| { |
| // Incoming level mismatch; update our level. |
| // This could be the case if the volume is changed manually, or if the |
| // sound device has a low volume resolution. |
| 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. |
| 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, (WebRtc_UWord32)(tmp32)); |
| stt->micVol = (WebRtc_Word32)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, (WebRtc_UWord32)(tmp32)); |
| stt->micVol = (WebRtc_Word32)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, (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel)); |
| stt->micVol = (WebRtc_Word32)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 */ |
| WebRtc_Word16 index, weightFIX; |
| WebRtc_Word16 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 = (WebRtc_Word16)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] |
| - (WebRtc_Word16)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 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel)); |
| stt->micVol = (WebRtc_Word32)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 */ |
| WebRtc_Word16 index, weightFIX; |
| WebRtc_Word16 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 = (WebRtc_Word16)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] |
| - (WebRtc_Word16)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 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel)); |
| stt->micVol = (WebRtc_Word32)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 WebRtc_Word16 *in_near, |
| const WebRtc_Word16 *in_near_H, WebRtc_Word16 samples, |
| WebRtc_Word16 *out, WebRtc_Word16 *out_H, WebRtc_Word32 inMicLevel, |
| WebRtc_Word32 *outMicLevel, WebRtc_Word16 echo, |
| WebRtc_UWord8 *saturationWarning) |
| { |
| Agc_t *stt; |
| WebRtc_Word32 inMicLevelTmp; |
| WebRtc_Word16 subFrames, i; |
| WebRtc_UWord8 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(WebRtc_Word16)); |
| } |
| if (stt->fs == 32000) |
| { |
| if (in_near_H != out_H) |
| { |
| memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16)); |
| } |
| } |
| |
| #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(WebRtc_Word32)); |
| memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(WebRtc_Word32)); |
| } |
| |
| 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, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel, |
| WebRtc_Word16 agcMode, WebRtc_UWord32 fs) |
| { |
| WebRtc_Word32 max_add, tmp32; |
| WebRtc_Word16 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((WebRtc_UWord32)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; |
| |
| /* 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) * (WebRtc_Word32)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] = (WebRtc_Word32)1000; /* -54dBm0 */ |
| } |
| stt->Rxx160w32 = 125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */ |
| |
| stt->Rxx16pos = 0; |
| stt->Rxx16_LPw32 = (WebRtc_Word32)16284; /* Q(-4) */ |
| |
| for (i = 0; i < 5; i++) |
| { |
| stt->Rxx16w32_array[0][i] = 0; |
| } |
| for (i = 0; i < 20; i++) |
| { |
| stt->env[0][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; |
| } |
| } |
