| /* |
| * Copyright (c) 2011 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. |
| */ |
| |
| /* digital_agc.c |
| * |
| */ |
| |
| #include "modules/audio_processing/agc/legacy/digital_agc.h" |
| |
| #include <string.h> |
| #ifdef WEBRTC_AGC_DEBUG_DUMP |
| #include <stdio.h> |
| #endif |
| |
| #include "rtc_base/checks.h" |
| #include "modules/audio_processing/agc/legacy/gain_control.h" |
| |
| // To generate the gaintable, copy&paste the following lines to a Matlab window: |
| // MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1; |
| // zeros = 0:31; lvl = 2.^(1-zeros); |
| // A = -10*log10(lvl) * (CompRatio - 1) / CompRatio; |
| // B = MaxGain - MinGain; |
| // gains = round(2^16*10.^(0.05 * (MinGain + B * ( |
| // log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / |
| // log(1/(1+exp(Knee*B)))))); |
| // fprintf(1, '\t%i, %i, %i, %i,\n', gains); |
| // % Matlab code for plotting the gain and input/output level characteristic |
| // (copy/paste the following 3 lines): |
| // in = 10*log10(lvl); out = 20*log10(gains/65536); |
| // subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input |
| // (dB)'); ylabel('Gain (dB)'); |
| // subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; |
| // xlabel('Input (dB)'); ylabel('Output (dB)'); |
| // zoom on; |
| |
| // Generator table for y=log2(1+e^x) in Q8. |
| enum { kGenFuncTableSize = 128 }; |
| static const uint16_t kGenFuncTable[kGenFuncTableSize] = { |
| 256, 485, 786, 1126, 1484, 1849, 2217, 2586, 2955, 3324, 3693, |
| 4063, 4432, 4801, 5171, 5540, 5909, 6279, 6648, 7017, 7387, 7756, |
| 8125, 8495, 8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449, 11819, |
| 12188, 12557, 12927, 13296, 13665, 14035, 14404, 14773, 15143, 15512, 15881, |
| 16251, 16620, 16989, 17359, 17728, 18097, 18466, 18836, 19205, 19574, 19944, |
| 20313, 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268, 23637, 24006, |
| 24376, 24745, 25114, 25484, 25853, 26222, 26592, 26961, 27330, 27700, 28069, |
| 28438, 28808, 29177, 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132, |
| 32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086, 35456, 35825, 36194, |
| 36564, 36933, 37302, 37672, 38041, 38410, 38780, 39149, 39518, 39888, 40257, |
| 40626, 40996, 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950, 44320, |
| 44689, 45058, 45428, 45797, 46166, 46536, 46905}; |
| |
| static const int16_t kAvgDecayTime = 250; // frames; < 3000 |
| |
| int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable, // Q16 |
| int16_t digCompGaindB, // Q0 |
| int16_t targetLevelDbfs, // Q0 |
| uint8_t limiterEnable, |
| int16_t analogTarget) // Q0 |
| { |
| // This function generates the compressor gain table used in the fixed digital |
| // part. |
| uint32_t tmpU32no1, tmpU32no2, absInLevel, logApprox; |
| int32_t inLevel, limiterLvl; |
| int32_t tmp32, tmp32no1, tmp32no2, numFIX, den, y32; |
| const uint16_t kLog10 = 54426; // log2(10) in Q14 |
| const uint16_t kLog10_2 = 49321; // 10*log10(2) in Q14 |
| const uint16_t kLogE_1 = 23637; // log2(e) in Q14 |
| uint16_t constMaxGain; |
| uint16_t tmpU16, intPart, fracPart; |
| const int16_t kCompRatio = 3; |
| const int16_t kSoftLimiterLeft = 1; |
| int16_t limiterOffset = 0; // Limiter offset |
| int16_t limiterIdx, limiterLvlX; |
| int16_t constLinApprox, zeroGainLvl, maxGain, diffGain; |
| int16_t i, tmp16, tmp16no1; |
| int zeros, zerosScale; |
| |
| // Constants |
| // kLogE_1 = 23637; // log2(e) in Q14 |
| // kLog10 = 54426; // log2(10) in Q14 |
| // kLog10_2 = 49321; // 10*log10(2) in Q14 |
| |
| // Calculate maximum digital gain and zero gain level |
| tmp32no1 = (digCompGaindB - analogTarget) * (kCompRatio - 1); |
| tmp16no1 = analogTarget - targetLevelDbfs; |
| tmp16no1 += |
| WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
| maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs)); |
| tmp32no1 = maxGain * kCompRatio; |
| zeroGainLvl = digCompGaindB; |
| zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1), |
| kCompRatio - 1); |
| if ((digCompGaindB <= analogTarget) && (limiterEnable)) { |
| zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft); |
| limiterOffset = 0; |
| } |
| |
| // Calculate the difference between maximum gain and gain at 0dB0v: |
| // diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio |
| // = (compRatio-1)*digCompGaindB/compRatio |
| tmp32no1 = digCompGaindB * (kCompRatio - 1); |
| diffGain = |
| WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
| if (diffGain < 0 || diffGain >= kGenFuncTableSize) { |
| RTC_DCHECK(0); |
| return -1; |
| } |
| |
| // Calculate the limiter level and index: |
| // limiterLvlX = analogTarget - limiterOffset |
| // limiterLvl = targetLevelDbfs + limiterOffset/compRatio |
| limiterLvlX = analogTarget - limiterOffset; |
| limiterIdx = 2 + WebRtcSpl_DivW32W16ResW16((int32_t)limiterLvlX * (1 << 13), |
| kLog10_2 / 2); |
| tmp16no1 = |
| WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio); |
| limiterLvl = targetLevelDbfs + tmp16no1; |
| |
| // Calculate (through table lookup): |
| // constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8) |
| constMaxGain = kGenFuncTable[diffGain]; // in Q8 |
| |
| // Calculate a parameter used to approximate the fractional part of 2^x with a |
| // piecewise linear function in Q14: |
| // constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14); |
| constLinApprox = 22817; // in Q14 |
| |
| // Calculate a denominator used in the exponential part to convert from dB to |
| // linear scale: |
| // den = 20*constMaxGain (in Q8) |
| den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8 |
| |
| for (i = 0; i < 32; i++) { |
| // Calculate scaled input level (compressor): |
| // inLevel = |
| // fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio) |
| tmp16 = (int16_t)((kCompRatio - 1) * (i - 1)); // Q0 |
| tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14 |
| inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14 |
| |
| // Calculate diffGain-inLevel, to map using the genFuncTable |
| inLevel = (int32_t)diffGain * (1 << 14) - inLevel; // Q14 |
| |
| // Make calculations on abs(inLevel) and compensate for the sign afterwards. |
| absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14 |
| |
| // LUT with interpolation |
| intPart = (uint16_t)(absInLevel >> 14); |
| fracPart = |
| (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part |
| tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8 |
| tmpU32no1 = tmpU16 * fracPart; // Q22 |
| tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14; // Q22 |
| logApprox = tmpU32no1 >> 8; // Q14 |
| // Compensate for negative exponent using the relation: |
| // log2(1 + 2^-x) = log2(1 + 2^x) - x |
| if (inLevel < 0) { |
| zeros = WebRtcSpl_NormU32(absInLevel); |
| zerosScale = 0; |
| if (zeros < 15) { |
| // Not enough space for multiplication |
| tmpU32no2 = absInLevel >> (15 - zeros); // Q(zeros-1) |
| tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13) |
| if (zeros < 9) { |
| zerosScale = 9 - zeros; |
| tmpU32no1 >>= zerosScale; // Q(zeros+13) |
| } else { |
| tmpU32no2 >>= zeros - 9; // Q22 |
| } |
| } else { |
| tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28 |
| tmpU32no2 >>= 6; // Q22 |
| } |
| logApprox = 0; |
| if (tmpU32no2 < tmpU32no1) { |
| logApprox = (tmpU32no1 - tmpU32no2) >> (8 - zerosScale); // Q14 |
| } |
| } |
| numFIX = (maxGain * constMaxGain) * (1 << 6); // Q14 |
| numFIX -= (int32_t)logApprox * diffGain; // Q14 |
| |
| // Calculate ratio |
| // Shift |numFIX| as much as possible. |
| // Ensure we avoid wrap-around in |den| as well. |
| if (numFIX > (den >> 8) || -numFIX > (den >> 8)) // |den| is Q8. |
| { |
| zeros = WebRtcSpl_NormW32(numFIX); |
| } else { |
| zeros = WebRtcSpl_NormW32(den) + 8; |
| } |
| numFIX *= 1 << zeros; // Q(14+zeros) |
| |
| // Shift den so we end up in Qy1 |
| tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 9); // Q(zeros - 1) |
| y32 = numFIX / tmp32no1; // in Q15 |
| // This is to do rounding in Q14. |
| y32 = y32 >= 0 ? (y32 + 1) >> 1 : -((-y32 + 1) >> 1); |
| |
| if (limiterEnable && (i < limiterIdx)) { |
| tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14 |
| tmp32 -= limiterLvl * (1 << 14); // Q14 |
| y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20); |
| } |
| if (y32 > 39000) { |
| tmp32 = (y32 >> 1) * kLog10 + 4096; // in Q27 |
| tmp32 >>= 13; // In Q14. |
| } else { |
| tmp32 = y32 * kLog10 + 8192; // in Q28 |
| tmp32 >>= 14; // In Q14. |
| } |
| tmp32 += 16 << 14; // in Q14 (Make sure final output is in Q16) |
| |
| // Calculate power |
| if (tmp32 > 0) { |
| intPart = (int16_t)(tmp32 >> 14); |
| fracPart = (uint16_t)(tmp32 & 0x00003FFF); // in Q14 |
| if ((fracPart >> 13) != 0) { |
| tmp16 = (2 << 14) - constLinApprox; |
| tmp32no2 = (1 << 14) - fracPart; |
| tmp32no2 *= tmp16; |
| tmp32no2 >>= 13; |
| tmp32no2 = (1 << 14) - tmp32no2; |
| } else { |
| tmp16 = constLinApprox - (1 << 14); |
| tmp32no2 = (fracPart * tmp16) >> 13; |
| } |
| fracPart = (uint16_t)tmp32no2; |
| gainTable[i] = |
| (1 << intPart) + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14); |
| } else { |
| gainTable[i] = 0; |
| } |
| } |
| |
| return 0; |
| } |
| |
| int32_t WebRtcAgc_InitDigital(DigitalAgc* stt, int16_t agcMode) { |
| if (agcMode == kAgcModeFixedDigital) { |
| // start at minimum to find correct gain faster |
| stt->capacitorSlow = 0; |
| } else { |
| // start out with 0 dB gain |
| stt->capacitorSlow = 134217728; // (int32_t)(0.125f * 32768.0f * 32768.0f); |
| } |
| stt->capacitorFast = 0; |
| stt->gain = 65536; |
| stt->gatePrevious = 0; |
| stt->agcMode = agcMode; |
| #ifdef WEBRTC_AGC_DEBUG_DUMP |
| stt->frameCounter = 0; |
| #endif |
| |
| // initialize VADs |
| WebRtcAgc_InitVad(&stt->vadNearend); |
| WebRtcAgc_InitVad(&stt->vadFarend); |
| |
| return 0; |
| } |
| |
| int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt, |
| const int16_t* in_far, |
| size_t nrSamples) { |
| RTC_DCHECK(stt); |
| // VAD for far end |
| WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples); |
| |
| return 0; |
| } |
| |
| int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt, |
| const int16_t* const* in_near, |
| size_t num_bands, |
| int16_t* const* out, |
| uint32_t FS, |
| int16_t lowlevelSignal) { |
| // array for gains (one value per ms, incl start & end) |
| int32_t gains[11]; |
| |
| int32_t out_tmp, tmp32; |
| int32_t env[10]; |
| int32_t max_nrg; |
| int32_t cur_level; |
| int32_t gain32, delta; |
| int16_t logratio; |
| int16_t lower_thr, upper_thr; |
| int16_t zeros = 0, zeros_fast, frac = 0; |
| int16_t decay; |
| int16_t gate, gain_adj; |
| int16_t k; |
| size_t n, i, L; |
| int16_t L2; // samples/subframe |
| |
| // determine number of samples per ms |
| if (FS == 8000) { |
| L = 8; |
| L2 = 3; |
| } else if (FS == 16000 || FS == 32000 || FS == 48000) { |
| L = 16; |
| L2 = 4; |
| } else { |
| return -1; |
| } |
| |
| for (i = 0; i < num_bands; ++i) { |
| if (in_near[i] != out[i]) { |
| // Only needed if they don't already point to the same place. |
| memcpy(out[i], in_near[i], 10 * L * sizeof(in_near[i][0])); |
| } |
| } |
| // VAD for near end |
| logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out[0], L * 10); |
| |
| // Account for far end VAD |
| if (stt->vadFarend.counter > 10) { |
| tmp32 = 3 * logratio; |
| logratio = (int16_t)((tmp32 - stt->vadFarend.logRatio) >> 2); |
| } |
| |
| // Determine decay factor depending on VAD |
| // upper_thr = 1.0f; |
| // lower_thr = 0.25f; |
| upper_thr = 1024; // Q10 |
| lower_thr = 0; // Q10 |
| if (logratio > upper_thr) { |
| // decay = -2^17 / DecayTime; -> -65 |
| decay = -65; |
| } else if (logratio < lower_thr) { |
| decay = 0; |
| } else { |
| // decay = (int16_t)(((lower_thr - logratio) |
| // * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10); |
| // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65 |
| tmp32 = (lower_thr - logratio) * 65; |
| decay = (int16_t)(tmp32 >> 10); |
| } |
| |
| // adjust decay factor for long silence (detected as low standard deviation) |
| // This is only done in the adaptive modes |
| if (stt->agcMode != kAgcModeFixedDigital) { |
| if (stt->vadNearend.stdLongTerm < 4000) { |
| decay = 0; |
| } else if (stt->vadNearend.stdLongTerm < 8096) { |
| // decay = (int16_t)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> |
| // 12); |
| tmp32 = (stt->vadNearend.stdLongTerm - 4000) * decay; |
| decay = (int16_t)(tmp32 >> 12); |
| } |
| |
| if (lowlevelSignal != 0) { |
| decay = 0; |
| } |
| } |
| #ifdef WEBRTC_AGC_DEBUG_DUMP |
| stt->frameCounter++; |
| fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, |
| logratio, decay, stt->vadNearend.stdLongTerm); |
| #endif |
| // Find max amplitude per sub frame |
| // iterate over sub frames |
| for (k = 0; k < 10; k++) { |
| // iterate over samples |
| max_nrg = 0; |
| for (n = 0; n < L; n++) { |
| int32_t nrg = out[0][k * L + n] * out[0][k * L + n]; |
| if (nrg > max_nrg) { |
| max_nrg = nrg; |
| } |
| } |
| env[k] = max_nrg; |
| } |
| |
| // Calculate gain per sub frame |
| gains[0] = stt->gain; |
| for (k = 0; k < 10; k++) { |
| // Fast envelope follower |
| // decay time = -131000 / -1000 = 131 (ms) |
| stt->capacitorFast = |
| AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast); |
| if (env[k] > stt->capacitorFast) { |
| stt->capacitorFast = env[k]; |
| } |
| // Slow envelope follower |
| if (env[k] > stt->capacitorSlow) { |
| // increase capacitorSlow |
| stt->capacitorSlow = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), |
| stt->capacitorSlow); |
| } else { |
| // decrease capacitorSlow |
| stt->capacitorSlow = |
| AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow); |
| } |
| |
| // use maximum of both capacitors as current level |
| if (stt->capacitorFast > stt->capacitorSlow) { |
| cur_level = stt->capacitorFast; |
| } else { |
| cur_level = stt->capacitorSlow; |
| } |
| // Translate signal level into gain, using a piecewise linear approximation |
| // find number of leading zeros |
| zeros = WebRtcSpl_NormU32((uint32_t)cur_level); |
| if (cur_level == 0) { |
| zeros = 31; |
| } |
| tmp32 = ((uint32_t)cur_level << zeros) & 0x7FFFFFFF; |
| frac = (int16_t)(tmp32 >> 19); // Q12. |
| tmp32 = (stt->gainTable[zeros - 1] - stt->gainTable[zeros]) * frac; |
| gains[k + 1] = stt->gainTable[zeros] + (tmp32 >> 12); |
| #ifdef WEBRTC_AGC_DEBUG_DUMP |
| if (k == 0) { |
| fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, |
| stt->capacitorFast, stt->capacitorSlow, zeros); |
| } |
| #endif |
| } |
| |
| // Gate processing (lower gain during absence of speech) |
| zeros = (zeros << 9) - (frac >> 3); |
| // find number of leading zeros |
| zeros_fast = WebRtcSpl_NormU32((uint32_t)stt->capacitorFast); |
| if (stt->capacitorFast == 0) { |
| zeros_fast = 31; |
| } |
| tmp32 = ((uint32_t)stt->capacitorFast << zeros_fast) & 0x7FFFFFFF; |
| zeros_fast <<= 9; |
| zeros_fast -= (int16_t)(tmp32 >> 22); |
| |
| gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm; |
| |
| if (gate < 0) { |
| stt->gatePrevious = 0; |
| } else { |
| tmp32 = stt->gatePrevious * 7; |
| gate = (int16_t)((gate + tmp32) >> 3); |
| stt->gatePrevious = gate; |
| } |
| // gate < 0 -> no gate |
| // gate > 2500 -> max gate |
| if (gate > 0) { |
| if (gate < 2500) { |
| gain_adj = (2500 - gate) >> 5; |
| } else { |
| gain_adj = 0; |
| } |
| for (k = 0; k < 10; k++) { |
| if ((gains[k + 1] - stt->gainTable[0]) > 8388608) { |
| // To prevent wraparound |
| tmp32 = (gains[k + 1] - stt->gainTable[0]) >> 8; |
| tmp32 *= 178 + gain_adj; |
| } else { |
| tmp32 = (gains[k + 1] - stt->gainTable[0]) * (178 + gain_adj); |
| tmp32 >>= 8; |
| } |
| gains[k + 1] = stt->gainTable[0] + tmp32; |
| } |
| } |
| |
| // Limit gain to avoid overload distortion |
| for (k = 0; k < 10; k++) { |
| // To prevent wrap around |
| zeros = 10; |
| if (gains[k + 1] > 47453132) { |
| zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]); |
| } |
| gain32 = (gains[k + 1] >> zeros) + 1; |
| gain32 *= gain32; |
| // check for overflow |
| while (AGC_MUL32((env[k] >> 12) + 1, gain32) > |
| WEBRTC_SPL_SHIFT_W32((int32_t)32767, 2 * (1 - zeros + 10))) { |
| // multiply by 253/256 ==> -0.1 dB |
| if (gains[k + 1] > 8388607) { |
| // Prevent wrap around |
| gains[k + 1] = (gains[k + 1] / 256) * 253; |
| } else { |
| gains[k + 1] = (gains[k + 1] * 253) / 256; |
| } |
| gain32 = (gains[k + 1] >> zeros) + 1; |
| gain32 *= gain32; |
| } |
| } |
| // gain reductions should be done 1 ms earlier than gain increases |
| for (k = 1; k < 10; k++) { |
| if (gains[k] > gains[k + 1]) { |
| gains[k] = gains[k + 1]; |
| } |
| } |
| // save start gain for next frame |
| stt->gain = gains[10]; |
| |
| // Apply gain |
| // handle first sub frame separately |
| delta = (gains[1] - gains[0]) * (1 << (4 - L2)); |
| gain32 = gains[0] * (1 << 4); |
| // iterate over samples |
| for (n = 0; n < L; n++) { |
| for (i = 0; i < num_bands; ++i) { |
| tmp32 = out[i][n] * ((gain32 + 127) >> 7); |
| out_tmp = tmp32 >> 16; |
| if (out_tmp > 4095) { |
| out[i][n] = (int16_t)32767; |
| } else if (out_tmp < -4096) { |
| out[i][n] = (int16_t)-32768; |
| } else { |
| tmp32 = out[i][n] * (gain32 >> 4); |
| out[i][n] = (int16_t)(tmp32 >> 16); |
| } |
| } |
| // |
| |
| gain32 += delta; |
| } |
| // iterate over subframes |
| for (k = 1; k < 10; k++) { |
| delta = (gains[k + 1] - gains[k]) * (1 << (4 - L2)); |
| gain32 = gains[k] * (1 << 4); |
| // iterate over samples |
| for (n = 0; n < L; n++) { |
| for (i = 0; i < num_bands; ++i) { |
| int64_t tmp64 = ((int64_t)(out[i][k * L + n])) * (gain32 >> 4); |
| tmp64 = tmp64 >> 16; |
| if (tmp64 > 32767) { |
| out[i][k * L + n] = 32767; |
| } |
| else if (tmp64 < -32768) { |
| out[i][k * L + n] = -32768; |
| } |
| else { |
| out[i][k * L + n] = (int16_t)(tmp64); |
| } |
| } |
| gain32 += delta; |
| } |
| } |
| |
| return 0; |
| } |
| |
| void WebRtcAgc_InitVad(AgcVad* state) { |
| int16_t k; |
| |
| state->HPstate = 0; // state of high pass filter |
| state->logRatio = 0; // log( P(active) / P(inactive) ) |
| // average input level (Q10) |
| state->meanLongTerm = 15 << 10; |
| |
| // variance of input level (Q8) |
| state->varianceLongTerm = 500 << 8; |
| |
| state->stdLongTerm = 0; // standard deviation of input level in dB |
| // short-term average input level (Q10) |
| state->meanShortTerm = 15 << 10; |
| |
| // short-term variance of input level (Q8) |
| state->varianceShortTerm = 500 << 8; |
| |
| state->stdShortTerm = |
| 0; // short-term standard deviation of input level in dB |
| state->counter = 3; // counts updates |
| for (k = 0; k < 8; k++) { |
| // downsampling filter |
| state->downState[k] = 0; |
| } |
| } |
| |
| int16_t WebRtcAgc_ProcessVad(AgcVad* state, // (i) VAD state |
| const int16_t* in, // (i) Speech signal |
| size_t nrSamples) // (i) number of samples |
| { |
| uint32_t nrg; |
| int32_t out, tmp32, tmp32b; |
| uint16_t tmpU16; |
| int16_t k, subfr, tmp16; |
| int16_t buf1[8]; |
| int16_t buf2[4]; |
| int16_t HPstate; |
| int16_t zeros, dB; |
| |
| // process in 10 sub frames of 1 ms (to save on memory) |
| nrg = 0; |
| HPstate = state->HPstate; |
| for (subfr = 0; subfr < 10; subfr++) { |
| // downsample to 4 kHz |
| if (nrSamples == 160) { |
| for (k = 0; k < 8; k++) { |
| tmp32 = (int32_t)in[2 * k] + (int32_t)in[2 * k + 1]; |
| tmp32 >>= 1; |
| buf1[k] = (int16_t)tmp32; |
| } |
| in += 16; |
| |
| WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState); |
| } else { |
| WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState); |
| in += 8; |
| } |
| |
| // high pass filter and compute energy |
| for (k = 0; k < 4; k++) { |
| out = buf2[k] + HPstate; |
| tmp32 = 600 * out; |
| HPstate = (int16_t)((tmp32 >> 10) - buf2[k]); |
| |
| // Add 'out * out / 2**6' to 'nrg' in a non-overflowing |
| // way. Guaranteed to work as long as 'out * out / 2**6' fits in |
| // an int32_t. |
| nrg += out * (out / (1 << 6)); |
| nrg += out * (out % (1 << 6)) / (1 << 6); |
| } |
| } |
| state->HPstate = HPstate; |
| |
| // find number of leading zeros |
| if (!(0xFFFF0000 & nrg)) { |
| zeros = 16; |
| } else { |
| zeros = 0; |
| } |
| if (!(0xFF000000 & (nrg << zeros))) { |
| zeros += 8; |
| } |
| if (!(0xF0000000 & (nrg << zeros))) { |
| zeros += 4; |
| } |
| if (!(0xC0000000 & (nrg << zeros))) { |
| zeros += 2; |
| } |
| if (!(0x80000000 & (nrg << zeros))) { |
| zeros += 1; |
| } |
| |
| // energy level (range {-32..30}) (Q10) |
| dB = (15 - zeros) * (1 << 11); |
| |
| // Update statistics |
| |
| if (state->counter < kAvgDecayTime) { |
| // decay time = AvgDecTime * 10 ms |
| state->counter++; |
| } |
| |
| // update short-term estimate of mean energy level (Q10) |
| tmp32 = state->meanShortTerm * 15 + dB; |
| state->meanShortTerm = (int16_t)(tmp32 >> 4); |
| |
| // update short-term estimate of variance in energy level (Q8) |
| tmp32 = (dB * dB) >> 12; |
| tmp32 += state->varianceShortTerm * 15; |
| state->varianceShortTerm = tmp32 / 16; |
| |
| // update short-term estimate of standard deviation in energy level (Q10) |
| tmp32 = state->meanShortTerm * state->meanShortTerm; |
| tmp32 = (state->varianceShortTerm << 12) - tmp32; |
| state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
| |
| // update long-term estimate of mean energy level (Q10) |
| tmp32 = state->meanLongTerm * state->counter + dB; |
| state->meanLongTerm = |
| WebRtcSpl_DivW32W16ResW16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
| |
| // update long-term estimate of variance in energy level (Q8) |
| tmp32 = (dB * dB) >> 12; |
| tmp32 += state->varianceLongTerm * state->counter; |
| state->varianceLongTerm = |
| WebRtcSpl_DivW32W16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
| |
| // update long-term estimate of standard deviation in energy level (Q10) |
| tmp32 = state->meanLongTerm * state->meanLongTerm; |
| tmp32 = (state->varianceLongTerm << 12) - tmp32; |
| state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
| |
| // update voice activity measure (Q10) |
| tmp16 = 3 << 12; |
| // TODO(bjornv): (dB - state->meanLongTerm) can overflow, e.g., in |
| // ApmTest.Process unit test. Previously the macro WEBRTC_SPL_MUL_16_16() |
| // was used, which did an intermediate cast to (int16_t), hence losing |
| // significant bits. This cause logRatio to max out positive, rather than |
| // negative. This is a bug, but has very little significance. |
| tmp32 = tmp16 * (int16_t)(dB - state->meanLongTerm); |
| tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm); |
| tmpU16 = (13 << 12); |
| tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16); |
| tmp32 += tmp32b >> 10; |
| |
| state->logRatio = (int16_t)(tmp32 >> 6); |
| |
| // limit |
| if (state->logRatio > 2048) { |
| state->logRatio = 2048; |
| } |
| if (state->logRatio < -2048) { |
| state->logRatio = -2048; |
| } |
| |
| return state->logRatio; // Q10 |
| } |
