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794 lines
28 KiB
C
794 lines
28 KiB
C
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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/* digital_agc.c
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*
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*/
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#include "webrtc/modules/audio_processing/agc/digital_agc.h"
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#include <assert.h>
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#include <string.h>
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#ifdef AGC_DEBUG
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#include <stdio.h>
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#endif
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#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
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// To generate the gaintable, copy&paste the following lines to a Matlab window:
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// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
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// zeros = 0:31; lvl = 2.^(1-zeros);
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// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
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// B = MaxGain - MinGain;
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// 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))))));
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// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
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// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines):
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// in = 10*log10(lvl); out = 20*log10(gains/65536);
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// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
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// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
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// zoom on;
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// Generator table for y=log2(1+e^x) in Q8.
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enum { kGenFuncTableSize = 128 };
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static const uint16_t kGenFuncTable[kGenFuncTableSize] = {
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256, 485, 786, 1126, 1484, 1849, 2217, 2586,
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2955, 3324, 3693, 4063, 4432, 4801, 5171, 5540,
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5909, 6279, 6648, 7017, 7387, 7756, 8125, 8495,
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8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449,
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11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404,
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14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359,
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17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313,
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20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268,
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23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222,
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26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177,
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29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
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32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086,
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35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041,
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38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996,
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41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950,
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44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905
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};
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static const int16_t kAvgDecayTime = 250; // frames; < 3000
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int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
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int16_t digCompGaindB, // Q0
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int16_t targetLevelDbfs,// Q0
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uint8_t limiterEnable,
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int16_t analogTarget) // Q0
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{
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// This function generates the compressor gain table used in the fixed digital part.
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uint32_t tmpU32no1, tmpU32no2, absInLevel, logApprox;
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int32_t inLevel, limiterLvl;
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int32_t tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
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const uint16_t kLog10 = 54426; // log2(10) in Q14
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const uint16_t kLog10_2 = 49321; // 10*log10(2) in Q14
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const uint16_t kLogE_1 = 23637; // log2(e) in Q14
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uint16_t constMaxGain;
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uint16_t tmpU16, intPart, fracPart;
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const int16_t kCompRatio = 3;
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const int16_t kSoftLimiterLeft = 1;
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int16_t limiterOffset = 0; // Limiter offset
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int16_t limiterIdx, limiterLvlX;
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int16_t constLinApprox, zeroGainLvl, maxGain, diffGain;
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int16_t i, tmp16, tmp16no1;
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int zeros, zerosScale;
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// Constants
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// kLogE_1 = 23637; // log2(e) in Q14
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// kLog10 = 54426; // log2(10) in Q14
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// kLog10_2 = 49321; // 10*log10(2) in Q14
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// Calculate maximum digital gain and zero gain level
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tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB - analogTarget, kCompRatio - 1);
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tmp16no1 = analogTarget - targetLevelDbfs;
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tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
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maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
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tmp32no1 = WEBRTC_SPL_MUL_16_16(maxGain, kCompRatio);
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zeroGainLvl = digCompGaindB;
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zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1),
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kCompRatio - 1);
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if ((digCompGaindB <= analogTarget) && (limiterEnable))
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{
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zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft);
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limiterOffset = 0;
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}
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// Calculate the difference between maximum gain and gain at 0dB0v:
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// diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio
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// = (compRatio-1)*digCompGaindB/compRatio
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tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
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diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
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if (diffGain < 0 || diffGain >= kGenFuncTableSize)
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{
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assert(0);
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return -1;
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}
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// Calculate the limiter level and index:
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// limiterLvlX = analogTarget - limiterOffset
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// limiterLvl = targetLevelDbfs + limiterOffset/compRatio
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limiterLvlX = analogTarget - limiterOffset;
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limiterIdx = 2
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+ WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)limiterLvlX, 13),
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(kLog10_2 / 2));
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tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
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limiterLvl = targetLevelDbfs + tmp16no1;
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// Calculate (through table lookup):
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// constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
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constMaxGain = kGenFuncTable[diffGain]; // in Q8
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// Calculate a parameter used to approximate the fractional part of 2^x with a
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// piecewise linear function in Q14:
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// constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
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constLinApprox = 22817; // in Q14
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// Calculate a denominator used in the exponential part to convert from dB to linear scale:
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// den = 20*constMaxGain (in Q8)
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den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8
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for (i = 0; i < 32; i++)
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{
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// Calculate scaled input level (compressor):
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// inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
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tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
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tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
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inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
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// Calculate diffGain-inLevel, to map using the genFuncTable
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inLevel = WEBRTC_SPL_LSHIFT_W32((int32_t)diffGain, 14) - inLevel; // Q14
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// Make calculations on abs(inLevel) and compensate for the sign afterwards.
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absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14
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// LUT with interpolation
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intPart = (uint16_t)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
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fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part
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tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
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tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
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tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((uint32_t)kGenFuncTable[intPart], 14); // Q22
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logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
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// Compensate for negative exponent using the relation:
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// log2(1 + 2^-x) = log2(1 + 2^x) - x
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if (inLevel < 0)
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{
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zeros = WebRtcSpl_NormU32(absInLevel);
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zerosScale = 0;
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if (zeros < 15)
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{
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// Not enough space for multiplication
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tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
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tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
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if (zeros < 9)
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{
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tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
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zerosScale = 9 - zeros;
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} else
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{
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tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
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}
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} else
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{
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tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
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tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
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}
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logApprox = 0;
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if (tmpU32no2 < tmpU32no1)
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{
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logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
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}
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}
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numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
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numFIX -= (int32_t)logApprox * diffGain; // Q14
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// Calculate ratio
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// Shift |numFIX| as much as possible.
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// Ensure we avoid wrap-around in |den| as well.
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if (numFIX > (den >> 8)) // |den| is Q8.
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{
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zeros = WebRtcSpl_NormW32(numFIX);
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} else
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{
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zeros = WebRtcSpl_NormW32(den) + 8;
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}
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numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
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// Shift den so we end up in Qy1
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tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
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if (numFIX < 0)
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{
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numFIX -= WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
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} else
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{
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numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
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}
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y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
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if (limiterEnable && (i < limiterIdx))
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{
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tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
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tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
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y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
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}
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if (y32 > 39000)
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{
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tmp32 = WEBRTC_SPL_MUL(y32 >> 1, kLog10) + 4096; // in Q27
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tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 13); // in Q14
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} else
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{
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tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
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tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
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}
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tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
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// Calculate power
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if (tmp32 > 0)
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{
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intPart = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 14);
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fracPart = (uint16_t)(tmp32 & 0x00003FFF); // in Q14
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if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
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{
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tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
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tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
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tmp32no2 *= tmp16;
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tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
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tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
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} else
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{
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tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
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tmp32no2 = fracPart * tmp16;
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tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
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}
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fracPart = (uint16_t)tmp32no2;
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gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
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+ WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
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} else
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{
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gainTable[i] = 0;
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}
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}
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return 0;
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}
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int32_t WebRtcAgc_InitDigital(DigitalAgc_t *stt, int16_t agcMode)
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{
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if (agcMode == kAgcModeFixedDigital)
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{
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// start at minimum to find correct gain faster
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stt->capacitorSlow = 0;
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} else
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{
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// start out with 0 dB gain
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stt->capacitorSlow = 134217728; // (int32_t)(0.125f * 32768.0f * 32768.0f);
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}
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stt->capacitorFast = 0;
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stt->gain = 65536;
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stt->gatePrevious = 0;
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stt->agcMode = agcMode;
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#ifdef AGC_DEBUG
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stt->frameCounter = 0;
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#endif
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// initialize VADs
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WebRtcAgc_InitVad(&stt->vadNearend);
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WebRtcAgc_InitVad(&stt->vadFarend);
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return 0;
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}
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int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const int16_t *in_far,
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int16_t nrSamples)
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{
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assert(stt != NULL);
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// VAD for far end
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WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
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return 0;
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}
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int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
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const int16_t *in_near_H, int16_t *out,
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int16_t *out_H, uint32_t FS,
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int16_t lowlevelSignal)
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{
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// array for gains (one value per ms, incl start & end)
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int32_t gains[11];
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int32_t out_tmp, tmp32;
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int32_t env[10];
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int32_t nrg, max_nrg;
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int32_t cur_level;
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int32_t gain32, delta;
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int16_t logratio;
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int16_t lower_thr, upper_thr;
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int16_t zeros = 0, zeros_fast, frac = 0;
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int16_t decay;
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int16_t gate, gain_adj;
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int16_t k, n;
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int16_t L, L2; // samples/subframe
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// determine number of samples per ms
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if (FS == 8000)
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{
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L = 8;
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L2 = 3;
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} else if (FS == 16000)
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{
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L = 16;
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L2 = 4;
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} else if (FS == 32000)
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{
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L = 16;
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L2 = 4;
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} else
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{
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return -1;
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}
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// TODO(andrew): again, we don't need input and output pointers...
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if (in_near != out)
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{
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// Only needed if they don't already point to the same place.
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memcpy(out, in_near, 10 * L * sizeof(int16_t));
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}
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if (FS == 32000)
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{
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if (in_near_H != out_H)
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{
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memcpy(out_H, in_near_H, 10 * L * sizeof(int16_t));
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}
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}
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// VAD for near end
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logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
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// Account for far end VAD
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if (stt->vadFarend.counter > 10)
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{
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tmp32 = WEBRTC_SPL_MUL_16_16(3, logratio);
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logratio = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32 - stt->vadFarend.logRatio, 2);
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}
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// Determine decay factor depending on VAD
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// upper_thr = 1.0f;
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// lower_thr = 0.25f;
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upper_thr = 1024; // Q10
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lower_thr = 0; // Q10
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if (logratio > upper_thr)
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{
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// decay = -2^17 / DecayTime; -> -65
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decay = -65;
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} else if (logratio < lower_thr)
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{
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decay = 0;
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} else
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{
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// decay = (int16_t)(((lower_thr - logratio)
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// * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
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// SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65
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tmp32 = WEBRTC_SPL_MUL_16_16((lower_thr - logratio), 65);
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decay = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
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}
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// adjust decay factor for long silence (detected as low standard deviation)
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// This is only done in the adaptive modes
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if (stt->agcMode != kAgcModeFixedDigital)
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{
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if (stt->vadNearend.stdLongTerm < 4000)
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{
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decay = 0;
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} else if (stt->vadNearend.stdLongTerm < 8096)
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{
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// decay = (int16_t)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12);
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tmp32 = WEBRTC_SPL_MUL_16_16((stt->vadNearend.stdLongTerm - 4000), decay);
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decay = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
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}
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if (lowlevelSignal != 0)
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{
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decay = 0;
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}
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}
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#ifdef AGC_DEBUG
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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++)
|
|
{
|
|
nrg = WEBRTC_SPL_MUL_16_16(out[k * L + n], out[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 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
|
|
frac = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
|
|
tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
|
|
gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
|
|
#ifdef AGC_DEBUG
|
|
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 = WEBRTC_SPL_LSHIFT_W16(zeros, 9) - WEBRTC_SPL_RSHIFT_W16(frac, 3);
|
|
// find number of leading zeros
|
|
zeros_fast = WebRtcSpl_NormU32((uint32_t)stt->capacitorFast);
|
|
if (stt->capacitorFast == 0)
|
|
{
|
|
zeros_fast = 31;
|
|
}
|
|
tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
|
|
zeros_fast = WEBRTC_SPL_LSHIFT_W16(zeros_fast, 9);
|
|
zeros_fast -= (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
|
|
|
|
gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
|
|
|
|
if (gate < 0)
|
|
{
|
|
stt->gatePrevious = 0;
|
|
} else
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL_16_16(stt->gatePrevious, 7);
|
|
gate = (int16_t)WEBRTC_SPL_RSHIFT_W32((int32_t)gate + tmp32, 3);
|
|
stt->gatePrevious = gate;
|
|
}
|
|
// gate < 0 -> no gate
|
|
// gate > 2500 -> max gate
|
|
if (gate > 0)
|
|
{
|
|
if (gate < 2500)
|
|
{
|
|
gain_adj = WEBRTC_SPL_RSHIFT_W16(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 = WEBRTC_SPL_RSHIFT_W32((gains[k+1] - stt->gainTable[0]), 8);
|
|
tmp32 = WEBRTC_SPL_MUL(tmp32, (178 + gain_adj));
|
|
} else
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL((gains[k+1] - stt->gainTable[0]), (178 + gain_adj));
|
|
tmp32 = WEBRTC_SPL_RSHIFT_W32(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 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
|
|
gain32 = WEBRTC_SPL_MUL(gain32, gain32);
|
|
// check for overflow
|
|
while (AGC_MUL32(WEBRTC_SPL_RSHIFT_W32(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] = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(gains[k+1], 8), 253);
|
|
} else
|
|
{
|
|
gains[k + 1] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(gains[k+1], 253), 8);
|
|
}
|
|
gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
|
|
gain32 = WEBRTC_SPL_MUL(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 = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
|
|
gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
|
|
// iterate over samples
|
|
for (n = 0; n < L; n++)
|
|
{
|
|
// For lower band
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out[n], WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
|
|
out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
if (out_tmp > 4095)
|
|
{
|
|
out[n] = (int16_t)32767;
|
|
} else if (out_tmp < -4096)
|
|
{
|
|
out[n] = (int16_t)-32768;
|
|
} else
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out[n], WEBRTC_SPL_RSHIFT_W32(gain32, 4));
|
|
out[n] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
}
|
|
// For higher band
|
|
if (FS == 32000)
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out_H[n],
|
|
WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
|
|
out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
if (out_tmp > 4095)
|
|
{
|
|
out_H[n] = (int16_t)32767;
|
|
} else if (out_tmp < -4096)
|
|
{
|
|
out_H[n] = (int16_t)-32768;
|
|
} else
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out_H[n],
|
|
WEBRTC_SPL_RSHIFT_W32(gain32, 4));
|
|
out_H[n] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
}
|
|
}
|
|
//
|
|
|
|
gain32 += delta;
|
|
}
|
|
// iterate over subframes
|
|
for (k = 1; k < 10; k++)
|
|
{
|
|
delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
|
|
gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
|
|
// iterate over samples
|
|
for (n = 0; n < L; n++)
|
|
{
|
|
// For lower band
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out[k * L + n],
|
|
WEBRTC_SPL_RSHIFT_W32(gain32, 4));
|
|
out[k * L + n] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
// For higher band
|
|
if (FS == 32000)
|
|
{
|
|
tmp32 = WEBRTC_SPL_MUL((int32_t)out_H[k * L + n],
|
|
WEBRTC_SPL_RSHIFT_W32(gain32, 4));
|
|
out_H[k * L + n] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
|
|
}
|
|
gain32 += delta;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void WebRtcAgc_InitVad(AgcVad_t *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 = WEBRTC_SPL_LSHIFT_W16(15, 10);
|
|
|
|
// variance of input level (Q8)
|
|
state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
|
|
|
|
state->stdLongTerm = 0; // standard deviation of input level in dB
|
|
// short-term average input level (Q10)
|
|
state->meanShortTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
|
|
|
|
// short-term variance of input level (Q8)
|
|
state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(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_t *state, // (i) VAD state
|
|
const int16_t *in, // (i) Speech signal
|
|
int16_t nrSamples) // (i) number of samples
|
|
{
|
|
int32_t out, nrg, 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 = WEBRTC_SPL_RSHIFT_W32(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 = WEBRTC_SPL_MUL(600, out);
|
|
HPstate = (int16_t)(WEBRTC_SPL_RSHIFT_W32(tmp32, 10) - buf2[k]);
|
|
tmp32 = WEBRTC_SPL_MUL(out, out);
|
|
nrg += WEBRTC_SPL_RSHIFT_W32(tmp32, 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 = WEBRTC_SPL_LSHIFT_W16(15 - zeros, 11);
|
|
|
|
// Update statistics
|
|
|
|
if (state->counter < kAvgDecayTime)
|
|
{
|
|
// decay time = AvgDecTime * 10 ms
|
|
state->counter++;
|
|
}
|
|
|
|
// update short-term estimate of mean energy level (Q10)
|
|
tmp32 = (WEBRTC_SPL_MUL_16_16(state->meanShortTerm, 15) + (int32_t)dB);
|
|
state->meanShortTerm = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
|
|
|
|
// update short-term estimate of variance in energy level (Q8)
|
|
tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
|
|
tmp32 += WEBRTC_SPL_MUL(state->varianceShortTerm, 15);
|
|
state->varianceShortTerm = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
|
|
|
|
// update short-term estimate of standard deviation in energy level (Q10)
|
|
tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
|
|
tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
|
|
state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
|
|
|
|
// update long-term estimate of mean energy level (Q10)
|
|
tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->counter) + (int32_t)dB;
|
|
state->meanLongTerm = WebRtcSpl_DivW32W16ResW16(
|
|
tmp32, WebRtcSpl_AddSatW16(state->counter, 1));
|
|
|
|
// update long-term estimate of variance in energy level (Q8)
|
|
tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
|
|
tmp32 += WEBRTC_SPL_MUL(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 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
|
|
tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
|
|
state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
|
|
|
|
// update voice activity measure (Q10)
|
|
tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
|
|
tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm));
|
|
tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
|
|
tmpU16 = (13 << 12);
|
|
tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
|
|
tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10);
|
|
|
|
state->logRatio = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
|
|
|
|
// limit
|
|
if (state->logRatio > 2048)
|
|
{
|
|
state->logRatio = 2048;
|
|
}
|
|
if (state->logRatio < -2048)
|
|
{
|
|
state->logRatio = -2048;
|
|
}
|
|
|
|
return state->logRatio; // Q10
|
|
}
|