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/*
* 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.
*/
#include "modules/audio_processing/agc/legacy/digital_agc.h"
#include <string.h>
#include "modules/audio_processing/agc/legacy/gain_control.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
// 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
// the 32 most significant bits of A(19) * B(26) >> 13
#define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13))
// C + the 32 most significant bits of A * B
#define AGC_SCALEDIFF32(A, B, C) \
((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16))
} // namespace
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;
int16_t limiterOffset = 0; // Limiter offset
int16_t limiterIdx, limiterLvlX;
int16_t constLinApprox, 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;
if ((digCompGaindB <= analogTarget) && (limiterEnable)) {
limiterOffset = 0;
}
// Calculate the difference between maximum gain and gain at 0dB0v
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;
// 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;
}
// Gains is an 11 element long array (one value per ms, incl start & end).
int32_t WebRtcAgc_ComputeDigitalGains(DigitalAgc* stt,
const int16_t* const* in_near,
size_t num_bands,
uint32_t FS,
int16_t lowlevelSignal,
int32_t gains[11]) {
int32_t tmp32;
int32_t env[10];
int32_t max_nrg;
int32_t cur_level;
int32_t gain32;
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, L;
// determine number of samples per ms
if (FS == 8000) {
L = 8;
} else if (FS == 16000 || FS == 32000 || FS == 48000) {
L = 16;
} else {
return -1;
}
// VAD for near end
logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, in_near[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;
}
}
// 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 = in_near[0][k * L + n] * in_near[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.
// Interpolate between gainTable[zeros] and gainTable[zeros-1].
tmp32 =
((stt->gainTable[zeros - 1] - stt->gainTable[zeros]) * (int64_t)frac) >>
12;
gains[k + 1] = stt->gainTable[zeros] + tmp32;
}
// 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++) {
// Find a shift of gains[k + 1] such that it can be squared without
// overflow, but at least by 10 bits.
zeros = 10;
if (gains[k + 1] > 47452159) {
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];
return 0;
}
int32_t WebRtcAgc_ApplyDigitalGains(const int32_t gains[11],
size_t num_bands,
uint32_t FS,
const int16_t* const* in_near,
int16_t* const* out) {
// Apply gain
// handle first sub frame separately
size_t 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 (size_t 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]));
}
}
// iterate over samples
int32_t delta = (gains[1] - gains[0]) * (1 << (4 - L2));
int32_t gain32 = gains[0] * (1 << 4);
for (size_t n = 0; n < L; n++) {
for (size_t i = 0; i < num_bands; ++i) {
int32_t out_tmp = (int64_t)out[i][n] * ((gain32 + 127) >> 7) >> 16;
if (out_tmp > 4095) {
out[i][n] = (int16_t)32767;
} else if (out_tmp < -4096) {
out[i][n] = (int16_t)-32768;
} else {
int32_t tmp32 = ((int64_t)out[i][n] * (gain32 >> 4)) >> 16;
out[i][n] = (int16_t)tmp32;
}
}
gain32 += delta;
}
// iterate over subframes
for (int k = 1; k < 10; k++) {
delta = (gains[k + 1] - gains[k]) * (1 << (4 - L2));
gain32 = gains[k] * (1 << 4);
// iterate over samples
for (size_t n = 0; n < L; n++) {
for (size_t 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;
int64_t tmp64;
// 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);
tmp64 = tmp32;
tmp64 += tmp32b >> 10;
tmp64 >>= 6;
// limit
if (tmp64 > 2048) {
tmp64 = 2048;
} else if (tmp64 < -2048) {
tmp64 = -2048;
}
state->logRatio = (int16_t)tmp64;
return state->logRatio; // Q10
}
} // namespace webrtc