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webrtc / src / 479a3c0f92b957a9ba1c242e7fe6e4d37b49ce32 / . / modules / audio_processing / aec / aec_core.cc
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/*
* 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.
*/
/*
* The core AEC algorithm, which is presented with time-aligned signals.
*/
#include "modules/audio_processing/aec/aec_core.h"
#include <math.h>
#include <stddef.h> // size_t
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <cmath>
#include "rtc_base/checks.h"
extern "C" {
#include "common_audio/ring_buffer.h"
}
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aec/aec_common.h"
#include "modules/audio_processing/aec/aec_core_optimized_methods.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "rtc_base/system/arch.h"
#include "system_wrappers/include/cpu_features_wrapper.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
// Buffer size (samples)
static const size_t kBufferSizeBlocks = 250; // 1 second of audio in 16 kHz.
// Metrics
static const size_t kSubCountLen = 4;
static const size_t kCountLen = 50;
static const int kDelayMetricsAggregationWindow = 1250; // 5 seconds at 16 kHz.
// Divergence metric is based on audio level, which gets updated every
// |kSubCountLen + 1| * PART_LEN samples. Divergence metric takes the statistics
// of |kDivergentFilterFractionAggregationWindowSize| audio levels. The
// following value corresponds to 1 second at 16 kHz.
static const int kDivergentFilterFractionAggregationWindowSize = 50;
// Quantities to control H band scaling for SWB input
static const float cnScaleHband = 0.4f; // scale for comfort noise in H band.
// Initial bin for averaging nlp gain in low band
static const int freqAvgIc = PART_LEN / 2;
// Matlab code to produce table:
// win = sqrt(hanning(63)); win = [0 ; win(1:32)];
// fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
ALIGN16_BEG const float ALIGN16_END WebRtcAec_sqrtHanning[65] = {
0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
1.00000000000000f};
// Matlab code to produce table:
// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65] = {
0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f, 0.1845f, 0.1926f,
0.2000f, 0.2069f, 0.2134f, 0.2195f, 0.2254f, 0.2309f, 0.2363f, 0.2414f,
0.2464f, 0.2512f, 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f, 0.3035f, 0.3070f,
0.3104f, 0.3138f, 0.3171f, 0.3204f, 0.3236f, 0.3268f, 0.3299f, 0.3330f,
0.3360f, 0.3390f, 0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f, 0.3752f, 0.3777f,
0.3803f, 0.3828f, 0.3854f, 0.3878f, 0.3903f, 0.3928f, 0.3952f, 0.3976f,
0.4000f};
// Matlab code to produce table:
// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65] = {
1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f, 1.3062f, 1.3307f,
1.3536f, 1.3750f, 1.3953f, 1.4146f, 1.4330f, 1.4507f, 1.4677f, 1.4841f,
1.5000f, 1.5154f, 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f, 1.6847f, 1.6960f,
1.7071f, 1.7181f, 1.7289f, 1.7395f, 1.7500f, 1.7603f, 1.7706f, 1.7806f,
1.7906f, 1.8004f, 1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f, 1.9186f, 1.9270f,
1.9354f, 1.9437f, 1.9520f, 1.9601f, 1.9682f, 1.9763f, 1.9843f, 1.9922f,
2.0000f};
// Delay Agnostic AEC parameters, still under development and may change.
static const float kDelayQualityThresholdMax = 0.07f;
static const float kDelayQualityThresholdMin = 0.01f;
static const int kInitialShiftOffset = 5;
#if !defined(WEBRTC_ANDROID)
static const int kDelayCorrectionStart = 1500; // 10 ms chunks
#endif
// Target suppression levels for nlp modes.
// log{0.001, 0.00001, 0.00000001}
static const float kTargetSupp[3] = {-6.9f, -11.5f, -18.4f};
// Two sets of parameters, one for the extended filter mode.
static const float kExtendedMinOverDrive[3] = {3.0f, 6.0f, 15.0f};
static const float kNormalMinOverDrive[3] = {1.0f, 2.0f, 5.0f};
const float WebRtcAec_kExtendedSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
{0.92f, 0.08f}};
const float WebRtcAec_kNormalSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
{0.93f, 0.07f}};
// Number of partitions forming the NLP's "preferred" bands.
enum { kPrefBandSize = 24 };
WebRtcAecFilterFar WebRtcAec_FilterFar;
WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
WebRtcAecOverdrive WebRtcAec_Overdrive;
WebRtcAecSuppress WebRtcAec_Suppress;
WebRtcAecComputeCoherence WebRtcAec_ComputeCoherence;
WebRtcAecUpdateCoherenceSpectra WebRtcAec_UpdateCoherenceSpectra;
WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
WebRtcAecWindowData WebRtcAec_WindowData;
__inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
return aRe * bRe - aIm * bIm;
}
__inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
return aRe * bIm + aIm * bRe;
}
// TODO(minyue): Due to a legacy bug, |framelevel| and |averagelevel| use a
// window, of which the length is 1 unit longer than indicated. Remove "+1" when
// the code is refactored.
PowerLevel::PowerLevel()
: framelevel(kSubCountLen + 1), averagelevel(kCountLen + 1) {}
BlockBuffer::BlockBuffer() {
buffer_ = WebRtc_CreateBuffer(kBufferSizeBlocks, sizeof(float) * PART_LEN);
RTC_CHECK(buffer_);
ReInit();
}
BlockBuffer::~BlockBuffer() {
WebRtc_FreeBuffer(buffer_);
}
void BlockBuffer::ReInit() {
WebRtc_InitBuffer(buffer_);
}
void BlockBuffer::Insert(const float block[PART_LEN]) {
WebRtc_WriteBuffer(buffer_, block, 1);
}
void BlockBuffer::ExtractExtendedBlock(float extended_block[PART_LEN2]) {
float* block_ptr = NULL;
RTC_DCHECK_LT(0, AvaliableSpace());
// Extract the previous block.
WebRtc_MoveReadPtr(buffer_, -1);
size_t read_elements = WebRtc_ReadBuffer(
buffer_, reinterpret_cast<void**>(&block_ptr), &extended_block[0], 1);
if (read_elements == 0u) {
std::fill_n(&extended_block[0], PART_LEN, 0.0f);
} else if (block_ptr != &extended_block[0]) {
memcpy(&extended_block[0], block_ptr, PART_LEN * sizeof(float));
}
// Extract the current block.
read_elements =
WebRtc_ReadBuffer(buffer_, reinterpret_cast<void**>(&block_ptr),
&extended_block[PART_LEN], 1);
if (read_elements == 0u) {
std::fill_n(&extended_block[PART_LEN], PART_LEN, 0.0f);
} else if (block_ptr != &extended_block[PART_LEN]) {
memcpy(&extended_block[PART_LEN], block_ptr, PART_LEN * sizeof(float));
}
}
int BlockBuffer::AdjustSize(int buffer_size_decrease) {
return WebRtc_MoveReadPtr(buffer_, buffer_size_decrease);
}
size_t BlockBuffer::Size() {
return static_cast<int>(WebRtc_available_read(buffer_));
}
size_t BlockBuffer::AvaliableSpace() {
return WebRtc_available_write(buffer_);
}
DivergentFilterFraction::DivergentFilterFraction()
: count_(0), occurrence_(0), fraction_(-1.0) {}
void DivergentFilterFraction::Reset() {
Clear();
fraction_ = -1.0;
}
void DivergentFilterFraction::AddObservation(const PowerLevel& nearlevel,
const PowerLevel& linoutlevel,
const PowerLevel& nlpoutlevel) {
const float near_level = nearlevel.framelevel.GetLatestMean();
const float level_increase =
linoutlevel.framelevel.GetLatestMean() - near_level;
const bool output_signal_active =
nlpoutlevel.framelevel.GetLatestMean() > 40.0 * nlpoutlevel.minlevel;
// Level increase should be, in principle, negative, when the filter
// does not diverge. Here we allow some margin (0.01 * near end level) and
// numerical error (1.0). We count divergence only when the AEC output
// signal is active.
if (output_signal_active && level_increase > std::max(0.01 * near_level, 1.0))
occurrence_++;
++count_;
if (count_ == kDivergentFilterFractionAggregationWindowSize) {
fraction_ = static_cast<float>(occurrence_) /
kDivergentFilterFractionAggregationWindowSize;
Clear();
}
}
float DivergentFilterFraction::GetLatestFraction() const {
return fraction_;
}
void DivergentFilterFraction::Clear() {
count_ = 0;
occurrence_ = 0;
}
// TODO(minyue): Moving some initialization from WebRtcAec_CreateAec() to ctor.
AecCore::AecCore(int instance_index)
: data_dumper(new ApmDataDumper(instance_index)) {}
AecCore::~AecCore() {}
static int CmpFloat(const void* a, const void* b) {
const float* da = (const float*)a;
const float* db = (const float*)b;
return (*da > *db) - (*da < *db);
}
static void FilterFar(int num_partitions,
int x_fft_buf_block_pos,
float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float y_fft[2][PART_LEN1]) {
int i;
for (i = 0; i < num_partitions; i++) {
int j;
int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
int pos = i * PART_LEN1;
// Check for wrap
if (i + x_fft_buf_block_pos >= num_partitions) {
xPos -= num_partitions * (PART_LEN1);
}
for (j = 0; j < PART_LEN1; j++) {
y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j], x_fft_buf[1][xPos + j],
h_fft_buf[0][pos + j], h_fft_buf[1][pos + j]);
y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j], x_fft_buf[1][xPos + j],
h_fft_buf[0][pos + j], h_fft_buf[1][pos + j]);
}
}
}
static void ScaleErrorSignal(float mu,
float error_threshold,
float x_pow[PART_LEN1],
float ef[2][PART_LEN1]) {
int i;
float abs_ef;
for (i = 0; i < (PART_LEN1); i++) {
ef[0][i] /= (x_pow[i] + 1e-10f);
ef[1][i] /= (x_pow[i] + 1e-10f);
abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
if (abs_ef > error_threshold) {
abs_ef = error_threshold / (abs_ef + 1e-10f);
ef[0][i] *= abs_ef;
ef[1][i] *= abs_ef;
}
// Stepsize factor
ef[0][i] *= mu;
ef[1][i] *= mu;
}
}
static void FilterAdaptation(
const OouraFft& ooura_fft,
int num_partitions,
int x_fft_buf_block_pos,
float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float e_fft[2][PART_LEN1],
float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
int i, j;
float fft[PART_LEN2];
for (i = 0; i < num_partitions; i++) {
int xPos = (i + x_fft_buf_block_pos) * (PART_LEN1);
int pos;
// Check for wrap
if (i + x_fft_buf_block_pos >= num_partitions) {
xPos -= num_partitions * PART_LEN1;
}
pos = i * PART_LEN1;
for (j = 0; j < PART_LEN; j++) {
fft[2 * j] = MulRe(x_fft_buf[0][xPos + j], -x_fft_buf[1][xPos + j],
e_fft[0][j], e_fft[1][j]);
fft[2 * j + 1] = MulIm(x_fft_buf[0][xPos + j], -x_fft_buf[1][xPos + j],
e_fft[0][j], e_fft[1][j]);
}
fft[1] =
MulRe(x_fft_buf[0][xPos + PART_LEN], -x_fft_buf[1][xPos + PART_LEN],
e_fft[0][PART_LEN], e_fft[1][PART_LEN]);
ooura_fft.InverseFft(fft);
memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
// fft scaling
{
float scale = 2.0f / PART_LEN2;
for (j = 0; j < PART_LEN; j++) {
fft[j] *= scale;
}
}
ooura_fft.Fft(fft);
h_fft_buf[0][pos] += fft[0];
h_fft_buf[0][pos + PART_LEN] += fft[1];
for (j = 1; j < PART_LEN; j++) {
h_fft_buf[0][pos + j] += fft[2 * j];
h_fft_buf[1][pos + j] += fft[2 * j + 1];
}
}
}
static void Overdrive(float overdrive_scaling,
const float hNlFb,
float hNl[PART_LEN1]) {
for (int i = 0; i < PART_LEN1; ++i) {
// Weight subbands
if (hNl[i] > hNlFb) {
hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
(1 - WebRtcAec_weightCurve[i]) * hNl[i];
}
hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
}
}
static void Suppress(const float hNl[PART_LEN1], float efw[2][PART_LEN1]) {
for (int i = 0; i < PART_LEN1; ++i) {
// Suppress error signal
efw[0][i] *= hNl[i];
efw[1][i] *= hNl[i];
// Ooura fft returns incorrect sign on imaginary component. It matters here
// because we are making an additive change with comfort noise.
efw[1][i] *= -1;
}
}
static int PartitionDelay(
int num_partitions,
float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
// Measures the energy in each filter partition and returns the partition with
// highest energy.
// TODO(bjornv): Spread computational cost by computing one partition per
// block?
float wfEnMax = 0;
int i;
int delay = 0;
for (i = 0; i < num_partitions; i++) {
int j;
int pos = i * PART_LEN1;
float wfEn = 0;
for (j = 0; j < PART_LEN1; j++) {
wfEn += h_fft_buf[0][pos + j] * h_fft_buf[0][pos + j] +
h_fft_buf[1][pos + j] * h_fft_buf[1][pos + j];
}
if (wfEn > wfEnMax) {
wfEnMax = wfEn;
delay = i;
}
}
return delay;
}
// Update metric with 10 * log10(numerator / denominator).
static void UpdateLogRatioMetric(Stats* metric,
float numerator,
float denominator) {
RTC_DCHECK(metric);
RTC_CHECK(numerator >= 0);
RTC_CHECK(denominator >= 0);
const float log_numerator = std::log10(numerator + 1e-10f);
const float log_denominator = std::log10(denominator + 1e-10f);
metric->instant = 10.0f * (log_numerator - log_denominator);
// Max.
if (metric->instant > metric->max)
metric->max = metric->instant;
// Min.
if (metric->instant < metric->min)
metric->min = metric->instant;
// Average.
metric->counter++;
// This is to protect overflow, which should almost never happen.
RTC_CHECK_NE(0, metric->counter);
metric->sum += metric->instant;
metric->average = metric->sum / metric->counter;
// Upper mean.
if (metric->instant > metric->average) {
metric->hicounter++;
// This is to protect overflow, which should almost never happen.
RTC_CHECK_NE(0, metric->hicounter);
metric->hisum += metric->instant;
metric->himean = metric->hisum / metric->hicounter;
}
}
// Threshold to protect against the ill-effects of a zero far-end.
const float WebRtcAec_kMinFarendPSD = 15;
// Updates the following smoothed Power Spectral Densities (PSD):
// - sd : near-end
// - se : residual echo
// - sx : far-end
// - sde : cross-PSD of near-end and residual echo
// - sxd : cross-PSD of near-end and far-end
//
// In addition to updating the PSDs, also the filter diverge state is
// determined.
static void UpdateCoherenceSpectra(int mult,
bool extended_filter_enabled,
float efw[2][PART_LEN1],
float dfw[2][PART_LEN1],
float xfw[2][PART_LEN1],
CoherenceState* coherence_state,
short* filter_divergence_state,
int* extreme_filter_divergence) {
// Power estimate smoothing coefficients.
const float* ptrGCoh =
extended_filter_enabled
? WebRtcAec_kExtendedSmoothingCoefficients[mult - 1]
: WebRtcAec_kNormalSmoothingCoefficients[mult - 1];
int i;
float sdSum = 0, seSum = 0;
for (i = 0; i < PART_LEN1; i++) {
coherence_state->sd[i] =
ptrGCoh[0] * coherence_state->sd[i] +
ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
coherence_state->se[i] =
ptrGCoh[0] * coherence_state->se[i] +
ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
// We threshold here to protect against the ill-effects of a zero farend.
// The threshold is not arbitrarily chosen, but balances protection and
// adverse interaction with the algorithm's tuning.
// TODO(bjornv): investigate further why this is so sensitive.
coherence_state->sx[i] =
ptrGCoh[0] * coherence_state->sx[i] +
ptrGCoh[1] *
WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
WebRtcAec_kMinFarendPSD);
coherence_state->sde[i][0] =
ptrGCoh[0] * coherence_state->sde[i][0] +
ptrGCoh[1] * (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
coherence_state->sde[i][1] =
ptrGCoh[0] * coherence_state->sde[i][1] +
ptrGCoh[1] * (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
coherence_state->sxd[i][0] =
ptrGCoh[0] * coherence_state->sxd[i][0] +
ptrGCoh[1] * (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
coherence_state->sxd[i][1] =
ptrGCoh[0] * coherence_state->sxd[i][1] +
ptrGCoh[1] * (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
sdSum += coherence_state->sd[i];
seSum += coherence_state->se[i];
}
// Divergent filter safeguard update.
*filter_divergence_state =
(*filter_divergence_state ? 1.05f : 1.0f) * seSum > sdSum;
// Signal extreme filter divergence if the error is significantly larger
// than the nearend (13 dB).
*extreme_filter_divergence = (seSum > (19.95f * sdSum));
}
// Window time domain data to be used by the fft.
__inline static void WindowData(float* x_windowed, const float* x) {
int i;
for (i = 0; i < PART_LEN; i++) {
x_windowed[i] = x[i] * WebRtcAec_sqrtHanning[i];
x_windowed[PART_LEN + i] =
x[PART_LEN + i] * WebRtcAec_sqrtHanning[PART_LEN - i];
}
}
// Puts fft output data into a complex valued array.
__inline static void StoreAsComplex(const float* data,
float data_complex[2][PART_LEN1]) {
int i;
data_complex[0][0] = data[0];
data_complex[1][0] = 0;
for (i = 1; i < PART_LEN; i++) {
data_complex[0][i] = data[2 * i];
data_complex[1][i] = data[2 * i + 1];
}
data_complex[0][PART_LEN] = data[1];
data_complex[1][PART_LEN] = 0;
}
static void ComputeCoherence(const CoherenceState* coherence_state,
float* cohde,
float* cohxd) {
// Subband coherence
for (int i = 0; i < PART_LEN1; i++) {
cohde[i] = (coherence_state->sde[i][0] * coherence_state->sde[i][0] +
coherence_state->sde[i][1] * coherence_state->sde[i][1]) /
(coherence_state->sd[i] * coherence_state->se[i] + 1e-10f);
cohxd[i] = (coherence_state->sxd[i][0] * coherence_state->sxd[i][0] +
coherence_state->sxd[i][1] * coherence_state->sxd[i][1]) /
(coherence_state->sx[i] * coherence_state->sd[i] + 1e-10f);
}
}
static void GetHighbandGain(const float* lambda, float* nlpGainHband) {
int i;
*nlpGainHband = 0.0f;
for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
*nlpGainHband += lambda[i];
}
*nlpGainHband /= static_cast<float>(PART_LEN1 - 1 - freqAvgIc);
}
static void GenerateComplexNoise(uint32_t* seed, float noise[2][PART_LEN1]) {
const float kPi2 = 6.28318530717959f;
int16_t randW16[PART_LEN];
WebRtcSpl_RandUArray(randW16, PART_LEN, seed);
noise[0][0] = 0;
noise[1][0] = 0;
for (size_t i = 1; i < PART_LEN1; i++) {
float tmp = kPi2 * randW16[i - 1] / 32768.f;
noise[0][i] = cosf(tmp);
noise[1][i] = -sinf(tmp);
}
noise[1][PART_LEN] = 0;
}
static void ComfortNoise(bool generate_high_frequency_noise,
uint32_t* seed,
float e_fft[2][PART_LEN1],
float high_frequency_comfort_noise[2][PART_LEN1],
const float* noise_spectrum,
const float* suppressor_gain) {
float complex_noise[2][PART_LEN1];
GenerateComplexNoise(seed, complex_noise);
// Shape, scale and add comfort noise.
for (int i = 1; i < PART_LEN1; ++i) {
float noise_scaling =
sqrtf(WEBRTC_SPL_MAX(1 - suppressor_gain[i] * suppressor_gain[i], 0)) *
sqrtf(noise_spectrum[i]);
e_fft[0][i] += noise_scaling * complex_noise[0][i];
e_fft[1][i] += noise_scaling * complex_noise[1][i];
}
// Form comfort noise for higher frequencies.
if (generate_high_frequency_noise) {
// Compute average noise power and nlp gain over the second half of freq
// spectrum (i.e., 4->8khz).
int start_avg_band = PART_LEN1 / 2;
float upper_bands_noise_power = 0.f;
float upper_bands_suppressor_gain = 0.f;
for (int i = start_avg_band; i < PART_LEN1; ++i) {
upper_bands_noise_power += sqrtf(noise_spectrum[i]);
upper_bands_suppressor_gain +=
sqrtf(WEBRTC_SPL_MAX(1 - suppressor_gain[i] * suppressor_gain[i], 0));
}
upper_bands_noise_power /= (PART_LEN1 - start_avg_band);
upper_bands_suppressor_gain /= (PART_LEN1 - start_avg_band);
// Shape, scale and add comfort noise.
float noise_scaling = upper_bands_suppressor_gain * upper_bands_noise_power;
high_frequency_comfort_noise[0][0] = 0;
high_frequency_comfort_noise[1][0] = 0;
for (int i = 1; i < PART_LEN1; ++i) {
high_frequency_comfort_noise[0][i] = noise_scaling * complex_noise[0][i];
high_frequency_comfort_noise[1][i] = noise_scaling * complex_noise[1][i];
}
high_frequency_comfort_noise[1][PART_LEN] = 0;
} else {
memset(high_frequency_comfort_noise, 0,
2 * PART_LEN1 * sizeof(high_frequency_comfort_noise[0][0]));
}
}
static void InitLevel(PowerLevel* level) {
const float kBigFloat = 1E17f;
level->averagelevel.Reset();
level->framelevel.Reset();
level->minlevel = kBigFloat;
}
static void InitStats(Stats* stats) {
stats->instant = kOffsetLevel;
stats->average = kOffsetLevel;
stats->max = kOffsetLevel;
stats->min = kOffsetLevel * (-1);
stats->sum = 0;
stats->hisum = 0;
stats->himean = kOffsetLevel;
stats->counter = 0;
stats->hicounter = 0;
}
static void InitMetrics(AecCore* self) {
self->stateCounter = 0;
InitLevel(&self->farlevel);
InitLevel(&self->nearlevel);
InitLevel(&self->linoutlevel);
InitLevel(&self->nlpoutlevel);
InitStats(&self->erl);
InitStats(&self->erle);
InitStats(&self->aNlp);
InitStats(&self->rerl);
self->divergent_filter_fraction.Reset();
}
static float CalculatePower(const float* in, size_t num_samples) {
size_t k;
float energy = 0.0f;
for (k = 0; k < num_samples; ++k) {
energy += in[k] * in[k];
}
return energy / num_samples;
}
static void UpdateLevel(PowerLevel* level, float power) {
level->framelevel.AddValue(power);
if (level->framelevel.EndOfBlock()) {
const float new_frame_level = level->framelevel.GetLatestMean();
if (new_frame_level > 0) {
if (new_frame_level < level->minlevel) {
level->minlevel = new_frame_level; // New minimum.
} else {
level->minlevel *= (1 + 0.001f); // Small increase.
}
}
level->averagelevel.AddValue(new_frame_level);
}
}
static void UpdateMetrics(AecCore* aec) {
const float actThresholdNoisy = 8.0f;
const float actThresholdClean = 40.0f;
const float noisyPower = 300000.0f;
float actThreshold;
if (aec->echoState) { // Check if echo is likely present
aec->stateCounter++;
}
if (aec->linoutlevel.framelevel.EndOfBlock()) {
aec->divergent_filter_fraction.AddObservation(
aec->nearlevel, aec->linoutlevel, aec->nlpoutlevel);
}
if (aec->farlevel.averagelevel.EndOfBlock()) {
if (aec->farlevel.minlevel < noisyPower) {
actThreshold = actThresholdClean;
} else {
actThreshold = actThresholdNoisy;
}
const float far_average_level = aec->farlevel.averagelevel.GetLatestMean();
// The last condition is to let estimation be made in active far-end
// segments only.
if ((aec->stateCounter > (0.5f * kCountLen * kSubCountLen)) &&
(aec->farlevel.framelevel.EndOfBlock()) &&
(far_average_level > (actThreshold * aec->farlevel.minlevel))) {
// ERL: error return loss.
const float near_average_level =
aec->nearlevel.averagelevel.GetLatestMean();
UpdateLogRatioMetric(&aec->erl, far_average_level, near_average_level);
// A_NLP: error return loss enhanced before the nonlinear suppression.
const float linout_average_level =
aec->linoutlevel.averagelevel.GetLatestMean();
UpdateLogRatioMetric(&aec->aNlp, near_average_level,
linout_average_level);
// ERLE: error return loss enhanced.
const float nlpout_average_level =
aec->nlpoutlevel.averagelevel.GetLatestMean();
UpdateLogRatioMetric(&aec->erle, near_average_level,
nlpout_average_level);
}
aec->stateCounter = 0;
}
}
static void UpdateDelayMetrics(AecCore* self) {
int i = 0;
int delay_values = 0;
int median = 0;
int lookahead = WebRtc_lookahead(self->delay_estimator);
const int kMsPerBlock = PART_LEN / (self->mult * 8);
int64_t l1_norm = 0;
if (self->num_delay_values == 0) {
// We have no new delay value data. Even though -1 is a valid |median| in
// the sense that we allow negative values, it will practically never be
// used since multiples of |kMsPerBlock| will always be returned.
// We therefore use -1 to indicate in the logs that the delay estimator was
// not able to estimate the delay.
self->delay_median = -1;
self->delay_std = -1;
self->fraction_poor_delays = -1;
return;
}
// Start value for median count down.
delay_values = self->num_delay_values >> 1;
// Get median of delay values since last update.
for (i = 0; i < kHistorySizeBlocks; i++) {
delay_values -= self->delay_histogram[i];
if (delay_values < 0) {
median = i;
break;
}
}
// Account for lookahead.
self->delay_median = (median - lookahead) * kMsPerBlock;
// Calculate the L1 norm, with median value as central moment.
for (i = 0; i < kHistorySizeBlocks; i++) {
l1_norm += abs(i - median) * self->delay_histogram[i];
}
self->delay_std = static_cast<int>((l1_norm + self->num_delay_values / 2) /
self->num_delay_values) *
kMsPerBlock;
// Determine fraction of delays that are out of bounds, that is, either
// negative (anti-causal system) or larger than the AEC filter length.
{
int num_delays_out_of_bounds = self->num_delay_values;
const int histogram_length =
sizeof(self->delay_histogram) / sizeof(self->delay_histogram[0]);
for (i = lookahead; i < lookahead + self->num_partitions; ++i) {
if (i < histogram_length)
num_delays_out_of_bounds -= self->delay_histogram[i];
}
self->fraction_poor_delays =
static_cast<float>(num_delays_out_of_bounds) / self->num_delay_values;
}
// Reset histogram.
memset(self->delay_histogram, 0, sizeof(self->delay_histogram));
self->num_delay_values = 0;
}
static void ScaledInverseFft(const OouraFft& ooura_fft,
float freq_data[2][PART_LEN1],
float time_data[PART_LEN2],
float scale,
int conjugate) {
int i;
const float normalization = scale / static_cast<float>(PART_LEN2);
const float sign = (conjugate ? -1 : 1);
time_data[0] = freq_data[0][0] * normalization;
time_data[1] = freq_data[0][PART_LEN] * normalization;
for (i = 1; i < PART_LEN; i++) {
time_data[2 * i] = freq_data[0][i] * normalization;
time_data[2 * i + 1] = sign * freq_data[1][i] * normalization;
}
ooura_fft.InverseFft(time_data);
}
static void Fft(const OouraFft& ooura_fft,
float time_data[PART_LEN2],
float freq_data[2][PART_LEN1]) {
int i;
ooura_fft.Fft(time_data);
// Reorder fft output data.
freq_data[1][0] = 0;
freq_data[1][PART_LEN] = 0;
freq_data[0][0] = time_data[0];
freq_data[0][PART_LEN] = time_data[1];
for (i = 1; i < PART_LEN; i++) {
freq_data[0][i] = time_data[2 * i];
freq_data[1][i] = time_data[2 * i + 1];
}
}
static int SignalBasedDelayCorrection(AecCore* self) {
int delay_correction = 0;
int last_delay = -2;
RTC_DCHECK(self);
#if !defined(WEBRTC_ANDROID)
// On desktops, turn on correction after |kDelayCorrectionStart| frames. This
// is to let the delay estimation get a chance to converge. Also, if the
// playout audio volume is low (or even muted) the delay estimation can return
// a very large delay, which will break the AEC if it is applied.
if (self->frame_count < kDelayCorrectionStart) {
self->data_dumper->DumpRaw("aec_da_reported_delay", 1, &last_delay);
return 0;
}
#endif
// 1. Check for non-negative delay estimate. Note that the estimates we get
// from the delay estimation are not compensated for lookahead. Hence, a
// negative |last_delay| is an invalid one.
// 2. Verify that there is a delay change. In addition, only allow a change
// if the delay is outside a certain region taking the AEC filter length
// into account.
// TODO(bjornv): Investigate if we can remove the non-zero delay change check.
// 3. Only allow delay correction if the delay estimation quality exceeds
// |delay_quality_threshold|.
// 4. Finally, verify that the proposed |delay_correction| is feasible by
// comparing with the size of the far-end buffer.
last_delay = WebRtc_last_delay(self->delay_estimator);
self->data_dumper->DumpRaw("aec_da_reported_delay", 1, &last_delay);
if ((last_delay >= 0) && (last_delay != self->previous_delay) &&
(WebRtc_last_delay_quality(self->delay_estimator) >
self->delay_quality_threshold)) {
int delay = last_delay - WebRtc_lookahead(self->delay_estimator);
// Allow for a slack in the actual delay, defined by a |lower_bound| and an
// |upper_bound|. The adaptive echo cancellation filter is currently
// |num_partitions| (of 64 samples) long. If the delay estimate is negative
// or at least 3/4 of the filter length we open up for correction.
const int lower_bound = 0;
const int upper_bound = self->num_partitions * 3 / 4;
const int do_correction = delay <= lower_bound || delay > upper_bound;
if (do_correction == 1) {
int available_read = self->farend_block_buffer_.Size();
// With |shift_offset| we gradually rely on the delay estimates. For
// positive delays we reduce the correction by |shift_offset| to lower the
// risk of pushing the AEC into a non causal state. For negative delays
// we rely on the values up to a rounding error, hence compensate by 1
// element to make sure to push the delay into the causal region.
delay_correction = -delay;
delay_correction += delay > self->shift_offset ? self->shift_offset : 1;
self->shift_offset--;
self->shift_offset = (self->shift_offset <= 1 ? 1 : self->shift_offset);
if (delay_correction > available_read - self->mult - 1) {
// There is not enough data in the buffer to perform this shift. Hence,
// we do not rely on the delay estimate and do nothing.
delay_correction = 0;
} else {
self->previous_delay = last_delay;
++self->delay_correction_count;
}
}
}
// Update the |delay_quality_threshold| once we have our first delay
// correction.
if (self->delay_correction_count > 0) {
float delay_quality = WebRtc_last_delay_quality(self->delay_estimator);
delay_quality =
(delay_quality > kDelayQualityThresholdMax ? kDelayQualityThresholdMax
: delay_quality);
self->delay_quality_threshold =
(delay_quality > self->delay_quality_threshold
? delay_quality
: self->delay_quality_threshold);
}
self->data_dumper->DumpRaw("aec_da_delay_correction", 1, &delay_correction);
return delay_correction;
}
static void RegressorPower(
int num_partitions,
int latest_added_partition,
float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float x_pow[PART_LEN1]) {
RTC_DCHECK_LT(latest_added_partition, num_partitions);
memset(x_pow, 0, PART_LEN1 * sizeof(x_pow[0]));
int partition = latest_added_partition;
int x_fft_buf_position = partition * PART_LEN1;
for (int i = 0; i < num_partitions; ++i) {
for (int bin = 0; bin < PART_LEN1; ++bin) {
float re = x_fft_buf[0][x_fft_buf_position];
float im = x_fft_buf[1][x_fft_buf_position];
x_pow[bin] += re * re + im * im;
++x_fft_buf_position;
}
++partition;
if (partition == num_partitions) {
partition = 0;
RTC_DCHECK_EQ(num_partitions * PART_LEN1, x_fft_buf_position);
x_fft_buf_position = 0;
}
}
}
static void EchoSubtraction(
const OouraFft& ooura_fft,
int num_partitions,
int extended_filter_enabled,
int* extreme_filter_divergence,
float filter_step_size,
float error_threshold,
float* x_fft,
int* x_fft_buf_block_pos,
float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float* const y,
float x_pow[PART_LEN1],
float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
float echo_subtractor_output[PART_LEN]) {
float s_fft[2][PART_LEN1];
float e_extended[PART_LEN2];
float s_extended[PART_LEN2];
float* s;
float e[PART_LEN];
float e_fft[2][PART_LEN1];
int i;
// Update the x_fft_buf block position.
(*x_fft_buf_block_pos)--;
if ((*x_fft_buf_block_pos) == -1) {
*x_fft_buf_block_pos = num_partitions - 1;
}
// Buffer x_fft.
memcpy(x_fft_buf[0] + (*x_fft_buf_block_pos) * PART_LEN1, x_fft,
sizeof(float) * PART_LEN1);
memcpy(x_fft_buf[1] + (*x_fft_buf_block_pos) * PART_LEN1, &x_fft[PART_LEN1],
sizeof(float) * PART_LEN1);
memset(s_fft, 0, sizeof(s_fft));
// Conditionally reset the echo subtraction filter if the filter has diverged
// significantly.
if (!extended_filter_enabled && *extreme_filter_divergence) {
memset(h_fft_buf, 0,
2 * kExtendedNumPartitions * PART_LEN1 * sizeof(h_fft_buf[0][0]));
*extreme_filter_divergence = 0;
}
// Produce echo estimate s_fft.
WebRtcAec_FilterFar(num_partitions, *x_fft_buf_block_pos, x_fft_buf,
h_fft_buf, s_fft);
// Compute the time-domain echo estimate s.
ScaledInverseFft(ooura_fft, s_fft, s_extended, 2.0f, 0);
s = &s_extended[PART_LEN];
// Compute the time-domain echo prediction error.
for (i = 0; i < PART_LEN; ++i) {
e[i] = y[i] - s[i];
}
// Compute the frequency domain echo prediction error.
memset(e_extended, 0, sizeof(float) * PART_LEN);
memcpy(e_extended + PART_LEN, e, sizeof(float) * PART_LEN);
Fft(ooura_fft, e_extended, e_fft);
// Scale error signal inversely with far power.
WebRtcAec_ScaleErrorSignal(filter_step_size, error_threshold, x_pow, e_fft);
WebRtcAec_FilterAdaptation(ooura_fft, num_partitions, *x_fft_buf_block_pos,
x_fft_buf, e_fft, h_fft_buf);
memcpy(echo_subtractor_output, e, sizeof(float) * PART_LEN);
}
static void FormSuppressionGain(AecCore* aec,
float cohde[PART_LEN1],
float cohxd[PART_LEN1],
float hNl[PART_LEN1]) {
float hNlDeAvg, hNlXdAvg;
float hNlPref[kPrefBandSize];
float hNlFb = 0, hNlFbLow = 0;
const int prefBandSize = kPrefBandSize / aec->mult;
const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
const int minPrefBand = 4 / aec->mult;
// Power estimate smoothing coefficients.
const float* min_overdrive = aec->extended_filter_enabled
? kExtendedMinOverDrive
: kNormalMinOverDrive;
hNlXdAvg = 0;
for (int i = minPrefBand; i < prefBandSize + minPrefBand; ++i) {
hNlXdAvg += cohxd[i];
}
hNlXdAvg /= prefBandSize;
hNlXdAvg = 1 - hNlXdAvg;
hNlDeAvg = 0;
for (int i = minPrefBand; i < prefBandSize + minPrefBand; ++i) {
hNlDeAvg += cohde[i];
}
hNlDeAvg /= prefBandSize;
if (hNlXdAvg < 0.75f && hNlXdAvg < aec->hNlXdAvgMin) {
aec->hNlXdAvgMin = hNlXdAvg;
}
if (hNlDeAvg > 0.98f && hNlXdAvg > 0.9f) {
aec->stNearState = 1;
} else if (hNlDeAvg < 0.95f || hNlXdAvg < 0.8f) {
aec->stNearState = 0;
}
if (aec->hNlXdAvgMin == 1) {
aec->echoState = 0;
aec->overDrive = min_overdrive[aec->nlp_mode];
if (aec->stNearState == 1) {
memcpy(hNl, cohde, sizeof(hNl[0]) * PART_LEN1);
hNlFb = hNlDeAvg;
hNlFbLow = hNlDeAvg;
} else {
for (int i = 0; i < PART_LEN1; ++i) {
hNl[i] = 1 - cohxd[i];
hNl[i] = std::max(hNl[i], 0.f);
}
hNlFb = hNlXdAvg;
hNlFbLow = hNlXdAvg;
}
} else {
if (aec->stNearState == 1) {
aec->echoState = 0;
memcpy(hNl, cohde, sizeof(hNl[0]) * PART_LEN1);
hNlFb = hNlDeAvg;
hNlFbLow = hNlDeAvg;
} else {
aec->echoState = 1;
for (int i = 0; i < PART_LEN1; ++i) {
hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
hNl[i] = std::max(hNl[i], 0.f);
}
// Select an order statistic from the preferred bands.
// TODO(peah): Using quicksort now, but a selection algorithm may be
// preferred.
memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
hNlFb = hNlPref[static_cast<int>(
std::floor(prefBandQuant * (prefBandSize - 1)))];
hNlFbLow = hNlPref[static_cast<int>(
std::floor(prefBandQuantLow * (prefBandSize - 1)))];
}
}
// Track the local filter minimum to determine suppression overdrive.
if (hNlFbLow < 0.6f && hNlFbLow < aec->hNlFbLocalMin) {
aec->hNlFbLocalMin = hNlFbLow;
aec->hNlFbMin = hNlFbLow;
aec->hNlNewMin = 1;
aec->hNlMinCtr = 0;
}
aec->hNlFbLocalMin =
WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1);
aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1);
if (aec->hNlNewMin == 1) {
aec->hNlMinCtr++;
}
if (aec->hNlMinCtr == 2) {
aec->hNlNewMin = 0;
aec->hNlMinCtr = 0;
aec->overDrive = WEBRTC_SPL_MAX(
kTargetSupp[aec->nlp_mode] /
static_cast<float>(std::log(aec->hNlFbMin + 1e-10f) + 1e-10f),
min_overdrive[aec->nlp_mode]);
}
// Smooth the overdrive.
if (aec->overDrive < aec->overdrive_scaling) {
aec->overdrive_scaling =
0.99f * aec->overdrive_scaling + 0.01f * aec->overDrive;
} else {
aec->overdrive_scaling =
0.9f * aec->overdrive_scaling + 0.1f * aec->overDrive;
}
// Apply the overdrive.
WebRtcAec_Overdrive(aec->overdrive_scaling, hNlFb, hNl);
}
static void EchoSuppression(const OouraFft& ooura_fft,
AecCore* aec,
float* nearend_extended_block_lowest_band,
float farend_extended_block[PART_LEN2],
float* echo_subtractor_output,
float output[NUM_HIGH_BANDS_MAX + 1][PART_LEN]) {
float efw[2][PART_LEN1];
float xfw[2][PART_LEN1];
float dfw[2][PART_LEN1];
float comfortNoiseHband[2][PART_LEN1];
float fft[PART_LEN2];
float nlpGainHband;
int i;
size_t j;
// Coherence and non-linear filter
float cohde[PART_LEN1], cohxd[PART_LEN1];
float hNl[PART_LEN1];
// Filter energy
const int delayEstInterval = 10 * aec->mult;
float* xfw_ptr = NULL;
// Update eBuf with echo subtractor output.
memcpy(aec->eBuf + PART_LEN, echo_subtractor_output,
sizeof(float) * PART_LEN);
// Analysis filter banks for the echo suppressor.
// Windowed near-end ffts.
WindowData(fft, nearend_extended_block_lowest_band);
ooura_fft.Fft(fft);
StoreAsComplex(fft, dfw);
// Windowed echo suppressor output ffts.
WindowData(fft, aec->eBuf);
ooura_fft.Fft(fft);
StoreAsComplex(fft, efw);
// NLP
// Convert far-end partition to the frequency domain with windowing.
WindowData(fft, farend_extended_block);
Fft(ooura_fft, fft, xfw);
xfw_ptr = &xfw[0][0];
// Buffer far.
memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
aec->delayEstCtr++;
if (aec->delayEstCtr == delayEstInterval) {
aec->delayEstCtr = 0;
aec->delayIdx = WebRtcAec_PartitionDelay(aec->num_partitions, aec->wfBuf);
}
aec->data_dumper->DumpRaw("aec_nlp_delay", 1, &aec->delayIdx);
// Use delayed far.
memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1,
sizeof(xfw[0][0]) * 2 * PART_LEN1);
WebRtcAec_UpdateCoherenceSpectra(aec->mult, aec->extended_filter_enabled == 1,
efw, dfw, xfw, &aec->coherence_state,
&aec->divergeState,
&aec->extreme_filter_divergence);
WebRtcAec_ComputeCoherence(&aec->coherence_state, cohde, cohxd);
// Select the microphone signal as output if the filter is deemed to have
// diverged.
if (aec->divergeState) {
memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
}
FormSuppressionGain(aec, cohde, cohxd, hNl);
aec->data_dumper->DumpRaw("aec_nlp_gain", PART_LEN1, hNl);
WebRtcAec_Suppress(hNl, efw);
// Add comfort noise.
ComfortNoise(aec->num_bands > 1, &aec->seed, efw, comfortNoiseHband,
aec->noisePow, hNl);
// Inverse error fft.
ScaledInverseFft(ooura_fft, efw, fft, 2.0f, 1);
// Overlap and add to obtain output.
for (i = 0; i < PART_LEN; i++) {
output[0][i] = (fft[i] * WebRtcAec_sqrtHanning[i] +
aec->outBuf[i] * WebRtcAec_sqrtHanning[PART_LEN - i]);
// Saturate output to keep it in the allowed range.
output[0][i] = WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, output[0][i],
WEBRTC_SPL_WORD16_MIN);
}
memcpy(aec->outBuf, &fft[PART_LEN], PART_LEN * sizeof(aec->outBuf[0]));
// For H band
if (aec->num_bands > 1) {
// H band gain
// average nlp over low band: average over second half of freq spectrum
// (4->8khz)
GetHighbandGain(hNl, &nlpGainHband);
// Inverse comfort_noise
ScaledInverseFft(ooura_fft, comfortNoiseHband, fft, 2.0f, 0);
// compute gain factor
for (j = 1; j < aec->num_bands; ++j) {
for (i = 0; i < PART_LEN; i++) {
output[j][i] = aec->previous_nearend_block[j][i] * nlpGainHband;
}
}
// Add some comfort noise where Hband is attenuated.
for (i = 0; i < PART_LEN; i++) {
output[1][i] += cnScaleHband * fft[i];
}
// Saturate output to keep it in the allowed range.
for (j = 1; j < aec->num_bands; ++j) {
for (i = 0; i < PART_LEN; i++) {
output[j][i] = WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, output[j][i],
WEBRTC_SPL_WORD16_MIN);
}
}
}
// Copy the current block to the old position.
memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
memmove(aec->xfwBuf + PART_LEN1, aec->xfwBuf,
sizeof(aec->xfwBuf) - sizeof(complex_t) * PART_LEN1);
}
static void ProcessNearendBlock(
AecCore* aec,
float farend_extended_block_lowest_band[PART_LEN2],
float nearend_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN],
float output_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN]) {
size_t i;
float fft[PART_LEN2];
float nearend_extended_block_lowest_band[PART_LEN2];
float farend_fft[2][PART_LEN1];
float nearend_fft[2][PART_LEN1];
float far_spectrum = 0.0f;
float near_spectrum = 0.0f;
float abs_far_spectrum[PART_LEN1];
float abs_near_spectrum[PART_LEN1];
const float gPow[2] = {0.9f, 0.1f};
// Noise estimate constants.
const int noiseInitBlocks = 500 * aec->mult;
const float step = 0.1f;
const float ramp = 1.0002f;
const float gInitNoise[2] = {0.999f, 0.001f};
float echo_subtractor_output[PART_LEN];
aec->data_dumper->DumpWav("aec_far", PART_LEN,
&farend_extended_block_lowest_band[PART_LEN],
std::min(aec->sampFreq, 16000), 1);
aec->data_dumper->DumpWav("aec_near", PART_LEN, &nearend_block[0][0],
std::min(aec->sampFreq, 16000), 1);
if (aec->metricsMode == 1) {
// Update power levels
UpdateLevel(
&aec->farlevel,
CalculatePower(&farend_extended_block_lowest_band[PART_LEN], PART_LEN));
UpdateLevel(&aec->nearlevel,
CalculatePower(&nearend_block[0][0], PART_LEN));
}
// Convert far-end signal to the frequency domain.
memcpy(fft, farend_extended_block_lowest_band, sizeof(float) * PART_LEN2);
Fft(aec->ooura_fft, fft, farend_fft);
// Form extended nearend frame.
memcpy(&nearend_extended_block_lowest_band[0],
&aec->previous_nearend_block[0][0], sizeof(float) * PART_LEN);
memcpy(&nearend_extended_block_lowest_band[PART_LEN], &nearend_block[0][0],
sizeof(float) * PART_LEN);
// Convert near-end signal to the frequency domain.
memcpy(fft, nearend_extended_block_lowest_band, sizeof(float) * PART_LEN2);
Fft(aec->ooura_fft, fft, nearend_fft);
// Power smoothing.
if (aec->refined_adaptive_filter_enabled) {
for (i = 0; i < PART_LEN1; ++i) {
far_spectrum = farend_fft[0][i] * farend_fft[0][i] +
farend_fft[1][i] * farend_fft[1][i];
// Calculate the magnitude spectrum.
abs_far_spectrum[i] = sqrtf(far_spectrum);
}
RegressorPower(aec->num_partitions, aec->xfBufBlockPos, aec->xfBuf,
aec->xPow);
} else {
for (i = 0; i < PART_LEN1; ++i) {
far_spectrum = farend_fft[0][i] * farend_fft[0][i] +
farend_fft[1][i] * farend_fft[1][i];
aec->xPow[i] =
gPow[0] * aec->xPow[i] + gPow[1] * aec->num_partitions * far_spectrum;
// Calculate the magnitude spectrum.
abs_far_spectrum[i] = sqrtf(far_spectrum);
}
}
for (i = 0; i < PART_LEN1; ++i) {
near_spectrum = nearend_fft[0][i] * nearend_fft[0][i] +
nearend_fft[1][i] * nearend_fft[1][i];
aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] * near_spectrum;
// Calculate the magnitude spectrum.
abs_near_spectrum[i] = sqrtf(near_spectrum);
}
// Estimate noise power. Wait until dPow is more stable.
if (aec->noiseEstCtr > 50) {
for (i = 0; i < PART_LEN1; i++) {
if (aec->dPow[i] < aec->dMinPow[i]) {
aec->dMinPow[i] =
(aec->dPow[i] + step * (aec->dMinPow[i] - aec->dPow[i])) * ramp;
} else {
aec->dMinPow[i] *= ramp;
}
}
}
// Smooth increasing noise power from zero at the start,
// to avoid a sudden burst of comfort noise.
if (aec->noiseEstCtr < noiseInitBlocks) {
aec->noiseEstCtr++;
for (i = 0; i < PART_LEN1; i++) {
if (aec->dMinPow[i] > aec->dInitMinPow[i]) {
aec->dInitMinPow[i] = gInitNoise[0] * aec->dInitMinPow[i] +
gInitNoise[1] * aec->dMinPow[i];
} else {
aec->dInitMinPow[i] = aec->dMinPow[i];
}
}
aec->noisePow = aec->dInitMinPow;
} else {
aec->noisePow = aec->dMinPow;
}
// Block wise delay estimation used for logging
if (aec->delay_logging_enabled) {
if (WebRtc_AddFarSpectrumFloat(aec->delay_estimator_farend,
abs_far_spectrum, PART_LEN1) == 0) {
int delay_estimate = WebRtc_DelayEstimatorProcessFloat(
aec->delay_estimator, abs_near_spectrum, PART_LEN1);
if (delay_estimate >= 0) {
// Update delay estimate buffer.
aec->delay_histogram[delay_estimate]++;
aec->num_delay_values++;
}
if (aec->delay_metrics_delivered == 1 &&
aec->num_delay_values >= kDelayMetricsAggregationWindow) {
UpdateDelayMetrics(aec);
}
}
}
// Perform echo subtraction.
EchoSubtraction(
aec->ooura_fft, aec->num_partitions, aec->extended_filter_enabled,
&aec->extreme_filter_divergence, aec->filter_step_size,
aec->error_threshold, &farend_fft[0][0], &aec->xfBufBlockPos, aec->xfBuf,
&nearend_block[0][0], aec->xPow, aec->wfBuf, echo_subtractor_output);
aec->data_dumper->DumpRaw("aec_h_fft", PART_LEN1 * aec->num_partitions,
&aec->wfBuf[0][0]);
aec->data_dumper->DumpRaw("aec_h_fft", PART_LEN1 * aec->num_partitions,
&aec->wfBuf[1][0]);
aec->data_dumper->DumpWav("aec_out_linear", PART_LEN, echo_subtractor_output,
std::min(aec->sampFreq, 16000), 1);
if (aec->metricsMode == 1) {
UpdateLevel(&aec->linoutlevel,
CalculatePower(echo_subtractor_output, PART_LEN));
}
// Perform echo suppression.
EchoSuppression(aec->ooura_fft, aec, nearend_extended_block_lowest_band,
farend_extended_block_lowest_band, echo_subtractor_output,
output_block);
if (aec->metricsMode == 1) {
UpdateLevel(&aec->nlpoutlevel,
CalculatePower(&output_block[0][0], PART_LEN));
UpdateMetrics(aec);
}
// Store the nearend signal until the next frame.
for (i = 0; i < aec->num_bands; ++i) {
memcpy(&aec->previous_nearend_block[i][0], &nearend_block[i][0],
sizeof(float) * PART_LEN);
}
aec->data_dumper->DumpWav("aec_out", PART_LEN, &output_block[0][0],
std::min(aec->sampFreq, 16000), 1);
}
AecCore* WebRtcAec_CreateAec(int instance_count) {
AecCore* aec = new AecCore(instance_count);
if (!aec) {
return NULL;
}
aec->nearend_buffer_size = 0;
memset(&aec->nearend_buffer[0], 0, sizeof(aec->nearend_buffer));
// Start the output buffer with zeros to be able to produce
// a full output frame in the first frame.
aec->output_buffer_size = PART_LEN - (FRAME_LEN - PART_LEN);
memset(&aec->output_buffer[0], 0, sizeof(aec->output_buffer));
aec->delay_estimator_farend =
WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
if (aec->delay_estimator_farend == NULL) {
WebRtcAec_FreeAec(aec);
return NULL;
}
// We create the delay_estimator with the same amount of maximum lookahead as
// the delay history size (kHistorySizeBlocks) for symmetry reasons.
aec->delay_estimator = WebRtc_CreateDelayEstimator(
aec->delay_estimator_farend, kHistorySizeBlocks);
if (aec->delay_estimator == NULL) {
WebRtcAec_FreeAec(aec);
return NULL;
}
#ifdef WEBRTC_ANDROID
aec->delay_agnostic_enabled = 1; // DA-AEC enabled by default.
// DA-AEC assumes the system is causal from the beginning and will self adjust
// the lookahead when shifting is required.
WebRtc_set_lookahead(aec->delay_estimator, 0);
#else
aec->delay_agnostic_enabled = 0;
WebRtc_set_lookahead(aec->delay_estimator, kLookaheadBlocks);
#endif
aec->extended_filter_enabled = 0;
aec->refined_adaptive_filter_enabled = false;
// Assembly optimization
WebRtcAec_FilterFar = FilterFar;
WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
WebRtcAec_FilterAdaptation = FilterAdaptation;
WebRtcAec_Overdrive = Overdrive;
WebRtcAec_Suppress = Suppress;
WebRtcAec_ComputeCoherence = ComputeCoherence;
WebRtcAec_UpdateCoherenceSpectra = UpdateCoherenceSpectra;
WebRtcAec_StoreAsComplex = StoreAsComplex;
WebRtcAec_PartitionDelay = PartitionDelay;
WebRtcAec_WindowData = WindowData;
#if defined(WEBRTC_ARCH_X86_FAMILY)
if (WebRtc_GetCPUInfo(kSSE2)) {
WebRtcAec_InitAec_SSE2();
}
#endif
#if defined(MIPS_FPU_LE)
WebRtcAec_InitAec_mips();
#endif
#if defined(WEBRTC_HAS_NEON)
WebRtcAec_InitAec_neon();
#endif
return aec;
}
void WebRtcAec_FreeAec(AecCore* aec) {
if (aec == NULL) {
return;
}
WebRtc_FreeDelayEstimator(aec->delay_estimator);
WebRtc_FreeDelayEstimatorFarend(aec->delay_estimator_farend);
delete aec;
}
static void SetAdaptiveFilterStepSize(AecCore* aec) {
// Extended filter adaptation parameter.
// TODO(ajm): No narrowband tuning yet.
const float kExtendedMu = 0.4f;
if (aec->refined_adaptive_filter_enabled) {
aec->filter_step_size = 0.05f;
} else {
if (aec->extended_filter_enabled) {
aec->filter_step_size = kExtendedMu;
} else {
if (aec->sampFreq == 8000) {
aec->filter_step_size = 0.6f;
} else {
aec->filter_step_size = 0.5f;
}
}
}
}
static void SetErrorThreshold(AecCore* aec) {
// Extended filter adaptation parameter.
// TODO(ajm): No narrowband tuning yet.
static const float kExtendedErrorThreshold = 1.0e-6f;
if (aec->extended_filter_enabled) {
aec->error_threshold = kExtendedErrorThreshold;
} else {
if (aec->sampFreq == 8000) {
aec->error_threshold = 2e-6f;
} else {
aec->error_threshold = 1.5e-6f;
}
}
}
int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
int i;
aec->data_dumper->InitiateNewSetOfRecordings();
aec->sampFreq = sampFreq;
SetAdaptiveFilterStepSize(aec);
SetErrorThreshold(aec);
if (sampFreq == 8000) {
aec->num_bands = 1;
} else {
aec->num_bands = (size_t)(sampFreq / 16000);
}
// Start the output buffer with zeros to be able to produce
// a full output frame in the first frame.
aec->output_buffer_size = PART_LEN - (FRAME_LEN - PART_LEN);
memset(&aec->output_buffer[0], 0, sizeof(aec->output_buffer));
aec->nearend_buffer_size = 0;
memset(&aec->nearend_buffer[0], 0, sizeof(aec->nearend_buffer));
// Initialize far-end buffer.
aec->farend_block_buffer_.ReInit();
aec->system_delay = 0;
if (WebRtc_InitDelayEstimatorFarend(aec->delay_estimator_farend) != 0) {
return -1;
}
if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
return -1;
}
aec->delay_logging_enabled = 0;
aec->delay_metrics_delivered = 0;
memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
aec->num_delay_values = 0;
aec->delay_median = -1;
aec->delay_std = -1;
aec->fraction_poor_delays = -1.0f;
aec->previous_delay = -2; // (-2): Uninitialized.
aec->delay_correction_count = 0;
aec->shift_offset = kInitialShiftOffset;
aec->delay_quality_threshold = kDelayQualityThresholdMin;
aec->num_partitions = kNormalNumPartitions;
// Update the delay estimator with filter length. We use half the
// |num_partitions| to take the echo path into account. In practice we say
// that the echo has a duration of maximum half |num_partitions|, which is not
// true, but serves as a crude measure.
WebRtc_set_allowed_offset(aec->delay_estimator, aec->num_partitions / 2);
// TODO(bjornv): I currently hard coded the enable. Once we've established
// that AECM has no performance regression, robust_validation will be enabled
// all the time and the APIs to turn it on/off will be removed. Hence, remove
// this line then.
WebRtc_enable_robust_validation(aec->delay_estimator, 1);
aec->frame_count = 0;
// Default target suppression mode.
aec->nlp_mode = 1;
// Sampling frequency multiplier w.r.t. 8 kHz.
// In case of multiple bands we process the lower band in 16 kHz, hence the
// multiplier is always 2.
if (aec->num_bands > 1) {
aec->mult = 2;
} else {
aec->mult = static_cast<int16_t>(aec->sampFreq) / 8000;
}
aec->farBufWritePos = 0;
aec->farBufReadPos = 0;
aec->inSamples = 0;
aec->outSamples = 0;
aec->knownDelay = 0;
// Initialize buffers
memset(aec->previous_nearend_block, 0, sizeof(aec->previous_nearend_block));
memset(aec->eBuf, 0, sizeof(aec->eBuf));
memset(aec->xPow, 0, sizeof(aec->xPow));
memset(aec->dPow, 0, sizeof(aec->dPow));
memset(aec->dInitMinPow, 0, sizeof(aec->dInitMinPow));
aec->noisePow = aec->dInitMinPow;
aec->noiseEstCtr = 0;
// Initial comfort noise power
for (i = 0; i < PART_LEN1; i++) {
aec->dMinPow[i] = 1.0e6f;
}
// Holds the last block written to
aec->xfBufBlockPos = 0;
// TODO(peah): Investigate need for these initializations. Deleting them
// doesn't change the output at all and yields 0.4% overall speedup.
memset(aec->xfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
memset(aec->wfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
memset(aec->coherence_state.sde, 0, sizeof(complex_t) * PART_LEN1);
memset(aec->coherence_state.sxd, 0, sizeof(complex_t) * PART_LEN1);
memset(aec->xfwBuf, 0,
sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
memset(aec->coherence_state.se, 0, sizeof(float) * PART_LEN1);
// To prevent numerical instability in the first block.
for (i = 0; i < PART_LEN1; i++) {
aec->coherence_state.sd[i] = 1;
}
for (i = 0; i < PART_LEN1; i++) {
aec->coherence_state.sx[i] = 1;
}
memset(aec->hNs, 0, sizeof(aec->hNs));
memset(aec->outBuf, 0, sizeof(float) * PART_LEN);
aec->hNlFbMin = 1;
aec->hNlFbLocalMin = 1;
aec->hNlXdAvgMin = 1;
aec->hNlNewMin = 0;
aec->hNlMinCtr = 0;
aec->overDrive = 2;
aec->overdrive_scaling = 2;
aec->delayIdx = 0;
aec->stNearState = 0;
aec->echoState = 0;
aec->divergeState = 0;
aec->seed = 777;
aec->delayEstCtr = 0;
aec->extreme_filter_divergence = 0;
// Metrics disabled by default
aec->metricsMode = 0;
InitMetrics(aec);
return 0;
}
void WebRtcAec_BufferFarendBlock(AecCore* aec, const float* farend) {
// Check if the buffer is full, and in that case flush the oldest data.
if (aec->farend_block_buffer_.AvaliableSpace() < 1) {
aec->farend_block_buffer_.AdjustSize(1);
}
aec->farend_block_buffer_.Insert(farend);
}
int WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(AecCore* aec,
int buffer_size_decrease) {
int achieved_buffer_size_decrease =
aec->farend_block_buffer_.AdjustSize(buffer_size_decrease);
aec->system_delay -= achieved_buffer_size_decrease * PART_LEN;
return achieved_buffer_size_decrease;
}
void FormNearendBlock(
size_t nearend_start_index,
size_t num_bands,
const float* const* nearend_frame,
size_t num_samples_from_nearend_frame,
const float nearend_buffer[NUM_HIGH_BANDS_MAX + 1]
[PART_LEN - (FRAME_LEN - PART_LEN)],
float nearend_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN]) {
RTC_DCHECK_LE(num_samples_from_nearend_frame, PART_LEN);
const int num_samples_from_buffer = PART_LEN - num_samples_from_nearend_frame;
if (num_samples_from_buffer > 0) {
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&nearend_block[i][0], &nearend_buffer[i][0],
num_samples_from_buffer * sizeof(float));
}
}
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&nearend_block[i][num_samples_from_buffer],
&nearend_frame[i][nearend_start_index],
num_samples_from_nearend_frame * sizeof(float));
}
}
void BufferNearendFrame(
size_t nearend_start_index,
size_t num_bands,
const float* const* nearend_frame,
size_t num_samples_to_buffer,
float nearend_buffer[NUM_HIGH_BANDS_MAX + 1]
[PART_LEN - (FRAME_LEN - PART_LEN)]) {
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&nearend_buffer[i][0],
&nearend_frame[i][nearend_start_index + FRAME_LEN -
num_samples_to_buffer],
num_samples_to_buffer * sizeof(float));
}
}
void BufferOutputBlock(
size_t num_bands,
const float output_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN],
size_t* output_buffer_size,
float output_buffer[NUM_HIGH_BANDS_MAX + 1][2 * PART_LEN]) {
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&output_buffer[i][*output_buffer_size], &output_block[i][0],
PART_LEN * sizeof(float));
}
(*output_buffer_size) += PART_LEN;
}
void FormOutputFrame(size_t output_start_index,
size_t num_bands,
size_t* output_buffer_size,
float output_buffer[NUM_HIGH_BANDS_MAX + 1][2 * PART_LEN],
float* const* output_frame) {
RTC_DCHECK_LE(FRAME_LEN, *output_buffer_size);
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&output_frame[i][output_start_index], &output_buffer[i][0],
FRAME_LEN * sizeof(float));
}
(*output_buffer_size) -= FRAME_LEN;
if (*output_buffer_size > 0) {
RTC_DCHECK_GE(2 * PART_LEN - FRAME_LEN, (*output_buffer_size));
for (size_t i = 0; i < num_bands; ++i) {
memcpy(&output_buffer[i][0], &output_buffer[i][FRAME_LEN],
(*output_buffer_size) * sizeof(float));
}
}
}
void WebRtcAec_ProcessFrames(AecCore* aec,
const float* const* nearend,
size_t num_bands,
size_t num_samples,
int knownDelay,
float* const* out) {
RTC_DCHECK(num_samples == 80 || num_samples == 160);
aec->frame_count++;
// For each frame the process is as follows:
// 1) If the system_delay indicates on being too small for processing a
// frame we stuff the buffer with enough data for 10 ms.
// 2 a) Adjust the buffer to the system delay, by moving the read pointer.
// b) Apply signal based delay correction, if we have detected poor AEC
// performance.
// 3) TODO(bjornv): Investigate if we need to add this:
// If we can't move read pointer due to buffer size limitations we
// flush/stuff the buffer.
// 4) Process as many partitions as possible.
// 5) Update the |system_delay| with respect to a full frame of FRAME_LEN
// samples. Even though we will have data left to process (we work with
// partitions) we consider updating a whole frame, since that's the
// amount of data we input and output in audio_processing.
// 6) Update the outputs.
// The AEC has two different delay estimation algorithms built in. The
// first relies on delay input values from the user and the amount of
// shifted buffer elements is controlled by |knownDelay|. This delay will
// give a guess on how much we need to shift far-end buffers to align with
// the near-end signal. The other delay estimation algorithm uses the
// far- and near-end signals to find the offset between them. This one
// (called "signal delay") is then used to fine tune the alignment, or
// simply compensate for errors in the system based one.
// Note that the two algorithms operate independently. Currently, we only
// allow one algorithm to be turned on.
RTC_DCHECK_EQ(aec->num_bands, num_bands);
for (size_t j = 0; j < num_samples; j += FRAME_LEN) {
// 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
// have enough far-end data for that by stuffing the buffer if the
// |system_delay| indicates others.
if (aec->system_delay < FRAME_LEN) {
// We don't have enough data so we rewind 10 ms.
WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aec, -(aec->mult + 1));
}
if (!aec->delay_agnostic_enabled) {
// 2 a) Compensate for a possible change in the system delay.
// TODO(bjornv): Investigate how we should round the delay difference;
// right now we know that incoming |knownDelay| is underestimated when
// it's less than |aec->knownDelay|. We therefore, round (-32) in that
// direction. In the other direction, we don't have this situation, but
// might flush one partition too little. This can cause non-causality,
// which should be investigated. Maybe, allow for a non-symmetric
// rounding, like -16.
int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
int moved_elements = aec->farend_block_buffer_.AdjustSize(move_elements);
aec->knownDelay -= moved_elements * PART_LEN;
} else {
// 2 b) Apply signal based delay correction.
int move_elements = SignalBasedDelayCorrection(aec);
int moved_elements = aec->farend_block_buffer_.AdjustSize(move_elements);
int far_near_buffer_diff =
aec->farend_block_buffer_.Size() -
(aec->nearend_buffer_size + FRAME_LEN) / PART_LEN;
WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
moved_elements);
// If we rely on reported system delay values only, a buffer underrun here
// can never occur since we've taken care of that in 1) above. Here, we
// apply signal based delay correction and can therefore end up with
// buffer underruns since the delay estimation can be wrong. We therefore
// stuff the buffer with enough elements if needed.
if (far_near_buffer_diff < 0) {
WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aec,
far_near_buffer_diff);
}
}
static_assert(
16 == (FRAME_LEN - PART_LEN),
"These constants need to be properly related for this code to work");
float output_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN];
float nearend_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN];
float farend_extended_block_lowest_band[PART_LEN2];
// Form and process a block of nearend samples, buffer the output block of
// samples.
aec->farend_block_buffer_.ExtractExtendedBlock(
farend_extended_block_lowest_band);
FormNearendBlock(j, num_bands, nearend, PART_LEN - aec->nearend_buffer_size,
aec->nearend_buffer, nearend_block);
ProcessNearendBlock(aec, farend_extended_block_lowest_band, nearend_block,
output_block);
BufferOutputBlock(num_bands, output_block, &aec->output_buffer_size,
aec->output_buffer);
if ((FRAME_LEN - PART_LEN + aec->nearend_buffer_size) == PART_LEN) {
// When possible (every fourth frame) form and process a second block of
// nearend samples, buffer the output block of samples.
aec->farend_block_buffer_.ExtractExtendedBlock(
farend_extended_block_lowest_band);
FormNearendBlock(j + FRAME_LEN - PART_LEN, num_bands, nearend, PART_LEN,
aec->nearend_buffer, nearend_block);
ProcessNearendBlock(aec, farend_extended_block_lowest_band, nearend_block,
output_block);
BufferOutputBlock(num_bands, output_block, &aec->output_buffer_size,
aec->output_buffer);
// Reset the buffer size as there are no samples left in the nearend input
// to buffer.
aec->nearend_buffer_size = 0;
} else {
// Buffer the remaining samples in the nearend input.
aec->nearend_buffer_size += FRAME_LEN - PART_LEN;
BufferNearendFrame(j, num_bands, nearend, aec->nearend_buffer_size,
aec->nearend_buffer);
}
// 5) Update system delay with respect to the entire frame.
aec->system_delay -= FRAME_LEN;
// 6) Form the output frame.
FormOutputFrame(j, num_bands, &aec->output_buffer_size, aec->output_buffer,
out);
}
}
int WebRtcAec_GetDelayMetricsCore(AecCore* self,
int* median,
int* std,
float* fraction_poor_delays) {
RTC_DCHECK(self);
RTC_DCHECK(median);
RTC_DCHECK(std);
if (self->delay_logging_enabled == 0) {
// Logging disabled.
return -1;
}
if (self->delay_metrics_delivered == 0) {
UpdateDelayMetrics(self);
self->delay_metrics_delivered = 1;
}
*median = self->delay_median;
*std = self->delay_std;
*fraction_poor_delays = self->fraction_poor_delays;
return 0;
}
int WebRtcAec_echo_state(AecCore* self) {
return self->echoState;
}
void WebRtcAec_GetEchoStats(AecCore* self,
Stats* erl,
Stats* erle,
Stats* a_nlp,
float* divergent_filter_fraction) {
RTC_DCHECK(erl);
RTC_DCHECK(erle);
RTC_DCHECK(a_nlp);
*erl = self->erl;
*erle = self->erle;
*a_nlp = self->aNlp;
*divergent_filter_fraction =
self->divergent_filter_fraction.GetLatestFraction();
}
void WebRtcAec_SetConfigCore(AecCore* self,
int nlp_mode,
int metrics_mode,
int delay_logging) {
RTC_DCHECK_GE(nlp_mode, 0);
RTC_DCHECK_LT(nlp_mode, 3);
self->nlp_mode = nlp_mode;
self->metricsMode = metrics_mode;
if (self->metricsMode) {
InitMetrics(self);
}
// Turn on delay logging if it is either set explicitly or if delay agnostic
// AEC is enabled (which requires delay estimates).
self->delay_logging_enabled = delay_logging || self->delay_agnostic_enabled;
if (self->delay_logging_enabled) {
memset(self->delay_histogram, 0, sizeof(self->delay_histogram));
}
}
void WebRtcAec_enable_delay_agnostic(AecCore* self, int enable) {
self->delay_agnostic_enabled = enable;
}
int WebRtcAec_delay_agnostic_enabled(AecCore* self) {
return self->delay_agnostic_enabled;
}
void WebRtcAec_enable_refined_adaptive_filter(AecCore* self, bool enable) {
self->refined_adaptive_filter_enabled = enable;
SetAdaptiveFilterStepSize(self);
SetErrorThreshold(self);
}
bool WebRtcAec_refined_adaptive_filter_enabled(const AecCore* self) {
return self->refined_adaptive_filter_enabled;
}
void WebRtcAec_enable_extended_filter(AecCore* self, int enable) {
self->extended_filter_enabled = enable;
SetAdaptiveFilterStepSize(self);
SetErrorThreshold(self);
self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
// Update the delay estimator with filter length. See InitAEC() for details.
WebRtc_set_allowed_offset(self->delay_estimator, self->num_partitions / 2);
}
int WebRtcAec_extended_filter_enabled(AecCore* self) {
return self->extended_filter_enabled;
}
int WebRtcAec_system_delay(AecCore* self) {
return self->system_delay;
}
void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
RTC_DCHECK_GE(delay, 0);
self->system_delay = delay;
}
} // namespace webrtc