
Note: estimation is turned OFF if config_.ep_strength.default_len is set >= 0 (in this case config_.ep_strength.default_len defines a constant echo decay factor), and hence turned ON if < 0. In case the echo tail estimation is turned ON, -config_.ep_strength.default_len is the starting point for the estimator. The estimation is done in two passes; first we go through all "sections" (corresponding to chunks of length kFftLengthBy2) of the filter impulse response to determine which sections correspond to a "stable" decay", and then the second pass we go through each stable decay section and estimate the decay. The actual decay estimation is based on linear regression of the log magnitude of the squared impulse response. A bunch of sanity checks are also performed continuously to avoid estimation error during e.g., filter adaptation. Bug: webrtc:8924 Change-Id: I686ce3f3e8b6b472348f8d6e01fb44c31e25145d Reviewed-on: https://webrtc-review.googlesource.com/48440 Commit-Queue: Christian Schuldt <cschuldt@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22247}
467 lines
17 KiB
C++
467 lines
17 KiB
C++
/*
|
|
* Copyright (c) 2017 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/aec3/aec_state.h"
|
|
|
|
#include <math.h>
|
|
|
|
#include <numeric>
|
|
#include <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "rtc_base/atomicops.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
// Computes delay of the adaptive filter.
|
|
int EstimateFilterDelay(
|
|
const std::vector<std::array<float, kFftLengthBy2Plus1>>&
|
|
adaptive_filter_frequency_response) {
|
|
const auto& H2 = adaptive_filter_frequency_response;
|
|
constexpr size_t kUpperBin = kFftLengthBy2 - 5;
|
|
RTC_DCHECK_GE(kMaxAdaptiveFilterLength, H2.size());
|
|
std::array<int, kMaxAdaptiveFilterLength> delays;
|
|
delays.fill(0);
|
|
for (size_t k = 1; k < kUpperBin; ++k) {
|
|
// Find the maximum of H2[j].
|
|
size_t peak = 0;
|
|
for (size_t j = 0; j < H2.size(); ++j) {
|
|
if (H2[j][k] > H2[peak][k]) {
|
|
peak = j;
|
|
}
|
|
}
|
|
++delays[peak];
|
|
}
|
|
|
|
return std::distance(delays.begin(),
|
|
std::max_element(delays.begin(), delays.end()));
|
|
}
|
|
|
|
float ComputeGainRampupIncrease(const EchoCanceller3Config& config) {
|
|
const auto& c = config.echo_removal_control.gain_rampup;
|
|
return powf(1.f / c.first_non_zero_gain, 1.f / c.non_zero_gain_blocks);
|
|
}
|
|
|
|
} // namespace
|
|
|
|
int AecState::instance_count_ = 0;
|
|
|
|
AecState::AecState(const EchoCanceller3Config& config)
|
|
: data_dumper_(
|
|
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
|
|
erle_estimator_(config.erle.min, config.erle.max_l, config.erle.max_h),
|
|
config_(config),
|
|
max_render_(config_.filter.main.length_blocks, 0.f),
|
|
reverb_decay_(fabsf(config_.ep_strength.default_len)),
|
|
gain_rampup_increase_(ComputeGainRampupIncrease(config_)) {}
|
|
|
|
AecState::~AecState() = default;
|
|
|
|
void AecState::HandleEchoPathChange(
|
|
const EchoPathVariability& echo_path_variability) {
|
|
const auto full_reset = [&]() {
|
|
blocks_since_last_saturation_ = 0;
|
|
usable_linear_estimate_ = false;
|
|
echo_leakage_detected_ = false;
|
|
capture_signal_saturation_ = false;
|
|
echo_saturation_ = false;
|
|
previous_max_sample_ = 0.f;
|
|
std::fill(max_render_.begin(), max_render_.end(), 0.f);
|
|
blocks_with_proper_filter_adaptation_ = 0;
|
|
capture_block_counter_ = 0;
|
|
filter_has_had_time_to_converge_ = false;
|
|
render_received_ = false;
|
|
blocks_with_active_render_ = 0;
|
|
initial_state_ = true;
|
|
};
|
|
|
|
// TODO(peah): Refine the reset scheme according to the type of gain and
|
|
// delay adjustment.
|
|
if (echo_path_variability.gain_change) {
|
|
full_reset();
|
|
}
|
|
|
|
if (echo_path_variability.delay_change !=
|
|
EchoPathVariability::DelayAdjustment::kBufferReadjustment) {
|
|
full_reset();
|
|
} else if (echo_path_variability.delay_change !=
|
|
EchoPathVariability::DelayAdjustment::kBufferFlush) {
|
|
active_render_seen_ = false;
|
|
full_reset();
|
|
} else if (echo_path_variability.delay_change !=
|
|
EchoPathVariability::DelayAdjustment::kDelayReset) {
|
|
full_reset();
|
|
} else if (echo_path_variability.delay_change !=
|
|
EchoPathVariability::DelayAdjustment::kNewDetectedDelay) {
|
|
full_reset();
|
|
} else if (echo_path_variability.gain_change) {
|
|
capture_block_counter_ = kNumBlocksPerSecond;
|
|
}
|
|
}
|
|
|
|
void AecState::Update(
|
|
const rtc::Optional<DelayEstimate>& delay_estimate,
|
|
const std::vector<std::array<float, kFftLengthBy2Plus1>>&
|
|
adaptive_filter_frequency_response,
|
|
const std::vector<float>& adaptive_filter_impulse_response,
|
|
bool converged_filter,
|
|
const RenderBuffer& render_buffer,
|
|
const std::array<float, kFftLengthBy2Plus1>& E2_main,
|
|
const std::array<float, kFftLengthBy2Plus1>& Y2,
|
|
const std::array<float, kBlockSize>& s,
|
|
bool echo_leakage_detected) {
|
|
// Store input parameters.
|
|
echo_leakage_detected_ = echo_leakage_detected;
|
|
|
|
// Estimate the filter delay.
|
|
filter_delay_ = EstimateFilterDelay(adaptive_filter_frequency_response);
|
|
const std::vector<float>& x = render_buffer.Block(-filter_delay_)[0];
|
|
|
|
// Update counters.
|
|
++capture_block_counter_;
|
|
const bool active_render_block = DetectActiveRender(x);
|
|
blocks_with_active_render_ += active_render_block ? 1 : 0;
|
|
blocks_with_proper_filter_adaptation_ +=
|
|
active_render_block && !SaturatedCapture() ? 1 : 0;
|
|
|
|
// Update the limit on the echo suppression after an echo path change to avoid
|
|
// an initial echo burst.
|
|
UpdateSuppressorGainLimit(render_buffer.GetRenderActivity());
|
|
|
|
// Update the ERL and ERLE measures.
|
|
if (converged_filter && capture_block_counter_ >= 2 * kNumBlocksPerSecond) {
|
|
const auto& X2 = render_buffer.Spectrum(filter_delay_);
|
|
erle_estimator_.Update(X2, Y2, E2_main);
|
|
erl_estimator_.Update(X2, Y2);
|
|
}
|
|
|
|
// Update the echo audibility evaluator.
|
|
echo_audibility_.Update(x, s, converged_filter);
|
|
|
|
// Detect and flag echo saturation.
|
|
// TODO(peah): Add the delay in this computation to ensure that the render and
|
|
// capture signals are properly aligned.
|
|
if (config_.ep_strength.echo_can_saturate) {
|
|
echo_saturation_ = DetectEchoSaturation(x);
|
|
}
|
|
|
|
// TODO(peah): Move?
|
|
filter_has_had_time_to_converge_ =
|
|
blocks_with_proper_filter_adaptation_ >= 1.5f * kNumBlocksPerSecond;
|
|
|
|
initial_state_ =
|
|
blocks_with_proper_filter_adaptation_ < 5 * kNumBlocksPerSecond;
|
|
|
|
// Flag whether the linear filter estimate is usable.
|
|
usable_linear_estimate_ =
|
|
!echo_saturation_ &&
|
|
(converged_filter && filter_has_had_time_to_converge_) &&
|
|
capture_block_counter_ >= 1.f * kNumBlocksPerSecond && !TransparentMode();
|
|
|
|
// After an amount of active render samples for which an echo should have been
|
|
// detected in the capture signal if the ERL was not infinite, flag that a
|
|
// transparent mode should be entered.
|
|
transparent_mode_ =
|
|
!converged_filter &&
|
|
(blocks_with_active_render_ == 0 ||
|
|
blocks_with_proper_filter_adaptation_ >= 5 * kNumBlocksPerSecond);
|
|
}
|
|
|
|
void AecState::UpdateReverb(const std::vector<float>& impulse_response) {
|
|
// Echo tail estimation enabled if the below variable is set as negative.
|
|
if (config_.ep_strength.default_len > 0.f) {
|
|
return;
|
|
}
|
|
|
|
if ((!(filter_delay_ && usable_linear_estimate_)) ||
|
|
(filter_delay_ >
|
|
static_cast<int>(config_.filter.main.length_blocks) - 4)) {
|
|
return;
|
|
}
|
|
|
|
constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
|
|
|
|
// Form the data to match against by squaring the impulse response
|
|
// coefficients.
|
|
std::array<float, GetTimeDomainLength(kMaxAdaptiveFilterLength)>
|
|
matching_data_data;
|
|
RTC_DCHECK_LE(GetTimeDomainLength(config_.filter.main.length_blocks),
|
|
matching_data_data.size());
|
|
rtc::ArrayView<float> matching_data(
|
|
matching_data_data.data(),
|
|
GetTimeDomainLength(config_.filter.main.length_blocks));
|
|
std::transform(impulse_response.begin(), impulse_response.end(),
|
|
matching_data.begin(), [](float a) { return a * a; });
|
|
|
|
if (current_reverb_decay_section_ < config_.filter.main.length_blocks) {
|
|
// Update accumulated variables for the current filter section.
|
|
|
|
const size_t start_index = current_reverb_decay_section_ * kFftLengthBy2;
|
|
|
|
RTC_DCHECK_GT(matching_data.size(), start_index);
|
|
RTC_DCHECK_GE(matching_data.size(), start_index + kFftLengthBy2);
|
|
float section_energy =
|
|
std::accumulate(matching_data.begin() + start_index,
|
|
matching_data.begin() + start_index + kFftLengthBy2,
|
|
0.f) *
|
|
kOneByFftLengthBy2;
|
|
|
|
section_energy = std::max(
|
|
section_energy, 1e-32f); // Regularization to avoid division by 0.
|
|
|
|
RTC_DCHECK_LT(current_reverb_decay_section_, block_energies_.size());
|
|
const float energy_ratio =
|
|
block_energies_[current_reverb_decay_section_] / section_energy;
|
|
|
|
main_filter_is_adapting_ = main_filter_is_adapting_ ||
|
|
(energy_ratio > 1.1f || energy_ratio < 0.9f);
|
|
|
|
// Count consecutive number of "good" filter sections, where "good" means:
|
|
// 1) energy is above noise floor.
|
|
// 2) energy of current section has not changed too much from last check.
|
|
if (!found_end_of_reverb_decay_ && section_energy > tail_energy_ &&
|
|
!main_filter_is_adapting_) {
|
|
++num_reverb_decay_sections_next_;
|
|
} else {
|
|
found_end_of_reverb_decay_ = true;
|
|
}
|
|
|
|
block_energies_[current_reverb_decay_section_] = section_energy;
|
|
|
|
if (num_reverb_decay_sections_ > 0) {
|
|
// Linear regression of log squared magnitude of impulse response.
|
|
for (size_t i = 0; i < kFftLengthBy2; i++) {
|
|
auto fast_approx_log2f = [](const float in) {
|
|
RTC_DCHECK_GT(in, .0f);
|
|
// Read and interpret float as uint32_t and then cast to float.
|
|
// This is done to extract the exponent (bits 30 - 23).
|
|
// "Right shift" of the exponent is then performed by multiplying
|
|
// with the constant (1/2^23). Finally, we subtract a constant to
|
|
// remove the bias (https://en.wikipedia.org/wiki/Exponent_bias).
|
|
union {
|
|
float dummy;
|
|
uint32_t a;
|
|
} x = {in};
|
|
float out = x.a;
|
|
out *= 1.1920929e-7f; // 1/2^23
|
|
out -= 126.942695f; // Remove bias.
|
|
return out;
|
|
};
|
|
RTC_DCHECK_GT(matching_data.size(), start_index + i);
|
|
float z = fast_approx_log2f(matching_data[start_index + i]);
|
|
accumulated_nz_ += accumulated_count_ * z;
|
|
++accumulated_count_;
|
|
}
|
|
}
|
|
|
|
num_reverb_decay_sections_ =
|
|
num_reverb_decay_sections_ > 0 ? num_reverb_decay_sections_ - 1 : 0;
|
|
++current_reverb_decay_section_;
|
|
|
|
} else {
|
|
constexpr float kMaxDecay = 0.95f; // ~1 sec min RT60.
|
|
constexpr float kMinDecay = 0.02f; // ~15 ms max RT60.
|
|
|
|
// Accumulated variables throughout whole filter.
|
|
|
|
// Solve for decay rate.
|
|
|
|
float decay = reverb_decay_;
|
|
|
|
if (accumulated_nn_ != 0.f) {
|
|
const float exp_candidate = -accumulated_nz_ / accumulated_nn_;
|
|
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
|
|
decay = std::min(decay, kMaxDecay);
|
|
decay = std::max(decay, kMinDecay);
|
|
}
|
|
|
|
// Filter tail energy (assumed to be noise).
|
|
|
|
constexpr size_t kTailLength = kFftLength;
|
|
constexpr float k1ByTailLength = 1.f / kTailLength;
|
|
const size_t tail_index =
|
|
GetTimeDomainLength(config_.filter.main.length_blocks) - kTailLength;
|
|
|
|
RTC_DCHECK_GT(matching_data.size(), tail_index);
|
|
tail_energy_ = std::accumulate(matching_data.begin() + tail_index,
|
|
matching_data.end(), 0.f) *
|
|
k1ByTailLength;
|
|
|
|
// Update length of decay.
|
|
num_reverb_decay_sections_ = num_reverb_decay_sections_next_;
|
|
num_reverb_decay_sections_next_ = 0;
|
|
// Must have enough data (number of sections) in order
|
|
// to estimate decay rate.
|
|
if (num_reverb_decay_sections_ < 5) {
|
|
num_reverb_decay_sections_ = 0;
|
|
}
|
|
|
|
const float N = num_reverb_decay_sections_ * kFftLengthBy2;
|
|
accumulated_nz_ = 0.f;
|
|
const float k1By12 = 1.f / 12.f;
|
|
// Arithmetic sum $2 \sum_{i=0}^{(N-1)/2}i^2$ calculated directly.
|
|
accumulated_nn_ = N * (N * N - 1.0f) * k1By12;
|
|
accumulated_count_ = -N * 0.5f;
|
|
// Linear regression approach assumes symmetric index around 0.
|
|
accumulated_count_ += 0.5f;
|
|
|
|
// Identify the peak index of the impulse response.
|
|
const size_t peak_index = std::distance(
|
|
matching_data.begin(),
|
|
std::max_element(matching_data.begin(), matching_data.end()));
|
|
|
|
current_reverb_decay_section_ = peak_index * kOneByFftLengthBy2 + 3;
|
|
// Make sure we're not out of bounds.
|
|
if (current_reverb_decay_section_ + 1 >=
|
|
config_.filter.main.length_blocks) {
|
|
current_reverb_decay_section_ = config_.filter.main.length_blocks;
|
|
}
|
|
size_t start_index = current_reverb_decay_section_ * kFftLengthBy2;
|
|
float first_section_energy =
|
|
std::accumulate(matching_data.begin() + start_index,
|
|
matching_data.begin() + start_index + kFftLengthBy2,
|
|
0.f) *
|
|
kOneByFftLengthBy2;
|
|
|
|
// To estimate the reverb decay, the energy of the first filter section
|
|
// must be substantially larger than the last.
|
|
// Also, the first filter section energy must not deviate too much
|
|
// from the max peak.
|
|
bool main_filter_has_reverb = first_section_energy > 4.f * tail_energy_;
|
|
bool main_filter_is_sane = first_section_energy > 2.f * tail_energy_ &&
|
|
matching_data[peak_index] < 100.f;
|
|
|
|
// Not detecting any decay, but tail is over noise - assume max decay.
|
|
if (num_reverb_decay_sections_ == 0 && main_filter_is_sane &&
|
|
main_filter_has_reverb) {
|
|
decay = kMaxDecay;
|
|
}
|
|
|
|
if (!main_filter_is_adapting_ && main_filter_is_sane &&
|
|
num_reverb_decay_sections_ > 0) {
|
|
decay = std::max(.97f * reverb_decay_, decay);
|
|
reverb_decay_ -= .1f * (reverb_decay_ - decay);
|
|
}
|
|
|
|
found_end_of_reverb_decay_ =
|
|
!(main_filter_is_sane && main_filter_has_reverb);
|
|
main_filter_is_adapting_ = false;
|
|
}
|
|
|
|
data_dumper_->DumpRaw("aec3_reverb_decay", reverb_decay_);
|
|
data_dumper_->DumpRaw("aec3_reverb_tail_energy", tail_energy_);
|
|
}
|
|
|
|
bool AecState::DetectActiveRender(rtc::ArrayView<const float> x) const {
|
|
const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
|
|
return x_energy > (config_.render_levels.active_render_limit *
|
|
config_.render_levels.active_render_limit) *
|
|
kFftLengthBy2;
|
|
}
|
|
|
|
// Updates the suppressor gain limit.
|
|
void AecState::UpdateSuppressorGainLimit(bool render_activity) {
|
|
const auto& rampup_conf = config_.echo_removal_control.gain_rampup;
|
|
if (!active_render_seen_ && render_activity) {
|
|
active_render_seen_ = true;
|
|
realignment_counter_ = rampup_conf.full_gain_blocks;
|
|
} else if (realignment_counter_ > 0) {
|
|
--realignment_counter_;
|
|
}
|
|
|
|
if (realignment_counter_ <= 0) {
|
|
suppressor_gain_limit_ = 1.f;
|
|
return;
|
|
}
|
|
|
|
if (realignment_counter_ > rampup_conf.non_zero_gain_blocks) {
|
|
suppressor_gain_limit_ = 0.f;
|
|
return;
|
|
}
|
|
|
|
if (realignment_counter_ == rampup_conf.non_zero_gain_blocks) {
|
|
suppressor_gain_limit_ = rampup_conf.first_non_zero_gain;
|
|
return;
|
|
}
|
|
|
|
RTC_DCHECK_LT(0.f, suppressor_gain_limit_);
|
|
suppressor_gain_limit_ =
|
|
std::min(1.f, suppressor_gain_limit_ * gain_rampup_increase_);
|
|
RTC_DCHECK_GE(1.f, suppressor_gain_limit_);
|
|
}
|
|
|
|
bool AecState::DetectEchoSaturation(rtc::ArrayView<const float> x) {
|
|
RTC_DCHECK_LT(0, x.size());
|
|
const float max_sample = fabs(*std::max_element(
|
|
x.begin(), x.end(), [](float a, float b) { return a * a < b * b; }));
|
|
previous_max_sample_ = max_sample;
|
|
|
|
// Set flag for potential presence of saturated echo
|
|
blocks_since_last_saturation_ =
|
|
previous_max_sample_ > 200.f && SaturatedCapture()
|
|
? 0
|
|
: blocks_since_last_saturation_ + 1;
|
|
|
|
return blocks_since_last_saturation_ < 20;
|
|
}
|
|
|
|
void AecState::EchoAudibility::Update(rtc::ArrayView<const float> x,
|
|
const std::array<float, kBlockSize>& s,
|
|
bool converged_filter) {
|
|
auto result_x = std::minmax_element(x.begin(), x.end());
|
|
auto result_s = std::minmax_element(s.begin(), s.end());
|
|
const float x_abs = std::max(fabsf(*result_x.first), fabsf(*result_x.second));
|
|
const float s_abs = std::max(fabsf(*result_s.first), fabsf(*result_s.second));
|
|
|
|
if (converged_filter) {
|
|
if (x_abs < 20.f) {
|
|
++low_farend_counter_;
|
|
} else {
|
|
low_farend_counter_ = 0;
|
|
}
|
|
} else {
|
|
if (x_abs < 100.f) {
|
|
++low_farend_counter_;
|
|
} else {
|
|
low_farend_counter_ = 0;
|
|
}
|
|
}
|
|
|
|
// The echo is deemed as not audible if the echo estimate is on the level of
|
|
// the quantization noise in the FFTs and the nearend level is sufficiently
|
|
// strong to mask that by ensuring that the playout and AGC gains do not boost
|
|
// any residual echo that is below the quantization noise level. Furthermore,
|
|
// cases where the render signal is very close to zero are also identified as
|
|
// not producing audible echo.
|
|
inaudible_echo_ = (max_nearend_ > 500 && s_abs < 30.f) ||
|
|
(!converged_filter && x_abs < 500);
|
|
inaudible_echo_ = inaudible_echo_ || low_farend_counter_ > 20;
|
|
}
|
|
|
|
void AecState::EchoAudibility::UpdateWithOutput(rtc::ArrayView<const float> e) {
|
|
const float e_max = *std::max_element(e.begin(), e.end());
|
|
const float e_min = *std::min_element(e.begin(), e.end());
|
|
const float e_abs = std::max(fabsf(e_max), fabsf(e_min));
|
|
|
|
if (max_nearend_ < e_abs) {
|
|
max_nearend_ = e_abs;
|
|
max_nearend_counter_ = 0;
|
|
} else {
|
|
if (++max_nearend_counter_ > 5 * kNumBlocksPerSecond) {
|
|
max_nearend_ *= 0.995f;
|
|
}
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|