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This CL increases the speech of the initial alignment in AEC3 by loosening the requirements on the accuracy of the initial estimates. Bug: webrtc:8784, chromium:804270 Change-Id: I86e2d97830843524090a1cf877965739f66dc058 Reviewed-on: https://webrtc-review.googlesource.com/40660 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21728}
335 lines
12 KiB
C++
335 lines
12 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/aec_state.h"
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#include <math.h>
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#include <numeric>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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// Computes delay of the adaptive filter.
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int EstimateFilterDelay(
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const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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adaptive_filter_frequency_response) {
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const auto& H2 = adaptive_filter_frequency_response;
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constexpr size_t kUpperBin = kFftLengthBy2 - 5;
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RTC_DCHECK_GE(kMaxAdaptiveFilterLength, H2.size());
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std::array<int, kMaxAdaptiveFilterLength> delays;
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delays.fill(0);
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for (size_t k = 1; k < kUpperBin; ++k) {
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// Find the maximum of H2[j].
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size_t peak = 0;
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for (size_t j = 0; j < H2.size(); ++j) {
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if (H2[j][k] > H2[peak][k]) {
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peak = j;
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}
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}
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++delays[peak];
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}
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return std::distance(delays.begin(),
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std::max_element(delays.begin(), delays.end()));
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}
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} // namespace
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int AecState::instance_count_ = 0;
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AecState::AecState(const EchoCanceller3Config& config)
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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erle_estimator_(config.erle.min, config.erle.max_l, config.erle.max_h),
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config_(config),
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max_render_(config_.filter.main.length_blocks, 0.f),
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reverb_decay_(config_.ep_strength.default_len) {}
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AecState::~AecState() = default;
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void AecState::HandleEchoPathChange(
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const EchoPathVariability& echo_path_variability) {
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const auto full_reset = [&]() {
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blocks_since_last_saturation_ = 0;
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usable_linear_estimate_ = false;
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echo_leakage_detected_ = false;
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capture_signal_saturation_ = false;
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echo_saturation_ = false;
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previous_max_sample_ = 0.f;
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std::fill(max_render_.begin(), max_render_.end(), 0.f);
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force_zero_gain_counter_ = 0;
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blocks_with_proper_filter_adaptation_ = 0;
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capture_block_counter_ = 0;
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filter_has_had_time_to_converge_ = false;
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render_received_ = false;
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force_zero_gain_ = true;
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blocks_with_active_render_ = 0;
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initial_state_ = true;
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};
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// TODO(peah): Refine the reset scheme according to the type of gain and
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// delay adjustment.
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if (echo_path_variability.gain_change) {
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full_reset();
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}
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if (echo_path_variability.delay_change !=
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EchoPathVariability::DelayAdjustment::kBufferReadjustment) {
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full_reset();
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} else if (echo_path_variability.delay_change !=
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EchoPathVariability::DelayAdjustment::kBufferFlush) {
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full_reset();
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} else if (echo_path_variability.delay_change !=
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EchoPathVariability::DelayAdjustment::kDelayReset) {
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full_reset();
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} else if (echo_path_variability.delay_change !=
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EchoPathVariability::DelayAdjustment::kNewDetectedDelay) {
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full_reset();
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} else if (echo_path_variability.gain_change) {
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capture_block_counter_ = kNumBlocksPerSecond;
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}
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}
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void AecState::Update(
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const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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adaptive_filter_frequency_response,
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const std::vector<float>& adaptive_filter_impulse_response,
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bool converged_filter,
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const RenderBuffer& render_buffer,
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const std::array<float, kFftLengthBy2Plus1>& E2_main,
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const std::array<float, kFftLengthBy2Plus1>& Y2,
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const std::array<float, kBlockSize>& s,
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bool echo_leakage_detected) {
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// Store input parameters.
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echo_leakage_detected_ = echo_leakage_detected;
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// Estimate the filter delay.
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filter_delay_ = EstimateFilterDelay(adaptive_filter_frequency_response);
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const std::vector<float>& x = render_buffer.Block(-filter_delay_)[0];
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// Update counters.
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++capture_block_counter_;
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const bool active_render_block = DetectActiveRender(x);
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blocks_with_active_render_ += active_render_block ? 1 : 0;
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blocks_with_proper_filter_adaptation_ +=
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active_render_block && !SaturatedCapture() ? 1 : 0;
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// Force zero echo suppression gain after an echo path change to allow at
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// least some render data to be collected in order to avoid an initial echo
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// burst.
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force_zero_gain_ = ++force_zero_gain_counter_ < kNumBlocksPerSecond / 5;
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// Update the ERL and ERLE measures.
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if (converged_filter && capture_block_counter_ >= 2 * kNumBlocksPerSecond) {
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const auto& X2 = render_buffer.Spectrum(filter_delay_);
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erle_estimator_.Update(X2, Y2, E2_main);
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erl_estimator_.Update(X2, Y2);
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}
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// Update the echo audibility evaluator.
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echo_audibility_.Update(x, s, converged_filter);
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// Detect and flag echo saturation.
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// TODO(peah): Add the delay in this computation to ensure that the render and
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// capture signals are properly aligned.
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if (config_.ep_strength.echo_can_saturate) {
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echo_saturation_ = DetectEchoSaturation(x);
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}
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// TODO(peah): Move?
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filter_has_had_time_to_converge_ =
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blocks_with_proper_filter_adaptation_ >= 0.5f * kNumBlocksPerSecond;
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initial_state_ =
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blocks_with_proper_filter_adaptation_ < 5 * kNumBlocksPerSecond;
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// Flag whether the linear filter estimate is usable.
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usable_linear_estimate_ =
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!echo_saturation_ &&
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(converged_filter && filter_has_had_time_to_converge_) &&
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capture_block_counter_ >= 1.f * kNumBlocksPerSecond && !TransparentMode();
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// After an amount of active render samples for which an echo should have been
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// detected in the capture signal if the ERL was not infinite, flag that a
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// transparent mode should be entered.
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transparent_mode_ =
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!converged_filter &&
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(blocks_with_active_render_ == 0 ||
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blocks_with_proper_filter_adaptation_ >= 5 * kNumBlocksPerSecond);
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// Update the room reverb estimate.
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UpdateReverb(adaptive_filter_impulse_response);
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}
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void AecState::UpdateReverb(const std::vector<float>& impulse_response) {
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if ((!(filter_delay_ && usable_linear_estimate_)) ||
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(filter_delay_ >
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static_cast<int>(config_.filter.main.length_blocks) - 4)) {
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return;
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}
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// Form the data to match against by squaring the impulse response
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// coefficients.
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std::array<float, GetTimeDomainLength(kMaxAdaptiveFilterLength)>
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matching_data_data;
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RTC_DCHECK_LE(GetTimeDomainLength(config_.filter.main.length_blocks),
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matching_data_data.size());
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rtc::ArrayView<float> matching_data(
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matching_data_data.data(),
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GetTimeDomainLength(config_.filter.main.length_blocks));
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std::transform(impulse_response.begin(), impulse_response.end(),
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matching_data.begin(), [](float a) { return a * a; });
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// Avoid matching against noise in the model by subtracting an estimate of the
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// model noise power.
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constexpr size_t kTailLength = 64;
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const size_t tail_index =
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GetTimeDomainLength(config_.filter.main.length_blocks) - kTailLength;
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const float tail_power = *std::max_element(matching_data.begin() + tail_index,
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matching_data.end());
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std::for_each(matching_data.begin(), matching_data.begin() + tail_index,
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[tail_power](float& a) { a = std::max(0.f, a - tail_power); });
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// Identify the peak index of the impulse response.
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const size_t peak_index = *std::max_element(
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matching_data.begin(), matching_data.begin() + tail_index);
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if (peak_index + 128 < tail_index) {
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size_t start_index = peak_index + 64;
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// Compute the matching residual error for the current candidate to match.
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float residual_sqr_sum = 0.f;
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float d_k = reverb_decay_to_test_;
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for (size_t k = start_index; k < tail_index; ++k) {
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if (matching_data[start_index + 1] == 0.f) {
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break;
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}
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float residual = matching_data[k] - matching_data[peak_index] * d_k;
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residual_sqr_sum += residual * residual;
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d_k *= reverb_decay_to_test_;
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}
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// If needed, update the best candidate for the reverb decay.
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if (reverb_decay_candidate_residual_ < 0.f ||
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residual_sqr_sum < reverb_decay_candidate_residual_) {
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reverb_decay_candidate_residual_ = residual_sqr_sum;
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reverb_decay_candidate_ = reverb_decay_to_test_;
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}
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}
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// Compute the next reverb candidate to evaluate such that all candidates will
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// be evaluated within one second.
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reverb_decay_to_test_ += (0.9965f - 0.9f) / (5 * kNumBlocksPerSecond);
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// If all reverb candidates have been evaluated, choose the best one as the
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// reverb decay.
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if (reverb_decay_to_test_ >= 0.9965f) {
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if (reverb_decay_candidate_residual_ < 0.f) {
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// Transform the decay to be in the unit of blocks.
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reverb_decay_ = powf(reverb_decay_candidate_, kFftLengthBy2);
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// Limit the estimated reverb_decay_ to the maximum one needed in practice
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// to minimize the impact of incorrect estimates.
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reverb_decay_ = std::min(config_.ep_strength.default_len, reverb_decay_);
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}
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reverb_decay_to_test_ = 0.9f;
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reverb_decay_candidate_residual_ = -1.f;
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}
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// For noisy impulse responses, assume a fixed tail length.
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if (tail_power > 0.0005f) {
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reverb_decay_ = config_.ep_strength.default_len;
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}
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data_dumper_->DumpRaw("aec3_reverb_decay", reverb_decay_);
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data_dumper_->DumpRaw("aec3_tail_power", tail_power);
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}
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bool AecState::DetectActiveRender(rtc::ArrayView<const float> x) const {
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const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
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return x_energy > (config_.render_levels.active_render_limit *
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config_.render_levels.active_render_limit) *
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kFftLengthBy2;
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}
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bool AecState::DetectEchoSaturation(rtc::ArrayView<const float> x) {
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RTC_DCHECK_LT(0, x.size());
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const float max_sample = fabs(*std::max_element(
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x.begin(), x.end(), [](float a, float b) { return a * a < b * b; }));
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previous_max_sample_ = max_sample;
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// Set flag for potential presence of saturated echo
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blocks_since_last_saturation_ =
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previous_max_sample_ > 200.f && SaturatedCapture()
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? 0
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: blocks_since_last_saturation_ + 1;
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return blocks_since_last_saturation_ < 20;
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}
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void AecState::EchoAudibility::Update(rtc::ArrayView<const float> x,
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const std::array<float, kBlockSize>& s,
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bool converged_filter) {
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auto result_x = std::minmax_element(x.begin(), x.end());
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auto result_s = std::minmax_element(s.begin(), s.end());
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const float x_abs =
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std::max(fabsf(*result_x.first), fabsf(*result_x.second));
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const float s_abs =
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std::max(fabsf(*result_s.first), fabsf(*result_s.second));
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if (converged_filter) {
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if (x_abs < 20.f) {
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++low_farend_counter_;
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} else {
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low_farend_counter_ = 0;
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}
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} else {
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if (x_abs < 100.f) {
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++low_farend_counter_;
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} else {
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low_farend_counter_ = 0;
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}
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}
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// The echo is deemed as not audible if the echo estimate is on the level of
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// the quantization noise in the FFTs and the nearend level is sufficiently
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// strong to mask that by ensuring that the playout and AGC gains do not boost
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// any residual echo that is below the quantization noise level. Furthermore,
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// cases where the render signal is very close to zero are also identified as
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// not producing audible echo.
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inaudible_echo_ = (max_nearend_ > 500 && s_abs < 30.f) ||
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(!converged_filter && x_abs < 500);
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inaudible_echo_ = inaudible_echo_ || low_farend_counter_ > 20;
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}
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void AecState::EchoAudibility::UpdateWithOutput(rtc::ArrayView<const float> e) {
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const float e_max = *std::max_element(e.begin(), e.end());
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const float e_min = *std::min_element(e.begin(), e.end());
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const float e_abs = std::max(fabsf(e_max), fabsf(e_min));
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if (max_nearend_ < e_abs) {
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max_nearend_ = e_abs;
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max_nearend_counter_ = 0;
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} else {
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if (++max_nearend_counter_ > 5 * kNumBlocksPerSecond) {
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max_nearend_ *= 0.995f;
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}
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}
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}
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} // namespace webrtc
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