| /* |
| * 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(kAdaptiveFilterLength, H2.size()); |
| std::array<int, kAdaptiveFilterLength> 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())); |
| } |
| |
| } // 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), |
| reverb_decay_(config_.ep_strength.default_len) {} |
| |
| AecState::~AecState() = default; |
| |
| void AecState::HandleEchoPathChange( |
| const EchoPathVariability& echo_path_variability) { |
| if (echo_path_variability.AudioPathChanged()) { |
| 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; |
| |
| if (echo_path_variability.delay_change) { |
| force_zero_gain_counter_ = 0; |
| blocks_with_filter_adaptation_ = 0; |
| render_received_ = false; |
| force_zero_gain_ = true; |
| capture_block_counter_ = 0; |
| } |
| if (echo_path_variability.gain_change) { |
| capture_block_counter_ = kNumBlocksPerSecond; |
| } |
| } |
| } |
| |
| void AecState::Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>& |
| adaptive_filter_frequency_response, |
| const std::array<float, kAdaptiveFilterTimeDomainLength>& |
| adaptive_filter_impulse_response, |
| bool converged_filter, |
| const rtc::Optional<size_t>& external_delay_samples, |
| const RenderBuffer& render_buffer, |
| const std::array<float, kFftLengthBy2Plus1>& E2_main, |
| const std::array<float, kFftLengthBy2Plus1>& Y2, |
| rtc::ArrayView<const float> x, |
| const std::array<float, kBlockSize>& s, |
| bool echo_leakage_detected) { |
| // Store input parameters. |
| echo_leakage_detected_ = echo_leakage_detected; |
| |
| // Update counters. |
| ++capture_block_counter_; |
| |
| // Force zero echo suppression gain after an echo path change to allow at |
| // least some render data to be collected in order to avoid an initial echo |
| // burst. |
| force_zero_gain_ = (++force_zero_gain_counter_) < kNumBlocksPerSecond / 5; |
| |
| // Estimate delays. |
| filter_delay_ = rtc::Optional<size_t>( |
| EstimateFilterDelay(adaptive_filter_frequency_response)); |
| external_delay_ = |
| external_delay_samples |
| ? rtc::Optional<size_t>(*external_delay_samples / kBlockSize) |
| : rtc::Optional<size_t>(); |
| |
| // 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. |
| 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; })); |
| |
| if (config_.ep_strength.echo_can_saturate) { |
| const bool saturated_echo = |
| (previous_max_sample_ > 200.f) && SaturatedCapture(); |
| |
| // Counts the blocks since saturation. |
| constexpr size_t kSaturationLeakageBlocks = 20; |
| blocks_since_last_saturation_ = |
| saturated_echo ? 0 : blocks_since_last_saturation_ + 1; |
| |
| echo_saturation_ = blocks_since_last_saturation_ < kSaturationLeakageBlocks; |
| } else { |
| echo_saturation_ = false; |
| } |
| previous_max_sample_ = max_sample; |
| |
| // Flag whether the linear filter estimate is usable. |
| usable_linear_estimate_ = |
| (!echo_saturation_) && (converged_filter || SufficientFilterUpdates()) && |
| capture_block_counter_ >= 2 * kNumBlocksPerSecond && external_delay_; |
| |
| // 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. |
| const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f); |
| const bool active_render_block = |
| x_energy > (config_.render_levels.active_render_limit * |
| config_.render_levels.active_render_limit) * |
| kFftLengthBy2; |
| if (active_render_block) { |
| render_received_ = true; |
| } |
| blocks_with_filter_adaptation_ += |
| (active_render_block && (!SaturatedCapture()) ? 1 : 0); |
| |
| transparent_mode_ = !converged_filter && |
| (!render_received_ || blocks_with_filter_adaptation_ >= |
| 5 * kNumBlocksPerSecond); |
| |
| // Update the room reverb estimate. |
| UpdateReverb(adaptive_filter_impulse_response); |
| } |
| |
| void AecState::UpdateReverb( |
| const std::array<float, kAdaptiveFilterTimeDomainLength>& |
| impulse_response) { |
| if ((!(filter_delay_ && usable_linear_estimate_)) || |
| (*filter_delay_ > kAdaptiveFilterLength - 4)) { |
| return; |
| } |
| |
| // Form the data to match against by squaring the impulse response |
| // coefficients. |
| std::array<float, kAdaptiveFilterTimeDomainLength> matching_data; |
| std::transform(impulse_response.begin(), impulse_response.end(), |
| matching_data.begin(), [](float a) { return a * a; }); |
| |
| // Avoid matching against noise in the model by subtracting an estimate of the |
| // model noise power. |
| constexpr size_t kTailLength = 64; |
| constexpr size_t tail_index = kAdaptiveFilterTimeDomainLength - kTailLength; |
| const float tail_power = *std::max_element(matching_data.begin() + tail_index, |
| matching_data.end()); |
| std::for_each(matching_data.begin(), matching_data.begin() + tail_index, |
| [tail_power](float& a) { a = std::max(0.f, a - tail_power); }); |
| |
| // Identify the peak index of the impulse response. |
| const size_t peak_index = *std::max_element( |
| matching_data.begin(), matching_data.begin() + tail_index); |
| |
| if (peak_index + 128 < tail_index) { |
| size_t start_index = peak_index + 64; |
| // Compute the matching residual error for the current candidate to match. |
| float residual_sqr_sum = 0.f; |
| float d_k = reverb_decay_to_test_; |
| for (size_t k = start_index; k < tail_index; ++k) { |
| if (matching_data[start_index + 1] == 0.f) { |
| break; |
| } |
| |
| float residual = matching_data[k] - matching_data[peak_index] * d_k; |
| residual_sqr_sum += residual * residual; |
| d_k *= reverb_decay_to_test_; |
| } |
| |
| // If needed, update the best candidate for the reverb decay. |
| if (reverb_decay_candidate_residual_ < 0.f || |
| residual_sqr_sum < reverb_decay_candidate_residual_) { |
| reverb_decay_candidate_residual_ = residual_sqr_sum; |
| reverb_decay_candidate_ = reverb_decay_to_test_; |
| } |
| } |
| |
| // Compute the next reverb candidate to evaluate such that all candidates will |
| // be evaluated within one second. |
| reverb_decay_to_test_ += (0.9965f - 0.9f) / (5 * kNumBlocksPerSecond); |
| |
| // If all reverb candidates have been evaluated, choose the best one as the |
| // reverb decay. |
| if (reverb_decay_to_test_ >= 0.9965f) { |
| if (reverb_decay_candidate_residual_ < 0.f) { |
| // Transform the decay to be in the unit of blocks. |
| reverb_decay_ = powf(reverb_decay_candidate_, kFftLengthBy2); |
| |
| // Limit the estimated reverb_decay_ to the maximum one needed in practice |
| // to minimize the impact of incorrect estimates. |
| reverb_decay_ = std::min(config_.ep_strength.default_len, reverb_decay_); |
| } |
| reverb_decay_to_test_ = 0.9f; |
| reverb_decay_candidate_residual_ = -1.f; |
| } |
| |
| // For noisy impulse responses, assume a fixed tail length. |
| if (tail_power > 0.0005f) { |
| reverb_decay_ = config_.ep_strength.default_len; |
| } |
| data_dumper_->DumpRaw("aec3_reverb_decay", reverb_decay_); |
| data_dumper_->DumpRaw("aec3_tail_power", tail_power); |
| } |
| |
| 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 |