/* * 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 "webrtc/modules/audio_processing/aec3/aec_state.h" #include #include #include #include "webrtc/base/array_view.h" #include "webrtc/base/atomicops.h" #include "webrtc/base/checks.h" #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" namespace webrtc { namespace { constexpr size_t kEchoPathChangeConvergenceBlocks = 2 * kNumBlocksPerSecond; constexpr size_t kSaturationLeakageBlocks = 20; // Computes delay of the adaptive filter. rtc::Optional EstimateFilterDelay( const std::vector>& adaptive_filter_frequency_response) { const auto& H2 = adaptive_filter_frequency_response; size_t reliable_delays_sum = 0; size_t num_reliable_delays = 0; constexpr size_t kUpperBin = kFftLengthBy2 - 5; constexpr float kMinPeakMargin = 10.f; const size_t kTailPartition = H2.size() - 1; for (size_t k = 1; k < kUpperBin; ++k) { // Find the maximum of H2[j]. int peak = 0; for (size_t j = 0; j < H2.size(); ++j) { if (H2[j][k] > H2[peak][k]) { peak = j; } } // Count the peak as a delay only if the peak is sufficiently larger than // the tail. if (kMinPeakMargin * H2[kTailPartition][k] < H2[peak][k]) { reliable_delays_sum += peak; ++num_reliable_delays; } } // Return no delay if not sufficient delays have been found. if (num_reliable_delays < 21) { return rtc::Optional(); } const size_t delay = reliable_delays_sum / num_reliable_delays; // Sanity check that the peak is not caused by a false strong DC-component in // the filter. for (size_t k = 1; k < kUpperBin; ++k) { if (H2[delay][k] > H2[delay][0]) { RTC_DCHECK_GT(H2.size(), delay); return rtc::Optional(delay); } } return rtc::Optional(); } constexpr int kEchoPathChangeCounterInitial = kNumBlocksPerSecond / 5; constexpr int kEchoPathChangeCounterMax = 2 * kNumBlocksPerSecond; } // namespace int AecState::instance_count_ = 0; AecState::AecState() : data_dumper_( new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))), echo_path_change_counter_(kEchoPathChangeCounterInitial) {} 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; echo_path_change_counter_ = kEchoPathChangeCounterMax; } if (echo_path_variability.gain_change) { echo_path_change_counter_ = kEchoPathChangeCounterInitial; } } } void AecState::Update(const std::vector>& adaptive_filter_frequency_response, const rtc::Optional& external_delay_samples, const RenderBuffer& render_buffer, const std::array& E2_main, const std::array& Y2, rtc::ArrayView x, bool echo_leakage_detected) { // Store input parameters. echo_leakage_detected_ = echo_leakage_detected; // Update counters. const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f); const bool active_render_block = x_energy > 10000.f * kFftLengthBy2; if (active_render_block) { render_received_ = true; } blocks_with_filter_adaptation_ += (active_render_block && (!SaturatedCapture()) ? 1 : 0); --echo_path_change_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. constexpr size_t kZeroGainBlocksAfterChange = kNumBlocksPerSecond / 5; force_zero_gain_ = (++force_zero_gain_counter_) < kZeroGainBlocksAfterChange; // Estimate delays. filter_delay_ = EstimateFilterDelay(adaptive_filter_frequency_response); external_delay_ = external_delay_samples ? rtc::Optional(*external_delay_samples / kBlockSize) : rtc::Optional(); // Update the ERL and ERLE measures. if (filter_delay_ && echo_path_change_counter_ <= 0) { const auto& X2 = render_buffer.Spectrum(*filter_delay_); erle_estimator_.Update(X2, Y2, E2_main); erl_estimator_.Update(X2, Y2); } // 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; })); const bool saturated_echo = previous_max_sample_ * kFixedEchoPathGain > 1600 && SaturatedCapture(); previous_max_sample_ = max_sample; // Counts the blocks since saturation. blocks_since_last_saturation_ = saturated_echo ? 0 : blocks_since_last_saturation_ + 1; echo_saturation_ = blocks_since_last_saturation_ < kSaturationLeakageBlocks; // Flag whether the linear filter estimate is usable. usable_linear_estimate_ = (!echo_saturation_) && (!render_received_ || blocks_with_filter_adaptation_ > kEchoPathChangeConvergenceBlocks) && filter_delay_ && echo_path_change_counter_ <= 0; // 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 // headset is used. headset_detected_ = !external_delay_ && !filter_delay_ && (!render_received_ || blocks_with_filter_adaptation_ >= kEchoPathChangeConvergenceBlocks); } } // namespace webrtc