| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |
| #define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |
| |
| #include <math.h> |
| |
| #include <algorithm> |
| #include <memory> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/audio/echo_canceller3_config.h" |
| #include "modules/audio_processing/aec3/aec3_common.h" |
| #include "modules/audio_processing/aec3/delay_estimate.h" |
| #include "modules/audio_processing/aec3/echo_audibility.h" |
| #include "modules/audio_processing/aec3/echo_path_variability.h" |
| #include "modules/audio_processing/aec3/erl_estimator.h" |
| #include "modules/audio_processing/aec3/erle_estimator.h" |
| #include "modules/audio_processing/aec3/filter_analyzer.h" |
| #include "modules/audio_processing/aec3/render_buffer.h" |
| #include "modules/audio_processing/aec3/suppression_gain_limiter.h" |
| #include "rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| |
| // Handles the state and the conditions for the echo removal functionality. |
| class AecState { |
| public: |
| explicit AecState(const EchoCanceller3Config& config); |
| ~AecState(); |
| |
| // Returns whether the echo subtractor can be used to determine the residual |
| // echo. |
| bool UsableLinearEstimate() const { return usable_linear_estimate_; } |
| |
| // Returns whether the echo subtractor output should be used as output. |
| bool UseLinearFilterOutput() const { return use_linear_filter_output_; } |
| |
| // Returns the estimated echo path gain. |
| float EchoPathGain() const { return filter_analyzer_.Gain(); } |
| |
| // Returns whether the render signal is currently active. |
| bool ActiveRender() const { return blocks_with_active_render_ > 200; } |
| |
| // Returns the appropriate scaling of the residual echo to match the |
| // audibility. |
| void GetResidualEchoScaling(rtc::ArrayView<float> residual_scaling) const { |
| echo_audibility_.GetResidualEchoScaling(residual_scaling); |
| } |
| |
| // Returns whether the stationary properties of the signals are used in the |
| // aec. |
| bool UseStationaryProperties() const { return use_stationary_properties_; } |
| |
| // Returns the ERLE. |
| const std::array<float, kFftLengthBy2Plus1>& Erle() const { |
| return erle_estimator_.Erle(); |
| } |
| |
| // Returns the time-domain ERLE. |
| float ErleTimeDomain() const { return erle_estimator_.ErleTimeDomain(); } |
| |
| // Returns the ERL. |
| const std::array<float, kFftLengthBy2Plus1>& Erl() const { |
| return erl_estimator_.Erl(); |
| } |
| |
| // Returns the time-domain ERL. |
| float ErlTimeDomain() const { return erl_estimator_.ErlTimeDomain(); } |
| |
| // Returns the delay estimate based on the linear filter. |
| int FilterDelayBlocks() const { return filter_delay_blocks_; } |
| |
| // Returns the internal delay estimate based on the linear filter. |
| absl::optional<int> InternalDelay() const { return internal_delay_; } |
| |
| // Returns whether the capture signal is saturated. |
| bool SaturatedCapture() const { return capture_signal_saturation_; } |
| |
| // Returns whether the echo signal is saturated. |
| bool SaturatedEcho() const { return echo_saturation_; } |
| |
| // Updates the capture signal saturation. |
| void UpdateCaptureSaturation(bool capture_signal_saturation) { |
| capture_signal_saturation_ = capture_signal_saturation; |
| } |
| |
| // Returns whether the transparent mode is active |
| bool TransparentMode() const { return transparent_mode_; } |
| |
| // Takes appropriate action at an echo path change. |
| void HandleEchoPathChange(const EchoPathVariability& echo_path_variability); |
| |
| // Returns the decay factor for the echo reverberation. |
| float ReverbDecay() const { return reverb_decay_; } |
| |
| // Returns the upper limit for the echo suppression gain. |
| float SuppressionGainLimit() const { |
| return suppression_gain_limiter_.Limit(); |
| } |
| |
| // Returns whether the suppression gain limiter is active. |
| bool IsSuppressionGainLimitActive() const { |
| return suppression_gain_limiter_.IsActive(); |
| } |
| |
| // Returns whether the linear filter should have been able to properly adapt. |
| bool FilterHasHadTimeToConverge() const { |
| return filter_has_had_time_to_converge_; |
| } |
| |
| // Returns whether the filter adaptation is still in the initial state. |
| bool InitialState() const { return initial_state_; } |
| |
| // Updates the aec state. |
| void Update(const absl::optional<DelayEstimate>& external_delay, |
| const std::vector<std::array<float, kFftLengthBy2Plus1>>& |
| adaptive_filter_frequency_response, |
| const std::vector<float>& adaptive_filter_impulse_response, |
| bool converged_filter, |
| bool diverged_filter, |
| const RenderBuffer& render_buffer, |
| const std::array<float, kFftLengthBy2Plus1>& E2_main, |
| const std::array<float, kFftLengthBy2Plus1>& Y2, |
| const std::array<float, kBlockSize>& s); |
| |
| // Returns the gain at the tail of the linear filter. |
| float GetFilterTailGain() const { return filter_analyzer_.GetTailGain(); } |
| |
| // Returns filter length in blocks. |
| int FilterLengthBlocks() const { |
| return filter_analyzer_.FilterLengthBlocks(); |
| } |
| |
| private: |
| void UpdateReverb(const std::vector<float>& impulse_response); |
| bool DetectActiveRender(rtc::ArrayView<const float> x) const; |
| void UpdateSuppressorGainLimit(bool render_activity); |
| bool DetectEchoSaturation(rtc::ArrayView<const float> x, |
| float echo_path_gain); |
| |
| static int instance_count_; |
| std::unique_ptr<ApmDataDumper> data_dumper_; |
| const EchoCanceller3Config config_; |
| const bool allow_transparent_mode_; |
| const bool use_stationary_properties_; |
| const bool enforce_delay_after_realignment_; |
| ErlEstimator erl_estimator_; |
| ErleEstimator erle_estimator_; |
| size_t capture_block_counter_ = 0; |
| size_t blocks_since_reset_ = 0; |
| size_t blocks_with_proper_filter_adaptation_ = 0; |
| size_t blocks_with_active_render_ = 0; |
| bool usable_linear_estimate_ = false; |
| bool capture_signal_saturation_ = false; |
| bool echo_saturation_ = false; |
| bool transparent_mode_ = false; |
| bool render_received_ = false; |
| int filter_delay_blocks_ = 0; |
| size_t blocks_since_last_saturation_ = 1000; |
| float tail_energy_ = 0.f; |
| float accumulated_nz_ = 0.f; |
| float accumulated_nn_ = 0.f; |
| float accumulated_count_ = 0.f; |
| size_t current_reverb_decay_section_ = 0; |
| size_t num_reverb_decay_sections_ = 0; |
| size_t num_reverb_decay_sections_next_ = 0; |
| bool found_end_of_reverb_decay_ = false; |
| bool main_filter_is_adapting_ = true; |
| std::array<float, kMaxAdaptiveFilterLength> block_energies_; |
| std::vector<float> max_render_; |
| float reverb_decay_ = fabsf(config_.ep_strength.default_len); |
| bool filter_has_had_time_to_converge_ = false; |
| bool initial_state_ = true; |
| const float gain_rampup_increase_; |
| SuppressionGainUpperLimiter suppression_gain_limiter_; |
| FilterAnalyzer filter_analyzer_; |
| bool use_linear_filter_output_ = false; |
| absl::optional<int> internal_delay_; |
| size_t diverged_blocks_ = 0; |
| bool filter_should_have_converged_ = false; |
| size_t blocks_since_converged_filter_; |
| size_t active_blocks_since_consistent_filter_estimate_; |
| bool converged_filter_seen_ = false; |
| bool consistent_filter_seen_ = false; |
| bool external_delay_seen_ = false; |
| absl::optional<DelayEstimate> external_delay_; |
| size_t frames_since_external_delay_change_ = 0; |
| size_t converged_filter_count_ = 0; |
| bool finite_erl_ = false; |
| size_t active_blocks_since_converged_filter_ = 0; |
| EchoAudibility echo_audibility_; |
| RTC_DISALLOW_COPY_AND_ASSIGN(AecState); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |