blob: 38ed3b047bea63ed1a9a8f0995a54f95746d1df5 [file] [log] [blame]
* 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 <algorithm>
#include <memory>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
#include "webrtc/modules/audio_processing/aec3/erl_estimator.h"
#include "webrtc/modules/audio_processing/aec3/erle_estimator.h"
#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
// Handles the state and the conditions for the echo removal functionality.
class AecState {
explicit AecState(const AudioProcessing::Config::EchoCanceller3& config);
// Returns whether the linear filter estimate is usable.
bool UsableLinearEstimate() const { return usable_linear_estimate_; }
// Returns whether there has been echo leakage detected.
bool EchoLeakageDetected() const { return echo_leakage_detected_; }
// Returns whether the render signal is currently active.
// TODO(peah): Deprecate this in an upcoming CL.
bool ActiveRender() const { return blocks_with_filter_adaptation_ > 200; }
// Returns the ERLE.
const std::array<float, kFftLengthBy2Plus1>& Erle() const {
return erle_estimator_.Erle();
// Returns the ERL.
const std::array<float, kFftLengthBy2Plus1>& Erl() const {
return erl_estimator_.Erl();
// Returns the delay estimate based on the linear filter.
rtc::Optional<size_t> FilterDelay() const { return filter_delay_; }
// Returns the externally provided delay.
rtc::Optional<size_t> ExternalDelay() const { return external_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 a probable headset setup has been detected.
bool HeadsetDetected() const { return headset_detected_; }
// 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 whether the echo suppression gain should be forced to zero.
bool ForcedZeroGain() const { return force_zero_gain_; }
// Returns whether the echo in the capture signal is audible.
bool InaudibleEcho() const { return echo_audibility_.InaudibleEcho(); }
// Updates the aec state with the AEC output signal.
void UpdateWithOutput(rtc::ArrayView<const float> e) {
// Updates the aec state.
void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
const std::array<float, kAdaptiveFilterTimeDomainLength>&
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_main,
bool echo_leakage_detected);
class EchoAudibility {
void Update(rtc::ArrayView<const float> x,
const std::array<float, kBlockSize>& s);
void UpdateWithOutput(rtc::ArrayView<const float> e);
bool InaudibleEcho() const { return inaudible_echo_; }
float max_nearend_ = 0.f;
size_t max_nearend_counter_ = 0;
size_t low_farend_counter_ = 0;
bool inaudible_echo_ = false;
void UpdateReverb(const std::array<float, kAdaptiveFilterTimeDomainLength>&
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
ErlEstimator erl_estimator_;
ErleEstimator erle_estimator_;
int echo_path_change_counter_;
size_t blocks_with_filter_adaptation_ = 0;
bool usable_linear_estimate_ = false;
bool echo_leakage_detected_ = false;
bool capture_signal_saturation_ = false;
bool echo_saturation_ = false;
bool headset_detected_ = false;
float previous_max_sample_ = 0.f;
bool force_zero_gain_ = false;
bool render_received_ = false;
size_t force_zero_gain_counter_ = 0;
rtc::Optional<size_t> filter_delay_;
rtc::Optional<size_t> external_delay_;
size_t blocks_since_last_saturation_ = 1000;
float reverb_decay_to_test_ = 0.9f;
float reverb_decay_candidate_ = 0.f;
float reverb_decay_candidate_residual_ = -1.f;
EchoAudibility echo_audibility_;
const AudioProcessing::Config::EchoCanceller3 config_;
float reverb_decay_;
} // namespace webrtc