Source code
Revision control
Copy as Markdown
Other Tools
/*
* 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_SUBTRACTOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
#include <math.h>
#include <stddef.h>
#include <array>
#include <vector>
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/aec3_fft.h"
#include "modules/audio_processing/aec3/aec_state.h"
#include "modules/audio_processing/aec3/block.h"
#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
#include "modules/audio_processing/aec3/echo_path_variability.h"
#include "modules/audio_processing/aec3/refined_filter_update_gain.h"
#include "modules/audio_processing/aec3/render_buffer.h"
#include "modules/audio_processing/aec3/render_signal_analyzer.h"
#include "modules/audio_processing/aec3/subtractor_output.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
namespace webrtc {
// Proves linear echo cancellation functionality
class Subtractor {
public:
Subtractor(const EchoCanceller3Config& config,
size_t num_render_channels,
size_t num_capture_channels,
ApmDataDumper* data_dumper,
Aec3Optimization optimization);
~Subtractor();
Subtractor(const Subtractor&) = delete;
Subtractor& operator=(const Subtractor&) = delete;
// Performs the echo subtraction.
void Process(const RenderBuffer& render_buffer,
const Block& capture,
const RenderSignalAnalyzer& render_signal_analyzer,
const AecState& aec_state,
rtc::ArrayView<SubtractorOutput> outputs);
void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
// Exits the initial state.
void ExitInitialState();
// Returns the block-wise frequency responses for the refined adaptive
// filters.
const std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>&
FilterFrequencyResponses() const {
return refined_frequency_responses_;
}
// Returns the estimates of the impulse responses for the refined adaptive
// filters.
const std::vector<std::vector<float>>& FilterImpulseResponses() const {
return refined_impulse_responses_;
}
void DumpFilters() {
data_dumper_->DumpRaw(
"aec3_subtractor_h_refined",
rtc::ArrayView<const float>(
refined_impulse_responses_[0].data(),
GetTimeDomainLength(
refined_filters_[0]->max_filter_size_partitions())));
if (ApmDataDumper::IsAvailable()) {
RTC_DCHECK_GT(coarse_impulse_responses_.size(), 0);
data_dumper_->DumpRaw(
"aec3_subtractor_h_coarse",
rtc::ArrayView<const float>(
coarse_impulse_responses_[0].data(),
GetTimeDomainLength(
coarse_filter_[0]->max_filter_size_partitions())));
}
refined_filters_[0]->DumpFilter("aec3_subtractor_H_refined");
coarse_filter_[0]->DumpFilter("aec3_subtractor_H_coarse");
}
private:
class FilterMisadjustmentEstimator {
public:
FilterMisadjustmentEstimator() = default;
~FilterMisadjustmentEstimator() = default;
// Update the misadjustment estimator.
void Update(const SubtractorOutput& output);
// GetMisadjustment() Returns a recommended scale for the filter so the
// prediction error energy gets closer to the energy that is seen at the
// microphone input.
float GetMisadjustment() const {
RTC_DCHECK_GT(inv_misadjustment_, 0.0f);
// It is not aiming to adjust all the estimated mismatch. Instead,
// it adjusts half of that estimated mismatch.
return 2.f / sqrtf(inv_misadjustment_);
}
// Returns true if the prediciton error energy is significantly larger
// than the microphone signal energy and, therefore, an adjustment is
// recommended.
bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; }
void Reset();
void Dump(ApmDataDumper* data_dumper) const;
private:
const int n_blocks_ = 4;
int n_blocks_acum_ = 0;
float e2_acum_ = 0.f;
float y2_acum_ = 0.f;
float inv_misadjustment_ = 0.f;
int overhang_ = 0.f;
};
const Aec3Fft fft_;
ApmDataDumper* data_dumper_;
const Aec3Optimization optimization_;
const EchoCanceller3Config config_;
const size_t num_capture_channels_;
const bool use_coarse_filter_reset_hangover_;
std::vector<std::unique_ptr<AdaptiveFirFilter>> refined_filters_;
std::vector<std::unique_ptr<AdaptiveFirFilter>> coarse_filter_;
std::vector<std::unique_ptr<RefinedFilterUpdateGain>> refined_gains_;
std::vector<std::unique_ptr<CoarseFilterUpdateGain>> coarse_gains_;
std::vector<FilterMisadjustmentEstimator> filter_misadjustment_estimators_;
std::vector<size_t> poor_coarse_filter_counters_;
std::vector<int> coarse_filter_reset_hangover_;
std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
refined_frequency_responses_;
std::vector<std::vector<float>> refined_impulse_responses_;
std::vector<std::vector<float>> coarse_impulse_responses_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_