Per Åhgren d4e6904d40 AEC3: Reducing the complexity and heap usage of the adaptive filter
This CL reduces the complexity and heap usage of the adaptive filter
in AEC3 by avoiding to compute these for the shadow
filter. In particular it
-Moves to compute the ERL, frequency response and impulse response
 on an on-demand basis.
-Stores the ERL, frequency response and impulse response outside
 of the adaptive filter.

All the changes have been tested for bitexactness on a sizeable
amount of recordings.

Bug: webrtc:10913
Change-Id: If83c236a6e3f2e489be129b9ebf6143a72f521d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151138
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29081}
2019-09-05 14:30:49 +00:00

132 lines
4.7 KiB
C++

/*
* 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/echo_path_variability.h"
#include "modules/audio_processing/aec3/main_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/shadow_filter_update_gain.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 rtc::ArrayView<const float> capture,
const RenderSignalAnalyzer& render_signal_analyzer,
const AecState& aec_state,
SubtractorOutput* output);
void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
// Exits the initial state.
void ExitInitialState();
// Returns the block-wise frequency response for the main adaptive filter.
const std::vector<std::array<float, kFftLengthBy2Plus1>>&
FilterFrequencyResponse() const {
return main_frequency_response_;
}
// Returns the estimate of the impulse response for the main adaptive filter.
const std::vector<float>& FilterImpulseResponse() const {
return main_impulse_response_;
}
void DumpFilters() {
size_t current_size = main_impulse_response_.size();
main_impulse_response_.resize(main_impulse_response_.capacity());
data_dumper_->DumpRaw("aec3_subtractor_h_main", main_impulse_response_);
main_impulse_response_.resize(current_size);
main_filter_.DumpFilter("aec3_subtractor_H_main");
shadow_filter_.DumpFilter("aec3_subtractor_H_shadow");
}
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_;
AdaptiveFirFilter main_filter_;
AdaptiveFirFilter shadow_filter_;
MainFilterUpdateGain G_main_;
ShadowFilterUpdateGain G_shadow_;
FilterMisadjustmentEstimator filter_misadjustment_estimator_;
size_t poor_shadow_filter_counter_ = 0;
std::vector<std::array<float, kFftLengthBy2Plus1>> main_frequency_response_;
std::vector<float> main_impulse_response_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_