93 lines
3.3 KiB
C++
93 lines
3.3 KiB
C++
/*
|
|
* Copyright (c) 2012 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_NS_NOISE_SUPPRESSOR_H_
|
|
#define MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/audio_buffer.h"
|
|
#include "modules/audio_processing/ns/noise_estimator.h"
|
|
#include "modules/audio_processing/ns/ns_common.h"
|
|
#include "modules/audio_processing/ns/ns_config.h"
|
|
#include "modules/audio_processing/ns/ns_fft.h"
|
|
#include "modules/audio_processing/ns/speech_probability_estimator.h"
|
|
#include "modules/audio_processing/ns/wiener_filter.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for suppressing noise in a signal.
|
|
class NoiseSuppressor {
|
|
public:
|
|
NoiseSuppressor(const NsConfig& config,
|
|
size_t sample_rate_hz,
|
|
size_t num_channels);
|
|
NoiseSuppressor(const NoiseSuppressor&) = delete;
|
|
NoiseSuppressor& operator=(const NoiseSuppressor&) = delete;
|
|
|
|
// Analyses the signal (typically applied before the AEC to avoid analyzing
|
|
// any comfort noise signal).
|
|
void Analyze(const AudioBuffer& audio);
|
|
|
|
// Applies noise suppression.
|
|
void Process(AudioBuffer* audio);
|
|
|
|
// Specifies whether the capture output will be used. The purpose of this is
|
|
// to allow the noise suppressor to deactivate some of the processing when the
|
|
// resulting output is anyway not used, for instance when the endpoint is
|
|
// muted.
|
|
void SetCaptureOutputUsage(bool capture_output_used) {
|
|
capture_output_used_ = capture_output_used;
|
|
}
|
|
|
|
private:
|
|
const size_t num_bands_;
|
|
const size_t num_channels_;
|
|
const SuppressionParams suppression_params_;
|
|
int32_t num_analyzed_frames_ = -1;
|
|
NrFft fft_;
|
|
bool capture_output_used_ = true;
|
|
|
|
struct ChannelState {
|
|
ChannelState(const SuppressionParams& suppression_params, size_t num_bands);
|
|
|
|
SpeechProbabilityEstimator speech_probability_estimator;
|
|
WienerFilter wiener_filter;
|
|
NoiseEstimator noise_estimator;
|
|
std::array<float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum;
|
|
std::array<float, kFftSize - kNsFrameSize> analyze_analysis_memory;
|
|
std::array<float, kOverlapSize> process_analysis_memory;
|
|
std::array<float, kOverlapSize> process_synthesis_memory;
|
|
std::vector<std::array<float, kOverlapSize>> process_delay_memory;
|
|
};
|
|
|
|
struct FilterBankState {
|
|
std::array<float, kFftSize> real;
|
|
std::array<float, kFftSize> imag;
|
|
std::array<float, kFftSize> extended_frame;
|
|
};
|
|
|
|
std::vector<FilterBankState> filter_bank_states_heap_;
|
|
std::vector<float> upper_band_gains_heap_;
|
|
std::vector<float> energies_before_filtering_heap_;
|
|
std::vector<float> gain_adjustments_heap_;
|
|
std::vector<std::unique_ptr<ChannelState>> channels_;
|
|
|
|
// Aggregates the Wiener filters into a single filter to use.
|
|
void AggregateWienerFilters(
|
|
rtc::ArrayView<float, kFftSizeBy2Plus1> filter) const;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
|