Source code
Revision control
Copy as Markdown
Other Tools
/*
* 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_