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15179a9986
This CL adds functionality in the noise suppressor that allows the computational complexity to be reduced when the output of APM is not used. Bug: b/177830919 Change-Id: I849351ba9559fae770e4667d78e38abde5230eed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211342 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33477}
92 lines
3.3 KiB
C++
92 lines
3.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
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#define MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/ns/noise_estimator.h"
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#include "modules/audio_processing/ns/ns_common.h"
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#include "modules/audio_processing/ns/ns_config.h"
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#include "modules/audio_processing/ns/ns_fft.h"
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#include "modules/audio_processing/ns/speech_probability_estimator.h"
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#include "modules/audio_processing/ns/wiener_filter.h"
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namespace webrtc {
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// Class for suppressing noise in a signal.
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class NoiseSuppressor {
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public:
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NoiseSuppressor(const NsConfig& config,
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size_t sample_rate_hz,
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size_t num_channels);
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NoiseSuppressor(const NoiseSuppressor&) = delete;
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NoiseSuppressor& operator=(const NoiseSuppressor&) = delete;
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// Analyses the signal (typically applied before the AEC to avoid analyzing
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// any comfort noise signal).
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void Analyze(const AudioBuffer& audio);
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// Applies noise suppression.
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void Process(AudioBuffer* audio);
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// Specifies whether the capture output will be used. The purpose of this is
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// to allow the noise suppressor to deactivate some of the processing when the
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// resulting output is anyway not used, for instance when the endpoint is
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// muted.
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void SetCaptureOutputUsage(bool capture_output_used) {
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capture_output_used_ = capture_output_used;
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}
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private:
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const size_t num_bands_;
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const size_t num_channels_;
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const SuppressionParams suppression_params_;
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int32_t num_analyzed_frames_ = -1;
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NrFft fft_;
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bool capture_output_used_ = true;
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struct ChannelState {
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ChannelState(const SuppressionParams& suppression_params, size_t num_bands);
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SpeechProbabilityEstimator speech_probability_estimator;
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WienerFilter wiener_filter;
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NoiseEstimator noise_estimator;
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std::array<float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum;
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std::array<float, kFftSize - kNsFrameSize> analyze_analysis_memory;
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std::array<float, kOverlapSize> process_analysis_memory;
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std::array<float, kOverlapSize> process_synthesis_memory;
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std::vector<std::array<float, kOverlapSize>> process_delay_memory;
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};
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struct FilterBankState {
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std::array<float, kFftSize> real;
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std::array<float, kFftSize> imag;
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std::array<float, kFftSize> extended_frame;
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};
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std::vector<FilterBankState> filter_bank_states_heap_;
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std::vector<float> upper_band_gains_heap_;
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std::vector<float> energies_before_filtering_heap_;
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std::vector<float> gain_adjustments_heap_;
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std::vector<std::unique_ptr<ChannelState>> channels_;
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// Aggregates the Wiener filters into a single filter to use.
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void AggregateWienerFilters(
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rtc::ArrayView<float, kFftSizeBy2Plus1> filter) const;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
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