webrtc/modules/audio_processing/ns/signal_model_estimator.h
Per Åhgren 87a7b82520 Refactoring of the noise suppressor and adding true multichannel support
This CL adds proper multichannel support to the noise suppressor.
To accomplish that in a safe way, a full refactoring of the noise
suppressor code has been done.

Due to floating point precision, the changes made are not entirely
bitexact. They are, however, very close to being bitexact.

As a safety measure, the former noise suppressor code is preserved
and a kill-switch is added to allow revering to the legacy noise
suppressor in case issues arise.

Bug: webrtc:10895, b/143344262
Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29646}
2019-10-29 23:23:38 +00:00

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/*
* Copyright (c) 2019 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_SIGNAL_MODEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_
#include <array>
#include "api/array_view.h"
#include "modules/audio_processing/ns/histograms.h"
#include "modules/audio_processing/ns/ns_common.h"
#include "modules/audio_processing/ns/prior_signal_model.h"
#include "modules/audio_processing/ns/prior_signal_model_estimator.h"
#include "modules/audio_processing/ns/signal_model.h"
namespace webrtc {
class SignalModelEstimator {
public:
SignalModelEstimator();
SignalModelEstimator(const SignalModelEstimator&) = delete;
SignalModelEstimator& operator=(const SignalModelEstimator&) = delete;
// Compute signal normalization during the initial startup phase.
void AdjustNormalization(int32_t num_analyzed_frames, float signal_energy);
void Update(
rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
float signal_spectral_sum,
float signal_energy);
const PriorSignalModel& get_prior_model() const {
return prior_model_estimator_.get_prior_model();
}
const SignalModel& get_model() { return features_; }
private:
float diff_normalization_ = 0.f;
float signal_energy_sum_ = 0.f;
Histograms histograms_;
int histogram_analysis_counter_ = 500;
PriorSignalModelEstimator prior_model_estimator_;
SignalModel features_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_