Speech Enhancement by Classification of Noisy Signals Decomposed Using NMF and Wiener Filtering (original) (raw)
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Digital Signal Processing, 2020
In this paper, we propose a novel single-channel speech enhancement algorithm that applies dualdomain transforms comprising of dual-tree complex wavelet transform (DTCWT) and short-time Fourier transform (STFT) with a sparse non-negative matrix factorization (SNMF). The first domain belongs to the DTCWT, which is utilized on the time domain signals to conquer the weakness of signal distortions brought about by the downsampling of the discrete wavelet packet transform (DWPT) and delivered a set of subband signals. The second domain alludes to the STFT, which is exploited to each subband signal and built a complex spectrogram. At last, we apply the SNMF to the magnitude spectrogram for extracting speech components. In short, the DTCWT decomposes the time-domain noisy signal into a set of subband signals and afterward applied STFT to each subband signal, and we get nonnegative matrices by taking the absolute value of the complex matrix. From this point forward, we apply SNMF to each nonnegative matrix and identify the speech components. Finally, the estimated signal can be achieved through a subband binary ratio mask (SBRM) by applying the inverse STFT (ISTFT) and, subsequently, the inverse DTCWT (IDTCWT). The proposed approach is assessed utilizing the GRID audiovisual and IEEE databases, and diverse kinds of noises such as stationary, non-stationary, and quasi-stationary. The exploratory outcomes demonstrate that the proposed algorithm improved objective speech quality and intelligibility altogether at all considered signal to noise ratios (SNRs), compared to the other seven speech enhancement methods of STFT-SNMF, STFT-SNMFSE, MLD-STFT-SNMF, STFT-GDL, STFT-CJSR, DTCWT-SNMF, and DWPT-STFT-SNMF.
Model Order Selection for Non-Negative Matrix Factorization with Application to Speech Enhancement
This report deals with the application of non-negative matrix factorization (NMF) in speech processing. A Bayesian NMF is used to find the optimal number of basis vectors for the speech signal. The result is validated by performing a speech enhancement task for a set of different number of basis vectors. The algorithm performance is measured with the Source to Distortion Ratio (SDR) that represents the overall quality of speech. The results show that for medium input SNRs, 60 basis vectors for each speaker are sufficient to model the speech spectrogram. NMF produced better SDR results than a recently developed version of Spectral Subtraction algorithm. The window length was found to have a great effect on the results, but zero padding did not influence the results.
Speech enhancement based on nonnegative matrix factorization with mixed group sparsity constraint
Proceedings of the Sixth International Symposium on Information and Communication Technology - SoICT 2015, 2015
This paper addresses a challenging single-channel speech enhancement problem in real-world environment where speech signal is corrupted by high level background noise. While most state-of-the-art algorithms tries to estimate noise spectral power and filter it from the observed one to obtain enhanced speech, the paper discloses another approach inspired from audio source separation technique. In the considered method, generic spectral characteristics of speech and noise are first learned from various training signals by non-negative matrix factorization (NMF). They are then used to guide the similar factorization of the observed power spectrogram into speech part and noise part. Additionally, we propose to combine two existing group sparsity-inducing penalties in the optimization process and adapt the corresponding algorithm for parameter estimation based on multiplicative update (MU) rule. Experiment results over different settings confirm the effectiveness of the proposed approach.
Wavelet Speech Enhancement Based on Nonnegative Matrix Factorization
IEEE Signal Processing Letters, 2016
For most of the state-of-the-art speech enhancement techniques, a spectrogram is usually preferred than the respective time-domain raw data since it reveals more compact presentation together with conspicuous temporal information over a long time span. However, the short-time Fourier transform (STFT) that creates the spectrogram in general distorts the original signal and thereby limits the capability of the associated speech enhancement techniques. In this study, we propose a novel speech enhancement method that adopts the algorithms of discrete wavelet packet transform (DWPT) and nonnegative matrix factorization (NMF) in order to conquer the aforementioned limitation. In brief, the DWPT is first applied to split a timedomain speech signal into a series of subband signals without introducing any distortion. Then we exploit NMF to highlight the speech component for each subband. Finally, the enhanced subband signals are joined together via the inverse DWPT to reconstruct a noisereduced signal in time domain. We evaluate the proposed DWPT-NMF based speech enhancement method on the MHINT task. Experimental results show that this new method behaves very well in prompting speech quality and intelligibility and it outperforms the convnenitional STFT-NMF based method.
2011 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), 2011
In this paper, a linear MMSE filter is derived for single-channel speech enhancement which is based on Nonnegative Matrix Factorization (NMF). Assuming an additive model for the noisy observation, an estimator is obtained by minimizing the mean square error between the clean speech and the estimated speech components in the frequency domain. In addition, the noise power spectral density (PSD) is estimated using NMF and the obtained noise PSD is used in a Wiener filtering framework to enhance the noisy speech. The results of the both algorithms are compared to the result of the same Wiener filtering framework in which the noise PSD is estimated using a recently developed MMSE-based method. NMF based approaches outperform the Wiener filter with the MMSE-based noise PSD tracker for different measures. Compared to the NMF-based Wiener filtering approach, Source to Distortion Ratio (SDR) is improved for the evaluated noise types for different input SNRs using the proposed linear MMSE filter.
Multiple-order non-negative matrix factorization for speech enhancement
Amongst the speech enhancement techniques, statistical models based on Non-negative Matrix Factorization (NMF) have received great attention. In a single channel configuration, NMF is used to describe the spectral content of both the speech and noise sources. As the number of components can have a crucial influence on separation quality, we here propose to investigate model order selection based on the variational Bayesian approximation to the marginal likelihood of models of different orders. To go further, we propose to use model averaging to combine several single-order NMFs and we show that a straightforward application of model averaging principles is inefficient as it turned out to be equivalent to model selection. We thus introduce a parameter to control the entropy of the model order distribution which makes the averaging effective. We also show that our probabilistic model nicely extends to a multipleorder NMF model where several NMFs are jointly estimated and averaged. Experiments are conducted on real data from the CHiME challenge and give an interesting insight on the entropic parameter and model order priors. Separation results are also promising as model averaging outperforms single-order model selection. Finally, our multiple-order NMF shows an interesting gain in computation time.
Recognize and separate approach for speech denoising using nonnegative matrix factorization
2015 23rd European Signal Processing Conference (EUSIPCO), 2015
This paper proposes a novel approach for denoising single-channel noisy speech signals. A speech dictionary and multiple noise dictionaries are trained using nonnegative matrix factorization (NMF). After observing the mixed signal, first the type of noise in the mixed signal is identified. The magnitude spectrogram of the noisy signal is decomposed using NMF with the concatenated trained dictionaries of noise and speech. Our results indicate that recognizing the noise type from the mixed signal and using the corresponding specific noise dictionary provides better results than using a general noise dictionary in the NMF approach. We also compare our algorithm with other state-of-the-art denoising methods and show that it has better performance than the competitors in most cases.
Journal of Signal Processing Systems, 2019
In this paper, we propose a novel speech enhancement method based on dual-tree complex wavelet transforms (DTCWT) and nonnegative matrix factorization (NMF) that exploits the subband smooth ratio mask (ssRM) through a joint learning process. The discrete wavelet packet transform (DWPT) suffers the absence of shift invariance, due to downsampling after the filtering process, resulting in a reconstructed signal with significant noise. The redundant stationary wavelet transform (SWT) can solve this shift invariance problem. In this respect, we use efficient DTCWT with a shift invariance property and limited redundancy and calculate the ratio masks (RMs) between the clean training speech and noisy speech (i.e., training noise mixed with clean speech). We also compute RMs between the noise and noisy speech and then learn both RMs with their corresponding clean training clean speech and noise. The auto-regressive moving average (ARMA) filtering process is applied before NMF in previously generated matrices for smooth decomposition. An ssRM is proposed to exploit the advantage of the joint use of the standard ratio mask (sRM) and square root ratio mask (srRM). In short, the DTCWT produces a set of subband signals employing the time-domain signal. Subsequently, the framing scheme is applied to each subband signal to form matrices and calculates the RMs before concatenation with the previously generated matrices. The ARMA filter is implemented in the nonnegative matrix, which is formed by considering the absolute value. Through ssRM, speech components are detected using NMF in each newly formed matrix. Finally, the enhanced speech signal is obtained via the inverse DTCWT (IDTCWT). The performances are evaluated by considering an IEEE corpus, the GRID audiovisual corpus, and different types of noises. The proposed approach significantly improves objective speech quality and intelligibility and outperforms the conventional STFT-NMF, DWPT-NMF, and DNN-IRM methods.
Speech denoising using nonnegative matrix factorization with priors
2008
Abstract We present a technique for denoising speech using nonnegative matrix factorization (NMF) in combination with statistical speech and noise models. We compare our new technique to standard NMF and to a state-of-the-art Wiener filter implementation and show improvements in speech quality across a range of interfering noise types.