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A new approach for speech enhancement based on a constrained Nonnegative Matrix Factorization

A new approach for speech enhancement based on a constrained...

引用

19th IEEE International Symposium on Intelligent Signal processing and Communication Systems, ISPACS 2011

作者： Mohammadiha, Nasser Gerkmann, Timo Leijon, Arne Sound and Image Processing Lab. KTH Royal Institute of Technology Stockholm Sweden

ISBN: (纸本)9781457721663

In this paper, a new approach is presented for single-channel speech enhancement which is based on Nonnegative Matrix Factorization (NMF). The proposed scheme combines the noise Power Spectral Density (PSD) estimation based on a constrained NMF and Wiener filtering to enhance the noisy speech. The imposed constraint is motivated by the time correlation of the underlying observations and enforces the NMF to give smoother estimates of the nonnegative factors. Compared to the standard NMF approach and Wiener filtering based on a recently developed noise PSD estimator, Source to Distortion Ratio (SDR) is improved for the evaluated noise types for different input SNRs. © 2011 IEEE.

关键词： Speech enhancement

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Fast amplitude compression in hearing aids improves audibility but degrades speech information transmission

Fast amplitude compression in hearing aids improves audibili...

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16th European Signal processing Conference, EUSIPCO 2008

作者： Leijon, Arne Stadler, Svante Sound and Image Processing Lab. School of Electrical Engineering KTH SE-10044 Stockholm Sweden

Common types of hearing impairment are caused mainly by a loss of nearly instantaneous compressive amplification in the inner ear. Therefore, it seems plausible that the loss might be compensated by fast frequency-dependent compression in the hearing aid. We simulated impaired listeners' auditory analysis of hearing-aid processed speech in noise using a functional auditory model. Using hidden Markov signal models, we estimated the mutual information between the phonetic structure of clean speech and the neural output from the auditory model, with fast and slow versions of hearing-aid compression. The long-term speech spectrum of amplified sound was identical in both systems, as specified individually by the widely accepted NAL prescription for the gain frequency response. The calculation showed clearly better speech-to-auditory information transmission with slow quasi-linear amplification than with fast hearing-aid compression, for speech in speech-shaped noise at signal-to-noise ratios ranging from -10 to +20 dB. copyright by EURASIP.

关键词： Signal to noise ratio

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PDF-optimized LSF vector quantization based on beta mixture models

PDF-optimized LSF vector quantization based on beta mixture ...

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作者： Ma, Zhanyu Leijon, Arne KTH Royal Institute of Technology Sound and Image Processing Lab. Stockholm Sweden

The line spectral frequencies (LSF) are known to be the most efficient representation of the linear predictive coding (LPC) parameters from both the distortion and perceptual point of view. By considering the bounded property of the LSF parameters, we apply beta mixture models (BMM) to model the distribution of the LSF parameters. Meanwhile, by following the principles of probability density function (PDF) optimized vector quantization (VQ), we derive the bit allocation strategy for the BMM. The LSF parameters are obtained from the TIMIT database and a practical VQ is designed. By taking the Bayesian information criterion (BIC), the square error (SE) and the spectral distortion (SD) as the criteria, the BMM based VQ outperforms the Gaussian mixture model based VQ with uncorrelated Gaussian component (UGMVQ) by about 1 2 bits/vector. © 2010 ISCA.

关键词： Vector quantization

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Modelling speech line spectral frequencies with Dirichlet mixture models

Modelling speech line spectral frequencies with Dirichlet mi...

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作者： Ma, Zhanyu Leijon, Arne KTH Royal Institute of Technology Sound and Image Processing Lab. Stockholm Sweden

In this paper, we model the underlying probability density function (PDF) of the speech line spectral frequencies (LSF) parameters with a Dirichlet mixture model (DMM). The LSF parameters have two special features: 1) the LSF parameters have a bounded range;2) the LSF parameters are in an increasing order. By transforming the LSF parameters to the ΔLSF parameters, the DMM can be used to model the ΔLSF parameters and take advantage of the features mentioned above. The distortion-rate (D-R) relation is derived for the Dirichlet distribution with the high rate assumption. A bit allocation strategy for DMM is also proposed. In modelling the LSF parameters extracted from the TIMIT database, the DMM shows a better performance compared to the Gaussian mixture model, in terms of D-R relation, likelihood and model complexity. Since modelling is the essential and prerequisite step in the PDF-optimized vector quantizer design, better modelling results indicate a superior quantization performance. © 2010 ISCA.

关键词： Probability density function

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Coding bounded support data with beta distribution

Coding bounded support data with beta distribution

引用

International Conference on Network Infrastructure and Digital Content

作者： Ma, Zhanyu Leijon, Arne Sound and Image Processing Lab. School of Electrical Engineering KTH-Royal Institute of Technoloy Stockholm Sweden

ISBN: (纸本)9781424468546

The probability density function (PDF) optimized quantization has been shown to be more efficient than the conventional quantization methods. In practical application, the data with bounded support can be modelled better with bounded support distribution (e.g. beta distribution, Dirichlet distribution) and a better quantization performance could be achieved by a more reasonable modelling. In this paper, we study the distortion rate (D-R) performance and the high rate quantization performance of the beta distribution. To implement a quantizer efficiently, a practical quantization scheme is proposed. The proposed scheme takes the advantages of conventional compander and exhaustive training. The advantage of the proposed scheme is verified with both theoretical experiment and practical application. ©2010 IEEE.

关键词： Probability density function

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Comparison of transmitter-based packet-loss recovery techniques for voice transmission 8

Comparison of transmitter-based packet-loss recovery techniq...

引用

8th International Conference on Spoken Language processing, ICSLP 2004

作者： Kim, Moo Young Kleijn, W. Bastiaan Sound and Image Processing Lab. Dept. Signals Sensors and Systems KTH Royal Institute of Technology Sweden

To facilitate real-time voice communication through the Internet, forward error correction (FEC) and multiple description coding (MDC) can be used as low-delay packet-loss recovery techniques. We use both a Gilbert channel model and data obtained from real IP connections to compare the rate-distortion performance of different variants of FEC and MDC. Using identical overall rates with stringent delay constraints, we find that side-distortion optimized MDC generally performs better than Reed-Solomon based FEC. If the channel condition is known from feedback through the Real-Time Control Protocol (RTCP), then channel-optimized MDC can be used to exploit this information, resulting in significantly improved performance.

关键词： Forward error correction

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Analysis of communication systems using iterative methods based on Banach's contraction principle

Analysis of communication systems using iterative methods ba...

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2007 6th International Conference on Information, Communications and Signal processing, ICICS

作者： Azari Soufiani, H. Saberian, M.J. Akhaee, M.A. Nasiri Mahallati, R. Marvasti, F. Multimedia Signal Sound and Image Processing Lab. Advanced Communications Research Institute of Sharif University of Technology Tehran Iran

ISBN: (纸本)1424409837

In this paper, the application of a well known mathematical theorem, Banach's fixed point theorem [1], is investigated in iterative signal processing in communications. In most practical communication systems some sort of a contraction mapping is used to enhance the operation of the system. Thus, using a suitable iterative approach, one can set the system in its fixed point and hence, the distortion produced in the transmitter, channel and the receiver can be compensated. In other words, a loosely designed transceiver can be enhanced by an iterative method. In order to verify the truth of the proposed iterative method, the distortion of A/D and D/A converters is compensated at the receiver. Although analyses are performed on a linear system, they could be generalized to non linear ones. ©2007 IEEE.

关键词： Iterative methods

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MATCHING PURSUIT FOR CHANNEL SELECTION IN COCHLEAR IMPLANTS BASED ON AN INTELLIGIBILITY METRIC

MATCHING PURSUIT FOR CHANNEL SELECTION IN COCHLEAR IMPLANTS ...

引用

20th European Signal processing Conference (EUSIPCO)

作者： Taal, C. H. Hendriks, R. C. Heusdens, R. Royal Inst Technol KTH Sound & Image Proc Lab Stockholm Sweden Delft Univ Technol Signal & Informat Processing Lab NL-2600 AA Delft Netherlands

ISBN: (纸本)9781467310680

In this paper the earlier proposed short-time objective intelligibility predictor (STOI) is simplified such that it can be expressed as a weighted l(2) norm in the auditory domain. Due to the mathematical properties of a norm, STOI can now be used with the matching pursuit algorithm in the n-of-m channel selection technique as found in several cochlear implant (CI) coding strategies. With this technique only a subset of frequency channels (electrodes) are stimulated, such that important channels can be updated more frequently and less significant channels are omitted. Intelligibility predictions with acoustic CI-simulations for normal-hearing listeners indicate that more intelligible speech is obtained with the proposed method compared to a conventional channel selection method based on peak picking. Reasons for this difference in performance are: (1) STOI considers an analysis window of a few hundreds of milliseconds in order to account for important low temporal modulations for speech intelligibility and (2) spectral leakage per channel is accounted for in the mathematical optimization process.

关键词： Speech intelligibility metric matching pursuit cochlear implants channel selection

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A Bayesian hierarchical mixture of experts approach to estimate speech quality

A Bayesian hierarchical mixture of experts approach to estim...

引用

2010 2nd International Workshop on Quality of Multimedia Experience, QoMEX 2010

作者： Mossavat, S. Iman Amft, Oliver De Vries, Bert Petkov, Petko N. Kleijn, W. Bastiaan Signal Processing Systems Department of Electrical Engineering Eindhoven University of Technology Netherlands Sound and Image Processing Lab. Department of Electrical Engineering KTH-Royal Institute of Technology Sweden

ISBN: (纸本)9781424469604

This paper demonstrates the potential of theoretically motivated learning methods in solving the problem of non-intrusive quality estimation for which the state-of-the-art is represented by ITU-T P.563 standard. To construct our estimator, we adopt the speech features from P.563, while we use a different mapping of features to form quality estimates. In contrast to P.563 which assumes distortion-classes to divide the feature space, our approach divides the feature space based on a clustering which is learned from the data using Bayesian inference. Despite using weaker modeling assumptions, we are still able to achieve comparable accuracy on predicting mean-opinion-scores with P.563. Our work suggests Bayesian model-evidence as an alternative metric to correlation-coefficient for determining the necessary number of experts for modeling the data. ©2010 IEEE.

关键词： Bayesian networks

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An investigation on mutual information for the linear predictive system and the extrapolation of speech signals 10

An investigation on mutual information for the linear predic...

引用

10th ITG Symposium on Speech Communication, ITGspeech 2012

作者： Taghia, Jalal Martin, Rainer Taghia, Jalil Leijon, Arne Institute of Communication Acoustics Ruhr-Universität Bochum Bochum Germany Sound and Image Processing Lab. KTH Royal Institute of Technology Stockholm Sweden

ISBN: (纸本)9783800734559

Mutual information (MI) is an important information theoretic concept which has many applications in telecommunications, in blind source separation, and in machine learning. More recently, it has been also employed for the instrumental assessment of speech intelligibility where traditionally correlation based measures are used. In this paper, we address the difference between MI and correlation from the viewpoint of discovering dependencies between variables in the context of speech signals. We perform our investigation by considering the linear predictive approximation and the extrapolation of speech signals as examples. We compare a parametric MI estimation approach based on a Gaussian mixture model (GMM) with the k-nearest neighbor (KNN) approach which is a well-known non-parametric method available to estimate the MI. We show that the GMM-based MI estimator leads to more consistent results. © VDE VERLAG GMBH • Berlin • Offenbach.

关键词： Extrapolation

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