By Topic

Signal Processing Letters, IEEE

Issue 8 • Date Aug. 2005

Filter Results

Displaying Results 1 - 18 of 18
  • Table of contents

    Publication Year: 2005 , Page(s): c1 - c4
    Save to Project icon | Request Permissions | PDF file iconPDF (37 KB)  
    Freely Available from IEEE
  • IEEE Signal Processing Letters publication information

    Publication Year: 2005 , Page(s): c2
    Save to Project icon | Request Permissions | PDF file iconPDF (46 KB)  
    Freely Available from IEEE
  • An adaptive robust LMS employing fuzzy step size and partial update

    Publication Year: 2005 , Page(s): 545 - 548
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB) |  | HTML iconHTML  

    We propose a fuzzy step size least-mean-square (LMS) algorithm, in which an appropriate step size is selected by a fuzzy inference system (FIS) to achieve faster convergence rate and lower steady-state fluctuation. An adaptive fuzzy sequential partial update scheme is developed to reduce system complexity without trading off bit-error rate (BER) and convergence/tracking performance. Simulations of a DS-CDMA interference suppression receiver illustrate the robust convergence and tracking behavior of the proposed LMS-based approaches with various fuzzy input vectors and fuzzy sets. The performance advantages of the proposed algorithms over other LMS algorithms in multipath Rayleigh-fading channels are investigated as well. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Low-complexity search for optimal delay in linear FIR MMSE equalization

    Publication Year: 2005 , Page(s): 549 - 552
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (152 KB) |  | HTML iconHTML  

    In this letter, an efficient algorithm to determine the optimal delay in linear finite impulse response equalizers based on the minimum mean-square-error criterion is proposed. The algorithm uses the Levinson-Durbin (L-D) recursion as a starting point to find the values of the mean-square error for equalizers with all nontrivial delays. Despite the exhaustive search approach, the complexity of the proposed algorithm is only doubled when compared to the calculation of the equalizer with one prescribed delay. Such increase in complexity may be fully justified in practice because it yields globally optimal equalizer's design. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A Bayesian approach for the estimation of model parameters from noisy data sets

    Publication Year: 2005 , Page(s): 553 - 556
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (144 KB) |  | HTML iconHTML  

    A Bayesian method is proposed for estimating model parameters from noisy data sets. The method is based on maximizing the posterior kernel, which enables priors on the model parameters to be incorporated. The posterior kernel is found by specifying hyperpriors and integrating the priors out, due to the use of conjugate priors. The use of probability models enables simultaneous data streams to be used to maximize the posterior kernel. The solution is found using an iterative scheme. The algorithm's performance is briefly illustrated using a real data set, demonstrating rapid convergence. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Iterative wavelet-based denoising methods and robust outlier detection

    Publication Year: 2005 , Page(s): 557 - 560
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (200 KB) |  | HTML iconHTML  

    The goal of this letter is to study convergence conditions for a previously presented iterative wavelet denoising method and to shed light on its relationship with outlier rejection. This method involves a user-defined parameter, which must fulfill certain conditions in order to ensure denoising. Using generalized Gaussian modeling for the wavelet coefficients distribution, we obtain a lower bound for this parameter, and the resulting threshold, both adapted to the shape of the distribution. The properties of this threshold are examined, and the proposed method is compared with other classical rejection methods. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Estimation of direction of arrival using information theory

    Publication Year: 2005 , Page(s): 561 - 564
    Cited by:  Papers (23)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB) |  | HTML iconHTML  

    Estimating the direction of arrival (DOA) of an acoustic source relies on the successful estimation of the relative delay between pairs of microphone signals. Processing is performed at the current time by operating on blocks of recorded data. When these recordings are performed in environments of strong multipath reflections, algorithms often fail to distinguish between the true DOA and that of a dominant reflection. In this letter, we assume Gaussianity of the source signal and use an information-theoretical measure, often met in blind source separation algorithms, to derive a robust DOA estimator, even under significant reverberant conditions. We discuss the most popular algorithm for time delay estimation, namely, the generalized cross-correlation method, and demonstrate under certain conditions its connection to the proposed one. Performance is demonstrated for both algorithms with sets of simulated results as a function of different reverberation times, microphone spacing, and data block size. The results indicate that the examined framework can accurately track the DOA of a typical acoustic source. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Reduced-complexity ML decoding of rate 6/8 and rate 1 linear complex space-time codes for up to eight transmit antennas with phase feedback

    Publication Year: 2005 , Page(s): 565 - 568
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB) |  | HTML iconHTML  

    It is shown that single symbol maximum-likelihood (ML) decoding for chosen rate 6/8 linear complex space-time codes for up to eight transmit antennas is possible in the presence of a phase feedback. The performance of the code is the same as that of an ideal code, for which the sufficient statistic of the channel is its Frobenius norm. For rate 1 codes for up to eight antennas, a two-phase feedback enables a symbol to see the interference from only one symbol instead of three, thereby reducing the complexity of ML decoding. Quantization of the feedback is also considered. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • MMSE precoder for unitary space-time codes in correlated time-varying channels

    Publication Year: 2005 , Page(s): 569 - 572
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (192 KB) |  | HTML iconHTML  

    This letter addresses the problem of the design of a precoder for multiple transmit antenna communication systems with spatially and temporally correlated fading channels. By using the asymptotic (high signal-to-noise ratio) mean-square error of the channel estimates, the letter derives a precoder for unitary space-time codes that can exploit the spatiotemporal correlation in the time-varying fading channels. Simulation results illustrate that significant performance gains can be achieved by using the new precoder. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Digital removal of power frequency artifacts using a Fourier space median filter

    Publication Year: 2005 , Page(s): 573 - 576
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1448 KB) |  | HTML iconHTML  

    An efficient technique to remove periodic noise from digital images via a novel implementation of the median filter is presented and discussed within this letter. The technique is based on applying the nonlinear filter to the Fourier amplitude spectrum of the image. Since the median filter is normally used to spatially reduce spiky noise in digital images, using this filter in Fourier space is especially useful for removing periodic frequency noise. Provisional experimental results are shown that demonstrate that the proposed filter provides good performance when compared with a similar Fourier filter: the notch filter. This letter, when considering a prototype version of a Fourier-based median filter, defines the set of user-defined options that are available and also defines a possible solution to solve the distortions caused around the dc component due to natural spectrum decay. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Prioritized transmission of data partitioned H.264 video with hierarchical QAM

    Publication Year: 2005 , Page(s): 577 - 580
    Cited by:  Papers (30)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB) |  | HTML iconHTML  

    In this letter, hierarchical quadrature amplitude modulation (HQAM) is used to provide unequal error protection (UEP) for layered (data partitioned) H.264 coded video. In a conventional HQAM system, the high-priority (HP) and low-priority (LP) capacities have a constant ratio, whereas in H.264 data partitioning, the corresponding parts do not necessarily have this constant ratio. This letter proposes a multilevel HQAM arrangement with adaptive constellation distances that provides a graceful degradation in the quality of the decoded video without requiring feedback from the receiver. The arrangement improves the quality relative to nonhierarchical transmission through poor signal-to-noise ratio (SNR) channels at the price of a modest quality reduction through good SNR channels. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Concurrent data transmission through analog speech channel using data hiding

    Publication Year: 2005 , Page(s): 581 - 584
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB) |  | HTML iconHTML  

    In this letter, we propose to perform concurrent data transmissions through analog speech channels using a data hiding technique. In particular, concurrent data are modulated into a noise-like signal and imperceptibly embedded into voices. Conventional embedding methods usually treat speech as a source of interferences to the simultaneously transmitted data. The proposed scheme exploits the knowledge of the speech signal during embedding. Therefore, such an interference can be effectively removed, and a higher data rate can be achieved. Besides, the proposed scheme is compatible with conventional phone terminals. A conventional telephone set is still able to access the basic voice service, while the added decoding mechanism can extract the embedded data and provide various concurrent services. The performance of the proposed method is analyzed. Experimental results demonstrate that an acceptable data rate and bit-error rate can be achieved. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Detection of confusable words in automatic speech recognition

    Publication Year: 2005 , Page(s): 585 - 588
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB) |  | HTML iconHTML  

    A new method to detect words that are likely to be confused by speech recognition systems is presented in this letter. A new dissimilarity measure between two words is calculated in two steps. First, the phonetic transcriptions of the words are aligned using only phonetic information. Two kinds of alignments are used: either with or without insertions and deletions. Second, the dissimilarity measure is calculated on the basis of the resulting alignment and acoustic information obtained from the hidden Markov models of the phones. In a classical false acceptance/false rejection framework, the equal error rate was measured to be less than 5%. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Variable step-size sign natural gradient algorithm for sequential blind source separation

    Publication Year: 2005 , Page(s): 589 - 592
    Cited by:  Papers (18)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB) |  | HTML iconHTML  

    A novel variable step-size sign natural gradient algorithm (VS-S-NGA) for online blind separation of independent sources is presented. A sign operator for the adaptation of the separation model is obtained from the derivation of a generalized dynamic separation model. A variable step size is also derived to better match the dynamics of the input signals and unmixing matrix. The proposed sign algorithm is appealing in practice due to its computational simplicity. Experimental results verify the superior convergence performance over conventional NGA in both stationary and nonstationary environments. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • IEEE Signal Processing Letters Information for authors

    Publication Year: 2005 , Page(s): 593 - 594
    Save to Project icon | Request Permissions | PDF file iconPDF (46 KB)  
    Freely Available from IEEE
  • IEEE Signal Processing Letters Edics

    Publication Year: 2005 , Page(s): 595
    Save to Project icon | Request Permissions | PDF file iconPDF (22 KB)  
    Freely Available from IEEE
  • Special issue on blind signal processing for speech and audio applications

    Publication Year: 2005 , Page(s): 596
    Save to Project icon | Request Permissions | PDF file iconPDF (104 KB)  
    Freely Available from IEEE
  • IEEE Signal Processing Society Information

    Publication Year: 2005 , Page(s): c3
    Save to Project icon | Request Permissions | PDF file iconPDF (34 KB)  
    Freely Available from IEEE

Aims & Scope

The IEEE Signal Processing Letters is a monthly, archival publication designed to provide rapid dissemination of original, cutting-edge ideas and timely, significant contributions in signal, image, speech, language and audio processing.

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Peter Willett
University of Connecticut
Storrs, CT 06269
peter.willett@uconn.edu