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Signal Processing, IEEE Transactions on

Issue 4 • Date April 2002

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Displaying Results 1 - 24 of 24
  • Quasi-maximum-likelihood multiuser detection using semi-definite relaxation with application to synchronous CDMA

    Publication Year: 2002 , Page(s): 912 - 922
    Cited by:  Papers (154)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (373 KB) |  | HTML iconHTML  

    The maximum-likelihood (ML) multiuser detector is well known to exhibit better bit-error-rate (BER) performance than many other multiuser detectors. Unfortunately,ML detection (MLD) is a nondeterministic polynomial-time hard (NP-hard) problem, for which there is no known algorithm that can find the optimal solution with polynomial-time complexity (in the number of users). In this paper, a polynomial-time approximation method called semi-definite (SD) relaxation is applied to the MLD problem with antipodal data transmission. SD relaxation is an accurate approximation method for certain NP-hard problems. The SD relaxation ML (SDR-ML) detector is efficient in that its complexity is of the order of K3.5, where K is the number of users. We illustrate the potential of the SDR-ML detector by showing that some existing detectors, such as the decorrelator and the linear-minimum-mean-square-error detector, can be interpreted as degenerate forms of the SDR-ML detector. Simulation results indicate that the BER performance of the SDR-ML detector is better than that of these existing detectors and is close to that of the true ML detector, even when the cross-correlations between users are strong or the near-far effect is significant. View full abstract»

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  • Correction to "Nonlinear kalman filtering with semi-parametric biscay distributions"

    Publication Year: 2002 , Page(s): 990
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (167 KB) |  | HTML iconHTML  

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  • Correction to "On the behavior of current second and higher order blind source separation methods for cyclostationary sources"

    Publication Year: 2002 , Page(s): 990
    Cited by:  Papers (2)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (167 KB) |  | HTML iconHTML  

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  • The fractional discrete cosine transform

    Publication Year: 2002 , Page(s): 902 - 911
    Cited by:  Papers (21)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (398 KB) |  | HTML iconHTML  

    The extension of the Fourier transform operator to a fractional power has received much attention in signal theory and is finding attractive applications. The paper introduces and develops the fractional discrete cosine transform (DCT) on the same lines, discussing multiplicity and computational aspects. Similarities and differences with respect to the fractional Fourier transform are pointed out View full abstract»

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  • Frequency estimation from proper sets of correlations

    Publication Year: 2002 , Page(s): 791 - 802
    Cited by:  Papers (27)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    As a complement to the periodogram, low-complexity frequency estimators are of interest. One such estimator is based on Prony's method and rely on phase information of the auto correlations. Without prior knowledge of the frequency (e.g., a given frequency interval), the frequency cannot be unambiguously estimated from a single correlation only. We introduce a new method of phase unwrapping using an arbitrary number (more than one) of correlations. From this arbitrary set of correlations, we propose a weighted average estimator. We derive the asymptotic performance and show how the correlation lags should be properly chosen. From a design aspect, there is often a restriction of using a fixed number of computations. In addition, we therefore propose a strategy to find a proper set of correlation lags subject to a given computational complexity. Finally, simulation results that lend support to the theoretical findings are included View full abstract»

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  • On maximum-likelihood detection and decoding for space-time coding systems

    Publication Year: 2002 , Page(s): 937 - 944
    Cited by:  Papers (33)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (318 KB) |  | HTML iconHTML  

    Space-time coding (STC) schemes for communication systems employing multiple transmit and receive antennas have been attracting increased attention. In this paper, we address two interrelated problems: detection of space-time codes under various interference conditions and information transfer from the STC detector to an error-correcting channel decoder. By taking a systematic maximum-likelihood (ML) approach to the joint detection and decoding problem, we show how to design optimal detectors and how to integrate them with a channel decoder. We also discuss various aspects of channel modeling for STC communication receivers. In particular, while many previous works on space-time coding assume that the channel is a stochastic quantity, we find that a deterministic channel model can have some advantages for the receiver design. Finally, we illustrate our results by numerical examples. Index Terms-Interference suppression, maximum-likelihood estimation, maximum-likelihood sequence detection, MIMO systems, space-time coding, soft information View full abstract»

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  • Blind high-resolution localization and tracking of multiple frequency hopped signals

    Publication Year: 2002 , Page(s): 889 - 901
    Cited by:  Papers (27)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (475 KB) |  | HTML iconHTML  

    This paper considers the problem of blind localization and tracking of multiple frequency-hopped spread-spectrum signals using a uniform linear antenna array without knowledge of hopping patterns or directions of arrival. As a preprocessing step, we propose to identify a hop-free subset of data by discarding high-entropy spectral slices from the spectrogram. High-resolution localization is then achieved via either quadrilinear regression of four-way data generated by capitalizing on both spatial and temporal shift invariance or a new maximum likelihood (ML)-based two-dimensional (2-D) harmonic retrieval algorithm. The latter option achieves the best-known model identifiability bound while remaining close to the Cramer-Rao bound even at low signal-to-noise ratios (SNRs). Following beamforming using the recovered directions, a dynamic programming approach is developed for joint ML estimation of signal frequencies and hop instants in single-user tracking. The efficacy of the proposed algorithms is illustrated in pertinent simulations View full abstract»

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  • Mutual information-based feature extraction on the time-frequency plane

    Publication Year: 2002 , Page(s): 779 - 790
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (451 KB)  

    A method is proposed for automatic extraction of effective features for class separability. It applies to nonstationary processes described only by sample sets of stochastic signals. The extraction is based on time-frequency representations (TFRs) that are potentially suited to the characterization of nonstationarities. The features are defined by parameterized mappings applied to a TFR. These mappings select a region of the time-frequency plane by using a two-dimensional (2-D) parameterized weighting function and provide a standard characteristic in the restricted representation obtained. The features are automatically drawn from the TFR by tuning the weighting function parameters. The extraction is driven to maximize the information brought by the features about the class membership. It uses a mutual information criterion, based on estimated probability distributions. The framework is developed for the extraction of a single feature and extended to several features. A classification scheme adapted to the extracted features is proposed. Finally, some experimental results are given to demonstrate the efficacy of the method View full abstract»

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  • Robust Kalman filters for linear time-varying systems with stochastic parametric uncertainties

    Publication Year: 2002 , Page(s): 803 - 813
    Cited by:  Papers (56)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (458 KB)  

    We present a robust recursive Kalman filtering algorithm that addresses estimation problems that arise in linear time-varying systems with stochastic parametric uncertainties. The filter has a one-step predictor-corrector structure and minimizes an upper bound of the mean square estimation error at each step, with the minimization reduced to a convex optimization problem based on linear matrix inequalities. The algorithm is shown to converge when the system is mean square stable and the state space matrices are time invariant. A numerical example consisting of equalizer design for a communication channel demonstrates that our algorithm offers considerable improvement in performance when compared with conventional Kalman filtering techniques View full abstract»

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  • Tracking properties of a gradient-based second-order adaptive IIR notch filter with constrained poles and zeros

    Publication Year: 2002 , Page(s): 878 - 888
    Cited by:  Papers (18)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (433 KB)  

    Gradient-type adaptive IIR notch filters have many attractive merits for various real-life applications since they require a small number of computations and yet demonstrate practical performance. However, it is generally quite difficult to assess their performance analytically. Their tracking properties, in particular, have not yet been investigated. In this paper, the tracking performance of a plain gradient (PG) algorithm is analyzed in detail for a second-order adaptive IIR notch filter with constrained poles and zeros, which takes a linear chirp signal as its input. First, two sets of difference equations for the frequency tracking error and mean square error (MSE) are established in the sense of convergence in the mean and convergence in the mean square, respectively. Closed-form expressions for the asymptotic tracking error and MSE are then derived from these difference equations. An optimum step-size parameter for the algorithm is also evaluated based on the minimization of the asymptotic tracking error or the tracking MSE. It is discovered that the asymptotic tracking error may be driven to zero for a positive chirp rate by selecting a proper step size, which is an interesting property for a real-valued adaptive filtering algorithm. Extensive simulations are performed to support the analytical findings View full abstract»

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  • Adaptive solution for blind identification/equalization using deterministic maximum likelihood

    Publication Year: 2002 , Page(s): 923 - 936
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (494 KB) |  | HTML iconHTML  

    A deterministic maximum likelihood (DML) approach is presented for the blind channel estimation problem. It is first proposed in a block version, which consists of iterating two steps, each one solving a least-squares problem either in the channel or in the symbols. In the noiseless case and under certain conditions, this algorithm gives the exact channel and the exact symbol vector with a finite number of samples. It is shown that even if the DML method has a single global minimum, the proposed iterative procedure can converge to spurious local minima. This problem can be detected (under some channel diversity conditions) by using a numerical test that is proposed in the paper. Based on these considerations, we extend the maximum likelihood block algorithm (MLBA) to recursive implementations [maximum likelihood recursive algorithm (MLRA)]. The MLRA is able to track variations of the system by the introduction of an exponential forgetting factor in the DML criterion. The link between the adaptive algorithm and a soft decision feedback equalizer (SDFE) is emphasized. Low-complexity versions of the recursive and adaptive algorithm are presented View full abstract»

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  • Enhanced recognition in hybrid systems

    Publication Year: 2002 , Page(s): 981 - 984
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (272 KB) |  | HTML iconHTML  

    Identifying the mode of operation of a hybrid system while simultaneously tracking the state is difficult without a direct modal measurement. Even with such a measurement, subtle biases have been observed in applications. It is shown here that for one estimation algorithm, modal identification can be improved with a simple adjustment in the measurement update View full abstract»

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  • Maximally linear FIR digital differentiators in frequencies of π/p-simplified formulas of weighting coefficients

    Publication Year: 2002 , Page(s): 978 - 981
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (263 KB) |  | HTML iconHTML  

    Simplified formulas of weighting coefficients of maximally linear FIR digital differentiators have been derived View full abstract»

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  • Two-dimensional Fourier series-based model for nonminimum-phase linear shift-invariant systems and texture image classification

    Publication Year: 2002 , Page(s): 945 - 955
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (554 KB) |  | HTML iconHTML  

    In this paper, Chi's (1997, 1999) real one-dimensional (1-D) parametric nonminimum-phase Fourier series-based model (FSBM) is extended to two-dimensional (2-D) FSBM for a 2-D nonminimum-phase linear shift-invariant system by using finite 2-D Fourier series approximations to its amplitude response and phase response, respectively. The proposed 2-D FSBM is guaranteed stable, and its complex cepstrum can be obtained from its amplitude and phase parameters through a closed-form formula without involving complicated 2-D phase unwrapping and polynomial rooting. A consistent estimator is proposed for the amplitude estimation of the 2-D FSBM using a 2-D half plane causal minimum-phase linear prediction error filter (modeled by a 2-D minimum-phase FSBM), and then, two consistent estimators are proposed for the phase estimation of the 2-D FSBM using the Chien et al. (1997) 2-D phase equalizer (modeled by a 2-D all-pass FSBM). The estimated 2-D FSBM can be applied to modeling of 2-D non-Gaussian random signals and 2-D signal classification using complex cepstra. Some simulation results are presented to support the efficacy of the three proposed estimators. Furthermore, classification of texture images (2-D non-Gaussian signals) using the estimated FSBM, second-, and higher order statistics is presented together with some experimental results. Finally, we draw some conclusions View full abstract»

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  • A study of two-channel complex-valued filterbanks and wavelets with orthogonality and symmetry properties

    Publication Year: 2002 , Page(s): 824 - 833
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (352 KB) |  | HTML iconHTML  

    We investigate two-channel complex-valued filterbanks and wavelets that simultaneously have orthogonality and symmetry properties. First, the conditions for the filterbank to be orthogonal, symmetric, and regular (for generating smooth wavelets) are presented. Then, a complete and minimal lattice structure is developed, which enables a general design approach for filterbanks and wavelets with arbitrary length and arbitrary order of regularity. Finally, two integer implementation methods that preserve the perfect reconstruction property of the filterbank are proposed. Their performances are evaluated via experimental results View full abstract»

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  • Minimum redundancy for ISI free FIR filterbank transceivers

    Publication Year: 2002 , Page(s): 842 - 853
    Cited by:  Papers (30)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (443 KB) |  | HTML iconHTML  

    There has been great interest in the design of filterbank transceivers. Usually, with proper time domain equalization, the channel is modeled as an FIR filter. It is known that for FIR channels, the introduction of certain redundancy allows the receiver to cancel intersymbol interference (ISI) completely, and channel equalization is performed implicitly using FIR transceivers. This scheme allows us to trade bandwidth for ISI cancellation. In this paper, we derive the minimum redundancy required for the existence of FIR transceivers for a given channel. We see that the minimum redundancy is directly related to the zeros of the channel and to the Smith form of an appropriately defined channel matrix View full abstract»

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  • Efficient arbitrary sampling rate conversion with recursive calculation of coefficients

    Publication Year: 2002 , Page(s): 854 - 865
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (363 KB) |  | HTML iconHTML  

    In this paper, we present a novel algorithm for sampling rate conversion by an arbitrary factor. Theoretically, sampling rate conversion of a discrete-time (DT) sequence can be performed by converting the sequence to a series of continuous-time (CT) impulses. This series of impulses is filtered with a CT lowpass filter, and the output is then sampled at the desired rate. If the CT filter is chosen to have a rational transfer function, then this system can be simulated using a DT algorithm for which both computation and memory requirements are low. The DT implementation is comprised of a parallel structure, where each branch consists of a time-varying filter with one or two taps, followed by a fixed recursive filter operating at the output sampling rate. The coefficients of the time-varying filters are calculated recursively. This eliminates the need to store a large table of coefficients, as is commonly done View full abstract»

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  • Eigenvalues and eigenvectors of generalized DFT, generalized DHT, DCT-IV and DST-IV matrices

    Publication Year: 2002 , Page(s): 866 - 877
    Cited by:  Papers (27)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (444 KB) |  | HTML iconHTML  

    In this paper, the eigenvalues and eigenvectors of the generalized discrete Fourier transform (GDFT), the generalized discrete Hartley transform (GDHT), the type-IV discrete cosine transform (DCT-IV), and the type-IV discrete sine transform (DST-IV) matrices are investigated in a unified framework. First, the eigenvalues and their multiplicities of the GDFT matrix are determined, and the theory of commuting matrices is applied to find the real, symmetric, orthogonal eigenvectors set that constitutes the discrete counterpart of Hermite Gaussian function. Then, the results of the GDFT matrix and the relationships among these four unitary transforms are used to find the eigenproperties of the GDHT, DCT-IV, and DST-IV matrices. Finally, the fractional versions of these four transforms are defined, and an image watermarking scheme is proposed to demonstrate the effectiveness of fractional transforms View full abstract»

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  • Allpass delay chain-based IIR PR filterbank and its application to multiple description subband coding

    Publication Year: 2002 , Page(s): 814 - 823
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (403 KB) |  | HTML iconHTML  

    A class of infinite impulse response (IIR) perfect reconstruction (PR) filterbank is obtained with an allpass delay chain and finite impulse response (FIR) matrices at the analysis side. Such a formulation leads to a filterbank that can be optimized to any desired response. For a first-order allpass, the synthesis bank becomes FIR. Design examples showing warping of the frequency scale and filters with unequal passband widths are presented. The latter part of the work deals with designing a filterbank for a multiple description coding scenario. This is achieved in two parts: by obtaining the MMSE synthesis bank for a given analysis bank and channel state and by optimizing the analysis response to minimize the average distortion in the presence of channel erasures and quantization. The proposed IIR filterbank, as well as the well-known FIR paraunitary filterbank, are optimized for various channels, and their performances for the two-channel case have been compared with the derived optimal performance View full abstract»

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  • A family of lapped regular transforms with integer coefficients

    Publication Year: 2002 , Page(s): 834 - 841
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (345 KB) |  | HTML iconHTML  

    Invertible transforms with integer coefficients are highly desirable because of their fast, efficient, VLSI-suitable implementations and their lossless coding capability. In this paper, a large class of lapped regular transforms with integer coefficients (ILT) is presented. Regularity constraints are also taken into account to provide smoother reconstructed signals. In other words, this ILT family can be considered to be an M-band biorthogonal wavelet with integer coefficients. The ILT also possesses a fast and efficient lattice that structurally enforces both linear-phase and exact reconstruction properties. Preliminary image coding experiments show that the ILT yields comparable objective and subjective performance to those of popular state-of-the-art transforms with floating-point coefficients View full abstract»

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  • Finite-length MIMO equalization using canonical correlation analysis

    Publication Year: 2002 , Page(s): 984 - 989
    Cited by:  Papers (7)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (330 KB) |  | HTML iconHTML  

    We propose finite-length multi-input multi-output (MIMO) equalization methods for "smart" antenna arrays using the statistical theory of canonical correlations. We show that the proposed methods are related to maximum likelihood (ML) reduced-rank channel and noise estimation algorithms in unknown spatially correlated noise as well as to several previously developed equalization schemes View full abstract»

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  • Tracking of signal subspace projectors

    Publication Year: 2002 , Page(s): 769 - 778
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (439 KB) |  | HTML iconHTML  

    A new subspace tracking approach that directly operates on the projection matrix onto the signal subspace instead of tracking its unitary eigenbasis is proposed. Therefore, the difficulties arising from introducing a calculus on the manifold of projection matrices are overcome by a proper parameterization of the projectors. Simulation results are presented from the field of angular frequency retrieval in array signal processing View full abstract»

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  • A VLSI architecture for lifting-based forward and inverse wavelet transform

    Publication Year: 2002 , Page(s): 966 - 977
    Cited by:  Papers (144)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (422 KB) |  | HTML iconHTML  

    We propose an architecture that performs the forward and inverse discrete wavelet transform (DWT) using a lifting-based scheme for the set of seven filters proposed in JPEG2000. The architecture consists of two row processors, two column processors, and two memory modules. Each processor contains two adders, one multiplier, and one shifter. The precision of the multipliers and adders has been determined using extensive simulation. Each memory module consists of four banks in order to support the high computational bandwidth. The architecture has been designed to generate an output every cycle for the JPEG2000 default filters. The schedules have been generated by hand and the corresponding timings listed. Finally, the architecture has been implemented in behavioral VHDL. The estimated area of the proposed architecture in 0.18-μ technology is 2.8 nun square, and the estimated frequency of operation is 200 MHz View full abstract»

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  • Wald statistic for model order selection in superposition models

    Publication Year: 2002 , Page(s): 956 - 965
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (396 KB) |  | HTML iconHTML  

    A consistent model selection algorithm is presented for superimposed signal models. The proposed method is motivated by the Wald statistic and reduces the computational complexity of procedures based on the minimum description length (MDL) principle. The procedure is suggested when a noncyclostationary signal model or short data length prevents use of covariance rank test. For maximum model-order K, the procedure provides O(K) computational savings over an MDL test. Additionally, a proof establishes the consistency of a least-squares estimator using overparametrized. models. Finite sample performance of the proposed model selection method is studied via Monte Carlo simulations for estimating the multipath delays and amplitudes of a chirp signal View full abstract»

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Aims & Scope

IEEE Transactions on Signal Processing covers novel theory, algorithms, performance analyses and applications of techniques for the processing, understanding, learning, retrieval, mining, and extraction of information from signals

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Editor-in-Chief
Sergios Theodoridis
University of Athens