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		<title><![CDATA[ Signal Processing, IEEE Transactions on - new TOC ]]></title>
		<link>http://ieeexplore.ieee.org</link>
		<description>TOC Alert for Publication# 78 </description>
		<year>2012</year>
		<month>February </month>
		<day>10</day>
		<item>
			<title><![CDATA[Table of Contents]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146822]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146822]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>C1</startPage>
			<endPage>C4</endPage>
			<fileSize>186</fileSize>
			<authors><![CDATA[]]></authors>
		</item>
		<item>
			<title><![CDATA[IEEE Transactions on Signal Processing publication information]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146825]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146825]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>C2</startPage>
			<endPage>C2</endPage>
			<fileSize>40</fileSize>
			<authors><![CDATA[]]></authors>
		</item>
		<item>
			<title><![CDATA[Locally Most Powerful Invariant Tests for the Properness of Quaternion Gaussian Vectors]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096451]]></link>
			<description><![CDATA[Previous works have addressed the second-order statistical characterization of quaternion random vectors, introducing different properness definitions, and presenting the generalized likelihood ratio tests (GLRTs) for determining the kind of quaternion properness. This paper considers the more challenging problem of deriving the locally most powerful invariant tests (LMPITs), which can be obtained, even without an explicit expression for the maximal invariants, thanks to the Wijsman's theorem. Specifically, we consider three binary hypothesis testing problems involving the two main kinds of quaternion properness, and show that the LMPIT statistics are given by the Frobenius norm of three previously defined sample coherence matrices. The proposed detectors exhibit interesting connections with the problem of testing for the properness of a complex vector, and with the problems of testing for the sphericity of a four-dimensional real (or two-dimensional complex proper) vector. Additionally, some numerical examples show that in general, the proposed LMPITs outperform their GLRT counterparts, and in some cases the performance gap is very noticeable.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096451]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>997</startPage>
			<endPage>1009</endPage>
			<fileSize>3148</fileSize>
			<authors><![CDATA[Via, J.;Vielva, L.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Orthogonal Matching Pursuit: A Brownian Motion Analysis]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086631]]></link>
			<description><![CDATA[A well-known analysis of Tropp and Gilbert shows that orthogonal matching pursuit (OMP) can recover a <formula formulatype="inline"> <tex Notation="TeX">$k$</tex></formula>-sparse <formula formulatype="inline"><tex Notation="TeX">$n$</tex></formula>-dimensional real vector from <formula formulatype="inline"> <tex Notation="TeX">$m=4klog(n)$</tex></formula> noise-free linear measurements obtained through a random Gaussian measurement matrix with a probability that approaches one as <formula formulatype="inline"><tex Notation="TeX">$nrightarrowinfty$</tex></formula>. This work strengthens this result by showing that a lower number of measurements, <formula formulatype="inline"><tex Notation="TeX">$m=2klog(n-k)$</tex> </formula>, is in fact sufficient for asymptotic recovery. More generally, when the sparsity level satisfies <formula formulatype="inline"><tex Notation="TeX">$k_{min}leq kleq k_{max}$</tex> </formula> but is unknown, <formula formulatype="inline"><tex Notation="TeX">$m=2k_{max}log(n-k_{min})$</tex> </formula> measurements is sufficient. Furthermore, this number of measurements is also sufficient for detection of the sparsity pattern (support) of the vector with measurement errors provided the signal-to-noise ratio (SNR) scales to infinity. The scaling <formula formulatype="inline"> <tex Notation="TeX">$m=2klog(n-k)$</tex></formula> exactly matches the number of measurements required by the more complex lasso method for signal recovery with a similar SNR scaling.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086631]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1010</startPage>
			<endPage>1021</endPage>
			<fileSize>2755</fileSize>
			<authors><![CDATA[Fletcher, A. K.;Rangan, S.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Bayesian Estimation for Nonstandard Loss Functions Using a Parametric Family of Estimators]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097071]]></link>
			<description><![CDATA[Bayesian estimation with other loss functions than the standard hit-or-miss loss or the quadratic loss often yields optimal Bayesian estimators (OBEs) that can only be formulated as optimization problems and which have to be solved for each new observation. The contribution of this paper is to introduce a new parametric family of estimators to circumvent this problem. By restricting the estimator to lie in this family, we split the estimation problem into two parts: In a first step, we have to find the best estimator with respect to the Bayes risk for a given nonstandard loss function, which has to be done only once. The second step then calculates the estimate for an observation using importance sampling. The computational complexity of this second step is therefore comparable to that of an MMSE estimator if the MMSE estimator also uses Monte Carlo integration. We study the proposed parametric family using two examples and show that the estimator family gives for both a good approximation of the OBE.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097071]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1022</startPage>
			<endPage>1031</endPage>
			<fileSize>3690</fileSize>
			<authors><![CDATA[Uhlich, S.;Yang, B.;]]></authors>
		</item>
		<item>
			<title><![CDATA[New Results on Deterministic Cram&#x00E9;r&#x2013;Rao Bounds for Real and Complex Parameters]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094230]]></link>
			<description><![CDATA[The Cram&#x00E9;r&#x2013;Rao bounds (CRB) is a lower bound of great interest for system analysis and design in the asymptotic region [high signal-to-noise ratio (SNR) and/or large number of snapshots], as it is simple to calculate and it is usually possible to obtain closed form expressions. The first part of the paper is a generalization to complex parameters of the Barankin rationale for deriving MSE lower bounds, that is the minimization of a norm under a set of linear constraints. With the norm minimization approach the study of Fisher information matrix (FIM) singularity, constrained CRB and regularity conditions become straightforward corollaries of the derivation. The second part provides new results useful for system analysis and design: a general reparameterization inequality, the equivalence between reparameterization and equality constraints, and an explicit relationship between parameters unidentifiability and FIM singularity.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094230]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1032</startPage>
			<endPage>1049</endPage>
			<fileSize>6767</fileSize>
			<authors><![CDATA[Menni, T.;Chaumette, E.;Larzabal, P.;Barbot, J. P.;]]></authors>
		</item>
		<item>
			<title><![CDATA[On the Product of Independent Complex Gaussians]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086771]]></link>
			<description><![CDATA[In this paper, we derive the joint (amplitude, phase) distribution of the product of two independent non-zero-mean Complex Gaussian random variables. We call this new distribution the complex Double Gaussian distribution. This probability distribution function (PDF) is a doubly infinite summation over modified Bessel functions of the first and second kind. We analyze the behavior of this sum and show that the number of terms needed for accuracy is dependent upon the Rician <formula formulatype="inline"><tex Notation="TeX">$k$</tex></formula>-factors of the two input variables. We derive an upper bound on the truncation error and use this to present an adaptive computational approach that selects the minimum number of terms required for accuracy. We also present the PDF for the special case where either one or both of the input complex Gaussian random variables is zero-mean. We demonstrate the relevance of our results by deriving the optimal Neyman&#x2013;Pearson detector for a time reversal detection scheme and computing the receiver operating characteristics through Monte Carlo simulations, and by computing the symbol error probability (SEP) for a single-channel <formula formulatype="inline"><tex Notation="TeX">$M$</tex></formula>-ary phase-shift-keying (M-PSK) communication system.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086771]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1050</startPage>
			<endPage>1063</endPage>
			<fileSize>3914</fileSize>
			<authors><![CDATA[O'Donoughue, N.;Moura, J. M. F.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Geometric Methods for Spectral Analysis]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097067]]></link>
			<description><![CDATA[This paper explores a geometric framework for modeling nonstationary but slowly varying time series, based on the assumption that short-windowed power spectra capture their spectral character, and that energy transference in the frequency domain has a physical significance. The framework relies on certain notions of transportation distance and their respective geodesics to model possible nonparametric changes in the power spectral density with respect to time. We discuss the relevance of this framework to applications in spectral tracking, spectral averaging, and speech morphing.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097067]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1064</startPage>
			<endPage>1074</endPage>
			<fileSize>2648</fileSize>
			<authors><![CDATA[Jiang, X.;Luo, Z.-Q.;Georgiou, T. T.;]]></authors>
		</item>
		<item>
			<title><![CDATA[EMD Revisited: A New Understanding of the Envelope and Resolving the Mode-Mixing Problem in AM-FM Signals]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104667]]></link>
			<description><![CDATA[Empirical mode decomposition (EMD) is an adaptive and data-driven approach for analyzing multicomponent nonlinear and nonstationary signals. The stop criterion, envelope technique, and mode-mixing problem are the most important topics that need to be addressed in order to improve the EMD algorithm. In this paper, we study the envelope technique and the mode-mixing problem caused by separating multicomponent AM-FM signals with the EMD algorithm. We present a new necessary condition on the envelope that questions the current assumption that the envelope passes through the extreme points of an intrinsic mode function (IMF). Then, we present a solution to the mode-mixing problem that occurs when multicomponent AM-FM signals are separated. We experiment on several signals, including simulated signals and real-life signals, to demonstrate the efficacy of the proposed method in resolving the mode-mixing problem.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104667]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1075</startPage>
			<endPage>1086</endPage>
			<fileSize>812</fileSize>
			<authors><![CDATA[Hu, X.;Peng, S.;Hwang, W.-L.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Testing for Parallelism Among Trends in Multiple Time Series]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094237]]></link>
			<description><![CDATA[This paper considers the inference of trends in multiple, nonstationary time series. To test whether trends are parallel to each other, we use a parallelism index based on the <formula formulatype="inline"><tex Notation="TeX">$L^{2}$</tex> </formula>-distances between nonparametric trend estimators and their average. A central limit theorem is obtained for the test statistic and the test's consistency is established. We propose a simulation-based approximation to the distribution of the test statistic, which significantly improves upon the normal approximation. The test is also applied to devise a clustering algorithm. Finally, the finite-sample properties of the test are assessed through simulations and the test methodology is illustrated by a cell phone download data collected in the United States.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094237]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1087</startPage>
			<endPage>1097</endPage>
			<fileSize>3005</fileSize>
			<authors><![CDATA[Degras, D.;Xu, Z.;Zhang, T.;Wu, W. B.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Time-Invariant Context for Sample Rate Conversion Systems]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6082465]]></link>
			<description><![CDATA[Digital systems dedicated to audio and speech processing usually require sample rate conversion units in order to adapt the sample rate from different signal flows: for instance 8 and 16 kHz for speech, 32 kHz for the broadcast rate, 44.1 kHz for CDs, and 48 kHz for studio work. The designer chooses the sample rate conversion (SRC) technology based on objective criteria, such as figures of complexity, development or integration cycle and of course performance characterization. The performances of the fractional SRC system include the in-band and the aliasing characterization plus its distortion behavior due to internal rounding errors. The paper shows the existence of a new compound time-invariant system made of multiple instances of the same SRC system. The characterization of the original SRC is obtained from the linear and distortion characteristics of this time-invariant system. Regular methods for characterizing time-invariant systems apply. The SRC system can be analyzed in black box conditions, either in batch processing or in real-time processing. Examples illustrate the capability of the method to fully recover characteristics and rounding noise behavior from actual SRC implementations.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6082465]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1098</startPage>
			<endPage>1107</endPage>
			<fileSize>2172</fileSize>
			<authors><![CDATA[Tassart, S.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Discrete Fourier-Invariant Signals: Design and Application]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096450]]></link>
			<description><![CDATA[In this paper, two methods for the design of discrete Fourier-invariant signals are proposed. The direct design method provides splitting between independent and dependent signal parts and calculation of the dependent part for any given independent part. The iterative design method generates a family of discrete Fourier-invariant signals by a successive approach. Further we show how the proposed direct design method can be combined with the Gabor uncertainty principle to generate discrete Fourier-invariant signals with the minimum product of their bandwidth (B) and their time-width (T). We show that these signals as well as signal families generated with the iterative design method achieve the theoretical lower BT bound. Also, it is shown that the BT product of discrete Hermite-Gauss signals converges to the theoretical lower bound. Finally, possible applications are illustrated in the case of time-frequency spectral analysis using the obtained discrete Fourier-invariant signals as the window that provides isoresolution.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096450]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1108</startPage>
			<endPage>1120</endPage>
			<fileSize>3012</fileSize>
			<authors><![CDATA[Temerinac-Ott, M.;Temerinac, M.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Xampling at the Rate of Innovation]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096447]]></link>
			<description><![CDATA[We address the problem of recovering signals from samples taken at their rate of innovation. Our only assumption is that the sampling system is such that the parameters defining the signal can be stably determined from the samples, a condition that lies at the heart of every sampling theorem. Consequently, our analysis subsumes previously studied nonlinear acquisition devices and nonlinear signal classes. In particular, we do not restrict attention to memoryless nonlinear distortions or to union-of-subspace models. This allows treatment of various finite-rate-of-innovation (FRI) signals that were not previously studied, including, for example, continuous phase modulation transmissions. Our strategy relies on minimizing the error between the measured samples and those corresponding to our signal estimate. This least-squares (LS) objective is generally nonconvex and might possess many local minima. Nevertheless, we prove that under the stability hypothesis, any optimization method designed to trap a stationary point of the LS criterion necessarily converges to the true solution. We demonstrate our approach in the context of recovering pulse streams in settings that were not previously treated. Furthermore, in situations for which other algorithms are applicable, we show that our method is often preferable in terms of noise robustness.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096447]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1121</startPage>
			<endPage>1133</endPage>
			<fileSize>3574</fileSize>
			<authors><![CDATA[Michaeli, T.;Eldar, Y. C.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Sub-Nyquist Sampling of Short Pulses]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086632]]></link>
			<description><![CDATA[We develop sub-Nyquist sampling systems for analog signals comprised of several, possibly overlapping, finite duration pulses with unknown shapes and time positions. Efficient sampling schemes when either the pulse shape or the locations of the pulses are known have been previously developed. To the best of our knowledge, stable and low-rate sampling strategies for continuous signals that are superpositions of unknown pulses without knowledge of the pulse locations have not been derived. The goal in this paper is to fill this gap. We propose a multichannel scheme based on Gabor frames that exploits the sparsity of signals in time and enables sampling multipulse signals at sub-Nyquist rates. Moreover, if the signal is additionally essentially multiband, then the sampling scheme can be adapted to lower the sampling rate without knowing the band locations. We show that, with proper preprocessing, the necessary Gabor coefficients, can be recovered from the samples using standard methods of compressed sensing. In addition, we provide error estimates on the reconstruction and analyze the proposed architecture in the presence of noise.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086632]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1134</startPage>
			<endPage>1148</endPage>
			<fileSize>3480</fileSize>
			<authors><![CDATA[Matusiak, E.;Eldar, Y. C.;]]></authors>
		</item>
		<item>
			<title><![CDATA[On Algorithms and Complexities of Cyclotomic Fast Fourier Transforms Over Arbitrary Finite Fields]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097072]]></link>
			<description><![CDATA[Discrete Fourier transforms over finite fields are significant due to their widespread applications in cryptography and error control codes, which in turn are used in all digital communication and storage systems. Cyclotomic fast Fourier transforms (CFFTs) are of great interest due to their low multiplicative complexities. However, all existing CFFTs are for characteristic-2 fields, and the computational complexities of CFFTs have not been analyzed theoretically. This paper addresses both problems for CFFTs, and has three main contributions to this end. First, we propose an efficient bilinear algorithm to compute Toeplitz matrix vector products (TMVPs), which has a lower computational complexity than existing algorithms, and works on all finite fields as well as the real and complex fields. Second, we propose an efficient algorithm for cyclic convolutions over arbitrary finite fields, which is essential in deriving efficient CFFTs over arbitrary finite fields. Finally, we derive bounds on the additive and multiplicative complexities of CFFTs over arbitrary finite fields. Our results confirm that CFFTs have the smallest multiplicative complexities among all known algorithms. Although their high additive complexities render them asymptotically suboptimal, CFFTs remain valuable since they have the smallest overall complexities for DFTs of up to thousands of symbols in most cases.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097072]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1149</startPage>
			<endPage>1158</endPage>
			<fileSize>2814</fileSize>
			<authors><![CDATA[Wu, X.;Wang, Y.;Yan, Z.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Parametrization of Linear Systems Using Diffusion Kernels]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094236]]></link>
			<description><![CDATA[Modeling natural and artificial systems has played a key role in various applications and has long been a task that has drawn enormous efforts. In this work, instead of exploring predefined models, we aim to identify implicitly the system degrees of freedom. This approach circumvents the dependency of a specific predefined model for a specific task or system and enables a generic data-driven method to characterize a system based solely on its output observations. We claim that each system can be viewed as a black box controlled by several independent parameters. Moreover, we assume that the perceptual characterization of the system output is determined by these independent parameters. Consequently, by recovering the independent controlling parameters, we find in fact a generic model for the system. In this work, we propose a supervised algorithm to recover the controlling parameters of natural and artificial linear systems. The proposed algorithm relies on nonlinear independent component analysis using diffusion kernels and spectral analysis. Employment of the proposed algorithm on both synthetic and practical examples has shown accurate recovery of parameters.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094236]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1159</startPage>
			<endPage>1173</endPage>
			<fileSize>3441</fileSize>
			<authors><![CDATA[Talmon, R.;Kushnir, D.;Coifman, R. R.;Cohen, I.;Gannot, S.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Quaternion Dynamic Time Warping]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094234]]></link>
			<description><![CDATA[Dynamic time warping (DTW) is used for the comparison and processing of nonlinear signals and constitutes a widely researched field of study. The method has been initially designed for, and applied to, signals representing audio data. Afterwords it has been successfully modified and applied to many other fields of study. In this paper, we present the results of researches on the generalized DTW method designed for use with rotational sets of data parameterized by quaternions. The need to compare and process quaternion time series has been gaining in importance recently. Three-dimensional motion data processing is one of the most important applications here. Specifically, it is applied in the context of motion capture, and in many cases all rotational signals are described in this way. We propose a construction of generalized method called quaternion dynamic time warping (QDTW), which makes use of specific properties of quaternion space. It allows for the creation of a family of algorithms that deal with the higher order features of the rotational trajectory. This paper focuses on the analysis of the properties of this new approach. Numerical results show that the proposed method allows for efficient element assignment. Moreover, when used as the measure of similarity for a clustering task, the method helps to obtain good clustering performance both for synthetic and real datasets.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094234]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1174</startPage>
			<endPage>1183</endPage>
			<fileSize>1361</fileSize>
			<authors><![CDATA[Jablonski, B.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Survival Information Potential: A New Criterion for Adaptive System Training]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096443]]></link>
			<description><![CDATA[Recently, the information potential (IP) of order <formula formulatype="inline"><tex Notation="TeX">$alpha$</tex></formula>, defined as the argument of the log in the <formula formulatype="inline"><tex Notation="TeX">$alpha$</tex> </formula>-order Renyi entropy, has been successfully used as an information theoretic criterion for supervised adaptive system training. In this paper, we use the survival function (or equivalently the distribution function) of an absolute value transformed random variable to define a new information potential, named the survival information potential (SIP). Compared with the IP, the SIP has some advantages, such as validity in a wide range of distributions, robustness, and the simplicity in computation. The properties of SIP and a simple formula for computing the empirical SIP are given in the paper. Finally, the SIP criterion is applied in adaptive system training, and simulation examples on FIR adaptive filtering, kernel adaptive filtering, and time delay neural networks (TDNNs) training are presented to demonstrate the performance.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096443]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1184</startPage>
			<endPage>1194</endPage>
			<fileSize>2820</fileSize>
			<authors><![CDATA[Chen, B.;Zhu, P.;Principe, J. C.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Placement Design of Microphone Arrays in Near-Field Broadband Beamformers]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096453]]></link>
			<description><![CDATA[In beamformer design, the microphone array configuration is often prescribed and the filter coefficients are varied in order to improve on the noise reduction performance. However, the positions of the microphone elements play an important role in the overall performance and should be optimized at the same time. This problem is addressed in this paper. In order to understand the performance improvement through location movements, we first look at the design with an infinite filter length, which gives the performance limit for finite filter length designs. When the filter length is finite, both the filter coefficients and the placement of the microphone array are decision variables. In both situations, the problems can be formulated as constrained optimization problems. As the filter length increases, we show that the performance converges quickly to the limit. For illustration, two numerical examples are solved. Comparing with several popular configurations, we show that the performance of the optimized configuration improves significantly.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096453]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1195</startPage>
			<endPage>1204</endPage>
			<fileSize>2939</fileSize>
			<authors><![CDATA[Feng, Z. G.;Yiu, K. F. C.;Nordholm, S. E.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Low-Complexity Addition or Removal of Sensors/Constraints in LCMV Beamformers]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094233]]></link>
			<description><![CDATA[We address the application of the linearly constrained minimum variance (LCMV) beamformer in sensor networks. In signal processing applications, it is common to have a redundancy in the number of nodes, fully covering the area of interest. Here we consider suboptimal LCMV beamformers utilizing only a subset of the available sensors for signal enhancement applications. Multiple desired and interfering sources scenarios in multipath environments are considered. We assume that an oracle entity determines the group of sensors participating in the spatial filtering, denoted as the active sensors. The oracle is also responsible for updating the constraints set according to either sensors or sources activity or dynamics. Any update of the active sensors or of the constraints set necessitates recalculation of the beamformer and increases the power consumption. As power consumption is a most valuable resource in sensor networks, it is important to derive efficient update schemes. In this paper, we derive procedures for adding or removing either an active sensor or a constraint from an existing LCMV beamformer. Closed-form, as well as generalized sidelobe canceller (GSC)-form implementations, are derived. These procedures use the previous beamformer to save calculations in the updating process. We analyze the computational burden of the proposed procedures and show that it is much lower than the computational burden of the straightforward calculation of their corresponding beamformers.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094233]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1205</startPage>
			<endPage>1214</endPage>
			<fileSize>2606</fileSize>
			<authors><![CDATA[Markovich-Golan, S.;Gannot, S.;Cohen, I.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Model-Based Subspace Projection Beamforming for Deep Interference Nulling]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093978]]></link>
			<description><![CDATA[This paper considers the problem of adaptive array processing for interference canceling to drive very deep nulls in difficult signal environments. In many practical scenarios, the achievable null depth is limited by covariance matrix estimation error leading to poor identification of the interference subspace. We address the particularly troublesome cases of low interference-to-noise ratio (INR), relatively rapid interference motion, and correlated noise across the receiving array. A polynomial-based model is incorporated in the proposed algorithm to track changes in the array covariance matrix over time, mitigate interference subspace estimation errors, and improve canceler performance. The application of phased array feeds for radio astronomical telescopes is used to illustrate the problem and proposed solution. Here even weak residual interference after cancelation may obscure a signal of interest, so very deep beampattern nulls are required. Performance for conventional subspace projection (SP) is compared with polynomial-augmented SP using simulated and real experimental data, showing null-depth improvement of 6 to 30 dB.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093978]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1215</startPage>
			<endPage>1228</endPage>
			<fileSize>2628</fileSize>
			<authors><![CDATA[Landon, J.;Jeffs, B. D.;Warnick, K. F.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Bayesian Parameter Estimation Using Periodic Cost Functions]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6062425]]></link>
			<description><![CDATA[In this paper, a new method for Bayesian periodic parameter estimation is derived using periodic cost functions. The method, named parameter estimation via root finding (PERF), is based on Fourier series representation of the Bayes periodic-risk functions. The PERF method is implemented for minimum cyclic error, minimum absolute periodic error, and minimum mean-square-periodic-error (MSPE) estimators and the corresponding estimators are derived. The periodic estimators are applied to direction-of-arrival and phase estimation problems and compared with the minimum mean-square-error and maximum a posteriori probability estimators, and the periodic Ziv-Zakai lower bound in terms of MSPE.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6062425]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1229</startPage>
			<endPage>1240</endPage>
			<fileSize>3804</fileSize>
			<authors><![CDATA[Routtenberg, T.;Tabrikian, J.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Efficient Time of Arrival Estimation Algorithm Achieving Maximum Likelihood Performance in Dense Multipath]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6062690]]></link>
			<description><![CDATA[Robust and accurate time-of-arrival (TOA) estimation in dense multipath channels such as those encountered in ultra-wideband (UWB) systems is a considerable challenge especially when the signal-to-noise ratio (SNR) is low. The exact maximum likelihood (EML) TOA estimator in dense multipath conditions has the potential to attain accurate TOA estimation, however, it is too complex for practical implementation. There is a substantial performance gap between the known practical algorithms for TOA estimation and the EML estimator. In this paper, a novel practical TOA estimation algorithm is developed that attains the EML performance when the multipath arrivals are dense. When the multipath arrivals density is low the estimator does not attain the maximum likelihood performance but still outperforms other known practical estimators. The estimator does not need to know the channel characteristics accurately, thus, it is robust to various multipath channels. The approach taken is to approximate the received multipath signal as a Gaussian process and derive the maximum likelihood estimator. In order to further decrease the computational load of the new algorithm, we develop a low complexity approximation with negligible performance degradation. The algorithm is useful for either single channel realization or multiple channel realizations using diversity either in time, frequency or space. When applying diversity technique a substantial performance gain is attained due to the optimal combining of the channel realizations and thus reliable TOA estimation is attainable even at low SNR. The estimator's performance can be closely predicted by a closed-form analytical error expression.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6062690]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1241</startPage>
			<endPage>1252</endPage>
			<fileSize>2763</fileSize>
			<authors><![CDATA[Bialer, O.;Raphaeli, D.;Weiss, A. J.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Multiple Level Nested Array: An Efficient Geometry for <formula formulatype="inline"> <img src="/images/tex/20180.gif" alt="2q"> </formula>th Order Cumulant Based Array Processing]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096449]]></link>
			<description><![CDATA[Recently, direction-of-arrival estimation (DOA) algorithms based on arbitrary even-order <formula formulatype="inline"><tex Notation="TeX">$(2q)$</tex> </formula> cumulants of the received data have been proposed, giving rise to new DOA estimation algorithms, namely the <formula formulatype="inline"> <tex Notation="TeX">$2q$</tex></formula> MUSIC algorithm. In particular, it has been shown that the <formula formulatype="inline"><tex Notation="TeX">$2q$</tex> </formula> MUSIC algorithm can identify <formula formulatype="inline"><tex Notation="TeX">$O(N^{q})$</tex></formula> statistically independent non-Gaussian sources. However, in this paper, it is demonstrated that the processing power of the <formula formulatype="inline"><tex Notation="TeX">$2q$</tex></formula>th-order cumulant based methods can potentially be even larger. It will be shown that the <formula formulatype="inline"><tex Notation="TeX">$2q$</tex></formula>th-order cumulant matrix of the data is directly related to the concept of a <formula formulatype="inline"><tex Notation="TeX">$2q$</tex></formula>th-order difference co-array which can potentially have <formula formulatype="inline"><tex Notation="TeX">$O(N^{2q})$</tex> </formula> virtual sensors, leading to identification of <formula formulatype="inline"> <tex Notation="TeX">$O(N^{2q})$</tex></formula> statistically independent non-Gaussian sources using <formula formulatype="inline"><tex Notation="TeX">$2q$</tex> </formula>th-order cumulants. However, the number of actually realizable virtual elements in the <formula formulatype="inline"><tex Notation="TeX">$2q$</tex> </formula>th-order difference co-array depends on the geometry of the physical array. In order to ensure that the co-array indeed has the desired degrees of freedom, a new generic class of linear (one dimensional) nonuniform arrays, namely the <formula formulatype="inline"><tex Notation="TeX">$2q$</tex></formula>th-order nested array, is proposed, whose <formula formulatype="inline">-
tex Notation="TeX">$2q$</tex> </formula>th-order difference co-array is proved to contain a uniform linear array with <formula formulatype="inline"><tex Notation="TeX">$O(N^{2q})$</tex> </formula> sensors. In order to exploit these increased degrees of freedom of the co-array, a new algorithm for DOA estimation is also developed, which acts on the same <formula formulatype="inline"><tex Notation="TeX">$2q$</tex> </formula>th-order cumulant matrix as the earlier methods and can yet identify <formula formulatype="inline"><tex Notation="TeX">$O(N^{2q})$</tex></formula> sources. It is proved that the proposed method can identify the maximum number of sources among all methods that use <formula formulatype="inline"><tex Notation="TeX">$2q$</tex> </formula>th-order cumulants.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096449]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1253</startPage>
			<endPage>1269</endPage>
			<fileSize>3366</fileSize>
			<authors><![CDATA[Pal, P.;Vaidyanathan, P. P.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Estimating the DOA and the Polarization of a Polynomial-Phase Signal Using a Single Polarized Vector-Sensor]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086774]]></link>
			<description><![CDATA[This paper introduces a novel algorithm to estimate the direction-of-arrival (DOA) and the polarization of a completely-polarized polynomial-phase signal of an arbitrary degree. The algorithm utilizes a polarized vector-sensor, comprising a spatially collocated six-component electromagnetic vector-sensor, a dipole triad, or a loop triad. This ESPRIT-based algorithm is based on a time-invariant matrix-pencil pair, derived from the time-delayed data-sets collected by a single polarized vector-sensor. The high-order difference-function of the signal's phase constructs the invariant-factor used in the ESPRIT algorithm. The steering vector is estimated from the signal-subspace eigenvector of the data-correlation matrix, following which the closed-form DOA and polarization can be obtained. Given the degree of the polynomial-phase signal, this approach resolves the two-dimensional azimuth-elevation angle and the polarization of the source, and requires neither a priori knowledge of the polynomial-phase signal's coefficients nor a priori knowledge of the polynomial-phase signal's frequency-spectrum. The efficacy of the proposed algorithm is verified by Monte Carlo simulations. Estimation accuracies of the DOA and the polarization parameters are evaluated by the closed-form Cram&#x00E9;r&#x2013;Rao bounds, which are independent of the polynomial coefficients, the degree of the polynomial-phase signal, and the azimuth-angle of the source.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086774]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1270</startPage>
			<endPage>1282</endPage>
			<fileSize>3499</fileSize>
			<authors><![CDATA[Yuan, X.;]]></authors>
		</item>
		<item>
			<title><![CDATA[A Robust Direct Data Domain Approach for STAP]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6084857]]></link>
			<description><![CDATA[In this paper, a novel approach for direct data domain space time adaptive processing (STAP) is presented. As already described in past literature, direct data domain STAP (also known as deterministic STAP) has several advantages compared to traditional stochastic STAP. In particular, being implicitly a single snapshot interference cancellation technique, deterministic STAP generally outperforms stochastic STAP in fast varying interference scenarios. On the other hand, in its classical derivation, target detection performances of deterministic STAP are severely deteriorated in case of uncertainty in the knowledge of exact target parameters as direction of arrival (DOA) and Doppler frequency. To overcome this problem, we propose a robust implementation of deterministic STAP in order to take into account a possible mismatch between the nominal and the actual target parameters. The proposed approach reformulates the deterministic STAP problem in the context of convex problem optimization. A detailed analysis of the maximum acceptable target parameters error is conducted, which ensures the existence of a numerical solution for the convex problem optimization. The proposed robust deterministic approach is defined for both the one dimensional (spatial-only) and the two dimensional (space-time) case. The effectiveness of the proposed approach is shown both in simulated scenarios and by direct application to real data taken from the experimental multichannel radar system PAMIR developed at Fraunhofer FHR.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6084857]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1283</startPage>
			<endPage>1294</endPage>
			<fileSize>4227</fileSize>
			<authors><![CDATA[Cristallini, D.;Burger, W.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Root Mean Square Decomposition for EST-Based Spatial Multiplexing Systems]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104174]]></link>
			<description><![CDATA[We consider the transceiver design for multiple-input&#x2013;multiple-output (MIMO) systems when the channel state information (CSI) is available at the transmitter as well as the receiver. First, we propose an open-loop low-complexity MIMO spatial multiplexing scheme based on the energy spreading transform (EST-SM). The EST-SM can spatially multiplex multiple data streams and iteratively detect the data streams with almost negligible interstream interference at sufficiently high SNR. Then, we propose a closed-loop precoding scheme suitable for the EST-SM called root mean square decomposition (RMSD) scheme. The RMSD precoding scheme combined with the EST-SM decomposes a MIMO channel into multiple subchannels with identical SNRs. This desired property minimizes bit error rate (BER) when different bit allocations on different subchannels, which cause a significant increase in system complexity, are not used. We show that when the EST-SM is used the RMSD scheme is optimal in BER performance and it achieves full diversity. Simulation results show that the RMSD scheme outperforms other existing techniques such as the geometric mean decomposition (GMD) scheme (Jiang <etal/>, IEEE Trans. Signal Process., vol. 53, no. 10, pp. 3791&#x2013;3803) and the uniform channel decomposition (UCD) scheme<footnoteref refid="fn1"/><footnote asterisk="no" id="fn1"><footnotepara>Throughout this paper, the GMD and UCD schemes mean GMD-VBLAST in <citerefgrp><citeref refid="ref12"/></citerefgrp> and UCD-VBLAST in <citerefgrp><citeref refid="ref13"/></citerefgrp>, respectively.</footnotepara></footnote> (Jiang <etal/>, IEEE Trans. Signal Process., vol. 53, no. 11, pp. 4283&#x2013;4294) in BER performance.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104174]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1295</startPage>
			<endPage>1306</endPage>
			<fileSize>2682</fileSize>
			<authors><![CDATA[Hwang, T.;Kwon, Y.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Widely Linear versus Conventional Subspace-Based Estimation of SIMO Flat-Fading Channels: Mean Squared Error Analysis]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086772]]></link>
			<description><![CDATA[We analyze the mean squared error (MSE) performance of widely linear (WL) and conventional subspace-based channel estimation for single-input multiple-output (SIMO) flat-fading channels employing binary phase-shift-keying (BPSK) modulation when the covariance matrix is estimated using a finite number of samples. The conventional estimator suffers from a phase ambiguity that reduces to a sign ambiguity for the WL estimator. We derive accurate closed-form expressions for the MSE of the two estimators under four ambiguity resolution scenarios. In the first three scenarios, the receiver resolves the ambiguity using some clairvoyant knowledge about the channel. The first scenario, used as a reference, is the ideal case of optimal resolution. The second scenario assumes that one of the channel coefficients is known and the third assumes knowledge of the coefficient with the largest magnitude. The fourth scenario considers the more realistic case where pilot symbols are employed for ambiguity resolution. Our work demonstrates that there is a strong relationship between the accuracy of ambiguity resolution and the relative performance of WL and conventional subspace-based estimators, showing that the WL estimator performs better when partial or inaccurate channel information is employed for ambiguity resolution.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086772]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1307</startPage>
			<endPage>1318</endPage>
			<fileSize>3128</fileSize>
			<authors><![CDATA[Abdallah, S.;Psaromiligkos, I. N.;]]></authors>
		</item>
		<item>
			<title><![CDATA[New Kurtosis Optimization Schemes for MISO Equalization]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094229]]></link>
			<description><![CDATA[This paper deals with efficient optimization of cumulant based contrast functions. Such a problem occurs in the blind source separation framework, where contrast functions are criteria to be maximized in order to retrieve the sources. More precisely, we focus on the extraction of one source signal and our method applies in deflation approaches, where the sources are extracted one by one. We propose new methods to maximize the kurtosis contrast function. These methods are intermediate between a gradient and an iterative &#x201C;fixed-point&#x201D; optimization of so-called reference contrasts. They rely on iterative updates of the parameters which monotonically increase the contrast function value: we point out the strong similarity with the Expectation-Maximization (EM) method and with recent generalizations referred to as Minimization-Maximization (MM). We also prove the global convergence of the algorithm to a stationary point. Simulations confirm the convergence of our methods to a separating solution. They also show experimentally that our methods have a much lower computational cost than former classical optimization methods. Finally, simulations suggest that the methods remain valid under weaker conditions than those required for proving convergence.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094229]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1319</startPage>
			<endPage>1330</endPage>
			<fileSize>2415</fileSize>
			<authors><![CDATA[Castella, M.;Moreau, E.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Achieving the Maximum Sum Rate Using D.C. Programming in Cellular Networks]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093977]]></link>
			<description><![CDATA[Intercell interference is the major limitation to the performance of future cellular systems. Despite the joint detection and joint transmission techniques aiming at interference cancelation which suffer from the limited possible cooperation among the nodes, power allocation is a promising approach for optimizing the system performance. If the interference is treated as noise, the power allocation optimization problem aiming at maximizing the sum rate with a total power constraint is nonconvex and up to now an open problem. In the present paper, the solution is found by reformulating the nonconvex objective function of the sum rate as a difference of two concave functions. A globally optimum power allocation is found by applying a branch-and-bound algorithm to the new formulation. In principle, the algorithm partitions the feasible region recursively into subregions where for every subregion the objective function is upper and lower bounded. For each subregion, a linear program is applied for estimating the upper bound of the sum rate which is derived from a convex maximization formulation of the problem with piecewise linearly approximated constraints. The performance is investigated by system-level simulations. The results show that the proposed algorithm outperforms the known conventional suboptimum schemes. Furthermore, it is shown that the algorithm asymptotically converges to a globally optimum power allocation.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093977]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1331</startPage>
			<endPage>1341</endPage>
			<fileSize>1832</fileSize>
			<authors><![CDATA[Al-Shatri, H.;Weber, T.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Optimal and Lattice Codes in a MIMO System With Complete Feedback]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096452]]></link>
			<description><![CDATA[Space time coding for a multiple-input, multiple-output (MIMO) system with complete feedback and maximum-likelihood (ML)-decoding is considered. First, a criterion for designing an optimal codebook with complete feedback in the case of high signal to noise ratio (SNR) is given; it turns out that designing an optimal codebook with complete feedback in the case of high SNR is equivalent to efficiently packing a finite number of points into a dynamic ellipsoid in a finite dimensional real Euclidean space. Second, based on this design criterion, an optimal tradeoff between diversity and multiplexing is then derived. Finally, practical design using lattice codes is addressed, and surprisingly, it is shown that the lattice codes can achieve the optimal tradeoff. It is also demonstrated that it is not necessary for an optimal lattice code to use all the subchannels generated by the system. This is in striking contrast with a general optimal code, which generally utilizes all the subchannels.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096452]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1342</startPage>
			<endPage>1351</endPage>
			<fileSize>2683</fileSize>
			<authors><![CDATA[Wang, H.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Joint Source and Relay Precoding Designs for MIMO Two-Way Relaying Based on MSE Criterion]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097070]]></link>
			<description><![CDATA[Properly designed precoders can significantly improve the spectral efficiency of multiple-input multiple-output (MIMO) relay systems. In this paper, we investigate joint source and relay precoding design based on the mean-square-error (MSE) criterion in MIMO two-way relay systems, where two multiantenna source nodes exchange information via a multiantenna amplify-and-forward relay node. This problem is non-convex and its optimal solution remains unsolved. Aiming to find an efficient way to solve the problem, we first decouple the primal problem into three tractable subproblems, and then propose an iterative precoding design algorithm based on alternating optimization. The solution to each subproblem is optimal and unique, thus the convergence of the iterative algorithm is guaranteed. Second, we propose a structured precoding design to lower the computational complexity. The proposed precoding structure is able to parallelize the channels in the multiple access (MAC) phase and broadcast (BC) phase. It thus reduces the precoding design to a simple power allocation problem. Last, for the special case where only a single data stream is transmitted from each source node, we present a source-antenna-selection (SAS)-based precoding design algorithm. This algorithm selects only one antenna for transmission from each source and thus requires lower signalling overhead. Comprehensive simulation is conducted to evaluate the effectiveness of all the proposed precoding designs.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6097070]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1352</startPage>
			<endPage>1365</endPage>
			<fileSize>3948</fileSize>
			<authors><![CDATA[Wang, R.;Tao, M.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Performance Assessment of MIMO-BICM Demodulators Based on Mutual Information]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093979]]></link>
			<description><![CDATA[We provide a comprehensive performance comparison of soft-output and hard-output demodulators in the context of non-iterative multiple-input multiple-output bit-interleaved coded modulation (MIMO-BICM). Coded bit error rate (BER), widely used in literature for demodulator comparison, has the drawback of depending strongly on the error correcting code being used. This motivates us to propose the mutual information of the equivalent modulation channel (comprising modulator, wireless channel, and demodulator) as a code-independent performance measure. We present extensive numerical results for spatially independent identically distributed (i.i.d.) ergodic and quasi-static fading channels under perfect and imperfect channel state information. These results reveal that the performance ranking of MIMO demodulators is rate-dependent and provide new insights regarding MIMO-BICM system design, i.e., the choice of antenna configuration, symbol constellation, and demodulator for a given target rate.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093979]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1366</startPage>
			<endPage>1382</endPage>
			<fileSize>3527</fileSize>
			<authors><![CDATA[Fertl, P.;Jalden, J.;Matz, G.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Combining Training and Quantized Feedback in Multiantenna Reciprocal Channels]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6099642]]></link>
			<description><![CDATA[The communication between a multiple-antenna transmitter and multiple receivers (users) with either a single or multiple-antenna each can be significantly enhanced by providing the channel state information at the transmitter (CSIT) of the users, as this allows for scheduling, beamforming and multiuser multiplexing gains. The traditional view on how to enable CSIT has been as follows: In time-division duplexed (TDD) systems, uplink (UL) and downlink (DL) channel reciprocity allows the use of a training sequence in the UL direction, which is exploited to obtain an UL channel estimate. This estimate is in turn recycled in the next downlink transmission. In frequency-division duplexed (FDD) systems, which lack the UL and DL reciprocity, the CSIT is provided via the use of a dedicated feedback link of limited capacity between the receivers and the transmitter. In this paper, we focus on TDD systems and put their classical approach in question. We show that the traditional TDD setup above fails to fully exploit the channel reciprocity in its true sense. In fact, we show that the system can benefit from a combined CSIT acquisition strategy mixing the use of limited feedback and that of a training sequence. This combining gives rise to a very interesting joint estimation and detection problem for which we propose two iterative algorithms. An outage rate based framework is also developed which gives the resource split between training and feedback. We demonstrate the potential of this hybrid combining in terms of the improved CSIT quality under a global training and feedback resource constraint.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6099642]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1383</startPage>
			<endPage>1396</endPage>
			<fileSize>2040</fileSize>
			<authors><![CDATA[Salim, U.;Gesbert, D.;Slock, D.;]]></authors>
		</item>
		<item>
			<title><![CDATA[QOS-Constrained Multiuser Peer-to-Peer Amplify-and-Forward Relay Beamforming]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086630]]></link>
			<description><![CDATA[A wireless communication scenario is considered with <formula formulatype="inline"><tex Notation="TeX">$K$</tex></formula> single-antenna source-destination pairs communicating through several half-duplex amplify-and-forward MIMO relays where each source is targeting only one destination. The relay beamforming matrices are designed in order to minimize the total power transmitted from the relays subject to quality of service constraints on the received signal to interference-plus-noise ratio at each destination node. Due to the nonconvexity of this problem, several approximations have been used in the literature to find a computationally efficient solution. A novel solution technique is developed in which the problem is decomposed into a group of second-order cone programs (SOCPs) parameterized by <formula formulatype="inline"><tex Notation="TeX">$K$</tex></formula> phase angles; each associated with one of the constraints. An iterative algorithm is proposed to search for the phase angles and the relay beamforming matrices sequentially. However, convergence to the global optimal beamforming matrices cannot be guaranteed. Two methods for searching for the optimal values of the phase angles are proposed (from which the optimal beamforming matrices can be obtained) using grid search and bisection and the convergence of these methods to the global optimal solution of the problem is proved. Numerical simulations are presented showing the superior performance of the proposed algorithms compared to earlier suboptimal approximations at the expense of a moderate increase in the computational complexity.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086630]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1397</startPage>
			<endPage>1408</endPage>
			<fileSize>3458</fileSize>
			<authors><![CDATA[Fadel, M.;El-Keyi, A.;Sultan, A.;]]></authors>
		</item>
		<item>
			<title><![CDATA[A New Framework for Distributed Detection With Conditionally Dependent Observations]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094231]]></link>
			<description><![CDATA[Distributed detection with conditionally dependent observations is known to be a challenging problem in decentralized inference. This paper attempts to make progress on this problem by proposing a new framework for distributed detection that builds on a hierarchical conditional independence model. Through the introduction of a hidden variable that induces conditional independence among the sensor observations, the proposed model unifies distributed detection with dependent or independent observations. This new framework allows us to identify several classes of distributed detection problems with dependent observations whose optimal decision rules resemble the ones for the independent case. The new framework induces a decoupling effect on the forms of the optimal local decision rules for these problems, much in the same way as the conditionally independent case. This is in sharp contrast to the general dependent case where the coupling of the forms of local sensor decision rules often renders the problem intractable. Such decoupling enables the use of, for example, the person-by-person optimization approach to find optimal local decision rules. Two classical examples in distributed detection with dependent observations are reexamined under this new framework: detection of a deterministic signal in dependent noises and detection of a random signal in independent noises.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094231]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1409</startPage>
			<endPage>1419</endPage>
			<fileSize>2371</fileSize>
			<authors><![CDATA[Chen, H.;Chen, B.;Varshney, P. K.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Downlink Coordinated Radio Resource Management in Cellular Networks With Partial CSI]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6111491]]></link>
			<description><![CDATA[We explore decentralized coordination of sectored cellular networks to adapt the usage of downlink resources to the instantaneous network conditions. The transmission frame consists of an orthogonal bandwidth usage phase, where sectors perform FDMA and power control over an agreed frequency chunk, and a shared bandwidth usage phase where each sector performs FDMA over the full available bandwidth without power control (interference is not controlled in this phase by any means). Decentralized network utility maximization with global optimality guarantee is enabled by fixing this structure of the transmission frame, which does not cause significant network-wide losses. Thus, the ability to better balance the resources gained from coordination generates some slack that can be used to either i) provide higher-quality access, ii) increase the number of active users, or iii) reduce deployment and maintenance costs by operating larger cells.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6111491]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1420</startPage>
			<endPage>1431</endPage>
			<fileSize>983</fileSize>
			<authors><![CDATA[Calvo, E.;Munoz, O.;Vidal, J.;Agustin, A.;]]></authors>
		</item>
		<item>
			<title><![CDATA[On Identifying Primary User Emulation Attacks in Cognitive Radio Systems Using Nonparametric Bayesian Classification]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096448]]></link>
			<description><![CDATA[Primary user emulation (PUE) attacks, where attackers mimic the signals of primary users (PUs), can cause significant performance degradation in cognitive radio (CR) systems. Detection of the presence of PUE attackers is thus an important problem. In this paper, using device-specific features, we propose a passive, nonparametric classification method to determine the number of transmitting devices in the PU spectrum. Our method, called DECLOAK, is passive since the sensing device listens and captures signals without injecting any signal to the wireless environment. It is nonparametric because the number of active devices needs not to be known as a priori. Channel independent features are selected forming fingerprints for devices, which cannot be altered postproduction. The infinite Gaussian mixture model (IGMM) is adopted and a modified collapsed Gibbs sampling method is proposed to classify the extracted fingerprints. Due to its unsupervised nature, there is no need to collect legitimate PU fingerprints. In combination with received power and device MAC address, we show through simulation studies that the proposed method can efficiently detect the PUE attack. The performance of DECLOAK is also shown to be superior than that of the classical non-parametric mean shift (MS) based clustering method.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096448]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1432</startPage>
			<endPage>1445</endPage>
			<fileSize>2204</fileSize>
			<authors><![CDATA[Nguyen, N. T.;Zheng, R.;Han, Z.;]]></authors>
		</item>
		<item>
			<title><![CDATA[An Effective Classification Framework for Brain&#x2013;Computer Interfacing Based on a Combinatoric Setting]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6069881]]></link>
			<description><![CDATA[This paper proposes a general framework that is able to define a set of classification algorithms for brain&#x2013;computer interfacing (BCI). We define a distributed representation of the EEG based on multichannel autoregressive models. In a subsequent step, we extend this multichannel modeling in a combinatoric setting, which is able to describe with a class of nonlinear combinatoric operators the embedded relationships that the EEG shows in the manifolds. The generality and the flexibility of the nonlinear combinatoric operators and their mathematical properties allow the design of an indefinite number of classification algorithms each displaying relevant properties, such as linearity with respect to the parameters, noise rejection, low computational complexity of the classification procedure. In such a way, we obtain an intuitive and rigorous way to design new BCI algorithms. As an example of this theoretical framework, we present a novel classification algorithm based on four properties of this nonlinear combinatoric operator. The method was validated on the classification of single-trial EEG signals recorded during motor imagination, and it was compared on two additional standard datasets obtained from the BCI competition, with other feature extraction and classification techniques based on common spatial pattern, common spatial subspace decomposition and Fisher discriminant analysis, linear discriminant analysis, Markov chains, and expectation maximization. In conclusion, the proposed framework is suited for a broad number of BCI applications.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6069881]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1446</startPage>
			<endPage>1459</endPage>
			<fileSize>3370</fileSize>
			<authors><![CDATA[Gianfelici, F.;Farina, D.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Universal Switching FIR Filtering]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086633]]></link>
			<description><![CDATA[We revisit recently considered universal finite-impulse-response (FIR) filtering problem and devise a scheme that asymptotically attains the expected mean-square error (MSE) of the best switching FIR filters for every underlying bounded, real-valued signal, provided that the switch rate of the best filters are sufficiently slow. As a performance metric, we consider adaptive expected regret, the maximum difference between the expected MSE of our filter and that of the best FIR filter over any contiguous time interval. Our algorithm is shown to have <formula formulatype="inline"><tex Notation="TeX">$O(log^{2} n)$</tex></formula> bound on the adaptive expected regret with <formula formulatype="inline"><tex Notation="TeX">$O(n^{2})$</tex> </formula> time-complexity, where <formula formulatype="inline"><tex Notation="TeX">$n$</tex> </formula> is the length of the signal; the bound implies that, regardless of the underlying signal, the expected MSE of our filter universally converges to that of the best switching FIR filters, if the number of switches is <formula formulatype="inline"><tex Notation="TeX">$o({{n} over {log^{2}n}})$</tex> </formula>. The experimental results show that our filter outperforms its stationary counterpart particularly when the underlying signal has time-varying characteristics. We also show a heuristic scheme with <formula formulatype="inline"> <tex Notation="TeX">$O(n)$</tex></formula> time-complexity works well without losing too much of the filtering performance.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086633]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1460</startPage>
			<endPage>1464</endPage>
			<fileSize>411</fileSize>
			<authors><![CDATA[Moon, T.;]]></authors>
		</item>
		<item>
			<title><![CDATA[RIP-Based Near-Oracle Performance Guarantees for SP, CoSaMP, and IHT]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6071013]]></link>
			<description><![CDATA[This correspondence presents an average case denoising performance analysis for SP, CoSaMP, and IHT algorithms. This analysis considers the recovery of a noisy signal, with the assumptions that it is corrupted by an additive random zero-mean white Gaussian noise and has a <formula formulatype="inline"><tex Notation="TeX">$K$</tex></formula>-sparse representation with respect to a known dictionary <formula formulatype="inline"><tex Notation="TeX">${bf D}$</tex> </formula>. The proposed analysis is based on the RIP, establishing a near-oracle performance guarantee for each of these algorithms. Beyond bounds for the reconstruction error that hold with high probability, in this work we also provide a bound for the average error.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6071013]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1465</startPage>
			<endPage>1468</endPage>
			<fileSize>202</fileSize>
			<authors><![CDATA[Giryes, R.;Elad, M.;]]></authors>
		</item>
		<item>
			<title><![CDATA[On Approximate Diagonalization of Correlation Matrices in Widely Linear Signal Processing]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096444]]></link>
			<description><![CDATA[The so called &#x201C;augmented&#x201D; statistics of complex random variables has established that both the covariance and pseudocovariance are necessary to fully describe second order properties of noncircular complex signals. To jointly decorrelate the covariance and pseudocovariance matrix, the recently proposed strong uncorrelating transform (SUT) requires two singular value decompositions (SVDs). In this correspondence, we further illuminate the structure of these matrices and demonstrate that for univariate noncircular data it is sufficient to diagonalize the pseudocovariance matrix&#x2014;this ensures that the covariance matrix is also approximately diagonal. The proposed approach is shown to result in lower computational complexity and enhanced numerical stability, and to enable elegant new formulations of performance bounds in widely linear signal processing. The analysis is supported by illustrative case studies and simulation examples.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096444]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1469</startPage>
			<endPage>1473</endPage>
			<fileSize>294</fileSize>
			<authors><![CDATA[Cheong Took, C.;Douglas, S. C.;Mandic, D. P.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Global Stabilization of the Least Mean Fourth Algorithm]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094232]]></link>
			<description><![CDATA[The least mean fourth algorithm has several stability problems. Its stability depends on the variance and distribution type of the adaptive filter input, the noise variance, and the initialization of the filter weights. The present correspondence provides a global solution to all these stability problems. This is achieved by normalizing the weight vector update term by a term that is fourth order in the regressor and second order in the estimation error. The former property stabilizes the algorithm against the variance and distribution type of the filter input, while the latter stabilizes the algorithm against the noise variance and the weight initialization. The obtained algorithm is stable for all values of the step-size between 0 and 2. The stability of the algorithm is supported by simulations.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094232]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1473</startPage>
			<endPage>1477</endPage>
			<fileSize>221</fileSize>
			<authors><![CDATA[Eweda, E.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Combination of Recursive Least <formula formulatype="inline">  <img src="/images/tex/387.gif" alt="p"> </formula>-Norm Algorithms for Robust Adaptive Filtering in Alpha-Stable Noise]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086634]]></link>
			<description><![CDATA[A method for adaptively minimizing the <formula formulatype="inline"> <tex Notation="TeX">$l_{p}$</tex></formula> norm relying on the convex combination of two recursive least <formula formulatype="inline"><tex Notation="TeX">$p$</tex> </formula>-norm <formula formulatype="inline"><tex Notation="TeX">$({rm RL}p{rm N})$</tex></formula> filters is presented. The approach is of interest when the noise is not Gaussian, for instance in the presence of impulsive or alpha-stable <formula formulatype="inline"><tex Notation="TeX">$(alpha {hbox{-}}{rm S})$</tex> </formula> distributed noise. In these cases, the <formula formulatype="inline"> <tex Notation="TeX">${rm RL}p{rm N}$</tex></formula> algorithm, aiming at recursively minimizing the <formula formulatype="inline"><tex Notation="TeX">$l_{p}$</tex> </formula> norm, offers a more stable and robust solution than adaptive filtering schemes based on the minimization of the squared error. However, since the <formula formulatype="inline"><tex Notation="TeX">${rm RL}p{rm N}$</tex></formula> solution cannot be obtained in closed form for <formula formulatype="inline"><tex Notation="TeX">$pne 2$</tex></formula>, it is necessary to introduce some approximations that critically affect the filter behavior. The main observed drawback is a poor convergence rate in nonstationary scenarios, especially in the presence of abrupt changes in the model. In this correspondence, we show how this problem can be overcome by relying on convex combinations of two <formula formulatype="inline"> <tex Notation="TeX">${rm RL}p{rm N}$</tex></formula> filters with long and short memories. The proposed methods are empirically shown to outperform state-of-the-art methods for this problem, requiring just slightly higher computation than its close competitors.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086634]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1478</startPage>
			<endPage>1482</endPage>
			<fileSize>403</fileSize>
			<authors><![CDATA[Navia-Vazquez, A.;Arenas-Garcia, J.;]]></authors>
		</item>
		<item>
			<title><![CDATA[A Technique for Efficient Realization of Wide-Band FIR LTI Systems]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096445]]></link>
			<description><![CDATA[This correspondence introduces a technique for efficient realization of wide-band finite-length impulse response (FIR) linear and time-invariant (LTI) systems. It divides the overall frequency region into three subregions through lowpass, bandpass, and highpass filters realized in terms of only one filter. The actual function to be approximated is in the low- and high-frequency regions realized using periodic subsystems. In this way, one can realize an overall wide-band LTI function in terms of three low-cost subblocks, leading to a reduced overall arithmetic complexity as compared to the regular realization. A systematic design technique is provided and a detailed example shows multiplication and addition savings of 62% and 48%, respectively, for a fractional-order differentiator with a 96% utilization of the bandwidth. Another example shows that the savings increase/decrease with increased/decreased bandwidth.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096445]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1482</startPage>
			<endPage>1486</endPage>
			<fileSize>211</fileSize>
			<authors><![CDATA[Sheikh, Z. U.;Johansson, H.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Spatio-Spectral Analysis on the Sphere Using Spatially Localized Spherical Harmonics Transform]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086770]]></link>
			<description><![CDATA[This correspondence studies a spatially localized spectral transform for signals on the unit sphere, which we call spatially localized spherical harmonics transform (SLSHT). For a systematic treatment, we explicitly express the transform in terms of rotated versions of an azimuthally symmetric window function and introduce the spatio-spectral SLSHT distribution with a succinct matrix representation. We present guidelines for the choice of the window function in the SLSHT, based on the inherent tradeoff between the spatial and spectral resolution of different window functions from the perspective of the uncertainty principle. We demonstrate the use of an eigenfunction window, obtained from the Slepian concentration problem on the sphere, as a good choice for window function. As an illustration, we apply the transform to the topographic map of Mars, which can reveal spatially localized spectral contributions that were not obtainable from traditional spherical harmonics analysis.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086770]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1487</startPage>
			<endPage>1492</endPage>
			<fileSize>704</fileSize>
			<authors><![CDATA[Khalid, Z.;Durrani, S.;Sadeghi, P.;Kennedy, R. A.;]]></authors>
		</item>
		<item>
			<title><![CDATA[LLL Algorithm and the Optimal Finite Wordlength FIR Design]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094235]]></link>
			<description><![CDATA[In practical finite-impulse-response (FIR) digital filter applications, it is often necessary to represent the filter coefficients with a finite number of bits. The finite wordlength restriction increases the filter deviation. This increase can be reduced substantially if the optimal finite wordlength coefficients are used. The time needed to compute these coefficients is greatly reduced with the help of a lower bound on the deviation increase. Derivation of an improved lower bound that uses the well-known LLL algorithm is presented in this correspondence.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094235]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1493</startPage>
			<endPage>1498</endPage>
			<fileSize>280</fileSize>
			<authors><![CDATA[Kodek, D. M.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Reconstruction of Uniformly Sampled Sequence From Nonuniformly Sampled Transient Sequence Using Symmetric Extension]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104416]]></link>
			<description><![CDATA[In this correspondence, reconstruction of a uniformly sampled sequence from a nonuniformly sampled transient sequence using symmetric extension is described. First, a relationship between the discrete Fourier transform (DFT) of a uniformly sampled sequence and the DFT of a nonuniformly sampled sequence is obtained. From the relationship, the formula to reconstruct the DFT of a uniformly sampled sequence from the DFT of a nonuniformly sampled sequence is derived when the nonuniform sampling ratios are known. Second, a symmetric extension of the nonuniformly sampled sequence is described to avoid discontinuity that adds high-frequency content in the DFT. Finally, experimental results are presented.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6104416]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1498</startPage>
			<endPage>1501</endPage>
			<fileSize>355</fileSize>
			<authors><![CDATA[Park, S.-W.;Hao, W.-D.;Leung, C. S.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Extrapolation of Bandlimited Signals in Linear Canonical Transform Domain]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6082464]]></link>
			<description><![CDATA[The linear canonical transform (LCT) has been shown to be a powerful analyzing tool in signal processing. Many results of this transform are already known, including bandlimited extrapolation. The existing algorithm for solving the problem of LCT bandlimited extrapolation is based on signal expansion into a series of generalized prolate spheroidal wave functions (GPSWFs). However, the requirement to compute and store the GPSWFs and the errors due to the series truncation render this algorithm ill-suitable for a practical implementation. In this correspondence, we first propose a new formulation of the Gerchberg&#x2013;Papoulis (GP) algorithm for LCT bandlimited extrapolation. Then, we present a fast convergence algorithm for the new formulation. The classical GP algorithm related to Fourier bandlimited signals is noted as a special case. Moreover, the comparison between the proposed algorithm and the one based on GPSWFs expansion is provided, and the validity of the theoretical derivations is demonstrated via simulations. Several potential applications of the achieved theory are also presented.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6082464]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1502</startPage>
			<endPage>1508</endPage>
			<fileSize>378</fileSize>
			<authors><![CDATA[Shi, J.;Sha, X.;Zhang, Q.;Zhang, N.;]]></authors>
		</item>
		<item>
			<title><![CDATA[The Consistency of MDL for Linear Regression Models With Increasing Signal-to-Noise Ratio]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6095654]]></link>
			<description><![CDATA[Recent work by Ding and Kay has demonstrated that the Bayesian information criterion (BIC) is an inconsistent estimator of model order in nested model selection as the noise variance <formula formulatype="inline"><tex Notation="TeX">$tau^{ast}rightarrow 0$</tex></formula>. Unfortunately, Ding and Kay have erroneously concluded that the minimum description length (MDL) principle also leads to inconsistent estimates of model order in this setting by equating BIC with MDL. This correspondence shows that only the earlier MDL criterion based on asymptotic assumptions has this problem, and proves that the new MDL linear regression criteria based on normalized maximum likelihood and Bayesian mixture codes satisfy the notion of consistency as <formula formulatype="inline"><tex Notation="TeX">$tau^{ast}rightarrow 0$</tex></formula>. The main result may be used as a basis to easily establish similar consistency results for other closely related information theoretic regression criteria.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6095654]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1508</startPage>
			<endPage>1510</endPage>
			<fileSize>133</fileSize>
			<authors><![CDATA[Schmidt, D. F.;Makalic, E.;]]></authors>
		</item>
		<item>
			<title><![CDATA[A Semi-Closed-Form Solution to Optimal Distributed Beamforming for Two-Way Relay Networks]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093981]]></link>
			<description><![CDATA[In this correspondence, we present a computationally simple semi-closed-form solution to the problem of designing distributed beamformer for two-way (bi-directional) multi-relay networks. In such a network, the relay nodes use amplify-and-forward relaying protocol to help two transceivers exchange information in a bidirectional manner. We consider a total power minimization approach to optimally find the relay beamforming weights and the transceiver transmit powers. This approach is based on the minimization of the total transmit power, consumed in the whole network, subject to SNR constraints at the two transceivers. We show that as far as the relay beamforming weight vector is concerned, this minimization problem is equivalent to the minimization of the total transmit power for a one-way relay network where the target SNR of the receiving transceiver is equal to the sum of the target SNRs of the two transceivers in the original two-way relay network. Based on this observation, we show that the relay beamforming weight vector can be obtained in a closed from given that an intermediate parameter, namely the transmit power of the transmitter in the equivalent one-way relay network, is available. This intermediate parameter is shown to be the solution to a one-dimensional optimization problem, and thus, it can be obtained using a simple bisection method. Our semi-closed-form solution not only reveals the structure of the optimal beamforming weight vector, but also leads to a one-dimensional search regardless of the number of relays. This provides the computational advantage over the gradient based numerical method of Havary-Nassab <etal/>, where the gradient dimension reflects the number of relays.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6093981]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1511</startPage>
			<endPage>1516</endPage>
			<fileSize>266</fileSize>
			<authors><![CDATA[Shahbazpanahi, S.;Dong, M.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Waveform Optimization for MIMO Radar in Colored Noise: Further Results for Estimation-Oriented Criteria]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086773]]></link>
			<description><![CDATA[Previous work on multiple-input multiple-output (MIMO) radar waveform design in colored noise based on minimizing minimum mean-square error (MMSE) and normalized mean-square error (NMSE) only considered the optimization of the singular value of the waveform matrix, while the optimization of singular vectors, which forms a basis of singular value optimization, was not stressed enough. In this correspondence, we revisit the waveform design problem for MIMO radar in colored noise based on MMSE and NMSE respectively. Further results for the optimal solutions under both criteria are derived. We prove the optimality of &#x201C;matching&#x201D; with the target and colored noise for the singular vectors of the transmitted waveform. Moreover, to obtain minimum MMSE/NMSE, the pairing of the eigenvectors of the target and noise should be carefully designed, where the optimal pairing of the eigenvectors based on MMSE is fixed while the optimal pairing based on NMSE changes according to the transmitted power, the strength of the target, and noise.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6086773]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1517</startPage>
			<endPage>1522</endPage>
			<fileSize>477</fileSize>
			<authors><![CDATA[Tang, B.;Tang, J.;Peng, Y.;]]></authors>
		</item>
		<item>
			<title><![CDATA[An SNR Dependent Model for the CDMA FLL]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6099641]]></link>
			<description><![CDATA[The tracking bandwidth and variance of the code-division multiple-access (CDMA) frequency-lock loop (FLL) are considered in this correspondence. It is shown that the behavior of the discriminator can vary significantly with varying <formula formulatype="inline"><tex Notation="TeX">${hbox{SNR}}_{c}$</tex> </formula> with corresponding variations in loop performance. A model is presented which reflects this signal-to-noise ration (SNR)-sensitivity and, thus, facilitates accurate loop design.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6099641]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1522</startPage>
			<endPage>1527</endPage>
			<fileSize>857</fileSize>
			<authors><![CDATA[Curran, J. T.;Lachapelle, G.;Murphy, C. C.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Minimum Euclidean Distance Based Precoders for MIMO Systems Using Rectangular QAM Modulations]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094228]]></link>
			<description><![CDATA[From the feedback of the channel state information (CSI), precoding techniques improve the performance of multiple-input multiple-output (MIMO) systems by optimizing various criteria. In this correspondence, an efficient precoder that maximizes the minimum distance <formula formulatype="inline"> <tex Notation="TeX">$(d_{min})$</tex></formula> of two received vectors is studied. This criterion leads to a nondiagonal precoding scheme and allows achieving a full diversity order. However, the optimized solution for MIMO systems using a high-order QAM modulation is rather complex and changes for different constellations. Therefore, we propose herein a general form of minimum Euclidean distance based precoders for all rectangular QAM modulations. It is shown that the new solution optimizes the distance <formula formulatype="inline"> <tex Notation="TeX">$d_{min}$</tex></formula> for small and large dispersive channels.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6094228]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1527</startPage>
			<endPage>1533</endPage>
			<fileSize>897</fileSize>
			<authors><![CDATA[Ngo, Q.-T.;Berder, O.;Scalart, P.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Adaptive Gating for Multitarget Tracking With Gaussian Mixture Filters]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096446]]></link>
			<description><![CDATA[In this correspondence, we use a generalization of the Bayesian approach to the multitarget problem that goes under the name of cardinalized probability hypothesis density (CPHD) filter to jointly estimate a time varying number of targets and their locations from sets of noisy range measurements. While in the case of Gaussian linear models a closed-form solution for the CPHD recursion exists in the form of a Gaussian mixture (GM), the more general case of nonlinear systems suboptimal solutions becomes necessary. Due to the Gaussianity assumption in the the GM-CPHD filter, we propose to integrate the square-root cubature Kalman filter (S-CKF) into the GM-CPHD recursion. A novel weighted gating strategy, which exploits the GM implementation of the proposed S-CKF-GM-CPHD filter, is offered to lower the computational time by adaptively increasing the gate sizes in proportion to the likelihood of the single GM components. The results reveal that the proposed gating yields considerable savings in processing requirements compared to no gating, without any significant degradation in performance. In addition, although the run time improvement achieved with elliptical or adaptive gating is equivalent, the latter does not degrade the results.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6096446]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1533</startPage>
			<endPage>1538</endPage>
			<fileSize>482</fileSize>
			<authors><![CDATA[Macagnano, D.;Freitas de Abreu, G. T.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Comments on &#x201C;Joint Detection and Estimation of Multiple Objects From Image Observations&#x201D;]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6061971]]></link>
			<description><![CDATA[The above article [1] introduced an algorithm for multitarget track-before-detect based on a multi-Bernoulli random finite set model (MB-TBD). This new algorithm was compared with the Histogram Probabilistic Multi-Hypothesis Tracker (H-PMHT) on simulated data examples containing multiple targets with non-linear dynamics. The authors reported poor performance from H-PMHT and described several deficiencies of the algorithm. This note highlights unnecessary assumptions made in the assessment of H-PMHT and repeats two of the simulation examples after relaxing them. We demonstrate a substantial improvement in performance compared with the originally published results. The simulation example is also shown to be a relatively high signal to noise problem and good performance is obtained from a conventional detect-then-track algorithm.]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6061971]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1539</startPage>
			<endPage>1540</endPage>
			<fileSize>520</fileSize>
			<authors><![CDATA[Davey, S. J.;]]></authors>
		</item>
		<item>
			<title><![CDATA[Reply to &#x201C;Comments on `Joint Detection and Estimation of Multiple Objects from Image Observations'&#x201D;]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6105580]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6105580]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1540</startPage>
			<endPage>1541</endPage>
			<fileSize>422</fileSize>
			<authors><![CDATA[Vo, B.-N.;Vo, B.-T.;Pham, N.-T.;Suter, D.;]]></authors>
		</item>
		<item>
			<title><![CDATA[IEEE Transactions on Signal Processing Edics]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146824]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146824]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1542</startPage>
			<endPage>1542</endPage>
			<fileSize>30</fileSize>
			<authors><![CDATA[]]></authors>
		</item>
		<item>
			<title><![CDATA[IEEE Transactions on Signal Processing information for authors]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146823]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146823]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>1543</startPage>
			<endPage>1544</endPage>
			<fileSize>46</fileSize>
			<authors><![CDATA[]]></authors>
		</item>
		<item>
			<title><![CDATA[IEEE Signal Processing Society Information]]></title>
			<link><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146826]]></link>
			<description><![CDATA[ ]]></description>
			<pubDate><![CDATA[March  2012]]></pubDate>
			<guid><![CDATA[http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=6146821&arnumber=6146826]]></guid>
			<volume>60</volume>
			<issue>3</issue>
			<startPage>C3</startPage>
			<endPage>C3</endPage>
			<fileSize>33</fileSize>
			<authors><![CDATA[]]></authors>
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