IEEE Transactions on Signal Processing

Using the Wigner distribution, we derive and analyze a matrix formulation for the chirplet transform, a signal analysis tool that generalizes the wavelet and short-time Fourier transforms. The formulation expresses the translations, scalings, and shears of the chirplet transform in terms of affine matrix transformations on the time-frequency plane. Our approach leads naturally to several new signa... View full abstract»

An algorithm has been developed to control the antenna array beam pattern for satellite communication applications where the uplink received antenna is desired to provide beamforming capability for selective earth coverage and jamming protection. The problem can be formulated to determine the antenna weight to meet the user gain and jammer null requirements in their known directions, subject to th... View full abstract»

By using block processing, partitioning, and fast Fourier transforms (FFTs), large filters perform efficiently in the frequency domain. For small processing delay the complexity can still be too large for implementation on a digital signal processor (DSP). A solution is to partition the filter into unequal-length subfilters. Application in adaptive filtering yields the nonuniform partitioned block... View full abstract»

Two-dimensional full-plane and half-plane filters are more general, and much better frequency responses can be obtained than the quarter-plane filters. In this correspondence, the analytical least squares method is generalized and extended for designing 2-D full-plane and half-plane linear phase complex FIR digital filters. The 2-D filter's coefficients can be effectively determined by use of a cl... View full abstract»

A global optimization method is introduced that minimize the rate of misclassification. We first derive the theoretical basis for the method, on which we base the development of a novel design algorithm and demonstrate its effectiveness and superior performance in the design of practical classifiers for some of the most popular structures currently in use. The method, grounded in ideas from statis... View full abstract»

Data flow languages are a natural and more formal way to describe the flow of computations in a DSP application. However, when the language contains extended array constructs, extra data dependency checks are needed. This correspondence describes a new model to represent multidimensional arrays and presents several data flow analysis techniques for multidimensional arrays. Results show very fast r... View full abstract»

Blind identification consists of estimating the impulse response of a linear, time-invariant channel used for transmission of digital data by observing the channel output without knowledge of the transmitted symbol sequence. We show how a previously proposed algorithm (Moulines et al., 1995) based on second-order statistics of the received signal can be modified to account for correlated noise View full abstract»

We investigate linear independence of steering vectors of one electromagnetic vector sensor. We show that every three steering vectors with distinct directions of arrival (DOAs) are linearly independent. We also show that four steering vectors with distinct DOAs are linearly independent if the ellipticity angles of the signals associated with any two of the four steering vectors are distinct. More... View full abstract»

The problem of finding the optimal set of causal pixels (support) for use in linear predictive models is addressed. After presenting counterexamples to popular intuitions about supports, a general result relating the distortion incurred with a small support to optimal coefficients of a larger support is derived. A geometrical interpretation is provided. Two algorithms that optimally increase/decre... View full abstract»

A parametric model and a corresponding parameter estimation algorithm for unwrapping 2-D phase functions are presented. The proposed algorithm performs global analysis of the observed signal. Since this analysis is based on parametric model fitting, the proposed phase unwrapping algorithm has low sensitivity to phase aliasing due to low sampling rates and noise, as well as to local errors. In its ... View full abstract»

This paper addresses the influence that the sampling locations have on the estimated frequencies of superimposed sinusoids. This problem has application in harmonic time-series analysis or direction-finding phased-array systems. Generalized mathematical bounds are developed in terms that are independent of the array locations and have an intuitively appealing physical interpretation. They establis... View full abstract»

A structure for implementing lapped transforms with time-varying block sizes that allows full orthogonality of the transient transforms is presented. The formulation is based on a factorization of the transfer matrix into orthogonal factors. Such an approach can be viewed as a sequence of stages with variable-block-size transforms separated by sample-shuffling (delay) stages. Details and design ex... View full abstract»

The discrete Laguerre transform (DLT) belongs to the family of unitary transforms known as Gauss-Jacobi transforms. Using classical methodology, the DLT is derived from the orthonormal set of Laguerre functions. By examining the basis vectors of the transform matrix, the types of signals that can be best represented by the DLT are determined. Simulation results are used to compare the DLT's effect... View full abstract»

Source separation consists of recovering a set of independent signals when only mixtures with unknown coefficients are observed. This paper introduces a class of adaptive algorithms for source separation that implements an adaptive version of equivariant estimation and is henceforth called equivariant adaptive separation via independence (EASI). The EASI algorithms are based on the idea of serial ... View full abstract»

Discusses the circular (or periodic) implementation of vector-channel least-squares lattice filters. The resulting algorithm eliminates matrix inversions and greatly reduces matrix multiplications. Complete residual and AR coefficient algorithms are provided, as well as initialization and other implementation issues. A full treatment of the prediction error form of the lattices and its relationshi... View full abstract»

Code division multiple access (CDMA) schemes allow a number of asynchronous users to share a transmission medium with minimum cooperation among them. However, sophisticated signal processing algorithms are needed at the receiver to combat interference from other users and multipath effects. A discrete-time multirate formulation is introduced for asynchronous CDMA systems, which can incorporate mul... View full abstract»

This paper presents an unsupervised method for restoration of sparse spike trains. These signals are modeled as random Bernoulli-Gaussian processes, and their unsupervised restoration requires (i) estimation of the hyperparameters that control the stochastic models of the input and noise signals and (ii) deconvolution of the pulse process. Classically, the problem is solved iteratively using a max... View full abstract»

Joint signal representations (JSRs) of arbitrary variables generalize time-frequency representations (TFRs) to a much broader class of nonstationary signal characteristics. Two main distributional approaches to JSRs of arbitrary variables have been proposed by Cohen (see Time-Frequency Analysis, Englewood Cliffs, NJ, Prentice Hall, 1995 and Proc. SPIE 1566, San Diego, 1991) and Baraniuk (see Proc.... View full abstract»

In an attempt to gain insight into the design of linear detectors for additive white noise channels (discrete-time case), we describe several procedures, both optimal and suboptimal. Using the Wiener representation for nonlinear systems, we derive an ad hoc suboptimal design procedure. Exact designs are found when the noise amplitude's probability distribution is stable and when the noise is Lapla... View full abstract»

An efficient implementation algorithm for the Koilpillai-Vaidyanathan (see ibid., vol.41, no.1, p.82-92, 1993) pseudo quadrature mirror filter (KVPQMF) bank, which is useful in audio compression schemes, is presented. The implementation employs a polyphase system with discrete cosine transforms (DCTs). Theoretical and practical results show a typical saving in computational load of 82% over the di... View full abstract»

In this correspondence, we present a result in probability that does not exist in the literature of probability theory. We show one application of this theorem in multidimensional signal processing, which generalizes some of the existing results View full abstract»

In several practical applications of the LMS algorithm, including certain VLSI implementations, the coefficient adaptation can be performed only after some fixed delay. The resulting algorithm is known as the delayed LMS (DLMS) algorithm in the literature. Previous published analyses of this algorithm are based on mean and moment convergence under the independence assumption between successive inp... View full abstract»

Describes a new adaptive linear-phase filter whose weights are updated by the normalized least-mean-square (LMS) algorithm in the transform domain. This algorithm provides a faster convergence rate compared with the time domain linear phase LMS algorithm. Various real-valued orthogonal transforms are investigated such as the discrete cosine transform (DCT), discrete Hartley transform (DHT), and po... View full abstract»