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Signal Processing, IEEE Transactions on

Issue 8 • Date Aug 2001

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Displaying Results 1 - 23 of 23
  • Differential space-code modulation for interference suppression

    Page(s): 1786 - 1795
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (308 KB) |  | HTML iconHTML  

    Space-time coding has been receiving much attention due to its potentials offered by fully exploiting the spatial and temporal diversities of multiple transmit and receive antennas. A differential space-time modulation (DSTM) scheme was previously proposed for demodulation without channel state information, which is attractive in fast fading channels where accurate channel estimates are difficult to obtain. However, this technique is sensitive to interference and is likely to deteriorate or even break down in a wireless environment, where interference (including intentional and unintentional jamming) signals exist. We propose a new coding and modulation scheme, referred to as the differential space-code modulation (DSCM), which is interference resistant. Our focus is on single-user communications. We show that DSCM outperforms DSTM significantly when interference is present. This advantage is achieved at the cost of a lower data rate or a wider bandwidth or a combination of both. To alleviate this problem, a high-rate DSCM (HR-DSCM) scheme is also presented, which increases the data rate considerably at the cost of a slightly higher bit-error rate (BER), while still maintaining the interference suppression capability View full abstract»

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  • Asymptotic properties of the algebraic constant modulus algorithm

    Page(s): 1796 - 1807
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB) |  | HTML iconHTML  

    The algebraic constant modulus algorithm (ACMA) is a noniterative blind source separation algorithm. It computes jointly beamforming vectors for all constant modulus sources as the solution of a joint diagonalization problem. We analyze its asymptotic properties and show that (unlike CMA) it converges to the Wiener beamformer when the number of samples or the signal-to-noise ratio (SNR) goes to infinity. We also sketch its connection to the related JADE algorithm and derive a version of ACMA that converges to a zero-forcing beamformer. This gives improved performance in applications that use the estimated mixing matrix, such as in direction finding View full abstract»

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  • Multiwindow time-varying spectrum with instantaneous bandwidth and frequency constraints

    Page(s): 1656 - 1666
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (348 KB)  

    We build on Cohen's work (Cohen and Lee 1988, 1989; Cohen 1990, 1995) on instantaneous bandwidth and frequency by extending it to a multiwindow framework for polynomial phase signals. Unlike the case with a single spectrogram, which Cohen considered, our multiwindow framework allows one to obtain a time-varying spectral estimate that simultaneously satisfies instantaneous bandwidth and frequency constraints. We then develop a method utilizing this new multiwindow time-varying spectral technique for estimating the instantaneous frequency of a signal. The method is computationally simple, asymptotically unbiased for noise-free signals, and provides a signal-to-noise ratio (SNR) improvement of more than 3 dB over other estimators, including the cross-polynomial Wigner distribution method, for quadratic and cubic FM signals View full abstract»

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  • A time-delay digital tanlock loop

    Page(s): 1808 - 1815
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (200 KB) |  | HTML iconHTML  

    We propose a nonuniform sampling digital tanlock loop (DTL) that utilizes a constant time-delay unit instead of the constant 90° phase shifter The new structure reduces the complexity of implementation and avoids many of the practical problems associated with the digital Hilbert transformer like the approximations and frequency limitations. The time-delay digital tanlock loop (TDTL) preserves the most important features of the conventional DTL (CDTL), such as reduced sensitivity to the variation of the signal power. It also introduces improvement over the first-order CDTL under suitable choice of the circuit parameters. The first- and second-order loops are analyzed for locking conditions and steady-state phase error View full abstract»

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  • Mixed time scale recursive algorithms

    Page(s): 1824 - 1830
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB)  

    We investigate the behavior of certain types of mixed time scale adaptive algorithms. These systems comprise a “fast” or quickly changing algorithm mutually coupled to a “slow” or slowly changing algorithm. They arise naturally in a variety of adaptive environments such as in IIR system identification, the training of recurrent neural networks, decision feedback equalization, and others. [These algorithms (despite their title) should not be confused with the mixed time scales of wavelet transforms or other algorithms associated with multiresolution signal processing]. We give conditions for when the system can be analyzed from the framework of a simpler “frozen state” system. This analysis extends some of the previous work of Solo (1995) and his coworkers View full abstract»

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  • On the convergence of Volterra filter equalizers using a pth-order inverse approach

    Page(s): 1734 - 1744
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    The pth-order inverse method is one of important approaches to Volterra equalization. However, when a pth-order Volterra equalizer instead of an exact Volterra equalizer is connected in cascade before (after) a nonlinear system, the existence of Volterra filter equalization and the approximation output error bound of the resulting system have yet to be reported. In this paper, the concept of local l 2 stability for a Volterra system is introduced, and the algorithmic formulae of a pth-order inverse equalizer via a multidimension z-transform are presented. The output error signal and the approximation output error bound of the resulting system are investigated as well. It is shown that the approximation output error tends to zero as p tends to infinity for a finite range of input amplitude values. Finally some simulation results are presented and discussed View full abstract»

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  • Performance analysis of blind carrier phase estimators for general QAM constellations

    Page(s): 1816 - 1823
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    Large quadrature amplitude modulation (QAM) constellations are currently used in throughput efficient high-speed communication applications such as digital TV. For such large signal constellations, carrier-phase synchronization is a crucial problem because for efficiency reasons, the carrier acquisition must often be performed blindly, without the use of training or pilot sequences. The goal of the paper is to provide thorough performance analysis of the blind carrier phase estimators that have been proposed in the literature and to assess their relative merits View full abstract»

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  • M-band compactly supported orthogonal symmetric interpolating scaling functions

    Page(s): 1704 - 1713
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB) |  | HTML iconHTML  

    In many applications, wavelets are usually expected to have the following properties: compact support, orthogonality, linear-phase, regularity, and interpolation. To construct such wavelets, it is crucial designing scaling functions with the above properties. In two- and three-band cases, except for the Haar functions, there exists no scaling function with the above five properties. In M-band case (M⩾4), more free degrees available in design enable us to construct such scaling functions. A novel approach to designing such scaling functions is proposed. First, we extend the two-band Dubuc (1986) filters to the M-band case. Next, the M-band FIR regular symmetric interpolating scaling filters are parameterized, and then, M-band FIR regular orthogonal symmetric interpolating scaling filters (OSISFs) are designed via optimal selection of parameters. Finally, two family of four-band and five-band OSISFs and scaling functions are developed, and their smoothness are estimated View full abstract»

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  • Eavesdropping in the synchronous CDMA channel: an EM-based approach

    Page(s): 1748 - 1756
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (236 KB) |  | HTML iconHTML  

    The problem of blind detection in a synchronous code division multiple access (CDMA) system when there is no knowledge of the users' spreading sequences is considered. An expectation maximization (EM)-based algorithm that exploits the finite alphabet (FA) property of the digital communications source is proposed. Simulations indicate that this approach, which makes use of knowledge of the subspace spanned by the signaling multiplex, achieves the Cramer-Rao lower bound (CRB). The issues of subspace estimation and timing acquisition are also considered View full abstract»

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  • Boundary filter optimization for segmentation-based subband coding

    Page(s): 1718 - 1727
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (308 KB)  

    This paper presents boundary optimization techniques for the nonexpansive decomposition of arbitrary-length signals with multirate filterbanks. Both biorthogonal and paraunitary filterbanks are considered. The paper shows how matching moments and orthonormality can be imposed as additional conditions during the boundary filter optimization process. It provides direct solutions to the problem of finding good boundary filters for the following cases: (a) biorthogonal boundary filters with exactly matching moments and (b) orthonormal boundary filters with almost matching moments. With the proposed methods, numerical optimization is only needed if orthonormality and exactly matching moments are demanded. The proposed direct solutions are applicable to systems with a large number of subbands and/or very long filter impulse responses. Design examples show that the methods allow the design of boundary filters with good frequency selectivity View full abstract»

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  • Performance analysis of Godard-based blind channel identification

    Page(s): 1757 - 1767
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (312 KB) |  | HTML iconHTML  

    We analyze a blind channel impulse response identification scheme based on the cross correlation of blind symbol estimates with the received signal. The symbol estimates specified are those minimizing the Godard (1980) (or constant modulus) criterion, for which mean-squared symbol estimation error bounds have been derived. We derive upper bounds for the average squared parameter estimation error (ASPE) of the blind identification scheme that depend on the mean-squared error of the Wiener equalizer, the kurtoses of the desired and interfering sources, and the channel impulse response. The effects of finite data length and stochastic gradient equalizer design on ASPE are also investigated. All results are derived in a general multiuser vector-channel context View full abstract»

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  • On the behavior of information theoretic criteria for model order selection

    Page(s): 1689 - 1695
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (168 KB) |  | HTML iconHTML  

    The Akaike (1974) information criterion (AIC) and the minimum description length (MDL) are two well-known criteria for model order selection in the additive white noise case. Our aim is to study the influence on their behavior of a large gap between the signal and the noise eigenvalues and of the noise eigenvalue dispersion. Our results are mostly qualitative and serve to explain the behavior of the AIC and the MDL in some cases of great practical importance. We show that when the noise eigenvalues are not clustered sufficiently closely, then the AIC and the MDL may lead to overmodeling by ignoring an arbitrarily large gap between the signal and the noise eigenvalues. For fixed number of data samples, overmodeling becomes more likely for increasing the dispersion of the noise eigenvalues. For fixed dispersion, overmodeling becomes more likely for increasing the number of data samples. Undermodeling may happen in the cases where the signal and the noise eigenvalues are not well separated and the noise eigenvalues are clustered sufficiently closely. We illustrate our results by using simulations from the effective channel order determination area View full abstract»

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  • Signal enhancement using beamforming and nonstationarity with applications to speech

    Page(s): 1614 - 1626
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (316 KB)  

    We consider a sensor array located in an enclosure, where arbitrary transfer functions (TFs) relate the source signal and the sensors. The array is used for enhancing a signal contaminated by interference. Constrained minimum power adaptive beamforming, which has been suggested by Frost (1972) and, in particular, the generalized sidelobe canceler (GSC) version, which has been developed by Griffiths and Jim (1982), are the most widely used beamforming techniques. These methods rely on the assumption that the received signals are simple delayed versions of the source signal. The good interference suppression attained under this assumption is severely impaired in complicated acoustic environments, where arbitrary TFs may be encountered. In this paper, we consider the arbitrary TF case. We propose a GSC solution, which is adapted to the general TF case. We derive a suboptimal algorithm that can be implemented by estimating the TFs ratios, instead of estimating the TFs. The TF ratios are estimated by exploiting the nonstationarity characteristics of the desired signal. The algorithm is applied to the problem of speech enhancement in a reverberating room. The discussion is supported by an experimental study using speech and noise signals recorded in an actual room acoustics environment View full abstract»

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  • A novel design of lifting scheme from general wavelet

    Page(s): 1714 - 1717
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (88 KB) |  | HTML iconHTML  

    A novel algorithm that is used to lift the vanishing moments of the wavelet from general wavelet, and not only from the Lazy wavelet, is proposed. It is based on the relationship between the vanishing moments of the wavelet and multiple of zeros of z=1 and only needs to solve simple linear equations to obtain the lifting coefficients. Moreover, the shortest lifting scheme and its uniqueness are introduced, from which an iterative algorithm for designing the lifting scheme is presented View full abstract»

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  • Robust ℋ filtering for uncertain discrete-time state-delayed systems

    Page(s): 1696 - 1703
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (204 KB)  

    This paper addresses the problem of robust ℋ filtering for linear discrete-time systems subject to parameter uncertainties in the system state-space model and with multiple time delays in the state variables. The uncertain parameters are supposed to belong to a given convex bounded polyhedral domain. A methodology is developed to design a stable linear filter that assures asymptotic stability and a prescribed ℋ performance for the filtering error, irrespective of the uncertainty and the time delays. The proposed design is given in terms of linear matrix inequalities, which has the advantage in that it can be implemented numerically very efficiently View full abstract»

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  • Frequency domain computations for nonlinear steady-state solutions

    Page(s): 1728 - 1733
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB) |  | HTML iconHTML  

    Steady-state analysis and Fourier analysis play a major role in linear signal processing. In response to a bounded input, a steady-state solution exists if all the poles of the discrete-time linear system are inside the unit circle. Despite the fact that there is no principle of superposition for nonlinear systems, under appropriate sufficient conditions (including all poles inside the unit circle for the linear part of the nonlinear system), there is a bounded solution for all time in response to a bounded input for all time for a time-varying nonlinear difference equation. All solutions that start sufficiently close to this unique solution converge to it as time goes to infinity. This steady-state solution can be computed by applying Fourier and inverse Fourier transforms to each step in a Picard process. In this paper, we develop an algorithm to compute (approximate) steady-state solutions for discrete-time, nonlinear difference equations by employing fast Fourier transforms and inverse fast Fourier transforms at each step of the iterative process. Simulations are provided to illustrate our algorithm View full abstract»

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  • Extensions of the weighted-sample method for digitizing continuous-time filters

    Page(s): 1627 - 1637
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB)  

    Methods continue to improve for taking the transfer function of a stable, continuous-time, single-input single-output system or filter and converting it to an “equivalent” discrete-time filter. The weighted-sample (WS) method of carrying out the digitizing process is one such method. It is a higher order method, compared with the first-order Tustin's method, so it usually achieves a smaller error for a given system, input, and sample time. Like the Tustin method, but unlike some higher order digitization methods, it allows the sample time to be selected without being constrained by stability considerations. This paper describes two extensions to the weighted-sample method. The first provides the ability to handle cases in which the continuous filter contains one or more series integrators. The second extension modifies the WS filter to a new form, termed the WS' filter, which turns out to be the same as the MSRP filter. Several examples are given, and accuracy is assessed View full abstract»

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  • Performance of a hybrid decision feedback equalizer structure for CAP-based DSL systems

    Page(s): 1768 - 1785
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (380 KB) |  | HTML iconHTML  

    We study the performance of a class of derision feedback equalizer (DFE) structures for high-speed digital transmission systems. We first present mathematical formulation of minimum mean-square error (MMSE) and the optimum tap coefficients for various finite-length phase-splitting equalizers over the loop in the presence of colored noise, such as near-end crosstalk (NEXT) and far-end crosstalk (FEXT). The performance of the equalizers is also analyzed in the presence of narrowband interference and the channel reflections introduced by bridged taps. The hybrid-type DFE (H-DFE) is presented as a practical equalizer structure for these applications. The results of analysis show that the H-DFE has advantages in the performance and/or in the implementation complexity as compared with the existing DFE structures. An additional advantage of the H-DFE is in the transmission systems that employ the precoding technique. The precoding for the H-DFE allows the system to track small changes in the channel View full abstract»

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  • Reflection coefficients counterpart of Cardan-Viete formulas

    Page(s): 1745 - 1747
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (60 KB) |  | HTML iconHTML  

    Polynomials can be represented by their coefficients or by their zeros. The link between these two representations is the Cardan-Viete formulas that let us express the coefficients as elementary symmetric functions in the zeros. In this paper, as a consequence of Levinson recursion, we present a counterpart to the Cardan-Viete formulas that expresses the polynomial coefficients or the regression vector of an autoregressive filter, when a lattice representation is considered, in terms of its reflection coefficients View full abstract»

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  • Relations between fractional operations and time-frequency distributions, and their applications

    Page(s): 1638 - 1655
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (548 KB) |  | HTML iconHTML  

    The fractional Fourier transform (FRFT) is a useful tool for signal processing. It is the generalization of the Fourier transform. Many fractional operations, such as fractional convolution, fractional correlation, and the fractional Hilbert transform, are defined from it. In fact, the FRFT can be further generalized into the linear canonical transform (LCT), and we can also use the LCT to define several canonical operations. In this paper, we discuss the relations between the operations described above and some important time-frequency distributions (TFDs), such as the Wigner distribution function (WDF), the ambiguity function (AF), the signal correlation function, and the spectrum correlation function. First, we systematically review the previous works in brief. Then, some new relations are derived and listed in tables. Then, we use these relations to analyze the applications of the FRPT/LCT to fractional/canonical filter design, fractional/canonical Hilbert transform, beam shaping, and then we analyze the phase-amplitude problems of the FRFT/LCT. For phase-amplitude problems, we find, as with the original Fourier transform, that in most cases, the phase is more important than the amplitude for the FRFT/LCT. We also use the WDF to explain why fractional/canonical convolution can be used for space-variant pattern recognition View full abstract»

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  • Frequency domain blind MIMO system identification based on second and higher order statistics

    Page(s): 1677 - 1688
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (340 KB) |  | HTML iconHTML  

    We present a novel frequency-domain framework for the identification of a multiple-input multiple-output (MIMO) system driven by white, mutually independent, unobservable inputs. The system frequency response is obtained based on singular value decomposition (SVD) of a matrix constructed based on the power-spectrum and slices of polyspectra of the system output. By appropriately selecting the polyspectra slices, we can create a set of such matrices, each of which could independently yield the solution, of they could all be combined in a joint diagonalization scheme to yield a solution with improved statistical performance. The freedom to select the polyspectra slices allows us to bypass the frequency-dependent permutation ambiguity that is usually associated with frequency domain SVD, while at the same time allows us compute and cancel the phase ambiguity. An asymptotic consistency analysis of the system magnitude response estimate is performed View full abstract»

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  • Adaptive Volterra filters for active control of nonlinear noise processes

    Page(s): 1667 - 1676
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (264 KB) |  | HTML iconHTML  

    This paper presents a Volterra filtered-X least mean square (LMS) algorithm for feedforward active noise control. The research has demonstrated that linear active noise control (ANC) systems can be successfully applied to reduce the broadband noise and narrowband noise, specifically, such linear ANC systems are very efficient in reduction of low-frequency noise. However, in some situations, the noise that comes from a dynamic system may he a nonlinear and deterministic noise process rather than a stochastic, white, or tonal noise process, and the primary noise at the canceling point may exhibit nonlinear distortion. Furthermore, the secondary path estimate in the ANC system, which denotes the transfer function between the secondary source (secondary speaker) and the error microphone, may have nonminimum phase, and hence, the causality constraint is violated. If such situations exist, the linear ANC system will suffer performance degradation. An implementation of a Volterra filtered-X LMS (VFXLMS) algorithm based on a multichannel structure is described for feedforward active noise control. Numerical simulation results show that the developed algorithm achieves performance improvement over the standard filtered-X LMS algorithm for the following two situations: (1) the reference noise is a nonlinear noise process, and at the same time, the secondary path estimate is of nonminimum phase; (2) the primary path exhibits the nonlinear behavior. In addition, the developed VFXLMS algorithm can also be employed as an alternative in the case where the standard filtered-X LMS algorithm does not perform well View full abstract»

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  • A subspace-based direction finding algorithm using fractional lower order statistics

    Page(s): 1605 - 1613
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB)  

    We propose several classes of fractional lower order moment (FLOM)-based matrices that can be used with MUSIC to estimate the DOAs of independent circular signals embedded in additive SαS (symmetric α stable) noise (e.g., sea clutter). We run simulations with different choices of the FLOM parameter p for our FLOM-based matrices and conclude that when the noise is SαS with unknown α≠2, FLOM-multiple signal classification (MUSIC) with p close to unity yields good performance. The performance of FLOM-MUSIC and robust covariation-based (ROC)-MUSIC are similar. Three scenarios that contain circular signals (phase modulation (PM), circularly symmetrical Gaussian, and quaternary phase-shift keying (QPSK)) and one scenario that contains noncircular signals (binary phase-shift keying (BPSK)), all embedded in the same SαS noise, are tested. These simulation results reveal that the scenario containing BPSK signals leads to poor performance, indicating that FLOM-MUSIC is presently limited to circular signals View full abstract»

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Aims & Scope

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

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Editor-in-Chief
Zhi-Quan (Tom) Luo
University of Minnesota