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Signal Processing, IEEE Transactions on

Issue 3 • Date March 1999

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Displaying Results 1 - 25 of 35
  • Lag-windowing and multiple-data-windowing are roughly equivalent for smooth spectrum estimation

    Publication Year: 1999 , Page(s): 839 - 843
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (187 KB)  

    There is no fundamental difference between lag-windowing a correlation sequence and multiple-windowing a data sequence when the objective is to reduce the mean-squared error of a spectrum estimator. By analyzing the approximate low-rank factorization of a bandlimiting Toeplitz operator, we find that lag-windowed (or spectrally smoothed) spectrum estimators have multiple-data-windowed implementations. This makes the Blackman-Tukey-Grenander-Rosenblatt spectrogram equivalent to the Thomson spectrum estimator (and vice-versa), meaning BTGR spectrograms may be implemented in a multichannel filterbank version of the Thomson estimator. View full abstract»

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  • Initialization and inner product computations of wavelet transform by interpolatory subdivision scheme

    Publication Year: 1999 , Page(s): 876 - 880
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (204 KB)  

    The initialization of wavelet transforms and the inner product computations of wavelets with their derivatives are very important in many applications. In this correspondence, the interpolatory subdivision scheme (ISS) is proposed to solve these problems efficiently. We introduce a general procedure to compute the exact values of derivatives of the interpolatory fundamental function and then derive a fast recursive algorithm for the realization of the initialization and inner product evaluations. Error analysis of the algorithm and its comparison with other approaches are discussed. Numerical experiments demonstrate the high performance of the algorithm View full abstract»

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  • A novel kurtosis driven variable step-size adaptive algorithm

    Publication Year: 1999 , Page(s): 864 - 872
    Cited by:  Papers (34)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (256 KB)  

    A new variable step-size LMS filter is introduced. The time-varying step-size sequence is adjusted, utilizing the kurtosis of the estimation error, therefore reducing performance degradation due to the existence of strong noise. The convergence properties of the algorithm are analyzed, and an adaptive kurtosis estimator that takes into account noise statistics and optimally adapts itself is also presented. Simulation results confirm the algorithm's improved performance and flexibility View full abstract»

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  • A maximum likelihood digital receiver using coordinate ascent and the discrete wavelet transform

    Publication Year: 1999 , Page(s): 813 - 825
    Cited by:  Papers (9)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (516 KB)  

    In this paper, a maximum likelihood (ML) method is presented for joint estimation of amplitude, phase, time delay, and data symbols in a single-user direct-sequence spread-spectrum communication system. Since maximization of the likelihood function is analytically intractable, a novel coordinate ascent algorithm is used to obtain sequential updates of the data symbols and all unknown nuisance parameters. The novelty of the algorithm is due to the use of a multiresolution expansion of the received signal and the use of polynomial rooting in the complex plane in place of a line search over the signal delay parameter. The multiresolution structure of the algorithm is exploited to reduce sensitivity to impulsive noise via wavelet thresholding. Computer simulations of the single-user system show that the algorithm has fast convergence, and comparison with theoretical lower bounds establishes that the algorithm achieves nearly optimal error performance View full abstract»

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  • Performance analysis of wavelets in embedded zerotree-based lossless image coding schemes

    Publication Year: 1999 , Page(s): 884 - 889
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (220 KB)  

    In this correspondence, we present a modification to the scanning approach in the set partitioning algorithm proposed by Said and Pearlman (1996) to exploit the correlation in a local neighborhood. The wavelet filters are characterized based on the wavelet coefficients obtained after the wavelet transform. Two new criteria are proposed for evaluating the performance of wavelets in lossless image compression applications: cumulative zerotree count and monotone spectral ordering of subbands produced after wavelet transform in a multiresolution scheme. Several wavelet filters are evaluated to test the evaluation criteria. The experimental results are presented to justify the proposed performance criteria View full abstract»

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  • A new approach to subband adaptive filtering

    Publication Year: 1999 , Page(s): 655 - 664
    Cited by:  Papers (52)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (428 KB)  

    Subband adaptive filtering has attracted much attention lately. In this paper, we propose a new structure and a new formulation for adapting the filter coefficients. This structure is based on polyphase decomposition of the filter to be adapted and is independent of the type of filter banks used in the subband decomposition. The new formulation yields improved convergence rate when the LMS algorithm is used for coefficient adaptation. As we increase the number of bands in the filter, the convergence rate increases and approaches the rate that can be obtained with a flat input spectrum. The computational complexity of the proposed scheme is nearly the same as that of the fullband approach. Simulation results are included to demonstrate the efficacy of the new approach View full abstract»

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  • Weighted least-squares implementation of Cohen-Posch time-frequency distributions with specified conditional and joint moment constraints

    Publication Year: 1999 , Page(s): 893 - 900
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (332 KB)  

    A positivity constrained iterative weighted least-squares (WLS) method for constructing non-negative joint time-frequency distributions (i.e., Cohen-Posch (1985) TFDs) satisfying marginal, joint moment, conditional moment, and generalized marginal constraints, is developed. The new algorithm solves the “leakage” problem of the least-squares approach and is computationally faster. It is also more computationally efficient than the MCE implementation of these constraints developed by Loughlin, Pitton, and Atlas (1994) View full abstract»

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  • Resolution in time-frequency

    Publication Year: 1999 , Page(s): 783 - 788
    Cited by:  Papers (23)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (260 KB)  

    We introduce a new measure Hp that is related to the Heisenberg uncertainty principle. The measure predicts the compactness of discrete-time signal descriptions in the sample-frequency phase plane. We conjecture a lower limit on the compaction in the phase plane and show that discretized Gaussians may not provide the most compact basis View full abstract»

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  • Bootstrapping bispectra: an application to testing for departure from Gaussianity of stationary signals

    Publication Year: 1999 , Page(s): 880 - 884
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (180 KB)  

    We propose a bootstrap version of a bispectrum-based test for departure from Gaussianity that achieves high power while maintaining the level of significance, even for small sample sizes. The proposed procedure can be also used to set confidence bands for a measure of the bicoherence of stationary random signals View full abstract»

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  • On the equivalence of blind equalizers based on MRE and subspace intersections

    Publication Year: 1999 , Page(s): 856 - 859
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (152 KB)  

    Two classes of algorithms for multichannel blind equalization are the mutually referenced equalizer (MRE) method by Gesbert et al. (see ibid. vol.45, p.2307-17, 1997) and the subspace intersection (SSI) method by van der Veen et al. (see ibid., vol. 45, no.1, p.173-90, 1997). Although these methods seem, at first sight, unrelated, we show that certain variants of the SSI and the MRE methods both optimize a new blind criterion, which is referred to as maximum coherence and, thus, are equivalent View full abstract»

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  • An effective memory addressing scheme for FFT processors

    Publication Year: 1999 , Page(s): 907 - 911
    Cited by:  Papers (48)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (220 KB)  

    The memory organization of FFT processors is considered. The new memory addressing assignment allows simultaneous access to all the data needed for butterfly calculations. The advantage of this memory addressing scheme lies in the fact that it reduces the delay of address generation nearly by half compared to existing ones View full abstract»

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  • Undermodeled equalization: a characterization of stationary points for a family of blind criteria

    Publication Year: 1999 , Page(s): 760 - 770
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (480 KB)  

    We attack specific problems related to equalizer performance in undermodeled cases in which assumptions of perfect equalizability are dismissed in favor of a more realistic situation in which no equalizer setting may achieve perfect channel equalization. We derive a characterization of candidate convergent points for a family of blind criteria which appeal, tacitly or wittingly, to maximizing the ratio of different sequence norms of the combined channel-equalizer impulse response. This may be accomplished in a practical implementation by using equalizer output cumulants of different orders. The popular Godard and Shalvi-Weinstein schemes are accommodated at one extreme of the family of criteria. We also show that each maximum at the other extreme of the family, involving progressively higher order output cumulants, yields, precisely, a Wiener response. This suggests that blind algorithms using progressively higher order statistics may converge more closely to a Wiener response than those using more modest order statistics. We show, moreover, that the superexponential family of algorithms is also included and establish a convergence proof for undermodeled cases that appeals to no approximation. Finally, some apparently novel bounds on attainable open-eye measures in undermodeled cases are also derived View full abstract»

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  • Minimum-noise-variance beamformer with an electromagnetic vector sensor

    Publication Year: 1999 , Page(s): 601 - 618
    Cited by:  Papers (19)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (992 KB)  

    We study the performance of the minimum-noise-variance beamformer employing a single electromagnetic (EM) vector sensor that is capable of measuring the complete electric and magnetic fields induced by EM signals at one point. Two types of signals are considered: one carries a single message, and the other carries two independent messages simultaneously. The state of polarization of the interference under consideration ranges from completely polarized to unpolarized. We first obtain explicit expressions for the signal to interference-plus-noise ratio (SINR) in terms of the parameters of the signal, interference, and noise. Then, we discuss some physical implications associated with the SINR expressions. These expressions provide a basis for effective interference suppression as well as generation of dual-message signals of which the two message signals have minimum interference effect on one another. We also analyze the characteristics of the main-lobe and side-lobe of the beampattern of an EM vector sensor and compare them with other types of sensor arrays View full abstract»

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  • Zolotarev polynomials and optimal FIR filters

    Publication Year: 1999 , Page(s): 717 - 730
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (728 KB)  

    The algebraic form of Zolotarev polynomials refraining from their parametric representation is introduced. A recursive algorithm providing the coefficients for a Zolotarev polynomial of an arbitrary order is obtained from a linear differential equation developed for this purpose. The corresponding narrowband, notch, and complementary pair FIR filters are optimal in the Chebyshev sense. A recursion giving an explicit access to the impulse response coefficients is also presented. Some design examples are included to demonstrate the efficiency of the presented approach View full abstract»

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  • Exploiting input cyclostationarity for blind channel identification in OFDM systems

    Publication Year: 1999 , Page(s): 848 - 856
    Cited by:  Papers (93)  |  Patents (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (452 KB)  

    Transmitter-induced cyclostationarity has been explored previously as an alternative to fractional sampling and antenna array methods for blind identification of FIR communication channels. An interesting application of these ideas is in OFDM systems, which induce cyclostationarity due to the cyclic prefix. We develop a novel subspace approach for blind channel identification using cyclic correlations at the OFDM receiver. Even channels with equispaced unit circle zeros are identifiable in the presence of any nonzero length cyclic prefix with adequate block length. Simulations of the proposed channel estimator along with its performance in OFDM systems combined with impulse response shortening and Reed-Solomon coding are presented View full abstract»

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  • A function of time, frequency, lag, and Doppler

    Publication Year: 1999 , Page(s): 789 - 799
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    In signal processing, four functions of one variable are commonly used. They are the signal in time, the spectrum, the auto-correlation function of the signal, and the auto-correlation function of the spectrum. The variables of these functions are denoted, respectively, as time, frequency, lag, and Doppler. In time-frequency analysis, these functions of one variable are extended to quadratic functions of two variables. In this paper, we investigate a method for creating quartic functions of three of these variables as well as a quartic function of all four variables. These quartic functions provide a meaningful representation of the signal that goes beyond the well-known quadratic functions. The quartic functions are applied to the design of signal-adaptive kernels for the Cohen class and shown to provide improvements over previous methods View full abstract»

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  • Implementation issues of the two-level residue number system with pairs of conjugate moduli

    Publication Year: 1999 , Page(s): 826 - 838
    Cited by:  Papers (25)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (640 KB)  

    One of the most important considerations when designing residue number systems (RNSs) is the choice of the moduli set; this is due to the fact that the dynamic range of the system, its speed, as well as its hardware complexity, depend on both the forms as well as the number of moduli chosen; In this paper, a new class of multimoduli RNSs based on sets of forms {2n(1)-1, 2n(1)+1, 2n2-1, 2n(2)+1, ···, 2n(L)-1, 2n(L)+1} is presented. The moduli 2n(i)-1 and 2 n(i)+1 are called conjugates of each other. The new RNSs that rely on pairs of conjugate moduli result in hardware-efficient two-level implementations for the weighted-to-RNS and RNS-to-weighted conversions, achieve very large dynamic ranges, and imply fast and efficient RNS processing. When compared with conventional systems of the same number of moduli and the same dynamic range, the proposed new systems offer the following benefits: (1) hardware savings of 25 to 40% for the weighted-to-RNS conversion and (2) a reduction of over 80% in the complexity of the final Chinese remainder theorem (CRT) involved in the RNS-to-weighted conversion. Thus, significant compromises between large dynamic ranges, fast internal processing, and low complexity are achieved by the new systems View full abstract»

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  • Evolutionary spectrum estimation by positivity constrained deconvolution

    Publication Year: 1999 , Page(s): 889 - 893
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (212 KB)  

    We present a deconvolution technique to obtain the evolutionary spectrum (ES) of nonstationary signals by deconvolving the blurring effects of the time-frequency distribution (TFD) kernel from bilinear TFDs. The resulting spectrum is non-negative and has desirable properties such as higher resolution and higher concentration in time frequency. The new technique is computationally more efficient compared with the previously proposed entropy-based deconvolution technique, and, unlike the entropy method, it is not restricted to deconvolution of spectrograms with Gaussian windows. This makes the method applicable to deconvolving many of the bilinear time-frequency distributions View full abstract»

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  • A refined fast 2-D discrete cosine transform algorithm

    Publication Year: 1999 , Page(s): 904 - 907
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (236 KB)  

    An index permutation-based fast two-dimensional discrete cosine transform (2-D DCT) algorithm is presented. It is shown that the N×N 2-D DCT, where N=2m, can be computed using only N 1-D DCTs and some post additions View full abstract»

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  • Blind equalization using least-squares lattice prediction

    Publication Year: 1999 , Page(s): 630 - 640
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (364 KB)  

    Second-order statistics of the received signal can be used to equalize a communication channel without knowledge of the transmitted sequence. Blind zero-forcing (ZF) and minimum mean-square error (MMSE) equalization can be achieved with linear prediction error filtering. The equivalence with the equalizers derived by Giannakis and Halford (see ibid., vol.45, p.2277-92, 1997) is shown, and adaptive predictors that result in a lattice filtering structure are applied. The required channel coefficient vector is obtained with adaptive eigen-pair tracking. Either forward or backward prediction errors can be used. The performance of the blind equalizer is examined by simulations. The MMSE of the optimum FSE is approached, and the algorithm exhibits robustness to channels with common subchannel zeros View full abstract»

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  • Multichannel linear and quadratic adaptive filtering based on the Chandrasekhar fast algorithm

    Publication Year: 1999 , Page(s): 860 - 864
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (188 KB)  

    A new fast algorithm for multichannel linear and quadratic adaptive filtering using the Chandrasekhar equations is presented. Based on the shift-invariance property, the multichannel linear model could be described by a time-invariant state-space model to which we apply the Chandrasekhar factorization technique, which provides interesting numerical properties. Furthermore, a new method for nonlinear filtering is given where the multichannel Chandrasekhar algorithm is applied on the second-order Volterra (SOV) filter after suitable transformations View full abstract»

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  • A nonparametric phase estimation method for SIMO systems based on second-order and higher order statistics

    Publication Year: 1999 , Page(s): 843 - 847
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (164 KB)  

    We present a nonparametric phase estimation algorithm for linear single-input multiple-output (SIMO) channels. Given an unknown stationary input signal with known statistics, our approach is to obtain the joint minimum mean square phase estimation based on the polyspectra and the cross-spectra of the SIMO channel outputs. By utilizing both higher order and second-order statistics of the channel outputs, our approach is shown to be more accurate and reliable than methods based on higher order statistics alone. It can be applied to SIMO channels with common zeros View full abstract»

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  • An adaptive noise canceller with low signal distortion for speech codecs

    Publication Year: 1999 , Page(s): 665 - 674
    Cited by:  Papers (21)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (412 KB)  

    This paper proposes an adaptive noise canceller (ANC) with low signal distortion for speech codecs. The proposed ANC has two adaptive filters: a main filter (MF) and a subfilter (SF). The signal-to-noise ratio (SNR) of input signals is estimated using the SF. To reduce signal distortion in the output signal of the ANC, a step size for coefficient update in the MF is controlled according to the estimated SNR. Computer simulation results using speech and diesel engine noise recorded in a special-purpose vehicle show that the proposed ANC reduces signal distortion in the output signal by up to 15 dB compared with a conventional ANC. Results of subjective listening tests show that the mean opinion scores (MOSs) for the proposed ANC with and without a speech codec are one point higher than the scores for the conventional ANC View full abstract»

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  • A low-power phase-splitting adaptive equalizer for high bit-rate communication systems

    Publication Year: 1999 , Page(s): 911 - 915
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (204 KB)  

    A low-power architecture for a phase-splitting passband equalizer (PSPE) is proposed for a transceiver in this correspondence. The Hilbert relationship between the in-phase and quadrature-phase equalizers in the PSPE is exploited to develop the proposed architecture. It is shown via analysis and simulations that in a 51.84-Mb/s ATM-LAN environment, the proposed receiver results in (1) a net saving in power if the length of the Hilbert filter is less than 130, and (2) a saving of up to 20% can be achieved with a degradation in signal-to-noise ratio of less than 0.5 dB View full abstract»

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  • A four-parameter atomic decomposition of chirplets

    Publication Year: 1999 , Page(s): 731 - 745
    Cited by:  Papers (81)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (840 KB)  

    A new four-parameter atomic decomposition of chirplets is developed for compact and precise representation of signals with chirp components. The four-parameter chirplet atom is obtained from the unit Gaussian function by successive applications of scaling, fractional Fourier transform (FRFT), and time-shift and frequency-shift operators. The application of the FRFT operator results in a rotation of the Wigner distribution of the Gaussian in the time-frequency plane by a specified angle. The decomposition is realized by using the matching pursuit algorithm. For this purpose, the four-parameter space is discretized to obtain a small but complete subset in the Hilbert space. A time-frequency distribution (TFD) is developed for clear and readable visualization of the signal components. It is observed that the chirplet decomposition and the related TFD provide more compact and precise representation of signal inner structures compared with the commonly used time-frequency representations 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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Meet Our Editors

Editor-in-Chief
Sergios Theodoridis
University of Athens