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Signal Processing, IEEE Transactions on

Issue 8 • Date Aug 1991

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Displaying Results 1 - 25 of 26
  • The quantization noise spectrum of a sinusoid in colored noise

    Publication Year: 1991 , Page(s): 1780 - 1787
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (580 KB)  

    The power density spectrum (PDS) of analog-to-digital (A/D) quantization noise is obtained for an input signal consisting of sinusoid and colored noise. While this type of quantization noise is often assumed to be uncorrelated or white, it may not be when the input signal occupies a relatively small number of quantization levels. Equations describing the PDS are derived and presented for combinations of deterministic and random A/D converter inputs View full abstract»

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  • Isolated-utterance speech recognition using hidden Markov models with bounded state durations

    Publication Year: 1991 , Page(s): 1743 - 1752
    Cited by:  Papers (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (824 KB)  

    Hidden Markov models (HMMs) with bounded state durations (HMM/BSD) are proposed to explicitly model the state durations of HMMs and more accurately consider the temporal structures existing in speech signals in a simple, direct, but effective way. A series of experiments have been conducted for speaker dependent applications using 408 highly confusing first-tone Mandarin syllables as the example vocabulary. It was found that in the discrete case the recognition rate of HMM/BSD (78.5%) is 9.0%, 6.3%, and 1.9% higher than the conventional HMMs and HMMs with Poisson and gamma distribution state durations, respectively. In the continuous case (partitioned Gaussian mixture modeling), the recognition rates of HMM/BSD (88.3% with 1 mixture, 88.8% with 3 mixtures, and 89.4% with 5 mixtures) are 6.3%, 5.0%, and 5.5% higher than those of the conventional HMMs, and 5.9% (with 1 mixture), 3.9% (with 3 mixtures) and 3.1% (with 1 mixture), 1.8% (with 3 mixtures) higher than HMMs with Poisson and gamma distributed state durations, respectively View full abstract»

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  • A geometric characterization of positive definite sequences and of the Fourier transform

    Publication Year: 1991 , Page(s): 1903 - 1907
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (444 KB)  

    If every positive-semidefinite normalized sequence of p autocorrelation coefficients is represented as a point in p-dimensional real space, then the set of such points forms a convex region: the positive semidefinite region. It is shown that the positive semidefinite region can be generated completely as the convex hull of a finite-length, one-dimensional curve that lies on the surface of the region. The curve is specified, several of its properties are given, and it is shown that its length is on the order of p3/2. The curve represents geometrically the kernel of the Fourier transform; computing the inverse Fourier transform of the spectrum then corresponds to taking the convex linear combination of points on this curve. It is shown that the surface of the positive semidefinite region can then be characterized by a set of polytopes with [p/2]+1 or fewer vertices View full abstract»

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  • On fast multivariable bilinear and Hadamard transforms

    Publication Year: 1991 , Page(s): 1788 - 1792
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (300 KB)  

    Applying the bilinear transformation to an n-variance polynomial, n⩾1, one arrives at a rational function whose numerator is the transformed polynomial. Based on I.J. Good's (1958) results and the symmetry properties of the single-dimensional bilinear transformation matrix, a fast implementation for computing the coefficients of such n-variable transformed polynomials is proposed. Letting Q denote the matrix relating the coefficients of the original polynomial to those of the transformed polynomials, it is shown that Sylvester-type Hadamard matrices coincide with the transformation matrices for the n-variable polynomials for which the highest degree of each variable is one. This new observation is used to derive W.R. Crowther and C.M. Rader's (1966) formulation of the fast Hadamard transform View full abstract»

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  • A new optimal window [signal processing]

    Publication Year: 1991 , Page(s): 1753 - 1769
    Cited by:  Papers (44)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1404 KB)  

    An optimal window that can provide the best tradeoff between the peak sidelobe level and the total sidelobe level is presented. Necessary and sufficient conditions are established for the optimal window. A design algorithm is presented, and examples are discussed. The examples demonstrate that the window is capable of much better performance than previously available windows, such as the Hamming window. Relationships between the maximum error and the mean-squared error are explored for the window optimization problem. It is concluded that the minimax and least squares approximations are both fundamentally inefficient. This conclusion is supported by experimental evidence and mathematical analyses View full abstract»

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  • A pipelined structure for QR adaptive LS system identification

    Publication Year: 1991 , Page(s): 1920 - 1923
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (264 KB)  

    A novel systolic array for LS system identification based on QR factorization via Givens rotations is proposed. The back-substitution step is circumvented, and the structure is fully pipelineable. Thus, the method is appropriate for continuous, sample-by-sample mode, adaptive operation. A modification of this structure is also suggested which is suitable for a wide range of linear algebraic operations and is solely based on Givens rotations View full abstract»

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  • The equivalence of the total least squares and minimum norm methods [signal processing]

    Publication Year: 1991 , Page(s): 1891 - 1892
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (156 KB)  

    It is shown that the minimum norm solution is equivalent to the total least squares solution. It is noted that two versions of the total least squares solution exist, one based on the signal subspace and another based on the noise subspace View full abstract»

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  • Signal enhancement of the spatial smoothing algorithm

    Publication Year: 1991 , Page(s): 1907 - 1911
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (368 KB)  

    A spatial smoothing algorithm to decorrelate highly correlated sources for direction-of-arrival (DOA) estimation in narrowband problems is developed. The rate of decorrelation of the coherent sources is very low and sensitive to the signal-to-noise ratio (SNR) for sources with close DOAs. A method to enhance the signal (to remove the effects of the sensor noise) and to make the spatial smoothing robust with respect to SNR is proposed. This approach will significantly improve the resolution of the algorithm. Statistical characteristics of the improved spatial smoothing are compared with that of the standard spatial smoothing View full abstract»

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  • Subband speech coding and matched convolutional channel coding for mobile radio channels

    Publication Year: 1991 , Page(s): 1717 - 1731
    Cited by:  Papers (58)  |  Patents (22)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1424 KB)  

    The effects of digital transmission errors on a family of variable-rate embedded subband speech coders (SBC) are analyzed in detail. It is shown that there is a difference in error sensitivity of four orders of magnitude between the most and the least sensitive bits of the speech coder. As a result, a family of rate-compatible punctured convolutional codes with flexible unequal error protection capabilities have been matched to the speech coder. These codes are optimally decoded with the Viterbi algorithm. Among the results, analysis and informal listening tests show that with a 4-level unequal error protection scheme transmission of 12 kb/s speech is possible with very little degradation in quality over a 16 kb/s channel with an average bit error rate (BER) of 2×10-2 at a vehicle speed of 60 m.p.h. and with interleaving over two 16 ms speech frames View full abstract»

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  • Statistical analysis of MUSIC and subspace rotation estimates of sinusoidal frequencies

    Publication Year: 1991 , Page(s): 1836 - 1847
    Cited by:  Papers (78)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (792 KB)  

    Consideration is given to the analysis of the large-sample second-order properties of multiple signal classification (MUSIC) and subspace rotation (SUR) methods, such as ESPRIT, for sinusoidal frequency estimation. Explicit expressions for the covariance elements of the estimation errors associated with either method are derived. These expressions of covariances are then used to analyze and compare the statistical performances of the MUSIC and SUR estimation (SURE) methods. Both MUSIC and SURE are based on the eigendecomposition of a sample data covariance matrix. The expressions for the estimation error variances derived are used to study the dependence of MUSIC and SURE performances on the dimension of the data covariance matrix used View full abstract»

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  • Interference cancellation matrix beamforming for 3-D beamspace ML/MUSIC bearing estimation

    Publication Year: 1991 , Page(s): 1858 - 1876
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1596 KB)  

    For the case of two closely spaced targets angularly located within the mainlobe of the transmitted beam, 3D-BDML in a computationally simplistic maximum likelihood bearing estimation scheme which operates in a 3-D beamspace generated by three orthogonal, classical beamformers. The presence of strong interferers angularly located outside the mainlobe encompassing the two targets of interest necessitates the use of adaptively formed left, center, and right beams. Let M denote the number of elements comprising the array. Novel procedures are developed for the construction of and M×3 interference cancellation matrix beamformer which retains those properties of the M×3 classical matrix beamformer critical to the computational simplicity of 3D-BDML. The most important of these is commonality of M-3 nulls among the left, center, and right beams. Simulations demonstrating the performance of both single frequency and multifrequency 3D-BDML are presented View full abstract»

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  • A novel approach for stabilizing recursive least squares filters

    Publication Year: 1991 , Page(s): 1770 - 1779
    Cited by:  Papers (28)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (820 KB)  

    A novel approach for stabilizing recursive least squares (RLS) filters is presented. The approach relies on a detailed fixed point analysis, which provides two important benefits. The analysis reveals a bias in the error propagation mechanism, providing an analytical basis for instability problems. The analysis then indicates which specific roundoff errors are causing instability. These roundoff errors are then biased in such a way that the overall filter is biased towards stable performance. Experimental results indicate that stability can be achieved with negligible loss in least squares performance View full abstract»

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  • Estimation and identification for 2-D block Kalman filtering

    Publication Year: 1991 , Page(s): 1885 - 1889
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (668 KB)  

    The development of a recursive identification and estimation procedure for two-dimensional block Kalman filtering is discussed. The recursive identification scheme can be used online to update the image model parameters at each iteration based on the local statistics within a block of the observed noisy image. The covariance matrix of the driving noise can also be estimated at each iteration of this algorithm. A recursive procedure for computing the parameters of the higher order models is given. Simulation results are also provided View full abstract»

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  • Iterative Wiener filters for image restoration

    Publication Year: 1991 , Page(s): 1892 - 1899
    Cited by:  Papers (31)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (960 KB)  

    The iterative Wiener filter, which successively uses the Wiener-filtered signal as an improved prototype to update the covariance estimates, is investigated. The convergence properties of this iterative filter are analyzed. It has been shown that this iterative process converges to a signal which does not correspond to the minimum mean-squared-error solution. Based on the analysis, an alternate iterative filter is proposed to correct for the convergence error. The theoretical performance of the filter has been shown to give minimum mean-squared error. In practical implementation when there is unavoidable error in the covariance computation, the filter may still result in undesirable restoration. Its performance has been investigated and a number of experiments in a practical setting were conducted to demonstrate its effectiveness View full abstract»

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  • The performance of minimax spatial resampling filters for focussing wide-band arrays

    Publication Year: 1991 , Page(s): 1899 - 1903
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (508 KB)  

    The design and performance of linear shift-variant filters for coherent wideband processing are examined via spatial resampling. In particular, a minimax error criterion is used to obtain realizable resampling filters, and an approximate statistical analysis of wideband spatially resampled minimum variance spatial spectral estimation is presented. Simulation results indicate that spatial resampling provides a computationally efficient means of reducing the threshold observation time required to obtain high-resolution estimates of source location View full abstract»

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  • Finite precision arithmetic and the split Schur algorithms

    Publication Year: 1991 , Page(s): 1805 - 1811
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (456 KB)  

    The split Schur algorithms of P. Delsarte and Y. Genin (1987) represent methods of computing reflection coefficients that are computationally more efficient, in terms of multiplications, than the conventional Schur algorithm by a constant factor. The authors investigate the use of fixed-point binary arithmetic, with quantization due to rounding, in the implementation of the symmetric and antisymmetric split Schur algorithms. It is shown, through a combination of analysis and simulation, that the errors in the reflection coefficient estimates due to quantization are large when the input signal is either a narrowband high-pass signal or a narrowband low-pass signal View full abstract»

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  • On a very tight truncation error bound for stationary stochastic processes

    Publication Year: 1991 , Page(s): 1918 - 1919
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (132 KB)  

    A very tight truncation error upper bound is established for bandlimited weakly stationary stochastic processes if the sampling interval is closed. In particular, the magnitude of the upper bound is O(N-2q ln2 N) for a symmetric sampling reconstruction from 2N+1 sampled values, where q is an arbitrary positive integer. The results are derived with the help of the Bernstein bound on the remainder of a symmetric complex Fourier series of the function exp (iλ t). Convergence rates are given for mean square and almost sure sampling reconstructions View full abstract»

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  • A VLSI for real-time linear operations and transforms

    Publication Year: 1991 , Page(s): 1914 - 1917
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (356 KB)  

    A VLSI system, utilizing 16 systolic array multipliers, designed to compute vector-matrix products at a rate of 640×106 MACs is presented. The 448,000-transistor, 1.6-μm CMOS device incorporates a dual timing scheme which allows multiplexing of hardware units over identical operations. This hardware balances maximum internal operating frequency with external data bandwidth and results in an improved ration of the signal throughput to silicon area. This system has wide application because of its ability to compute correlation, convolution, linear transforms, and connections in multilayer perceptrons View full abstract»

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  • Filtering of colored noise for speech enhancement and coding

    Publication Year: 1991 , Page(s): 1732 - 1742
    Cited by:  Papers (107)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (868 KB)  

    Scalar and vector Kalman filters are implemented for filtering speech contaminated by additive white noise or colored noise, and an iterative signal and parameter estimator which can be used for both noise types is presented. Particular emphasis is placed on the removal of colored noise, such as helicopter noise, by using state-of-the-art colored-noise-assumption Kalman filters. The results indicate that the colored noise Kalman filters provide a significant gain in signal-to-noise ratio (SNR), a visible improvement in the sound spectrogram, and an audible improvement in output speech quality, none of which are available with white-noise-assumption Kalman and Wiener filters. When the filter is used as a prefilter for linear predictive coding, the coded output speech quality and intelligibility are enhanced in comparison to direct coding of the noisy speech View full abstract»

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  • A parametric method for determining the number of signals in narrow-band direction finding

    Publication Year: 1991 , Page(s): 1848 - 1857
    Cited by:  Papers (52)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (708 KB)  

    A novel and more accurate method to determine the number of signals in the multisource direction finding problem is developed. The information-theoretic criteria of Y. Yin and P. Krishnaiah (1988) are applied to a set of quantities which are evaluated from the log-likelihood function. Based on proven asymptotic properties of the maximum likelihood estimation, these quantities have the properties required by the criteria. Since the information-theoretic criteria use these quantities instead of the eigenvalues of the estimated correlation matrix, this approach possesses the advantage of not requiring a subjective threshold, and also provides higher performance than when eigenvalues are used. Simulation results are presented and compared to those obtained from the nonparametric method given by H. Wax and T. Kailath (1985) View full abstract»

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  • LD2-ARMA identification algorithm

    Publication Year: 1991 , Page(s): 1822 - 1835
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1032 KB)  

    The authors present the LD2-ARMA identifier, a novel algorithm that solves the essentially nonlinear autoregressive moving-averaged (ARMA) identification problem with a linear procedure, in two steps: an order selection algorithm followed by an ARMA parameter estimator. The determination of the AR and MA coefficients involves the solution of two dual systems of linear equations. These systems decouple the estimation of the autoregressive component from the estimation of the moving average component. The selection of the number of poles and of the number of zeros is accomplished by a scheme that minimizes the mismatch of the data to each proposed model. Simulated experiments on the proposed order selection procedure are presented View full abstract»

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  • The application of nonlinear structures to the reconstruction of binary signals

    Publication Year: 1991 , Page(s): 1877 - 1884
    Cited by:  Papers (67)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (664 KB)  

    The problem of reconstructing digital signals which have been passed through a dispersive channel and corrupted with additive noise is discussed. The problems encountered by linear equalizers under adverse conditions on the signal-to-noise ratio and channel phase are described. By considering the equalization problem as a geometric classification problem the authors demonstrate how these difficulties can be overcome by utilizing nonlinear classifiers as channel equalizers. The manner in which neural networks can be utilized as adaptive channel equalizers is described, and simulation results which suggest that the neural network equalizers offer a performance which exceeds that of the linear structures, particularly in the high-noise environment, are presented View full abstract»

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  • Statistical analysis of initialization methods for RLS adaptive filters

    Publication Year: 1991 , Page(s): 1793 - 1804
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (956 KB)  

    Theoretical analysis is used to evaluate the mean and second-moment properties of recursive least squares algorithms incorporating the fast exact initialization and soft constrained initialization methods during the initialization period. It is shown that the weight vector mean and covariance produced by fast exact initialization are undefined for this period. Theoretical results are derived for soft constrained initialization that show that the weight vector mean and covariance are finite, and expressions are given for these quantities. Simulations for various cases are presented to support the accuracy of these theoretical results View full abstract»

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  • A note on `Efficient numerically stabilized rank-one eigenstructure updating' [and reply]

    Publication Year: 1991 , Page(s): 1911 - 1914
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB)  

    In the above-named paper, R.D. DeGroat and R.A. Roberts (see ibid., vol.38, no.2, p.301-16, Feb. 1990) developed a reorthogonalization scheme for stabilizing eigenstructure updating algorithms. The commenters show that only part of this scheme, namely the renormalization, is essential for the stability, so that a cheaper scheme, with roughly half as many computations, can perform equally well. DeGroat addresses two points raised by the commenters' work View full abstract»

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  • Design of the unimodular shaping filter

    Publication Year: 1991 , Page(s): 1889 - 1891
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    A unimodular shaping filter for noisy input is designed. It is stable, and shapes the wavelet into (a) the desired output with least squares error and (b) the actual output with a unit area. Stability was achieved by using the prewhitening parameter. A numerical example shows that the prewhitening parameter reduces the error energy in the unimodular constrained shaping filter as compared to the unconstrained shaping filter 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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Sergios Theodoridis
University of Athens