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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date April 1986

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Displaying Results 1 - 22 of 22
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Adaptive spectral estimation by the conjugate gradient method

    Page(s): 272 - 284
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    This paper proposes an alternative technique for adaptive spectral estimation. The new technique applies the method of conjugate gradient, which is used for iteratively finding the generalized eigenvector corresponding to the minimum generalized eigenvalue of a semidefinite Hermitian matrix, to the adaptive spectral analysis problem. Computer simulations have been performed to compare the new method to existing ones. From the limited examples presented, it is seen that the new method is computationally more efficient at the expense of more core storage. Also, this method is effective for small data records and can implement noise correction to yield unbiased spectral estimates if an estimate of the noise covariance matrix is available. The technique performs well for both narrow-band and wide-band signals. View full abstract»

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  • Authors' reply

    Page(s): 379
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    First Page of the Article
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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • A variable step (VS) adaptive filter algorithm

    Page(s): 309 - 316
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    In recent work, a new version of an LMS algorithm has been developed which implements a variable feedback constant μ for each weight of an adaptive transversal filter. This technique has been called the VS (variable step) algorithm and is an extension of earlier ideas in stochastic approximation for varying the step size in the method of steepest descents. The method may be implemented in hardware with only modest increases in complexity (approx 15percent) over the LMS Widrow-Hoff algorithm. It is shown that an upper bound for the convergence time is the classical mean-square-error time constant, and examples are given to demonstrate that for broad signal classes (both narrow-band and broad-band) the convergence time is reduced by a factor of up to 50 in noise canceller applications for the proper selection of variable step parameters. Finally, the VS algorithm is applied to an IIR filter and simulations are presented for applications of the VS FIR and IIR adaptive filters. View full abstract»

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  • A new approach to the realization of low-sensitivity IIR digital filters

    Page(s): 350 - 361
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    A new implementation of an IIR digital filter transfer function is presented that is structurally passive and, hence, has extremely low pass-band sensitivity. The structure is based on a simple parallel interconnection of two all-pass sections, with each section implemented in a structurally lossless manner. The structure shares a number of properties in common with wave lattice digital filters. Computer simulation results verifying the low-sensitivity feature are included, along with results on roundoff noise/dynamic range interaction. A large number of alternatives is available for the implementation of the all-pass sections, giving rise to the well-known wave lattice digital filters as a specific instance of the implementation. View full abstract»

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  • Fast biased polynominal transforms

    Page(s): 383 - 385
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    The fast biased polynomial transforms (FBPT's) are defined directly on the ZN- 1 ring, then two-dimensional convolutions can be carried out without using the Chinese remainder theorem (CRT), complex mapping, and column-row reordering processes. Furthermore, for N prime, these FBPT's are used for the evaluation of 2-D prime length DFT's very efficiently. View full abstract»

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  • A fast local maximum likelihood estimator for time delay estimation

    Page(s): 375 - 378
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    A fast algorithm for the local maximum likelihood determination of the difference of arrival time of a common signal at two spatially separated sensors with uncorrelated noise is given. The fast algorithm consists of locally maximizing the cross-correlation function from the two wide-band signals by using Newton's method for finding the root of an equation. The probability density function of one iteration of Newton's method is explicitly computed in terms of exponential and error functions. Using a theorem by Rice on the probability density of local maxima of Gaussian processes, the probability density of the local maxima of the cross correlator is obtained. These results are new. When both the signal and the noises have flat power spectral densities, the mean-square error (MSE) of two iterates of Newton's method is practically equal to the MSE computed from the probability density of the local maxima of the cross correlator (via Rice's theorem). The above holds if the starting point used in Newton's method is within a quarter signal resolution binwidth from the true delay and the signal-to-noise ratio (SNR) at the cross-correlator output is 15 dB or higher. The MSE of the local maximum estimator obtained from Rice's theorem is almost equal to the Cramer-Rao bound even for low SNR, i.e., 5 dB. View full abstract»

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  • Robust adapative Kalman filtering for systems with unknown step inputs and non-Gaussian measurement errors

    Page(s): 252 - 263
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    Target tracking with Kalman filters is hampered by target maneuvering and unknown process and measurement noises. We show that moving data windows may be used to analyze state and measurement error sequences, determining robust estimates of bias and covariance. For steps in the system forcing functions and non-Gaussian measurement errors, the robust estimators yield improvements over linear bias and covariance estimators. Extensive simulations compare conventional, linear adaptive, and robust adaptive average step responses of a first-order system filter. Quantities examined are state estimate, state error, process and measurement covariance estimates, Kalman gain, and input step estimate. View full abstract»

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  • A multiprocessor digital signal processing system for real-time audio applications

    Page(s): 225 - 231
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    This paper describes the design and application of a multiprocessor digital signal processing system. Incorporating six VLSI digital signal processors, the fully programmable system is a powerful laboratory instrument that provides real-time digital signal processing to facilitate speech and audio research. The system capabilities include: up to 15 million 16-bit multiply-accumulate operations per second; two analog input channels sampled and converted to 16-bit values each at the rate of up to 50 kHz; and two 16-bit digital-to-analog output channels with conversion rates to 50 kHz. The implementation of a four-channel hearing aid is described to provide an understanding of a typical application to which this system is well suited. View full abstract»

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  • Improvement of the fast recursive least-squares algorithms via normalization: A comparative study

    Page(s): 296 - 308
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    This paper deals with the derivation and the properties of fast optimal least-squares algorithms, and particularly with their normalization. It is shown how the well-known fast Kalman algorithm, written in the most general form, can be normalized through a purely algebraic point of view, leading to the normalized least-squares transversal filter derived by Cioffi, Kailath, and Lev-Ari from the geometric approach. An improved form of the algorithm is presented. The different algorithms have been compared from a practical point of view as regards their convergence, initialization procedures, complexity, and numerical properties. Normalized transversal algorithms are shown to be interesting because of their nice structured form, simplicity of conception, and improved good numerical behavior. View full abstract»

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  • Signal--A data flow-oriented language for signal processing

    Page(s): 362 - 374
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    We present the language SIGNAL which is a data flow-oriented real-time, synchronous, side effect-free language suited to the expression and recovery of the parallelism in signal or image processing algorithms. The language is intended to be, at the same time, an executable simulation language, and a specification of a virtual machine implementing the algorithm. The language is semantically sound, and is suitable to perform program transforms-a major requirement when the ultimate goal is an aid to the architecture design. View full abstract»

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  • Authors' reply to "Comments on 'direct Fourier reconstruction in computer tomography'"

    Page(s): 379 - 380
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    In their comments, Fan and Sanz discuss an interpolation formula used by us in our papers on direct Fourier transform tomography [1], [2]. They raise questions relating to: 1) the relation between our work and others'; and 2) the optimality of the interpolation procedure in direct Fourier reconstruction in CT. We show that: 1) most of their points are irrelevant; and 2) other points follow from an error which was already corrected by the authors in [3], prior to the publication of the comments. View full abstract»

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  • Implementation of "Split-radix" FFT algorithms for complex, real, and real-symmetric data

    Page(s): 285 - 295
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    A new algorithm is presented for the fast computation of the discrete Fourier transform. This algorithm belongs to that class of recently proposed 2n-FFT's which present the same arithmetic complexity (the lowest among any previously published one). Moreover, this algorithm has the advantage of being performed "in-place," by repetitive use of a "butterfly"-type structure, without any data reordering inside the algorithm. Furthermore, it can easily be applied to real and real-symmetric data with reduced arithmetic complexity by removing all redundancy in the algorithm. View full abstract»

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  • A new FFT algorithm of radix 3,6, and 12

    Page(s): 380 - 383
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    A new algorithm for implementation of radix 3, 6, and 12 FFT is introduced. An FFT using this algorithm is computed in an ordinary (1,j) complex plane and the number of additions can be significantly reduced; the number of multiplication is also reduced. High efficiency of the algorithm is derived from the fact that, if an input sequence is favorably reordered, rotating factors can be treated in pairs so that the rotating factors are conjugate to each other. View full abstract»

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  • Large and small error performance limits for multipath time delay estimation

    Page(s): 245 - 251
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    The Cramer-Rao lower bound (CRLB) and a lower bound on the probability of large errors for multipath time delay estimation, for a single resolvable multipath, are developed. These theoretical results are compared to the performance of the maximum likelihood estimator (MLE) and an autocorrelator, via computer simulation. For small errors and large SNR, the MLE reaches the CRLB; the variance of the time delay estimate for the autocorrelator is much larger than the CRLB if the reflection coefficient for the delayed path is near unity. For small errors and small SNR, the MLE reduces to an autocorrelator, hence, their performance is identical. For large errors and bandwidth time products greater than 500, the error probability for the autocorrelator is within 1 dB in input SNR of the lower bound on large error probability. Thus, the autocorrelator is nearly an optimal instrumentation from the point of view of large errors, at least for large bandwidth time products and the flat, low-pass spectrum considered here. View full abstract»

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  • State-space modeling and estimation of time differences of arrival

    Page(s): 232 - 244
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    We propose a new method for time differences of arrival (TDOA's) estimation of multiple sources whose spectra are unknown. In the first part, we show how state-space techniques can be used to easily arrive at an exact discrete global model of the transfer between the source inputs and sensor outputs for sources having rational spectra and for integer and noninteger delays. This part is of interest by itself-e.g., for simulation purposes-but it also leads in a natural way to an original TDOA's estimation procedure presented in the second part of the paper. A condensed parametric model of the covariance information is the unique input to what can be seen as a generalized parametric covariance method. The difficulties inherent to these methods in the presence of several sources are easily turned around by a decoupling scheme which isolates the contributions of the individual, elementary sources. Simulation results are presented. View full abstract»

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  • Adaptive interpolation of discrete-time signals that can be modeled as autoregressive processes

    Page(s): 317 - 330
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    This paper presents an adaptive algorithm for the restoration of lost sample values in discrete-time signals that can locally be described by means of autoregressive processes. The only restrictions are that the positions of the unknown samples should be known and that they should be embedded in a sufficiently large neighborhood of known samples. The estimates of the unknown samples are obtained by minimizing the sum of squares of the residual errors that involve estimates of the autoregressive parameters. A statistical analysis shows that, for a burst of lost samples, the expected quadratic interpolation error per sample converges to the signal variance when the burst length tends to infinity. The method is in fact the first step of an iterative algorithm, in which in each iteration step the current estimates of the missing samples are used to compute the new estimates. Furthermore, the feasibility of implementation in hardware for real-time use is established. The method has been tested on artificially generated auto-regressive processes as well as on digitized music and speech signals. View full abstract»

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  • A systematic approach to the extraction of diphone elements from natural speech

    Page(s): 264 - 271
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    Synthetic speech can be generated with an unrestricted vocabulary by concatenating stored units such as diphone elements. When joining speech segments that were not adjacent in the original context they were taken from, discontinuities in the spectral envelope may arise that impair intelligibility. The method proposed here attempts to find optimum diphone boundaries in order to minimize these discontinuities, Steady-state zones of all phones carrying a diphone boundary are specified by means of a centroid vector. Based on the centroids and on an objective distance measure, hypothetical boundary cost functions are defined. Their minimization together with the evaluation of a set of additional rules determines the boundary locations. A rhyme test carried out with speech generated by concatenating diphone elements extracted according to this method yielded an intelligibility score of 96.7 percent for isolated words. View full abstract»

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  • Some properties of lattice autoregressive filters

    Page(s): 342 - 349
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    An autoregressive filter is defined either by the components of the regression vector or by the reflection coefficients appearing in its lattice representation. The mathematical expression of the regression vector in terms of the reflection coefficients is very complex but many structural properties can be obtained without this exact expression. In this paper, we present some examples of such structural properties, and we apply these results to prove some extremal properties of stable filters such as the maximum value of the components of the regression vector or the maximum value of its norm. Moreover, some properties of the boundary of the stability domain are discussed. View full abstract»

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  • The statistical performance of the MUSIC and the minimum-norm algorithms in resolving plane waves in noise

    Page(s): 331 - 341
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    This paper presents an asymptotic statistical analysis of the null-spectra of two eigen-assisted methods, MUSIC [1] and Minimum-Norm [2], for resolving independent closely spaced plane waves in noise. Particular attention is paid to the average deviation of the null-spectra from zero at the true angles of arrival for the plane waves. These deviations are expressed as functions of signal-to-noise ratios, number of array elements, angular separation of emitters, and the number of snapshots. In the case of MUSIC. an approximate expression is derived for the resolution threshold of two plane waves with equal power in noise. This result is validated by Monte Carlo simulations. View full abstract»

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Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope