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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 8 • Date Aug 1989

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  • A second improved digit-reversal permutation algorithm for fast transforms

    Publication Year: 1989 , Page(s): 1288 - 1291
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (308 KB)  

    Based on three previously published theorems and an algorithm for the digit-reversal permutation required by fast transform algorithms, a fourth theorem is given, indicating an alternate order of generating the index pairs for swapping, and a second permutation algorithm results. This algorithm uses the same principles as the first and differs principally in that in its innermost loop, one of the pair of indexes is usually generated by an integer increment (i:=i+n ). This will result in slightly faster execution on most computers View full abstract»

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  • A fast algorithm for computing the Hankel transform of order 1

    Publication Year: 1989 , Page(s): 1291 - 1293
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (264 KB)  

    A fast algorithm for computing the Hankel transform of order one is derived by slightly modifying the algorithm developed by E.W. Hansen (1985). Since the algorithm uses the formal equivalency between a Hankel transform and an Abel transform followed by a Fourier transform, it enjoys computational advantages using a rapid Abel transform with shift-variant recursive filter and a fast Fourier transform. Good agreement between actual and computer transforms was obtained in the simulation with a known transform pair View full abstract»

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  • Conditions for the 2-D characteristic polynomial of a matrix to be very strict Hurwitz

    Publication Year: 1989 , Page(s): 1284 - 1286
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (264 KB)  

    Conditions for the bivariate characteristic polynomial of a matrix to be very strict Hurwitz are presented. These conditions are based on the necessary and sufficient conditions for the existence of positive definite solutions to the 2-D continuous Lyapunov equation. It is shown that such an existence is only sufficient but not necessary for the characteristic polynomial to be very strict Hurwitz. Further, the testing of zeros at infinite distant points requires the use of a class of very strict positive real functions View full abstract»

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  • Stabilized hyperbolic Householder transformations

    Publication Year: 1989 , Page(s): 1286 - 1288
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB)  

    A modification of the hyperbolic Householder scheme is introduced which is demonstrably stable theoretically (according to an established stability criterion) and which exhibits superior numerical behavior in simulations. The modified transform scheme effects downdating by applying conventional orthonormal, rather than hyperbolic, Householder transformations to the data. The latter have preferable numerical properties. However, the construction of these orthonormal operators itself requires hyperbolic computations. Thus, the proposed method is, in some sense, half hyperbolic and half orthonormal. There is no computational penalty incurred with these stabilized hyperbolic Householder transforms; they enjoy an operation count identical to their conventional counterparts View full abstract»

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  • Double-window Hodges-Lehman (D) filter and hybrid D-median filter for robust image smoothing

    Publication Year: 1989 , Page(s): 1293 - 1298
    Cited by:  Papers (8)  |  Patents (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (620 KB)  

    The double window Hodges-Lehman filter (DWD-filter) and a hybrid D-median filter (HDM-filter) for robust image smoothing are proposed. An adaptive mixture of the median and the D-filter, the HDM filter first makes decisions about the presence of edges on the basis of a two-way classification of pixels near and around the pixel to be filtered. Subsequently, straightforward D-filtering is used in the absence of edges, and median filtering is used in the presence of edges. The DWD filter uses two windows and D-filtering. The smaller window is used to preserve the details, then the larger window to provide for sufficient smoothing. Detailed simulation results show that the HDM-filter, while retaining all the good properties of the DWD filter, consistently performs better, in terms of signal-to-noise ratio, than the DWD filter and a number of other filters, including the median filter. The DWD filter is shown to have simpler structure, although not necessarily lesser computational complexity View full abstract»

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  • Detection of the number of coherent signals by the MDL principle

    Publication Year: 1989 , Page(s): 1190 - 1196
    Cited by:  Papers (164)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (472 KB)  

    An approach is presented to the problem of detecting the number of sources impinging on a passive sensor array that is based on J. Rissanen's (1983) minimum description length (MDL) principle. The approach is applicable to any type of sources, including the case of sources which are fully correlated, referred to as the coherent signals case. Two slightly different detection criteria are derived, both requiring the estimation of the locations of the sources. The first is tailored to the detection problem per se, whereas the second is tailored to the combined detection/estimation problem. Consistency of the two criteria is proved and their performance is demonstrated by computer simulations View full abstract»

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  • On the two-dimensional vector split-radix FFT algorithm

    Publication Year: 1989 , Page(s): 1302 - 1304
    Cited by:  Papers (6)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    The complete equations are presented for the first stage of the two-dimensional vector split-radix decimation-in-frequency fast Fourier transform algorithm using a structural approach. The computational complexity of the algorithm is discussed and compared to other published results. The author states that generally, the vector split-radix method provides a significant reduction in the number of complex multiplications required to implement a two-dimensional discrete Fourier transform View full abstract»

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  • High performance connected digit recognition using hidden Markov models

    Publication Year: 1989 , Page(s): 1214 - 1225
    Cited by:  Papers (51)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1096 KB)  

    The authors use an enhanced analysis feature set consisting of both instantaneous and transitional spectral information and test the hidden-Markov-model (HMM)-based connected-digit recognizer in speaker-trained, multispeaker, and speaker-independent modes. For the evaluation, both a 50-talker connected-digit database recorded over local, dialed-up telephone lines, and the Texas Instruments, 225-adult-talker, connected-digits database are used. Using these databases, the performance achieved was 0.35, 1.65, and 1.75% string error rates for known-length strings, for speaker-trained, multispeaker, and speaker-independent modes, respectively, and 0.78, 2.85, and 2.94% string error rates for unknown-length strings of up to seven digits in length for the three modes. Several experiments were carried out to determine the best set of conditions (e.g., training, recognition, parameters, etc.) for recognition of digits. The results and the interpretation of these experiments are described View full abstract»

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  • Semisystolic incremental realization of FIR digital filters using ternary arithmetic

    Publication Year: 1989 , Page(s): 1231 - 1240
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (720 KB)  

    A fully incremental realization of finite impulse response (FIR) filters is presented in which ternary, incremental representations are used for all the signals in the structure and the impulse response. Delta modulation (DM) is used for this purpose, allowing a considerable simplification of the arithmetic involved in the filtering operations. Simulation showed that the proposed DM-adder has lower mean-square error (MSE) and reduced processing delay with respect to existing DM-adders, while demonstrating the feasibility of the proposed modular incremental realization for FIR filters View full abstract»

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  • Performance analysis of MUSIC-type high resolution estimators for direction finding in correlated and coherent scenes

    Publication Year: 1989 , Page(s): 1176 - 1189
    Cited by:  Papers (57)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (960 KB)  

    An asymptotic analysis is presented of a class of high-resolution estimators for resolving correlated and coherent plane waves in noise. These estimators are in turn constructed from certain eigenvectors associated with spatially smoothed (or unsmoothed) covariance matrices generated from a uniform array. The analysis is first carried out for the smoothed case, and from this the conventional (i.e., unsmoothed) multiple signal classification (MUSIC) scheme follows as a special case. Independent of the total number of sources present in the scene, the variance of the conventional MUSIC estimator along the true arrival angles is shown to be zero within a first-order approximation. The bias expressions in the smoothed case are used to obtain a resolution threshold for two coherent, equipowered plane waves in white noise, and the result is compared to the one derived by Kaveh et al. (1986) for two uncorrelated, equipowered plane waves View full abstract»

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  • An improved FFT digit-reversal algorithm

    Publication Year: 1989 , Page(s): 1298 - 1300
    Cited by:  Papers (9)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (252 KB)  

    Several improvements on the bit-reversal algorithm of B. Gold and C.M. Rader (1969) are presented. The savings in computation are obtained by observing that not all indexes need to be reversed. In particular, a closed-form expression is derived for the largest index that must be digit-reversed (for an arbitrary radix). To limit the number of unnecessary digit-reversals, a closed-form expression (in terms of N and R) is derived for the largest array index that must be reversed. In addition, it is shown that the smallest index that must be reversed is always 1, not 0, as is commonly implemented. A computational analysis is given, comparing the original and modified algorithms. The modifications to the algorithm led to an improved efficiency with almost no increase in the size of the program or its working storage View full abstract»

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  • Acquisition of spread spectrum signals by an adaptive array

    Publication Year: 1989 , Page(s): 1253 - 1270
    Cited by:  Papers (22)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1196 KB)  

    Discrete-time processing techniques are discussed for the acquisition of direct-sequence spread-spectrum signals by an antenna array. Both constant data and random data modulation of the code are considered. The maximum-likelihood (ML) procedures for estimating the received code lag are described, assuming an unknown channel and interference signal of either known or unknown covariance. Analytic and simulation results for performance of the optimum estimators are presented. Simulation results for the ML data estimators are shown to be close to analytical predictions of bit error rate (BER) performance. The ML procedure for data demodulation is also described View full abstract»

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  • Source range and depth estimation from multipath range difference measurements

    Publication Year: 1989 , Page(s): 1157 - 1165
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (620 KB)  

    A prefilter is developed that converts range differences into position estimates for the case of multipath range differences or simple range differences measured to a collinear element array. Lower bounds are placed on the variance of position estimates. The sensor placement providing the minimum bound variance is discussed, and an estimator-achieving the variance lower bound in the limit of small estimation errors-is presented. The estimator is the minimizer of a weighted equation error norm, and is a closed-form function of the measured range differences; analytic expressions for its bias and variance are given View full abstract»

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  • An IIR parallel-type adaptive algorithm using the fast least squares method

    Publication Year: 1989 , Page(s): 1226 - 1230
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (272 KB)  

    A parallel-type adaptive algorithm is proposed which utilizes the fast least-squares method. Its computational complexity is much less than that of L. Landau's (1978) method, which is based on hyperstability theory Hyperstability requires much computation because it involves matrix operations. The proposed method has nothing to do with hyperstability. It is also shown theoretically and using computer simulation, that the convergence performance of the proposed method is better than that of the series-parallel-type fast least-squares method View full abstract»

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  • Bidiagonal factorization of Fourier matrices and systolic algorithms for computing discrete Fourier transforms

    Publication Year: 1989 , Page(s): 1280 - 1283
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    An algorithm is presented for factoring Fourier matrices into products of bidiagonal matrices. These factorizations have the same structure for every n and make possible discrete Fourier transform (DFT) computation via a sequence of local, regular computations. A parallel pipeline technique for computing sequences of k-point DFTs, for every kn, on a systolic array is proposed View full abstract»

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  • Lexical access to large vocabularies for speech recognition

    Publication Year: 1989 , Page(s): 1197 - 1213
    Cited by:  Papers (9)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1292 KB)  

    A large-vocabulary isolated-word recognition system based on the hypothesize-and-test paradigm is described. Word preselection is achieved by segmenting and classifying the input signal in terms of broad phonetic classes. A lattice of phonetic segments is generated and organized as a graph. Word hypothesization is obtained by matching this graph against the models of all vocabulary words, where a word model is itself a phonetic representation made in terms of a graph. A modified dynamic programming matching procedure gives an efficient solution to this graph-to-graph matching problem. Hidden Markov models (HMMs) of subword units are used as a more detailed knowledge in the verification step. The word candidates generated by the previous step are represented as sequences of diphone-like subword units, and the Viterbi algorithm is used for evaluating their likelihood. Lexical knowledge is organized in a tree structure where the initial common subsequences of word descriptions are shared, and a beam-search strategy carries on the most promising paths only. The results show that a complexity reduction of about 73% can be achieved by using the two-pass approach with respect to the direct approach, while the recognition accuracy remains comparable View full abstract»

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  • System level reliability in convolution computations

    Publication Year: 1989 , Page(s): 1241 - 1252
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (868 KB)  

    The author demonstrates how generalized cyclic codes, defined over the rings and fields usually used in high-speed convolution of discrete-time sequence processing, can be incorporated directly and quite naturally within the data processing of such arithmetic systems. Design methods for one class of real cyclic codes are detailed, along with examples. Encoding and parity manipulation methods are developed which permit straightforward and efficient mechanizations. Errors are detected immediately at the conclusion of the processing pass, allowing appropriate error control actions to be initiated. Typical responses to detected errors may be to retry the calculation, reconfigure the overall system, or enter a subsystem testing mode. A complexity analysis of the protection subsystems shows that protection requires additional arithmetic complexity on the order of the number of parity positions squared View full abstract»

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  • Fast cepstrum analysis using the Hartley transform

    Publication Year: 1989 , Page(s): 1300 - 1302
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (228 KB)  

    The use of the Hartley transform (HT) in cepstrum analysis, as a substitute for the more commonly used Fourier transform (FT), is examined. With this substitution, the input to the cepstrum must be in the real domain only. The benefits of using the HT are approximately 50% less data memory required and approximately 40% faster program execution, at no loss in accuracy View full abstract»

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  • Synthesis of two-dimensional binary images through band-limited systems: a slicing method

    Publication Year: 1989 , Page(s): 1271 - 1279
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (828 KB)  

    A method is presented for solving the problem of synthesis of a two-dimensional (2-D) band limited function with prescribed level crossing. This method relies on decomposing a 2-D band limited function into the sum of products of one-dimensional (1-D) band limited functions along two orthogonal directions. The 1-D functions in one direction represent slices of the desired 2-D function and have zero crossings at the desired locations. They are produced by inserting and deleting zeros in a known band limited function (a sinc function). The orthogonal 1-D functions are used as interslice interpolation functions. The generated 2-D function is band-limited and has correct zeros at the slices. Its zeros between slices may not be correct, but the interslice distance can be made arbitrarily small. Nonuniform slicing can be used to improve the resolution of the approximations. Interpolation functions for the nonuniform slices are generated by the same method used to produce the slices View full abstract»

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  • Robust adaptive Kalman filtering with unknown inputs

    Publication Year: 1989 , Page(s): 1166 - 1175
    Cited by:  Papers (16)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (844 KB)  

    A method is proposed to adapt the Kalman filter to the changes in the input forcing functions and the noise statistics. The resulting procedure is stable in the sense that the duration of divergences caused by external disturbances are finite and short and, also, the procedure is robust with respect to impulsive noise (outlier). The input forcing functions are estimated by a running window curve-fitting algorithm, which concurrently provides estimates of the measurement noise covariance matrix and the time instant of any significant change in the input forcing functions. In addition, an independent technique for estimating the process noise covariance matrix is suggested which establishes a negative feedback in the overall adaptive Kalman filter. This procedure is based on the residual characteristics of the standard optimum Kalman filter and a stochastic approximation method. The performance of the proposed method is demonstrated by simulations and compared to the conventional sequential adaptive Kalman filter algorithm 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