# IEEE Transactions on Signal Processing

## Filter Results

Displaying Results 1 - 25 of 41
• ### Correction to 'Comments on 'A curiosum concerning discrete time convolution',' plus a remark

Publication Year: 1992
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In the author's comments to an article by Hall and Wise (see ibid., vol.38, no.6, p.1059, 1990) included a typographic error which is corrected. He also points out that the sequences used in the example become L/sub 1/ norm sequences.<> View full abstract»

• ### A novel two-stage algorithm for DCT and IDCT

Publication Year: 1992, Page(s):1610 - 1612
Cited by:  Papers (4)
| | PDF (180 KB)

On the basis of the Mobius function, a two-stage algorithm for the discrete cosine transform (DCT) and the inverse DCT (IDCT) is proposed. In this approach, the DCT matrix is factorized into the preprocessing and postprocessing matrices. The preprocessing matrix has elements of values 1 and -1, and the postprocessing matrix is a circular convolution/correlation matrix View full abstract»

• ### A neural network model for invariant pattern recognition

Publication Year: 1992, Page(s):1595 - 1599
Cited by:  Papers (10)  |  Patents (1)
| | PDF (356 KB)

A neural network is proposed to achieve invariant pattern recognition to binary inputs based on a computation of the invariance net and the ART1 neural network. Computer simulations show that the new network is plastic to unfamiliar inputs and requires no explicit training of the inputs View full abstract»

• ### Fully sigma-delta modulation encoded FIR filters

Publication Year: 1992, Page(s):1605 - 1610
Cited by:  Papers (35)  |  Patents (2)
| | PDF (464 KB)

The author suggests an FIR filter structure where both the input signal and the impulse response are encoded using sigma-delta modulation. As a result, the convolution can be obtained by summing a collection of binary or ternary numbers, depending on the quantizer in the sigma-delta modulator. This filter structure can accept analog input signals directly and can be built using modular hardware. T... View full abstract»

• ### Phonemic recognition using a large hidden Markov model

Publication Year: 1992, Page(s):1590 - 1595
Cited by:  Papers (5)  |  Patents (1)
| | PDF (528 KB)

The authors present a novel method for using the state sequence output of a large hidden Markov model as input to a phonemic recognition system. It thereby demonstrates that a significant amount of speech information is preserved in the most likely state sequences produced by such a model. Two different system formulations are presented, both achieving recognitions results equivalent to those achi... View full abstract»

• ### Digital filters for attenuating interference arriving from a wide range of angles

Publication Year: 1992, Page(s):1499 - 1507
Cited by:  Papers (1)
| | PDF (592 KB)

Digital filters for the offline multichannel processing of the signals recorded by an array of sensors are derived with the objective of extracting a desired signal arriving from a known direction. Both the desired signal and coherent interference have arbitrary waveforms. Since the direction of arrival of the interference is unknown, the array response to it is treated as a random variable, and t... View full abstract»

• ### Polynomial array for F(z1, z 2) on |z1|=1 and 2-D filter stability test

Publication Year: 1992, Page(s):1579 - 1581
Cited by:  Papers (6)
| | PDF (188 KB)

A polynomial array for a complex polynomial F(z 1, z2) on |z1|=1 is developed to overcome the difficulty in the implementation of some algebraic methods for the 2-D filter stability test. A simple and systematic algorithm to construct this polynomial array is given. The algorithm removes the requirement for conjugate polynomials ... View full abstract»

• ### Cartesian hidden Markov models with applications

Publication Year: 1992, Page(s):1601 - 1604
Cited by:  Papers (9)
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The author introduces the concept of a Cartesian hidden Markov model (CHMM), which consists of a Markov chain assuming values in the Cartesian product of a finite number of elementary state sets. The states are observed via a multivariable probabilistic mapping, again assuming values in a Cartesian product of finite sets of observables. The CHMM can be reduced to an ordinary (i.e., scalar) HMM by ... View full abstract»

• ### Adaptive code excited predictive coding

Publication Year: 1992, Page(s):1317 - 1326
Cited by:  Papers (7)  |  Patents (2)
| | PDF (740 KB)

A novel way to use the code excited linear prediction (CELP) concept that decreases the processing load while keeping the same speech quality is discussed. Rather than performing individual weighting of each candidate sequence, a global implementation of the perceptual weighting function at the codebook level is proposed. As a result, the analysis-by-synthesis procedure does not require the proces... View full abstract»

• ### Uniform approximation with doubly finite Volterra series

Publication Year: 1992, Page(s):1438 - 1442
Cited by:  Papers (23)
| | PDF (316 KB)

The assumption that a system possesses a certain discrete-time Volterra series representation frequently forms the basis for studies in the areas of signal processing and communication theory. A further assumption often made, always without discussion, is that the representation can be suitably approximated by a corresponding double finite' series. It is shown that, for a very large class of nonl... View full abstract»

• ### On local maxima of the likelihood function for Toeplitz matrix estimation

Publication Year: 1992, Page(s):1563 - 1565
Cited by:  Papers (3)
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It is shown that in estimating a Toeplitz covariance matrix, the likelihood function can present a local maximum close to the global maximum. An analytical proof is given for the matrices of order 3, and numerical examples are shown for orders 3, 4, and 5 View full abstract»

• ### Signal detection in a multiple time delay environment

Publication Year: 1992, Page(s):1587 - 1590
| | PDF (316 KB)

Signal detection methods for an uncertain multipath environment are described. The multipath transmission channel is modeled considering a different transfer function from the source to each hydrophone of the receiving array. The signal-to-noise ratio (SNR) is assumed to be low. Based on a simple example, which is given together with Monte Carlo simulation results, the statistical behavior of the ... View full abstract»

• ### Efficient order recursive algorithms for multichannel least squares filtering

Publication Year: 1992, Page(s):1354 - 1374
Cited by:  Papers (23)
| | PDF (1412 KB)

Four efficient order-recursive algorithms for least-squares (LS) multichannel FIR filtering and multivariable system identification are developed. The need for such algorithms arises when the system model assigns an unequal number of delay elements to each input channel. All proposed schemes provide considerable improvements over overparametrization or the zero padding approach. First, a block-str... View full abstract»

• ### On focusing matrices for wide-band array processing

Publication Year: 1992, Page(s):1295 - 1302
Cited by:  Papers (86)
| | PDF (672 KB)

A general class of transformation matrices for coherent signal-subspace processing is presented. These signal-subspace transformation (SST) matrices are shown to generate a sufficient statistic for maximum-likelihood bearing estimation. Two general forms for calculating SST matrices are presented, and the rotational signal-subspace (RSS) focusing matrices proposed by H. Hung and M. Kaveh (1988) ar... View full abstract»

• ### Two methods for Toeplitz-plus-Hankel approximation to a data covariance matrix

Publication Year: 1992, Page(s):1490 - 1498
Cited by:  Papers (5)
| | PDF (572 KB)

Recently, fast algorithms have been developed for computing the optimal linear least squares prediction filters for nonstationary random processes (fields) whose covariances have (block) Toeplitz-Hankel form. If the covariance of the random process (field) must be estimated from the data, the following problem is presented: given a data covariance matrix, computer from the available data, find the... View full abstract»

• ### Spectral distortion in sampling rate conversion by zero-order polynomial interpolation

Publication Year: 1992, Page(s):1576 - 1579
Cited by:  Papers (4)
| | PDF (336 KB)

For such applications as digital beamforming in sonar and sampling rate conversion in digital audio systems, there is interest in fast, approximate interpolation techniques. Computationally, the simplest of these is to approximate the desired time series sample by the available sample which is nearest in time. This procedure is variously known as sample-and-hold interpolation, zero-order polynomia... View full abstract»

• ### Time-domain filter bank analysis: a new design theory

Publication Year: 1992, Page(s):1412 - 1429
Cited by:  Papers (124)  |  Patents (16)
| | PDF (1440 KB)

The authors present a new time-domain approach for the analysis and design of a broad class of general analysis/synthesis systems based on M-band filter banks. They derive a set of time-domain conditions for reconstruction which can be used directly in a filter bank design procedure. The general and unrestricted nature of this framework allows for the design of many useful banks. In addit... View full abstract»

• ### Design of windows with steerable sidelobe dips

Publication Year: 1992, Page(s):1452 - 1459
Cited by:  Papers (11)
| | PDF (504 KB)

Based on the frequency sampling method, a novel and simple window design technique was established upon three theorems. The technique produces a set of windows which have simple expressions and the steerable sidelobe-dip (SSLD) property. By steering the dips in the sidelobes, it is easy to reduce the edge of the mainlobe, or to minimize the height of sidelobes, or to create a hole' in the sidelob... View full abstract»

• ### On unique localization of constrained-signals sources

Publication Year: 1992, Page(s):1542 - 1547
Cited by:  Papers (17)
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Conditions for unique localization of radiating sources by passive sensor arrays are presented. Unlike in previous analyses, the case wherein the signals are constrained to certain loci in the complex plane is addressed. Two types of constraints are considered. The first includes the case of constant-amplitude signals and the second includes the case of complex-exponential signals. The conditions ... View full abstract»

• ### Suppression of limit cycles in the first-order two-dimensional direct form digital filter with a controlled rounding arithmetic

Publication Year: 1992, Page(s):1599 - 1601
Cited by:  Papers (2)
| | PDF (240 KB)

The first-order two-dimensional direct-form digital filter with magnitude truncation is known to be free from limit cycles for a limited range of allowed filter coefficients. In this correspondence, the quantization technique of controlled rounding is used to extend the region of filter coefficients for which suppression of limit cycles can be proved. With controlled rounding a signal is quantized... View full abstract»

• ### The Cramer-Rao bound on frequency estimates of signals closely spaced in frequency

Publication Year: 1992, Page(s):1507 - 1517
Cited by:  Papers (53)
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This paper examines the Cramer-Rao (CR) lower bound on the variance of frequency estimates for the problem of n signals closely spaced in frequency. The main results presented are simple analytic expressions for the CR bound in terms of the maximum frequency separation, δω, SNR, and the number of data vectors, N, that are valid for small δω. The results are applica... View full abstract»

• ### An adaptive array robust to beam pointing error

Publication Year: 1992, Page(s):1582 - 1584
Cited by:  Papers (24)
| | PDF (252 KB)

The authors propose an algorithm that is robust to beam pointing error. In the algorithm, the weight vector of a beam steered adaptive array corresponds to the maximum eigenvector of a data covariance matrix which is filtered by the projection operator composed of the eigenvectors of noise subspace of the covariance matrix in which the desired signal is removed by subtractive preprocessing. The re... View full abstract»

• ### Least squared error FIR filter design with transition bands

Publication Year: 1992, Page(s):1327 - 1340
Cited by:  Papers (52)
| | PDF (1136 KB)

The authors propose the use of transition bands and transition functions in the ideal amplitude frequency response to allow the analytical design of optimal least-squared-error FIR digital filters with an explicit control of the transition band edges. Design formulas are derived for approximations to ideal frequency responses which use pth-order spline transition functions. A mixed analyt... View full abstract»

• ### An adaptive detector for deterministic signals in noise of unknown spectra using the Rao test

Publication Year: 1992, Page(s):1460 - 1468
Cited by:  Papers (10)
| | PDF (612 KB)

An adaptive detector for a known deterministic signal of unknown amplitude in Gaussian noise of unknown spectra is described. The detector is based on the Rao test, which is asymptotically equivalent to the generalized likelihood ratio. The detector achieves constant false alarm probability in the presence of large changes in input noise bandwidth and variance while providing optimum detection per... View full abstract»

• ### The optimum scalar data nonlinearity in LMS adaptation for arbitrary IID inputs

Publication Year: 1992, Page(s):1566 - 1570
Cited by:  Papers (8)
| | PDF (396 KB)

The authors show that the optimum nonlinear scale operation upon the elements of the observation vector in the LMS algorithm is exactly x/(1+μx2) for any independent stochastic data input and any noise density. Moreover, use of such a nonlinearity can yield a significant performance improvement in fast adaptation situations View full abstract»

## 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

Full Aims & Scope

## Meet Our Editors

Editor-in-Chief
Pier Luigi Dragotti
Imperial College London