Scheduled System Maintenance:
Some services will be unavailable Sunday, March 29th through Monday, March 30th. We apologize for the inconvenience.
By Topic

Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 5 • Date October 1986

Filter Results

Displaying Results 1 - 25 of 46
  • [Front cover and table of contents]

    Publication Year: 1986 , Page(s): 0
    Save to Project icon | Request Permissions | PDF file iconPDF (244 KB)  
    Freely Available from IEEE
  • Comments on "A general method of minimum cross-entropy spectral estimation"

    Publication Year: 1986 , Page(s): 1324 - 1326
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (284 KB)  

    In a recent correspondence Tzannes et al. [1] introduced a new form of minimum cross-entropy (MCE) spectral analysis based on the observation that normalized spectra and symmetric probability density functions (pdf's) are axiomatically indistinguishable. Thus, the form of the minimum cross-entropy posterior pdf can be used as a model for spectral estimation. In this correspondence we derive the same result without using a Lagrange multiplier methodology. We also establish the connection of this estimate to the cepstral representation of the signal and propose an efficient algorithm not requiring explicit solution of nonlinear equations. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Corrections to "Results on nonrecursive digital filters with nonequidistant taps"

    Publication Year: 1986 , Page(s): 1336
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (140 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Radar data processing: Vol. I - Introduction and tracking

    Publication Year: 1986 , Page(s): 1350 - 1351
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (221 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • [Back cover]

    Publication Year: 1986 , Page(s): c4
    Save to Project icon | Request Permissions | PDF file iconPDF (1045 KB)  
    Freely Available from IEEE
  • Comparison of various time delay estimation methods by computer simulation

    Publication Year: 1986 , Page(s): 1329 - 1330
    Cited by:  Papers (27)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    This correspondence provides qualitative estimates of the magnitude of the error in the measured delay time resulting from the error on the observed cross-correlation curve. The variances of five time delay estimators have been obtained by computer simulation to demonstrate the accuracy of all five methods. The comparison of the different correlation techniques shows that the average magnitude difference function gives results almost as accurate as direct correlation. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On the performance of signal-subspace processing-- Part I: Narrow-band systems

    Publication Year: 1986 , Page(s): 1201 - 1209
    Cited by:  Papers (62)  |  Patents (19)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (800 KB)  

    This paper presents an analytical evaluation of detection (determination of the number of sources) and estimation performances of narrow-band signal-subspace processing for multiple-source direction finding. The probabilities of underestimating and overestimating the number of sources are derived, under asymptotic conditions and around the threshold regions, in terms of the choice of a penalty function and signal, noise, and array parameters for the cases of at most two closely spaced sources in the spatially white noise. A scalar measure is introduced for the evaluation of the quality of the estimated signal subspace. Based on the statistics of this measure, performance thresholds are demonstrated for the signal-to-noise ratio, angle separation, and correlation between two equipowered sources. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A recursive estimation algorithm using selective updating for spectral analysis and adaptive signal processing

    Publication Year: 1986 , Page(s): 1331 - 1334
    Cited by:  Papers (32)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (424 KB)  

    This correspondence presents a recursive estimation algorithm which, unlike conventional ones, updates the estimates only when a sufficient improvement can be obtained. With a bounded noise, assumption, the resulting sequence of estimates is a sequence of convex sets (ellipsoids) in the parameter space. For the cases studied, the algorithm used less than 20 percent of the data to update the estimates and still acquired very good accuracy for spectral estimation. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Nonlinear space-variant postprocessing of block coded images

    Publication Year: 1986 , Page(s): 1258 - 1268
    Cited by:  Papers (137)  |  Patents (30)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2256 KB)  

    An important application of spatial filtering techniques is in the postprocessing of images degraded by coding. Linear, space-invariant filters are inadequate to reduce the noise produced by block coders. The noise in block coded images is correlated with the local characteristics of the signal, and such filters are unable to exploit this correlation to reduce the noise. We propose a new nonlinear, space-variant filtering algorithm which smooths jagged edges without blurring them, and smooths out abrupt intensity changes in monotone areas. Edge sharpness is preserved because near edges the filtering of the signal is negligible. Consequently, in-band noise is not reduced, but the well-known masking effect reduces the visibility of this in-band noise. The algorithm is only slightly more complex to implement than simple linear filtering. We present examples of processed images and SNR figures to demonstrate that a significant improvement in subjective and objective quality is achieved. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Synthesis of natural sounding pitch contours in isolated utterances using hidden Markov models

    Publication Year: 1986 , Page(s): 1074 - 1080
    Cited by:  Papers (8)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (736 KB)  

    A novel technique is introduced for characterizing prosodic structure and is used for speech synthesis. The mechanism consists of modeling a set of observations as a probabilistic function of a hidden Markov chain. It uses mixtures of Gaussian continuous probability density functions to represent the essential, perceptually relevant structure of intonation by observing movements of fundamental frequency in monosyllabic words of varying phonetic structure. High-quality speech synthesis, using multipulse excitation, is used to demonstrate the power of the HMM in preserving the naturalness of the intonational meaning, conveyed by the variation of fundamental frequency and duration. The fundamental frequency contours are synthesized using a random number generator from the models, and are imposed on a synthesized prototype word which had the intonation of a low fall. The resulting monosyllabic words with imposed synthesized fundamental frequency contours show a high level of naturalness and are found to be perceptually indistinguishable from the original recordings with the same intonation. The results clearly show the high potential of hidden Markov models as a mechanism for the representation of prosodic structure by naturally capturing its essentials. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Simple, effective computation of principal eigenvectors and their eigenvalues and application to high-resolution estimation of frequencies

    Publication Year: 1986 , Page(s): 1046 - 1053
    Cited by:  Papers (47)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (784 KB)  

    We present the results of an investigation of the Prony-Lanczos (P-L) method [14], [38] and the power method [39] for simple computation of approximations to a few eigenvectors and eigenvalues of a Hermitian matrix. We are motivated by realization of high-resolution signal processing in an integrated circuit. The computational speeds of the above methods are analyzed. They are completely dependent on the speed of a matrix-vector product operation. If only a few eigenvalues or eigenvectors are needed, the suggested methods can substitute for methods of the LINPACK or EISPACK subroutine libraries. The accuracies of the suggested methods are evaluated using matrices formed from simulated data consisting of two sinusoids plus Gaussian noise. Comparisons are made to the corresponding eigenvalues and eigenvectors obtained using LINPACK as a reference. Also, the accuracies of frequency estimates obtained from the eigenvectors are compared. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new scheme for designing IIR filters with finite wordlength coefficients

    Publication Year: 1986 , Page(s): 1335 - 1336
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    A scheme for designing IIR digital filters, with specified coefficient wordlengths based on successive digitization and reoptimization, is presented and compared to related approaches via a number of design examples. The ordered digitization of the coefficients based on their sensitivity, with each digitization followed by reoptimization of the remaining coefficients, results in a much reduced computation effort in comparison to related design approaches. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Adaptive comb filtering for harmonic signal enhancement

    Publication Year: 1986 , Page(s): 1124 - 1138
    Cited by:  Papers (103)  |  Patents (52)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1224 KB)  

    A new algorithm is presented for adaptive comb filtering and parametric spectral estimation of harmonic signals with additive white noise. The algorithm is composed of two cascaded parts. The first estimates the fundamental frequency and enhances the harmonic component in the input, and the second estimates the harmonic amplitudes and phases. Performance analysis provides new results for the asymptotic Cramer-Rao bound (CRB) on the parameters of harmonic signals with additive white noise. Results of simulations indicate that the variances of the estimates are of the same order of magnitude as the CRB for sufficiently large data sets, and illustrate the performance in enhancing noisy artificial periodic signals. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Multichannel AR spectrum estimation: The optimum approach in the reflection coefficient domain

    Publication Year: 1986 , Page(s): 1139 - 1152
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1016 KB)  

    A new approach for multichannel autoregressive (AR) spectrum estimation is introduced based on an iterative gradient algorithm. In the context of dynamic programming, the approach is shown to be optimum in the reflection coefficient domain, whereas the multichannel AR Nuttall-Strand algorithm is shown to be suboptimum. The approach guarantees a stable multichannel AR filter and a nonnegative spectrum matrix estimate. It is demonstrated via simulation examples that such drawbacks of the Nuttall-Strand method as frequency bias and line splitting may be alleviated by the optimum approach. However, the improved performance is achieved at the expense of considerably more computational effort and time. The Single-channel Fougere descent method may be seen as a special case of the proposed multichannel AR approach. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Some new aspects of filters for filter banks

    Publication Year: 1986 , Page(s): 1182 - 1200
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (976 KB)  

    The present paper deals with the design of filters for use in filter banks, i.e., with certain constraints on their transition bands. The goal is to achieve a baseband frequency response of a back-to-back analysis/synthesis filter bank system that is ideally flat in magnitude and linear in phase. Using the proposed method, this is possible for conventional polyphase filter banks, single-sideband (SSB) filter banks, and generalized quadrature mirror filter (GQMF) banks. The method provides several options, one of them even yielding subsystems with linear phase. In contrast to most previous approaches to this topic, the described design procedure is an analytical one and is not based on optimizations or iterative algorithms. Furthermore, two families of window sequences are proposed for use in the design procedure mentioned above, but are also suitable for any windowed Fourier filter design. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Stochastic operator norms for two-parameter adaptive lattice filters

    Publication Year: 1986 , Page(s): 1162 - 1164
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    Expressions are derived for bounds on average output power in lattice filters with unequal forward and backward reflection coefficients, such as arise in the adaptive least-squares lattice algorithms [3]. It is shown that under stationary, noisy conditions, these bounds will in general be larger than in the case of equal forward and backward reflection coefficients, as in the adaptive gradient lattice filters. Expressions obtained here can be used to derive the average output power for several (adaptive) lattice filters. They suggest that misadjustment may be higher under noisy conditions for the ALSL than for the AGL. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Separating multipaths by global optimization of a multidimensional matched filter

    Publication Year: 1986 , Page(s): 1029 - 1037
    Cited by:  Papers (25)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (840 KB)  

    A transmitted signal can arrive at a receiver via several refracted Fermat paths. If the paths are independent in the Fresnel sense, then the received signal can be modeled as the sum of amplitude scaled and time shifted copies of a predetermined replica plus white noise. We present an algorithm that uses the replica to determine the time shifts and amplitudes for each path. It is referred to as an n-dimensional matched filter algorithm by analogy with the well-known matched filter algorithm. The cross correlation between the received signal and the replica oscillates near the center frequency of the transmitted signal. This causes the n-dimensional matched filter output to have many local maxima that are not globally optimal. The time shifts and amplitude scalings for the Fermat paths are determined by maximizing the output of the n-dimensional matched filter. The algorithm is more robust and efficient than others currently available. Simulated realizations of received signals were generated with multipath and noise characteristics similar to an ocean acoustic transmission case. These realizations were then separated into arrival times and corresponding amplitudes by the algorithm. The results of these tests and the general limitations of the algorithm are discussed. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Subband coding of images

    Publication Year: 1986 , Page(s): 1278 - 1288
    Cited by:  Papers (478)  |  Patents (94)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2200 KB)  

    Subband coding has become quite popular for the source encoding of speech. This paper presents a simple yet efficient extension of this concept to the source coding of images. We specify the constraints for a set of two-dimensional quadrature mirror filters (QMF's) for a particular frequency-domain partition, and show that these constraints are satisfied by a separable combination of one-dimensional QMF's. Bits are then optimally allocated among the subbands to minimize the mean-squared error for DPCM coding of the subbands. Also, an adaptive technique is developed to allocate the bits within each subband by means of a local variance mask. Optimum quantization is employed with quantizers matched to the Laplacian distribution. Subband coded images are presented along with their signal-to-noise ratios (SNR's). The SNR performance of the subband coder is compared to that of the adaptive discrete cosine transform (DCT), vector quantization, and differential vector quantization for bit rates of 0.67, 1.0, and 2.0 bits per pixel for 256 × 256 monochrome images. The adaptive subband coder has the best SNR performance. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Exact maximum likelihood parameter estimation of superimposed exponential signals in noise

    Publication Year: 1986 , Page(s): 1081 - 1089
    Cited by:  Papers (290)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1104 KB)  

    A unified framework for the exact maximum likelihood estimation of the parameters of superimposed exponential signals in noise, encompassing both the time series and the array problems, is presented. An exact expression for the ML criterion is derived in terms of the linear prediction polynomial of the signal, and an iterative algorithm for the maximization of this criterion is presented. The algorithm is equally applicable in the case of signal coherence in the array problem. Simulation shows the estimator to be capable of providing more accurate frequency estimates than currently existing techniques. The algorithm is similar to those independently derived by Kumaresan et al. In addition to its practical value, the present formulation is used to interpret previous methods such as Prony's, Pisarenko's, and modifications thereof. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A complex adaptive algorithm for IIR filtering

    Publication Year: 1986 , Page(s): 1342 - 1344
    Cited by:  Papers (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (312 KB)  

    This correspondence generalizes the Gauss-Newton algorithm [1] for adaptive IIR filters to include complex coefficients. The resulting algorithm simultaneously updates the real and imaginary parts of the filter coefficients to minimize the average squared estimation error. It has application in frequency-domain adaptive IIR filtering [2] where the signals and filter coefficients are complex. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Broad-band beamforming and the generalized sidelobe canceller

    Publication Year: 1986 , Page(s): 1322 - 1323
    Cited by:  Papers (16)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    This correspondence identifies general conditions under which the generalized sidelobe canceller (GSC) structure implements a linearly constrained minimum variance (LCMV) broad-band beam-former. In previously published work, conditions have been presented for the case in which the linear constraint is restricted to fixed gain in the steer direction. Here, that work is extended to include general linear constraints. A geometric interpretation is presented to provide insight into the relationship between the GSC structure and LCMV beamformers, and several recent developments in broad-band beam-forming are reviewed and related to the GSC structure. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Image reconstruction from zero crossings

    Publication Year: 1986 , Page(s): 1269 - 1277
    Cited by:  Papers (22)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1360 KB)  

    This study is concerned with the information in zero crossings (ZC) of images. Logan's conditions, specifying when a one-dimensional signal may be recovered (within a multiplicative constant) from its ZC, are extended for various cases of two-dimensional signals. An algorithm is implemented in reconstruction of context-free band-pass images. It is also successfully applied in reconstruction of contextual images by first dissecting the image into appropriate bandpass-band-limited two-dimensional signals. In almost all cases of spectrum confined to less than an octave in one dimension, the reconstruction algorithm converges, whereas it does not converge for any signal exceeding one octave in bandwidth. This paper substantiates the proposition that images are well represented by the partial information confined to ZC. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Roundoff noise minimization and pole-zero sensitivity in fixed-point digital filters using residue feedback

    Publication Year: 1986 , Page(s): 1210 - 1220
    Cited by:  Papers (45)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1088 KB)  

    The problem of designing a finite wordlength fixed-point realization of an Nth-order digital filter, which uses residue feedback to minimize the output roundoff noise subject to l2-scaling, is considered. The new structures require N extra additions, but no more multiplications than the earlier low noise structures of Mullis and Roberts, and have lower roundoff noise for sufficiently narrow bandwidth filters. A new set of filter invariants, called the residue modes, are defined which characterize the new low noise structures and determine the output noise variance. If the sum of the residue modes is less than the sum of the second-order modes of Mullis and Roberts, then lower roundoff noise is achieved. Every filter structure is shown to define a unique (symmetrizing) matrix Q. Pole-zero sensitivities and a new noise measure are defined in terms of Q. Numerical results are included which compare the noise and pole-zero sensitivity characteristics of different filter structures. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Unbiased parameter estimation of nonstationary signals in noise

    Publication Year: 1986 , Page(s): 1319 - 1322
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (400 KB)  

    Recent approaches to the modeling of nonstationary signals by means of AR or ARMA models use a representation with time-varying parameters. The time-varying parameters are assumed to be linear combinations of a set of basis time functions so that the model is specified by constant parameters. For stationary signals disturbed by white noise, an approach based upon a modified least-squares method leads to a good unbiased estimator of the parameters. In this correspondence, a similar algorithm deriving the unbiased parameters for nonstationary signals in white noise is given. The experimental results show the good performance of the proposed estimator. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Modeling and identification of symmetric noncausal impulse responces

    Publication Year: 1986 , Page(s): 1171 - 1181
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1096 KB)  

    Spectrally equivalent causal (SEC) models for single-input/single-output noncausal systems are derived with the objective of estimating the noncausal impulse response. Attention is restricted to symmetric, noncausal impulse responses, and the measurements of the system output are assumed to be noisy. It is shown that by using structured SEC models, either state-space or ARMA-plus-noise, consistent estimates of the noncausal impulse response can be obtained by exploiting only the second-order statistics of the measurements. A new version of the correlation technique particularly suited to structured models is presented and analyzed. Computer simulation results are also presented to illustrate the performance of the proposed approach. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.

Aims & Scope

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

Full Aims & Scope