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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 1 • Date February 1979

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

    Publication Year: 1979 , Page(s): 0
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    Freely Available from IEEE
  • Correction of a reported result

    Publication Year: 1979 , Page(s): 97
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  • [Back cover]

    Publication Year: 1979 , Page(s): c4
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  • A new proof of the minimum-phase property of the unit prediction error operator

    Publication Year: 1979 , Page(s): 99 - 100
    Cited by:  Papers (3)
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    It is well known that the unit prediction error operator has the minimum-phase property. We give a new proof which is based on the eigenvector-eigenvalue representation of the associated autocorrelation matrix. View full abstract»

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  • Data compression: Benchmark papers in EECS

    Publication Year: 1979 , Page(s): 101
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  • Optimal design and comparison of wide-band digital on-line differentiators

    Publication Year: 1979 , Page(s): 46 - 52
    Cited by:  Papers (10)
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    The problem of wide-band digital on-line differentiators is of increasing importance. The frequency approach is used to design and compare different filters. First, an error criterion that represents numerical error is derived, combining amplitude and phase errors. Conditions that ensure good low frequency behavior are included. Two design procedures to obtain the filter coefficients are discussed: an indirect approach based on modeling with polynomials and modified splines and a direct approach using an optimization scheme. Finally, the frequency characteristics of the main optimal and suboptimal wide-band differentiators are compared. The full-band requirement restricts improvements to about 33 percent with respect to the first-order difference formula. View full abstract»

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  • Recent advances in residue number techniques for recursive digital filtering

    Publication Year: 1979 , Page(s): 19 - 30
    Cited by:  Papers (41)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1192 KB)  

    A new algorithm for scaling in residue number systems (RNS's) is presented for applying residue number theory to recursive digital filtering. The algorithm provides an efficient method for scaling the output of each recursive filter section for use in subsequent iterations of the recursion. Four classes of residue systems are described in which scaling is simple and quantization errors are minimized, thereby combining good quantization error performance with the advantages of high-speed residue arithmetic. A computer analysis of the scaling quantization errors is presented, as well as some results from a recursive residue simulation. Three hardware architectures are described for the realization of recursive residue filters. View full abstract»

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  • Variable digital filters having a recursive structure

    Publication Year: 1979 , Page(s): 98 - 99
    Cited by:  Papers (4)  |  Patents (3)
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    Variable digital filters allow the frequency characteristics of a filter to be manipulated. Frequency transformations have been successfully used to obtain a variable filter from a nonrecursive prototype. However, this technique cannot be applied directly to filters having a recursive structure. A variation of this method is presented which allows variable filters to be obtained from both recursive and nonrecursive prototypes. This method has special advantages when the prototype has an equal number of poles and zeros (e.g., digital filters obtained by the bilinear transformation from an analog filter). View full abstract»

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  • Surface wave filters

    Publication Year: 1979 , Page(s): 101 - 102
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  • A procedure for using pattern classification techniques to obtain a voiced/Unvoiced classifier

    Publication Year: 1979 , Page(s): 83 - 89
    Cited by:  Papers (20)  |  Patents (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (880 KB)  

    A classifier to make the voiced/unvoiced (V/UV) decision in speech analysis which performs with an error rate of less than half of a percent is presented. The decision making process is viewed as a pattern recognition problem in which a number of features can be used to make the classification. Training is accomplished using a nonparametric, nonstatistical technique. In order to obtain a classifier which would make the correct decision for a variety of speakers and to determine which of the features under consideration should be used, a procedure for interleaving the contributions of the feature and speaker sets was developed. This procedure is presented in terms of the notions of covering and satisfaction. The failure of a classifier to cover a set of speakers indicates that more training information from those speakers is necessary to define the classifier. The failure of the classifier to satisfy a set of speakers indicates that the performance of the classifier could be improved by the use of more features in making the V/UV decision. In the training procedure, covering and satisfaction were attained on successively larger sets of speakers, with the result that a classifier was obtained which could correctly make the V/UV decision for all of the speakers used in testing, including those not used in the training process. View full abstract»

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  • A fast two-dimensional median filtering algorithm

    Publication Year: 1979 , Page(s): 13 - 18
    Cited by:  Papers (212)  |  Patents (33)
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    We present a fast algorithm for two-dimensional median filtering. It is based on storing and updating the gray level histogram of the picture elements in the window. The algorithm is much faster than conventional sorting methods. For a window size of m × n, the computer time required is 0(n). View full abstract»

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  • Implementation of a new stability test for n-D filters

    Publication Year: 1979 , Page(s): 1 - 4
    Cited by:  Papers (8)
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    The implementation procedure for a new stability test for two-dimensional (2-D) digital filters, recently reported, has been extended to the n-D case. The procedure dwells on the generation of a sequence of polynomials in an increasing number (up to n - 1) of real variables, which need be tested for local positivity. Corresponding multidimensional continuous filter results can be routinely developed. View full abstract»

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  • Text-independent speaker recognition from a large linguistically unconstrained time-spaced data base

    Publication Year: 1979 , Page(s): 74 - 82
    Cited by:  Papers (19)  |  Patents (1)
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    A very large data base consisting of over 36 h of unconstrained extemporaneous speech, from 17 speakers, recorded over a period of more than three months, has been analyzed to determine the effectiveness of long-term average features for speaker recognition. Results are shown to be strongly dependent on the voiced speech averaging interval Lε. Monotonic increases in the probability of correct identification and monotonic decreases in the equal error probability for speaker verification were obtained as Lεincreased, even with substantial time periods between successive sessions. For Lεcorresponding to approximately 39 s of speech, text-independent results (no linguistic constraints embedded into the data base) of 98.05 percent for speaker identification and 4.25 percent for equal error speaker verification were obtained. View full abstract»

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  • Second-order output statistics of the adaptive line enhancer

    Publication Year: 1979 , Page(s): 31 - 39
    Cited by:  Papers (46)
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    The adaptive line enhancer (ALE) is an adaptive digital filter designed to suppress uncorrelated components of its input, while passing any narrow-band components with little attenuation. The purpose of this paper is to analyze the second-order output statistics of the ALE in steady-state operation, for input samples consisting of weak narrow-band signals in white Gaussian noise. The ALE output is shown to be the sum of two uncorrelated components, one arising from optimum finite-lag Wiener filtering of the narrow-band input components, and the other arising from the misadjustment error associated with the adaptation process. General expressions are given for the output auto-correlation function and power spectrum with arbitrary narrow-band input signals, and the case of a single sinusoid in white noise is worked out as an example. Finally, the significance of these results to practical applications of the ALE is mentioned. View full abstract»

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  • A random-search method for designing finite-wordlength recursive digital filters

    Publication Year: 1979 , Page(s): 40 - 46
    Cited by:  Papers (16)
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    A new approach is described for finding the optimal finite-wordlength coefficients of recursive digital filters. A theoretical justification for the new method is presented and computational experience is reported, allowing comparison with some other methods proposed in the literature. The method appears to be suitable for designing filters of considerable complexity, using a realistic amount of computer time. View full abstract»

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  • Transient and convergent behavior of the adaptive line enhancer

    Publication Year: 1979 , Page(s): 53 - 62
    Cited by:  Papers (119)
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    The adaptive line enhancer (ALE) was first described as a practical technique for separating the periodic from the broad-band components of an input signal and for detecting the presence of a sinusoid in white noise. Subsequent work has shown that this adaptive filtering structure is applicable to spectral estimation, predictive deconvolution, speech processing, interference rejection, and other applications which have historically used matrix inversion or Levinson's algorithm techniques. This paper uses an eigenvalue-eigenvector analysis of the expected ALE impulse response vector to demonstrate properties of the convergent filter and to quantify the convergence time and characteristics of the ALE. In particular the ALE's response to a sinusoid plus white noise input is derived and compared to a computer simulation of the ALE with such an input. The eigenvalue-eigenvector technique is then used to evaluate the ALE's performance as an adaptive prewhitener for autoregressive (AR) models with white observation noise. A method is demonstrated which prevents the problem of spectral estimation bias which usually accrues from the observation noise. View full abstract»

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  • Composite signal decomposition by digital inverse filtering

    Publication Year: 1979 , Page(s): 95 - 97
    Cited by:  Papers (2)
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    Inverse filters are conventionally used for resolving overlapping signals of identical waveshape. However, the inverse filtering approach is shown to be useful for resolving overlapping signals, identical or otherwise, of unknown waveshapes. Digital inverse filter design based on autocorrelation formulation of linear prediction is known to perform optimum spectral flattening of the input signal for which the filter is designed. This property of the inverse filter is used to accomplish composite signal decomposition. The theory has been presented assuming constituent signals to be responses of all-pole filters. However, the approach may be used for a general situation. View full abstract»

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  • An extrapolation procedure for band-limited signals

    Publication Year: 1979 , Page(s): 4 - 12
    Cited by:  Papers (72)
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    In this paper, the task of extrapolating a time-truncated version of a band-limited signal shall be considered. It will be shown that the basic extrapolation operation is feasible for only a particular subset of the class of band-limited signals (i.e., the operation is well-posed mathematically). An efficient algorithmic method for achieving the desired extrapolation on this subset is then presented. This algorithm is structured so that all necessary signal manipulations involve signals which are everywhere zero except possibly on a finite "observation time" set. As a consequence, its implementation is straightforward and can be carried out in real time. This is to be contrasted with many existing extrapolation algorithms which theoretically involve operations on signals that are nonzero for almost all values of time. Their numerical implementation thereby necessitates an error producing time-truncation and a resultant deleterious effect on the corresponding extrapolation. Using straightforward algebraic operations, a convenient one-step extrapolation procedure is next developed. This is noteworthy in that this procedure thereby enables one to effectively circumvent any potentially slow convergence rate difficulties which typically characterize extrapolation algorithms. The effectiveness of this one-step procedure is demonstrated by means of two examples. View full abstract»

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  • Linear systems, Fourier transforms and optics

    Publication Year: 1979 , Page(s): 102
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  • Adaptive noise spectral shaping and entropy coding in predictive coding of speech

    Publication Year: 1979 , Page(s): 63 - 73
    Cited by:  Papers (49)  |  Patents (11)
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    We report on research to code speech at 16 kbit/s with the goal of having the quality of the coded speech be equal to that of the original. Some of the original speech had been corrupted by noise and distortions typical of long-distance telephone lines. The basic structure chosen for our system was adaptive predictive coding. However, the rigorous requirements of this work led to a new outlook on the different aspects of adaptive predictive coding. We have found that the pitch predictor is not cost-effective on balance and may be eliminated. Solutions are presented to deal with the two types of quantization noise: clipping and granular noise. The clipping problem is completely eliminated by allowing the number of quantizer levels to increase indefinitely. An appropriate self-synchronizing variable-length code is proposed to minimize the average data rate; the coding scheme seems to be adequate for all speech and all conditions tested. The granular noise problem is treated by modifying the predictive coding system in a novel manner to include an adaptive noise spectral shaping filter. A design for such a filter is proposed that effectively eliminates the perception of granular noise. View full abstract»

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  • Approaches to adaptive transform speech coding at low bit rates

    Publication Year: 1979 , Page(s): 89 - 95
    Cited by:  Papers (20)  |  Patents (7)
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    This paper discusses problems of adaptive transform coding schemes at bit rates of 12 kbit/s and below. Objective and subjective performance reductions, like low-pass filtering effects as one of the main sources of perceptual distortion, are investigated and proposals are made how to improve the performance of the coder at low and medium bit rates. Additionally, the needed transmission of side information reduces the efficiency of the scheme. Various methods to lower the rate of this supplementary data signal are given as well as modifications of the scheme which lead to a more easily implemented coder structure. View full abstract»

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  • Applied acoustics

    Publication Year: 1979 , Page(s): 103
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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