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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date August 1976

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Displaying Results 1 - 19 of 19
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Introduction to digital filtering

    Page(s): 344 - 345
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    Freely Available from IEEE
  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • A comparison of several speech-spectra classification methods

    Page(s): 289 - 295
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    An important consideration in speech processing involves classification of speech spectra. Several methods for performing this classification are discussed. A number of these were selected for comparative evaluation. Two measures of performance-accuracy and stability-were derived through the use of an automatic performance evaluation system. Over 3000 hand-labeled spectra were used. Of those evaluated, a linearly mean-corrected minimum distance measure, on a 40-point spectral representation with a square (or cube) norm was consistently superior to the other methods. View full abstract»

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  • An ADPCM realization of nonrecursive digital filters

    Page(s): 312 - 320
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    A realization of nonrecursive digital filters using adaptive differential pulse-code modulation (ADPCM) as the analog-to-digital conversion scheme is presented. Output samples are computed by a sequence of read-only memory (ROM) lookups, shifts, and additions, thereby eliminating multipliers from the filter structure. Such ADPCM filters can typically operate three times faster than comparable PCM filters realized using high-speed hardware multipliers, with power consumption reduced by 25 percent. Equations are derived for computation time per output sample, signal-to-noise ratio, and coefficient accuracy necessary to achieve the optimum predicted performance. Computer simulation results agree well with theoretical predictions. View full abstract»

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  • An algorithm to compute the sequency ordered Walsh transform

    Page(s): 335 - 336
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    A fast sequency ordered Walsh Transform algorithm is presented, which is the complement to one developed by Manz. It is in place, is its own inverse, and accepts data in normal order, returning the coefficients in bit-reversed sequency order. View full abstract»

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  • An error analysis for a vector model of two-dimensional recursive filter

    Page(s): 339 - 341
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    The problem of bounding the output quantization errors in linear two-dimensional recursive filters is examined. A two-dimensional filter is analyzed by developing a model for the filter based on a vector formed from the two-dimensional output sequence. A deterministic bound is obtained and then applied to a special class of second-order filters. View full abstract»

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  • The enhancement of speech intelligibility in high noise levels by high-pass filtering followed by rapid amplitude compression

    Page(s): 277 - 282
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    This paper presents the results of an examination of rapid amplitude compression following high-pass filtering as a method for processing speech, prior to reception by the listener, as a means of enhancing the intelligibility of speech in high noise levels. Arguments supporting this particular signal processing method are based on the results of previous perceptual studies of speech in noise. In these previous studies, it has been shown that high-pass filtered/clipped speech offers a significant gain in the intelligibility of speech in white noise over that for unprocessed speech at the same signal-to-noise ratios. Similar results have also been obtained for speech processed by high-pass filtering alone. The present paper explores these effects and it proposes the use of high-pass filtering followed by rapid amplitude compression as a signal processing method for enhancing the intelligibility of speech in noise. It is shown that this new method resuits in a substantial improvement in the intelligibility of speech in white noise over normal speech and over previously implemented methods. View full abstract»

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  • Further considerations in the design of decimators and interpolators

    Page(s): 296 - 311
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    In this paper several issues concerning the design and implementation of multistage decimators, interpolators, and narrow-band filters are discussed. In particular, the question of designing these systems in terms of minimum storage rather than minimum computation rate is examined. It is shown that a design which uses finite impulse response (FIR) filters for each stage, and which is minimized for storage is essentially minimized in terms of computation rate as well. The problem of further improvements in designing decimators and interpolators by taking advantage of DON'T CARE frequency bands is also discussed. For the early stages in a multistage design it is shown that fairly significant reductions in filter order can be achieved in this manner. A third issue in the design process is the question of practical schemes for efficient implementation of multistage decimators and interpolators in both hardware and software. One such efficient implementation is discussed in this paper. Finally, the problem of designing multistage decimators and interpolators using elliptic infinite impulse response (IIR) filters is discussed. It is shown that multistage IIR designs can be somewhat more efficient computationally than single-stage designs; however, the storage efficiency of the multistage IIR design is worse than that of the single-stage IIR design. View full abstract»

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  • Speaker recognition using orthogonal linear prediction

    Page(s): 283 - 289
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    Recent experiments in speech synthesis have shown that, by an appropriate eigenvector analysis, a set of orthogonal parameters can be obtained that is essentially independent of all linguistic information across an analyzed utterance, but highly indicative of the identity of the speaker. The orthogonal parameters are formed by a linear transformation of the linear prediction parameters, and can achieve their recognition potential without the need of any time-normalization procedure. The speaker discrimination potential of the linear prediction orthogonal parameters was formally tested in both a speaker identification and a speaker verification experiment. The speech data for these experiments consisted of six repetitions of the same sentence spoken by 21 male speakers on six separate occasions. For both identification and verification, the recognition accuracy of the orthogonal parameters exceeded 99 percent for high-quality speech inputs. For telephone inputs, the accuracy exceeded 96 percent. In a separate text-independent speaker identification experiment, an accuracy of 94 percent was achieved for high-quality speech inputs. View full abstract»

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  • On least-squars design of recursive digital filters

    Page(s): 337 - 339
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    Time-domain methods for the design of recursive digital filters using a squared error criterion are compared with a frequency-domain technique. Levy's method, which has been used to estimate transfer functions of continuous-time systems is modified to obtain design equations for digital filters. A special case of Levy's method is shown to be essentially equivalent to the time-domain methods. View full abstract»

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  • Computer-aided filter design

    Page(s): 345 - 346
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    First Page of the Article
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  • Signal analysis by homomorphic prediction

    Page(s): 327 - 332
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    Two commonly used signal analysis techniques are linear prediction and homomorphic filtering. Each has particular advantages and limitations. This paper considers several ways of combining these methods to capitalize on the advantages of both. The resulting techniques, referred to collectively as homomorphic prediction, are potentially useful for pole-zero modeling and inverse filtering of mixed phase signals. Two of these techniques are illustrated by means of synthetic examples. View full abstract»

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  • An improved algorithm for ordering poles and zeros of fixed-point recursive digital filter

    Page(s): 341 - 343
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    An improved algorithm is presented for ordering the poles and zeros of fixed-point recursive digital filters so as to minimize the effect of roundoff noise. The method proceeds by successive improvement, and yields local optima which cannot be improved further by interchanging a pair of poles or zeros. The resulting assignments found for four filters of orders 8-22 are the best known. View full abstract»

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  • Pseudorandom arrays generated by two-dimensional digital filtering

    Page(s): 332 - 334
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    This correspondence describes properties of multilevel pseudorandom arrays obtained by two-dimensional digital filtering of binary pseudonoise (PN) arrays derived from maximal-length linear binary sequences. View full abstract»

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  • Fast complex convolution in finite rings

    Page(s): 343 - 344
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    Number theoretic transforms (NTT's) that can be used for the convolution of complex integer sequences are defined. A unified setting is provided wherein these transforms may be defined for all odd moduli, thus extending recent results on this topic. Multiplication-free implementation of certain of these transforms is possible. When these transforms are used to implement convolution, the resulting computation is exact. View full abstract»

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  • Coefficient sensitivities of cascaded lattices at zero frequency

    Page(s): 336 - 337
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    Expressions for the coefficient sensitivities of cascaded digital lattices introduced by Itakura and Saito and further developed by Gray and Markel are calculated using chain parameters. It is shown that a cascade of lattices with ε = +1 exhibits zero coefficient sensitivities at zero frequency. View full abstract»

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  • The generalized correlation method for estimation of time delay

    Page(s): 320 - 327
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    A maximum likelihood (ML) estimator is developed for determining time delay between signals received at two spatially separated sensors in the presence of uncorrelated noise. This ML estimator can be realized as a pair of receiver prefilters followed by a cross correlator. The time argument at which the correlator achieves a maximum is the delay estimate. The ML estimator is compared with several other proposed processors of similar form. Under certain conditions the ML estimator is shown to be identical to one proposed by Hannan and Thomson [10] and MacDonald and Schultheiss [21]. Qualitatively, the role of the prefilters is to accentuate the signal passed to the correlator at frequencies for which the signal-to-noise (S/N) ratio is highest and, simultaneously, to suppress the noise power. The same type of prefiltering is provided by the generalized Eckart filter, which maximizes the S/N ratio of the correlator output. For low S/N ratio, the ML estimator is shown to be equivalent to Eckart prefiltering. 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