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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date August 1983

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Displaying Results 1 - 25 of 42
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Comments on "Hardware for two-dimensional digital filtering using Fermat number transforms

    Page(s): 1034 - 1037
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    In the above-mentioned paper it was shown how two-dimensional convolutions of arrays of size μn by μn could be calculated by means of number theoretic transforms of length n by n. Some improvements can be made in this algorithm in the sense that the number of general multiplications can be reduced. Systematic techniques to accomplish this reduction are presented in this comment. View full abstract»

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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • Noise spectra of digital sine-generators using the table-lookup method

    Page(s): 1037 - 1039
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    The table-lookup method is a convenient and very flexible way to generate high-quality sinusoidal test signals for measurements in psychoacoustics, speech perception, etc. Exact spectra of such signals, which are of great interest in experiments, are derived, and the relation between spectral shape, table length, and sine frequency is shown. View full abstract»

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  • Steady-state response of a delay-constrained adaptive linear predictor filter to a sinusoid in white noise

    Page(s): 1039 - 1043
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    An analysis is presented for the steady-state response of a delay-constrained adaptive linear predictor filter to a sinusoid in white noise. Expressions are derived for the steady-state adaptive weights, frequency response, and line suppression. A useful bound is also derived for the worst-case output signal-to-noise ratio as a function of normalized frequency and the order of the filter. View full abstract»

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  • Comparison of three correlation coefficient estimators for gaussian stationary processes

    Page(s): 1023 - 1025
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    Using independent samples from two stationary Gaussian processes, second-order statistics are examined for the following three correlation estimators: 1) the direct estimator, 2) a "hybrid-sign" estimator, and 3) the polarity coincidence estimator. In most cases, increasing the number of samples by a factor of two or three allows one to use the simple polarity coincidence estimator with the same accuracy as the other two estimators. View full abstract»

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  • Calculation of the impulse response of two-dimensional digital filter

    Page(s): 1050 - 1052
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    A method for the calculation of the impulse response of quarter-plane and half-plane filters will be presented. The basic result is that the functions Hl(Z) defined byH(z, w) = summin{l=-infin}max{+infin} H_{l}(z)w^{-1}obey a simple difference equation. The necessary initial conditions can be obtained by restrictions on the support of the impulse response. The proposed method is of theoretical and practical interest. It provides a way for the calculation of the impulse response of half-plane filters having a total half plane as support. View full abstract»

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  • On efficient implementations of 2-D digital filters using logarithmic number systems

    Page(s): 877 - 885
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    The implementation of 2-D digital filters using logarithmic number systems (LNS's) is discussed in detail. The properties of such systems are derived with reference to an iterative rounding scheme, which allows increasing the accuracy of the filter coefficient representation by using more than one power of the base a. The optimization of the base is proposed as an efficient tool for the design of accurate 2-D filters. Different design strategies are considered, together with a discussion of the degree of precision required for the circuits which determine the base of the LNS. The efficiency and versatility of this approach are demonstrated with several design examples in 2-D. View full abstract»

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  • Two-dimensional IIR filter design with magnitude and phase error criteria

    Page(s): 886 - 894
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    Two-dimensional IIR filters are designed using an error criterion consisting of the weighted sum of magnitude, phase, and stability errors, thereby extending the spectral factorization-based method to complex approximation. The weights are adjusted to effectively constrain stability while minimizing a weighted sum of frequency domain magnitude and phase errors. We use a derivative-free variant of the Marquardt optimization algorithm augmented by a one-dimensional search. We present examples of approximation of both linear and nonlinear phase ideal transfer functions, and conduct comparisons across a range of numerator and denominator orders, from FIR to all-pole, while keeping the total number of coefficients approximately constant. We find that the designed IIR filters offer better magnitude response in the case of linear phase ideal functions, even considering coefficient reflection symmetry, but of course, the FIR filter has the better (zero) phase error. When the ideal transfer function has nonlinear phase, and hence no coefficient reflection symmetry, we find that the designed IIR filter can perform better than the FIR filter with regard to both magnitude and phase error, when magnitude error is weighted more heavily than phase error. View full abstract»

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  • Relative linear power contribution with estimation statistics

    Page(s): 1025 - 1028
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    The relative contribution by a noiselessly observed input signal to the power of a possibly disturbed observed stationary output signal from a linear system is expressed into signal spectral densities. Approximations of estimator statistics and derived confidence limits agree fairly well with simulation results for white signals. View full abstract»

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  • On temporal alignment of sentences of natural and synthetic speech

    Page(s): 807 - 813
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    One way to improve the quality of synthetic speech, and to learn about temporal aspects of speech recognition, is to study the problem of time aligning pairs of spoken sentences. For example, one could evaluate various sets of duration rules for synthesis by comparing the time alignments of speech sounds within synthetic sentences to those of naturally spoken sentences. In this manner, an improved set of sound duration rules could be obtained by applying some objective measure to the alignment scores. For speech recognition applications, one could obtain automatic labeling of continuous speech from a hand-marked prototype to obtain models and/or statistical data on sounds within sentences. A key question in the use of automatic alignment of sentence length utterances is whether the time warping methods, developed for isolated word recognition, could be extended to the problem of time aligning sentence length utterances (up to several seconds long). A second key question is the reliability and accuracy of such an alignment. In this paper we investigate these questions. It is shown that, with some simple modifications, the dynamic time warping procedures used for isolated word recognition apply almost as well to alignment of sentence length utterances. It is also shown that, on the average, the uncertainty in the location of significant events within the sentence is much smaller than the event durations although the largest errors are longer than some event durations. Hence, one must apply caution in using the time alignment contour for synthesis or recognition applications. View full abstract»

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  • Arrays of synthetic acoustic well logging waveforms: Computation and source design

    Page(s): 946 - 955
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    Arrays of synthetic acoustic well logging waveforms are useful in the initial development of signal processing methods, as an aid in understanding actual well logging data, and in an analysis-by-synthesis approach to data inversion. In this paper, we present a fast and accurate method for the numerical evaluation of arrays of synthetic acoustic well logging waveforms associated with a homogeneous solid formation model. The method evaluates the two-dimensional Fourier transform of the modal coefficient which is expressed in terms of the cylindrical wave reflection coefficient and cylindrical wave impedances. A table look-up and interpolation scheme for the evaluation of ratios of Hankel functions for complex-valued arguments is responsible for a factor of three reduction in the overall computation time required to generate synthetic data, relative to the evaluation of these ratios using a Hankel function evaluation algorithm. This substantial savings easily outweighs the modest increase in storage which is required for the table. A low frequency constraint on the temporal response of the source is presented to ensure that the synthetic waveforms tend to zero for large time. View full abstract»

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  • An algorithm for the numerical evaluation of the Hankel and Abel transforms

    Page(s): 979 - 985
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    A procedure for the efficient numerical evaluation of the Hankel and Abel transforms is proposed. The Abel transform is reduced to a convolution which is evaluated in part analytically and in part with an FFT. The Hankel transform is obtained by following the Abel transform with an FFT. View full abstract»

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  • Asymptotically optimal detection in unknown colored noise via autoregressive modeling

    Page(s): 927 - 940
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    The problem of detecting a known signal in colored Gaussian noise of unknown covariance is addressed. The noise is modeled as an autoregressive process of known order but unknown coefficients. By employing the theory of generalized likelihood ratio testing, a detector structure is derived and then analyzed for performance. It is proven that for large data records the detection performance is identical to that of an optimal prewhitener and matched filter, and therefore the detector itself is optimal. Simulation results indicate that the data record length necessary for the asymptotic results to apply can be quite small. Thus, the proposed detector is well suited for practical applications. View full abstract»

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  • On the effects of varying filter bank parameters on isolated word recognition

    Page(s): 793 - 807
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    The vast majority of commercially available isolated word recognizers use a filter bank analysis as the front end processing for recognition. It is not well understood how the parameters of different filter banks (e.g., number of filters, types of filters, filter spacing, etc.) affect recognizer performance. In this paper we present results of performance evaluation of several types of filter bank analyzers in a speaker trained isolated word recognition test using dialed-up telephone line recordings. We have studied both DFT (discrete Fourier transform) and direct form implementations of the filter banks. We have also considered uniform and nonuniform filter spacings. The results indicate that the best performance (highest word accuracy) is obtained by both a 15-channel uniform filter bank and a 13-channel nonuniform filter bank (with channels spacing along a critical band scale). The performance of a 7-channel critical band filter bank is almost as good as that of the two best filter banks. In comparison to a conventional linear predictive coding (LPC) word recognizer, the performance of the best filter bank recognizers was, on average, several percent worse than that of an eighth-order LPC-based recognizer. A discussion as to why some filter banks performed better than others, and why the LPC-based system did the best, is given in this paper. View full abstract»

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  • A new design method of optimal finite wordlength linear phase FIR digital filters

    Page(s): 1032 - 1034
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    The branch and bound technique is applied to the design of optimal finite wordlength linear phase FIR digital filters in a new efficient way. Compared to other reported methods there is a great saving of storage area used by the designing program; reductions in computation time are also to be expected. View full abstract»

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  • Effects of additive noise on signal reconstruction from Fourier transform phase

    Page(s): 894 - 898
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    The effects of additive noise in the given phase on signal reconstruction from the Fourier transform phase are experimentally studied. Specifically, the effects on the sequence reconstruction of different methods of sampling the degraded phase of the number of nonzero points in the sequence, and of the noise level, are examined. A sampling method that significantly reduces the error in the reconstructed sequence is obtained, and the error is found to increase as the number of nonzero points in the sequence increases and as the noise level increases. In addition, an averaging technique is developed which reduces the effects of noise when the continuous phase function is known. Finally, as an illustration of how the results in this paper may be applied in practice, Fourier transform signal coding is considered. Coding only the Fourier transform phase and reconstructing the signal from the coded phase is found to be considerably less efficient (i.e., a higher bit rate is required for the same mean-square error) than reconstructing from both the coded phase and magnitude. View full abstract»

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  • Time-dependent ARMA modeling of nonstationary signals

    Page(s): 899 - 911
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    Modeling of nonstationary signals can be achieved through time-dependent autoregressive moving-average models and lattices, by the use of a limited series expansion of the time-varying coefficients in the models. This method leads to an extension of several well-known techniques of stationary spectral estimation to the nonstationary case. Time-varying AR models are identified by means of a fast (Levinson) algorithm which is also suitable for the AR part of a mixed ARMA model. An alternative to this method is given by the extension of Cadzow's method. Lattices with time-dependent reflection coefficients are identified through an algorithm which is similar to Burg's. Finally, the Prony-Pisarenko estimator is adapted to this nonstationary context, the signal considered in this case being the output of a zero-input time-varying system corrupted by an additive white noise. In all these methods the estimation is global in the sense that the parameters are estimated over a time interval [0, T], given the observations [y0... yT]. The maximum likelihood method which falls within the same framework is also briefly studied in this paper. Simulations of these algorithms on chirp signals and on transitions between phonemes in speech conclude the paper. View full abstract»

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  • Signal reconstruction from short-time Fourier transform magnitude

    Page(s): 986 - 998
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    In this paper, a signal is shown to be uniquely represented by the magnitude of its short-time Fourier transform (STFT) under mild restrictions on the signal and the analysis window of the STFT. Furthermore, various algorithms are developed which reconstruct signal from appropriate samples of the STFT magnitude. Several of the algorithms can also be used to obtain signal estimates from the processed STFT magnitude, which generally does not have a valid short-time structure. These algorithms are successfully applied to the time-scale modification and noise reduction problems in speech processing. Finally, the results presented here have similar potential for other application areas, including those with multidimensional signals. View full abstract»

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  • Noise sensitivity of band-limited signal derivative interpolation

    Page(s): 1028 - 1032
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    The sensitivity of interpolation of the pth derivative of a band-limited signal directly from the signal's samples in the presence of additive stationary noise is considered. Oversampling and filtering generally decrease the interpolation noise level when the data noise is not band-limited. A lower bound on the interpolation noise level can be approached arbitrarily closely by increasing the sampling rate. The lower bound is equivalent to the noise level obtained by low-pass filtering and pth-order differentiation of the unsampled additive input noise. View full abstract»

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  • K-cyclic symmetries in multidimensional sampled signals

    Page(s): 847 - 860
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    In this paper, K-cyclic frequency domain symmetry conditions for N-dimensional sampled signals are introduced. It is shown that these frequency domain symmetry conditions result in certain interrelationships between the sample values. A theorem is given that specifies these interrelationships. Several interesting properties of K-cyclic symmetries are discussed. Also, a systematic approach is given for finding the parameters of several commonly encountered symmetry conditions. Certain implications to multidimensional digital filtering are discussed, including Lpoptimality and the reduction in computation for design and implementation. View full abstract»

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  • Number theoretic transforms for the calculation of convolutions

    Page(s): 969 - 978
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    In this paper, we present new algorithms for the calculation of convolutions by means of number theoretic transforms over modulo rings. Not only are these algorithms more efficient than currently used methods, but they are also very flexible. Indeed, using special algorithms for short convolutions allows trading computational efficiency for structural simplicity. View full abstract»

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  • The forrest mims circuit scrapbook

    Page(s): 1053
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    First Page of the Article
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  • Adaptive beam forming using a cascade configuration

    Page(s): 940 - 945
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    A new technique for adaptive beam forming based on the Davies cascade array configuration [11] is presented. First of all it is shown that an n-element linear array can be reconfigured as a number of subarrays of two elements each. This two-element array is capable of steering one null in an arbitrary direction while maintaining a desired gain in some other prespecified look direction. To adaptively perform this function at any cascade stage, the inverse of a (2 × 2) correlation matrix of the signal vector at a subarray is computed and substituted in the expression for the constrained optimum weight vector. Similar processing is performed at subsequent stages of the cascade configuration. At each stage of adaptation a noise source contributing maximum power at that stage is nulled out. Simulation studies carried out verify the validity of this approach and demonstrate the fast convergence property of the proposed technique. 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