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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 6 • Date December 1978

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Displaying Results 1 - 21 of 21
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Author's reply

    Page(s): 604
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    First Page of the Article
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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • A pitch estimation algorithm for speech and music

    Page(s): 597 - 604
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    We present an efficient pitch estimation algorithm which operates in the time domain by comparing features extracted from the signal being analyzed. The algorithm uses both primary features (extracted directly from the signal) and secondary features (derived by combining several primary features to reduce the effect of variations in the signal envelope). We relate the algorithm both analytically and quantitatively to autocorrelation analysis. The algorithm is shown to function successfully for both speech and music over a six octave pitch range, and to be appropriate for operation in real time. View full abstract»

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  • Memory and time improvements in a dynamic programming algorithm for matching speech patterns

    Page(s): 583 - 586
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    Recently, dynamic programming has been found useful for performing nonlinear time warping in speech recognition. Although considerably faster than exhaustive search procedures, the dynamic programming procedure nevertheless requires substantial computation. Also, considerable storage is normally required for reference prototypes necessary in the matching process. This paper is concerned with methods for reducing this storage and computation. Empirical results indicate that one method yields 50 to 60 percent storage reduction and a factor of 4 to 6 in computational savings relative to conventional dynamic programming procedures without degradation in recognition accuracy. View full abstract»

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  • A fast computation of complex convolution using a hybrid transform

    Page(s): 566 - 570
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    In this paper it is shown that the cyclic convolution of complex values can be performed by a hybrid transform. This transform is a combination of a Winograd algorithm and a fast complex integer transform developed previously by the authors. This new hybrid algorithm requires fewer multiplications than any previously known algorithm. View full abstract»

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  • Cal-sal Walsh-Hadamard transform

    Page(s): 605 - 607
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    Walsh-Hadamard matrices are rearranged such that the first half of the rows represents cal functions in increasing order of sequency whereas the second half represents sal functions in decreasing order of sequency. The transform based on this rearrangement is called the Cal-Sal Walsh-Hadamard transform or (WHT)cs. General expressions for developing the elements of these matrices are developed. These matrices are decomposed into sparse matrix factors which lead directly to the fast algorithms similar to those for other forms of the WHT. The (WHT)csis useful in mapping an even or odd sequence since, in this case, at least one-half of the transform components will be zero. View full abstract»

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  • A new realization method for 2-D digital transfer functions

    Page(s): 544 - 550
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    This paper presents a general method for realizing an arbitrary 2-D digital transfer function. The realizations are obtained in terms of two types of delay blocks, adders and multipliers. The techniques presented here can be used to obtain infinitely many realizations of the same transfer function. View full abstract»

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  • Acoustic transfer function of the ocean for a motional source

    Page(s): 493 - 501
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    The motional-source transfer characteristics of long-range acoustic propagations are derived based on the discrete multipath propagation model predicted by geometrical acoustics. Parameters are those associated with the characteristic solutions of the eikonal equation. A modified transfer function is formulated which relates the received signal to a particularly useful transformation of the source signal. The modified transfer function shows that a remotely received signal consists of the weighted superposition of a number of modified source signals both slightly compressed (or expanded) and translated in time. The time variable compression factor is a stochastic process which reflects the fluctuations inherent in both the medium and the source motion. Application of the model transfer function to a specific ocean profile exemplifies the ocean filter characteristics and the complex nature of the received signal from a CW source in motion. It is concluded that the model transfer function will be a useful tool for signal processing applications in underwater acoustics. View full abstract»

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  • The design of optimal multirate bandpass and bandstop filters

    Page(s): 534 - 543
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    Narrow-band bandpass and bandstop filters are inherently of high order and require a large computation rate. A multirate filter using decimators and interpolators can be designed to have bandpass or bandstop characteristics, often with a much smaller computation rate. This paper develops rules for designing such a filter by placing constraints on the filter approximation error and the aliasing error. The question of admissible decimation factors is investigated in detail. A method to minimize the computation rate is described. Several examples are presented. View full abstract»

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  • Short-time spectral analysis with the conventional and sliding CZT

    Page(s): 561 - 566
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    Two sequential short-time spectral analysis techniques, amenable to nonrecursive filter implementation, are the conventional chirp-z-transform (CZT) realization of the discrete Fourier transform and the sliding CZT realization of the discrete sliding Fourier transform. This paper presents a comparative study of frame rate limitations, windowing, time and frequency resolution, spectral correlation, complexity, and inverse structures for these methods, with particular emphasis on the recently proposed sliding transform. The sliding transform and its CZT realization are viewed as skewed output samples of a filter bank, an approach which aids in understanding the relationship between the conventional and sliding schemes. Numerous forward and inverse CZT formulations are presented to improve resolution, frame rates, and compactness. View full abstract»

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  • Cubic splines for image interpolation and digital filtering

    Page(s): 508 - 517
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    This paper presents the use of B-splines as a tool in various digital signal processing applications. The theory of B-splines is briefly reviewed, followed by discussions on B-spline interpolation and B-spline filtering. Computer implementation using both an efficient software viewpoint and a hardware method are discussed. Finally, experimental results are presented for illustrative purposes in two-dimensional image format. Applications to image and signal processing include interpolation, smoothing, filtering, enlargement, and reduction. View full abstract»

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  • Image restoration of space-variant blurs by sectioned methods

    Page(s): 608 - 609
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    Previous work on sectional methods in image processing is extended to the processing of degradations produced by space-variant point spread functions. View full abstract»

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  • A posteriori estimation of vocal tract length

    Page(s): 571 - 574
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    Various estimators of vocal tract length, both iterative and noniterative, have been proposed and used for various speech processing and speech pathological purposes. The results of the accuracy of these methods are compared with an a posteriori estimator and others. The a posteriori estimator is shown to provide nearly the same accuracy as the more complex and computation-time costly iterative algorithms. View full abstract»

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  • Stability of general two-dimensional recursive digital filters

    Page(s): 550 - 560
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    Two-dimensional recursive filters are defined from a different point of view. A general stability preserving mapping theorem is presented which allows most recursive filters of a particular type to be mapped into any other type of recursive filter. In particular, any type of filter can be mapped into a first-quadrant filter. This mapping is used to prove a number of general stability theorems. Among these is a theorem which relates the stability of any digital filter to its two-dimensional phase function. Furthermore, other stability theorems which are valid for any type of recursive filter are presented. Finally, a number of practical stability tests are developed including one which requires the testing of only several one-dimensional polynomial root distributions with respect to the unit circle. View full abstract»

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  • Spectral smoothing technique in PARCOR speech analysis-synthesis

    Page(s): 587 - 596
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    In linear predictive analysis of speech, voice periodicity influences formant frequency and bandwidth estimation accuracy. One of the most serious errors in estimating formant parameters is bandwidth underestimation that causes a quality difference between synthetic and natural speech. In this paper, the spectral smoothing technique (SST), using a lag window, is introduced in an autocorrelation method of the linear predictive analysis. In order to assess the effectiveness of the SST to reduce estimation errors, experimental comparisons of the usual autocorrelation method and the SST are presented. Spectral sensitivity analysis is also presented to evaluate the SST from the viewpoint of parameter quantization properties. SST features are summarized as follows: 1) Bandwidth underestimation elimination. 2) Spectral sensitivity reduction of PARCOR coefficients. 3) Simplicity in hardware implementation. View full abstract»

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  • Calculation of narrow-band spectra by direct decimation

    Page(s): 529 - 534
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    The calculation of the spectrum of a narrow-band signal which is embedded in a broad-band sequence usually requites substantial computation and storage if executed by performing an FFT or DFT's directly on the broad-band sequence. In this paper a new scheme using direct decimation is proposed which computes the narrow-band spectrum with good resolution while requiring only modest computation and storage. The performance of the proposed scheme is analyzed. Examples are presented which demonstrate the efficiency of this scheme when compared with the FFT, DFT, zoom transform, and complex modulation scheme. View full abstract»

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  • Considerations in dynamic time warping algorithms for discrete word recognition

    Page(s): 575 - 582
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    The technique of dynamic time warping for time registration of a reference and test utterance has found widespread use in the areas of speaker verification and discrete word recognition. As originally proposed, the algorithm placed strong constraints on the possible set of dynamic paths-namely it was assumed that the initial and final frames of both the test and reference utterances were in exact time synchrony. Because of inherent practical difficulties with satisfying the assumptions under which the above constraints are valid, we have considered some modifications to the dynamic time warping algorithm. In particular, an algorithm in which an uncertainty exists in the registration both for initial and final frames was studied. Another modification constrains the dynamic path to follow (within a given range) the path which is locally optimum at each frame. This modification tends to work well when the location of the final frame of the test utterance is significantly in error due to breath noise, etc. To test the different time warping algorithms a set of ten isolated words spoken by 100 speakers was used. Probability density functions of the distances from each of the 100 versions of a word to a reference version of the word were estimated for each of three dynamic warping algorithms. From these data, it is shown that, based on a set of assumptions about the distributions of the distances, the warping algorithm that minimizes the overall probability of making a word error is the modified time warping algorithm with unconstrained endpoints. A discussion of this key result along with some ideas on where the other modifications would be most useful is included. View full abstract»

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  • Linear prediction in cascade form

    Page(s): 518 - 528
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    The autocorrelation and covariance methods of linear prediction are formulated in terms of an inverse digital filter in cascade form, rather than the traditional direct form, to allow pole locations in the system model to be readily estimated and constrained. Iterative solution of the corresponding nonlinear normal equations is described. Applications to speech analysis and the compensation of biomedical signals are briefly discussed. View full abstract»

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  • Some novel windows and a concise tutorial comparison of window families

    Page(s): 501 - 507
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    Some novel windows are introduced. A comparison of these and the well-known windows in terms of their frequency domain properties is given. It is concluded that Kaiser, modified Kaiser, Tukey, and three-coefficient window families appear to be the best of the known windows of 6, 12, and 18 dB/oct decay rates. 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