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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1976

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

    Publication Year: 1976 , Page(s): 0
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1976 , Page(s): c4
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    Freely Available from IEEE
  • Efficient computation of mixed higher partials of digital filter transfer functions

    Publication Year: 1976 , Page(s): 262 - 263
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    A concise formula is derived for the computation of mixed higher partial derivatives of digital filter transfer functions with respect to any combination of multiplier coefficients. If only one multiplier coefficient is involved in the differentiation, our formula specializes to a recent result of Crochiere [1]. View full abstract»

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  • Benchmark papers in acoustics--Physical acoustics

    Publication Year: 1976 , Page(s): 273
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    First Page of the Article
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  • Transformation matrices for bilinear transformation of multivariable polynomials

    Publication Year: 1976 , Page(s): 266 - 267
    Cited by:  Papers (15)
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    The Q-matrix technique of bilinear transformation of a single-variable polynomial is extended to multivariable polynomials. A computer program for the transformation is included in the Appendix. View full abstract»

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  • A digital correlator using delta modulation

    Publication Year: 1976 , Page(s): 238 - 243
    Cited by:  Papers (2)  |  Patents (12)
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    Correlators are powerful instruments for analyzing signals. Recently many digital correlators have been developed. Their main parts are delays, a multiplier, and an integrator, the basic principles being the same in most cases. In this paper, the author proposes a new method (the M.I. Unit) which provides a multiplication and an integration all in one circuit by using delta modulation. Due to using this method, it is suggested that simpler digital correlators can be constructed. View full abstract»

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  • A new principle for fast Fourier transformation

    Publication Year: 1976 , Page(s): 264 - 266
    Cited by:  Papers (39)
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    An alternative form of the fast Fourier transform (FFT) is developed. The new algorithm has the peculiarity that none of the multiplying constants required are complex-most are pure imaginary. The advantages of the new form would, therefore, seem to be most pronounced in systems for which multiplication are most costly. View full abstract»

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  • A pattern recognition approach to voiced-unvoiced-silence classification with applications to speech recognition

    Publication Year: 1976 , Page(s): 201 - 212
    Cited by:  Papers (121)  |  Patents (37)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1192 KB)  

    In speech analysis, the voiced-unvoiced decision is usually performed in conjunction with pitch analysis. The linking of voiced-unvoiced (V-UV) decision to pitch analysis not only results in unnecessary complexity, but makes it difficult to classify short speech segments which are less than a few pitch periods in duration. In this paper, we describe a pattern recognition approach for deciding whether a given segment of a speech signal should be classified as voiced speech, unvoiced speech, or silence, based on measurements made on the signal. In this method, five different measurements are made on the speech segment to be classified. The measured parameters are the zero-crossing rate, the speech energy, the correlation between adjacent speech samples, the first predictor coefficient from a 12-pole linear predictive coding (LPC) analysis, and the energy in the prediction error. The speech segment is assigned to a particular class based on a minimum-distance rule obtained under the assumption that the measured parameters are distributed according to the multidimensional Gaussian probability density function. The means and covariances for the Gaussian distribution are determined from manually classified speech data included in a training set. The method has been found to provide reliable classification with speech segments as short as 10 ms and has been used for both speech analysis-synthesis and recognition applications. A simple nonlinear smoothing algorithm is described to provide a smooth 3-level contour of an utterance for use in speech recognition applications. Quantitative results and several examples illustrating the performance of the method are included in the paper. View full abstract»

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  • Real-time composite signal decomposition

    Publication Year: 1976 , Page(s): 267 - 270
    Cited by:  Papers (2)
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    Real-time decomposition is investigated via cepstrum techniques for a signal composed of a basic wavelet and its echoes. Methods suitable for implementation as a specialized signal processor are outlined, and the real-time system performance is estimated. A system which utilizes the algorithms suggested is capable of wavelet recovery at sampling rates on the order of 105Hz. View full abstract»

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  • Implementation of the digital phase vocoder using the fast Fourier transform

    Publication Year: 1976 , Page(s): 243 - 248
    Cited by:  Papers (62)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (624 KB)  

    This paper discusses a digital formulation of the phase vocoder, an analysis-synthesis system providing a parametric representation of a speech waveform by its short-time Fourier transform. Such a system is of interest both for data-rate reduction and for manipulating basic speech parameters. The system is designed to be an identity system in the absence of any parameter modifications. Computational efficiency is achieved by employing the fast Fourier transform (FFT) algorithm to perform the bulk of the computation in both the analysis and synthesis procedures, thereby making the formulation attractive for implementation on a minicomputer. View full abstract»

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  • Limit cycles in the combinatorial implementation of digital filters

    Publication Year: 1976 , Page(s): 248 - 256
    Cited by:  Papers (20)  |  Patents (2)
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    The existence of limit cycles in combinatorial filters using two's complement truncation arithmetic is investigated in this paper. Exact results for limit cycles of period one and two are presented. Some results for longer period limit cycles are obtained using an effective value linear model. Bounds on these limit cycles are also derived. The accessability of the limit cycles is briefly discussed. View full abstract»

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  • Hardware realization of a Fermat number transform

    Publication Year: 1976 , Page(s): 216 - 225
    Cited by:  Papers (27)  |  Patents (2)
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    The hardware design and implementation of a Fermat number transform (FNT) is described. The arithmetic logic design is treated in detail and a new data representation for integers modulo a Fermat number is derived. In addition, the FNT is compared with the fast Fourier transform (FFT) on the basis of hardware required for a pipeline convolver. View full abstract»

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  • Generalized Paul-Koch basis functions

    Publication Year: 1976 , Page(s): 263 - 264
    Cited by:  Patents (1)
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    The set of piecewise-linear (PL) basis function described by Paul and Koch is generalized to a class of functions providing piecewise approximation with n continuous derivatives and generated by multiple integration of Walsh functions. View full abstract»

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  • The use of second-order information in the approximation of discreate-time linear systems

    Publication Year: 1976 , Page(s): 226 - 238
    Cited by:  Papers (111)
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    It is common practice to partially characterize a filter with a finite portion of its impulse response, with the objective of generating a recursive approximation. This paper discusses the use of mixed first and second information, in the form of a finite portion of the impulse response and autocorrelation sequences. The discussion encompasses a number of techniques and algorithms for this purpose. Two approximation problems are studied: an interpolation problem and a least squares problem. These are shown to be closely related. The linear systems which form the solutions to these problems are shown to be stable. An efficient algorithm for obtaining solutions is given and shown to be closely related to a well-known algorithm of Levinson and the Jury stability test. The close connection between these algorithms suggests that they are fundamental in the numerical analysis of stable discrete-time linear systems. View full abstract»

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  • Tone detection for automatic control of audio tape drives

    Publication Year: 1976 , Page(s): 212 - 215
    Cited by:  Papers (1)
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    This paper describes digital hardware for automatically stopping a cassette recorder upon detection of a prerecorded tone. This hardware is used in conjunction with experiments on computer assisted voice wiring experiments being performed at Western Electric locations [1]. For these experiments a sequence of instructions is automatically recorded on a cassette tape by a computer voice response system. At the end of each instruction, a tone is recorded. The hardware detects this tone and stops the cassette recorder. The operator, after performing the prescribed wiring instruction, manually restarts the cassette recorder for the next instruction. The technique used to detect the tone is a simple digital method comparing the axis crossings of the signal to a fixed threshold. This threshold is determined based on knowledge of the tone frequency, duration, and amplitude. When the signal axis crossings exceeds this threshold during two consecutive 40 ms nonoverlapping intervals the tone is detected and the tape recorder is stopped. The method described is a robust one which is rather insensitive to normal tape recorder problems, e.g., wow and loss of signal level due to battery drainage. The tone detection hardware requires nominal power and is portable. View full abstract»

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  • Residual energy of linear prediction applied to vowel and speaker recognition

    Publication Year: 1976 , Page(s): 270 - 271
    Cited by:  Papers (10)  |  Patents (1)
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    Recognition of steady-state vowels based on the residual energy of linear prediction was ascertained to be useful for a recognition system in which the reference data are taken from the intended speaker. Sharp speaker selectivity based on a threshold criterion suggests that the use of the residual signal energy may also be useful for speaker identification, especially for speaker screening in a large population. View full abstract»

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  • Roundoff noise in state-space digital filtering: A general analysis

    Publication Year: 1976 , Page(s): 256 - 262
    Cited by:  Papers (63)
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    A new noise expression is formulated for the class of fixed-point digital filters described by the state equations, and two methods of its computation are discussed. The effects of possible structure transformation and state-amplitude scalings are then incorporated in this expression, and the results have been analyzed. In particular, it is shown that the output noise and state amplitudes are inversely proportional, and that an elementary transformation is well suited for a step-by-step generation of a low-noise filter. View full abstract»

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  • Acoustics: Historical and philosophical development

    Publication Year: 1976 , Page(s): 272 - 273
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    First Page of the Article
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  • Digital filtering and signal processing

    Publication Year: 1976 , Page(s): 273 - 274
    Cited by:  Papers (1)
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    First Page of the Article
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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