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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date August 1975

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

    Publication Year: 1975 , Page(s): 0
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    Freely Available from IEEE
  • Comments on "An lpdesign techinque for two-dimensional digital recursive filters"

    Publication Year: 1975 , Page(s): 389 - 390
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (263 KB)  

    It is shown that examples can be found for which the lpdesign algorithm in two dimensions will attempt to converge to an unstable solution. View full abstract»

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  • Digital signal processing

    Publication Year: 1975 , Page(s): 392 - 394
    Cited by:  Papers (9)
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    First Page of the Article
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  • [Back cover]

    Publication Year: 1975 , Page(s): c4
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    Freely Available from IEEE
  • On multistage finite impulse response (FIR)filters with decimation

    Publication Year: 1975 , Page(s): 353 - 357
    Cited by:  Papers (31)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    Nonrecursive, finite impulse response (FIR) digital filters are known to require a reduced rate of arithmetic operations relative to recursive infinite impulse response filters (IIR) in many applications where the sample rate is to be diminished. FIR symmetry and the absence of state variables contribute to the efficiency. A significant further economy in both arithmetic and memory space can be achieved by realizing the desired filter as a cascade of FIR stages, with interstage decimation of sample rate. In this paper, an approach to optimally allocate (i.e., factor) the composite decimation among filter stages (and hence determine the filter state characteristics) is described. View full abstract»

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  • On the efficient design of bandpass digial filter structures

    Publication Year: 1975 , Page(s): 380 - 381
    Cited by:  Papers (4)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    Several advantages are observed in designing recursive band-pass digital filter structures according to a bandpass transformation that is incorporated directly in the structure. In particular, it is shown that such structures can be implemented with fewer multiplies than in conventional designs and in some cases, with shorter coefficient word lengths. View full abstract»

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  • Direct determination of vocal tract wall impedance

    Publication Year: 1975 , Page(s): 370 - 373
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (472 KB)  

    The side wall of the human vocal tract yields and vibrates in response to contained sound pressure. This motion influences the acoustic resonances of the vocal tract and the vibratory behavior of the vocal cords. Accurate computer modeling of the cord/tract system therefore requires knowledge of the mechanical impedance of the yielding walls. Heretofore this factor has only been estimated indirectly from sound wave measurements. We describe here a technique for direct measurement of the wall impedance. We also give a rationale for interpreting the low-frequency impedance in terms of a simple mass, compliance, viscous-loss combination. Finally, we give mechanical parameters for the vocal-tract wall which we presently use in computer simulations. View full abstract»

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  • The variable law detector

    Publication Year: 1975 , Page(s): 357 - 362
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    Digital processing of signals, especially fast Fourier transform (FFT) hardware, has reemphasized the role of energy detectors in spectrum analysis. The cost associated with the size of words in special-purpose computers causes a search for methods of representing detected data effectively with a modest number of bits for either further processing or display. A method of accomplishing a reduction in word size is proposed here as a variable law detector. The variable law detector algorithm is described in detail and the detectability performance as a function of the detector parameters is given. View full abstract»

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  • Minimax optimization of recursive digital filters using recent minimax results

    Publication Year: 1975 , Page(s): 333 - 345
    Cited by:  Papers (15)
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    The purpose of this paper is to use some recent work by the author on nonlinear minimax optimization to derive an efficient algorithm for the minimax optimization problem. This is followed by the application of the algorithm to the problem of choosing the coefficients of a recursive digital filter to meet arbitrary design specifications on the magnitude or the group delay characteristics. View full abstract»

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  • A mathematical formulation and comparison of zero-crossing analysis techniques which have been applied to automatioc speech recognition

    Publication Year: 1975 , Page(s): 373 - 380
    Cited by:  Papers (8)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (912 KB)  

    Zero-crossing analysis techniques have long been applied to speech analysis, to automatic speech recognition, and to many other signal-processing and pattern-recognition tasks. In this paper, a mathematical formulation for each of several zero-crossing feature extraction techniques is derived and related (where possible) to each of the other zero-crossing methods. Based upon this mathematical formulation, a physical interpretation of each analysis technique is effected, as is a discussion of the properties of each method. It is shown that four of these methods are a description of a short-time waveform in which essentially the same information is preserved. Each turns out to be a particular normalization of a count of zero-crossing intervals method. The effects of the various forms of normalization are discussed. A fifth method is shown to be a different type of measure; one which preserves information concerning the duration of zero-crossing intervals rather than their absolute number. Although reference is made as to how each of the zero-crossing methods has been applied to automatic speech recognition, an attempt is made to enumerate general characteristics of each of the techniques so as to make the mathematical analysis generally applicable. View full abstract»

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  • A fast-correlation algorithm for swept-frequency seismic data

    Publication Year: 1975 , Page(s): 346 - 352
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    The controlled-source seismic reflection method using a so-called "swept-frequency" requires the application of a digital matched filter in the data preprocessing. The filter is of the swept-frequency form and it is normally long. With a 125 Hz aliasing frequency, 4 ms-sampled filters can commonly consists of up to 5000 samples. For the professor unfortunate enough to not possess a hard-wire correlator in his computer, the application of the matched filter is almost prohibitively time consuming and costly. The subject algorithm ignores any end tapering of the filter, and it concentrates only on the positions of the filter peaks and troughs. The computational savings to be realized can be around an order of magnitude, with essentially no loss in accuracy resulting. View full abstract»

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  • Digital speech analysis using sequential estimation techniques

    Publication Year: 1975 , Page(s): 362 - 369
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (896 KB)  

    Two new digital speech analysis methods for sequentially identifying the coefficients of the linear prediction model are presented; the methods are based on the stochastic approximation and Kalman filter sequential estimation algorithms. Speech synthesized using the predictor coefficients identified by the Kalman filter algorithm is highly intelligible and comparable in quality to that obtained by the autocorrelation and covariance methods. Speech synthesized using predictor coefficients identified by the stochastic approximation algorithm is also highly intelligible but of lower quality. The analysis and synthesis procedures use hand-picked pitch and voiced/unvoiced information, and the predictor coefficients are converted to PARCOR coefficients for checking stability and transmission to the receiver. The sequential techniques are shown to be real-time feasible and closely related to the more familiar autocorrelation and covariance methods for speech analysis. View full abstract»

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  • On the relationship between digital Hilbert transformers and certain low-pass filters

    Publication Year: 1975 , Page(s): 381 - 383
    Cited by:  Papers (12)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB)  

    Designs of symmetric Hilbert transformers are shown to be easily derived from corresponding designs for symmetric half-band low-pass filters, and vice versa. The latter filter type is particularly useful for interpolation or smoothing in sample-rate alteration. View full abstract»

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  • Theory and application of digital signal processing

    Publication Year: 1975 , Page(s): 394 - 395
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    First Page of the Article
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  • Speech amplitude and zero crossings for automated identification of human speakers

    Publication Year: 1975 , Page(s): 390 - 392
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    Speech amplitude, low-pass (below 1 kHz) and high-pass (above 1 kHz) zero-crossing rates were represented as weighted sums of orthonormal functions. The weighting coefficients were used to identify unknown speakers. For a ten speaker population recognition rates as high as 96.6 percent were obtained using the same data for training and testing. View full abstract»

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  • A note on implementation of digital filters

    Publication Year: 1975 , Page(s): 387 - 389
    Cited by:  Papers (2)  |  Patents (2)
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    A recently proposed method to implement digital filters using ROM can also be employed to implement multiplication by a constant. This note discusses alternate implementations for digital filter sections substituting memory for logic and points out the possible advantage of doing so. View full abstract»

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  • Digital generation of equal temperament

    Publication Year: 1975 , Page(s): 329 - 333
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    The generation of pitches of an equally tempered musical scale through nonbinary frequency division is explored, including an analysis of the resulting frequency errors. Useful approximations to 21/12, in addition to the well known 196/185 ratio, are sought. A criterion for allowable errors is developed and applied to ratios involving integers up to 10 000. Based on an optimum tuning algorithm, it is determined that there are other possible ratios that could be implemented more simply and at a lower cost than the 196/185 ratio. View full abstract»

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  • A note on real-time linear prediction of speech waveforms

    Publication Year: 1975 , Page(s): 386 - 387
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (312 KB)  

    A well-known, real-time computational algorithm for use in adaptive linear prediction of speech waveforms is discussed and is related to known research in other fields. View full abstract»

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  • An analytical approach to the design of nonrecursive digital filters

    Publication Year: 1975 , Page(s): 383 - 385
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    This correspondence describes an analytical approach to the design of nonrecursive digital filters where Chebyshev functions are used to obtain filter transfer characteristics with equiripple behavior in both the passband and stopband. The resulting filters are of length 2npns+ 1 compared with optimal filters of length 2(np+ ns) - 1. 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