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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 5 • Date October 1976

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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
  • Comments on "A Fast Algorithm for the estimation of autocorrelation functions"

    Page(s): 432 - 434
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    A recently published algorithm for the estimation of arithmetic autocorrelation functions may be further refined. The premultiplications may be converted into simple additions which are efficiently computed within the algorithm. Savings in storage and computational effort are realized. View full abstract»

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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • Finite wordlength effects in the ROM digital filter

    Page(s): 436 - 437
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    The finite wordlength effects in the read-only memory (ROM) digital filter are studied. By quantizing the coefficients before the programming of the ROM, the two problems of coefficient accuracy and roundoff error are decoupled. This leads to a filter which is asymptotically linear. View full abstract»

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  • Comparison of the cosine and Fourier transforms of Markov-1 signals

    Page(s): 428 - 429
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    This correspondence compares the effectiveness of the discrete cosine and Fourier transforms in decorrelating sampled signals with Markov-1 statistics. It is shown that the discrete cosine transform (DCT) offers a higher (or equal) effectiveness than the discrete Fourier transform (DFT) for all values of the correlation coefficient. The mean residual correlation is shown to vanish as the inverse square root of the sample size. View full abstract»

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  • Recursive digital filter synthesis via gradient based algorithms

    Page(s): 349 - 355
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    The three gradient-based algorithms of 1) steepest descent, 2) Newton's method, and 3) the linearization algorithm are applied to the problem of synthesizing linear recursive filters in the time domain. It is shown that each of these algorithms requires knowledge of the associated recursive filter's first-order sensitivity vectors, and, in the case of the Newton method, second-order sensitivity vectors as well. Systematic procedures for generating these sensitivity vectors by computing the response of a companion filter structure are then presented. Using the ideal low-pass filter as a design objective, it is then demonstrated that the linearization algorithm is particularly well suited for recursive filter design. On the other hand, the steepest descent and Newton methods are found to work rather poorly for this class of problems. Reasons for these empirical observed results are postulated. View full abstract»

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  • A simplified binary arithmetic for the Fermat number transform

    Page(s): 356 - 359
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    A binary arithmetic that permits the exact computation of the Fermat number transform (FNT) is described. This technique involves arithmetic in a binary code corresponding to the simplest one of a set of code translations from the normal binary representation of each integer in the ring of integers modulo a Fermat number Ft= 2b+ 1, b = 2t. The resulting FNT binary arithmetic operations are of the complexity of 1's complement arithmetic as in the case of a previously proposed technique which corresponds to another one of the set of code translations. The general multiplication of two integers modulo Ftrequired in the computation of FNT convolution is discussed. View full abstract»

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  • Structure of a phonological rule component for a synthesis-by-rule program

    Page(s): 391 - 398
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    A synthesis-by-rule program can be thought of as a functional model of human sentence production. The phonological component of this model accepts as input a linear string of symbols that have been produced by the semantics component, syntactic component, and lexical component of a grammar of English. This abstract representation for an utterance is transformed by the phonological component into a narrow phonetic transcription and a specification of stress levels, segmental durations, and aspects of the fundamental frequency contour. View full abstract»

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  • Digital signal analysis

    Page(s): 440
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    First Page of the Article
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  • Design of minimum noise digital filters subject to inequality constraints using quadratic programming

    Page(s): 434 - 436
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    A numerical method is presented for designing digital filters. The method allows one to minimize the mean-square error or noise power over some intervals of frequency, while simultaneously constraining the maximum error in other intervals of frequency. Thus, for example, one can minimize noise power from a stopband of frequencies while constraining signal fidelity in a passband of frequencies by limiting the maximum passband deviation. View full abstract»

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  • Separator functions for homomorphic filtering

    Page(s): 359 - 364
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    A procedure is presented to test an arbitrary function of two variables N(x, y) to see if a separator function exists such that S[N(x, y)] = A (x) + B(y). If so, the procedure evaluates S[.],A,B. Necessary conditions for the existence of S[.] are established and some limitations are given in a theorem on sufficient conditions. Examples and a canonic representation of the test and procedure are also presented. View full abstract»

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  • Maximum likelihood pitch estimation

    Page(s): 418 - 423
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    A method for estimating the pitch period of voiced speech sounds is developed based on a maximum likelihood (ML) formulation. It is capable of resolution finer than one sampling period and is shown to perform better in the presence of noise than the cepstrum method. View full abstract»

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  • Automatic detection and description of syllabic features in continuous speech

    Page(s): 365 - 379
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    The details of the implementation of a syntax-controlled acoustic encoder of a speech understanding system (SUS) are presented. Finite-state automata operating on artificial descriptions of suprasegmentals and global spectral features isolate syllables in continuous speech. Then a combinational algorithm tracks the formants for the voiced intervals of each syllable, and other algorithms provide a complete structural description of spectral and prosodic features for a spoken sentence. Such a description consists of a string of symbols and numerical attributes and is a representation of speech in terms of perceptually significant primitive forms. It contains all the information required to reconstruct the analyzed sentence with a formant synthesizer; it can be used directly either for emitting or verifying hypotheses at the lexical level of an SUS and for automatically learning phonetic features by grammatical inference. View full abstract»

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  • A correct proof of Huang's theorem on stability

    Page(s): 425 - 426
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    A correct proof of Huang's theorem on the stability of two-dimensional causal recursive digital filters is developed using a maximum modulus theorem for algebraic functions. View full abstract»

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  • A group of linear spectral transformation for two-dimensional digital filters

    Page(s): 424 - 425
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    There are eight linear transformations of the spectral plane which map the frequency axes onto themselves. These transformations can be used to change the pass and stop regions of a two-dimensional digital filter. A stable realization is assured by transforming the data rather than the system transfer function. View full abstract»

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  • An alternate proof of Huang's stability theorem

    Page(s): 426 - 427
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    In this correspondence a complete proof of Huang's stability theorem for two-dimensional discrete filters is presented. The proof is based on certain properties of algebraic functions. It appears that Huang's original proof is not complete and it is hoped that this proof will fill the gap. View full abstract»

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  • A comparative performance study of several pitch detection algorithms

    Page(s): 399 - 418
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    A comparative performance study of seven pitch detection algorithms was conducted. A speech data base, consisting of eight utterances spoken by three males, three females, and one child was constructed. Telephone, close talking microphone, and wideband recordings were made of each of the utterances. For each of the utterances in the data base; a "standard" pitch contour was semiautomatically measured using a highly sophisticated interactive pitch detection program. The "standard" pitch contour was then compared with the pitch contour that was obtained from each of the seven programmed pitch detectors. The algorithms used in this study were 1) a center clipping, infinite-peak clipping, modified autocorrelation method (AUTOC), 2) the cepstral method (CEP), 3) the simplified inverse filtering technique (SIFT) method, 4) the parallel processing time-domain method (PPROC), 5) the data reduction method (DARD), 6) a spectral flattening linear predictive coding (LPC) method, and 7) the average magnitude difference function (AMDF) method. A set of measurements was made on the pitch contours to quantify the various types of errors which occur in each of the above methods. Included among the error measurements were the average and standard deviation of the error in pitch period during voiced regions, the number of gross errors in the pitch period, and the average number of voiced-unvoiced classification errors. For each of the error measurements, the individual pitch detectors could be rank ordered as a measure of their relative performance as a function of recording condition, and pitch range of the various speakers. Performance scores are presented for each of the seven pitch detectors based on each of the categories of error. View full abstract»

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  • Distance measures for speech processing

    Page(s): 380 - 391
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    The properties and interrelationships among four measures of distance in speech processing are theoretically and experimentally discussed. The root mean square (rms) log spectral distance, cepstral distance, likelihood ratio (minimum residual principle or delta coding (DELCO) algorithm), and a cosh measure (based upon two nonsymmetrical likelihood ratios) are considered. It is shown that the cepstral measure bounds the rms log spectral measure from below, while the cosh measure bounds it from above. A simple nonlinear transformation of the likelihood ratio is shown to be highly correlated with the rms log spectral measure over expected ranges. Relationships between distance measure values and perception are also considered. The likelihood ratio, cepstral measure, and cosh measure are easily evaluated recursively from linear prediction filter coefficients, and each has a meaningful and interrelated frequency domain interpretation. Fortran programs are presented for computing the recursively evaluated distance measures. View full abstract»

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  • Two preliminary studies of the intelligibility of predictor-coefficient and formant-coded speech

    Page(s): 429 - 432
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    Two preliminary studies comparing the intelligibilities of predictor-coefficient versus formant-frequency-coded speech and the intelligibilities of predictor-coefficient-coded speech using different numbers of coefficients are reported. 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