ICASSP '77. IEEE International Conference on Acoustics, Speech, and Signal Processing

9-11 May 1977

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  • [Front cover and table of contents]

    Publication Year: 1977, Page(s): 0
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    Freely Available from IEEE
  • Inverse covariance-invariant signal processing

    Publication Year: 1977, Page(s):54 - 57
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (109 KB)

    Techniques are outlined for synthesizing continuous-time systems and signals which are covariance-invariant (CI) with corresponding discrete-time systems and signals. The property of covariance-invariance (CI) insures that the covariance sequence characterizing the response of a stable, linear discrete-time system to "white-noise" equals the sampled covariance function that characterizes the sampl... View full abstract»

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  • On the variance of time and frequency averages over modified periodograms

    Publication Year: 1977, Page(s):58 - 62
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (92 KB)

    Expressions of a common form will be given for the variance of spectral estimators which are either time or frequency averages over modified periodograms. A modified periodogram is a generalization of the usual periodogram in which the time sample is premultiplied by a data window. Such estimates include almost all in common use and this discussion should clarify the relationships between practica... View full abstract»

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  • Efficient cubic spline fit

    Publication Year: 1977, Page(s):109 - 111
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (50 KB)

    A new method is presented for fitting a cubic spline (piecewise cubic polynomial with two derivatives continuous at the joints) to a set of n equispaced data. The method exploits the eigenfunction-eigenvalue expansion of a particular sparse matrix, and determines the piecewise polynomial coefficients in2nlog_{2}noperations. View full abstract»

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  • Speed enhancement of digital signal processing software via microprogramming a general purpose minicomputer

    Publication Year: 1977, Page(s): 163
    Cited by:  Papers (4)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (10 KB)

    Summary form only given, as follows. Execution time of digital signal processing software can be substantially reduced using automatically generated code which incorporates precomputed, data independent, control and data access parameters. In this paper, further speed enhancement of such software via microprogramming of computational kernels is discussed and demonstrated. It is shown that, in addi... View full abstract»

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  • Performance evaluation of the canonical signed-digit code (CSC)

    Publication Year: 1977, Page(s):168 - 172
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (141 KB)

    This paper discusses algorithms and performance, in terms of average number of operative symbols for shift right and shift left Canonical Signed-digit Codes [CSC] which can be applied to generation of multiplications on sequential type microprocessing machines. The reference algorithms are based on application of codes which minimize the number of operative symbols required to perform binary multi... View full abstract»

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  • The application of a functional perceptual model of speech to variable-rate LPC systems

    Publication Year: 1977, Page(s):219 - 222
    Cited by:  Papers (5)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (137 KB)

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  • The creation of psycho-acoustic traditions for synthesized music

    Publication Year: 1977, Page(s):231 - 234
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

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  • Microprocessor-based signal processing for the perceptually deaf

    Publication Year: 1977, Page(s):255 - 256
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (42 KB)

    A two-band compression system is being designed to improve the intelligibility of speech in noise for persons suffering from sensori-neural hearing impairment, The compressor is microprocessor based allowing variable thresholds, compression ratios above threshold, and expansion below threshold. The compressor will have significantly lower cost than commercial units based completely on analog devic... View full abstract»

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  • A variable band coding scheme for speech encoding at 4.8 kb/s

    Publication Year: 1977, Page(s):444 - 447
    Cited by:  Papers (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (141 KB)

    First Page of the Article
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  • A microprogrammable signal processor

    Publication Year: 1977, Page(s):494 - 497
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (110 KB)

    Digital signal processing has cone full circle from software algorithms performed on general purpose computers through special purpose, dedicated, real time processors to the programmable signal/array processor. The Programmable Number Thresher (PNTJT) is an addition to the list and its organization and features is described. The PNUT is intended primarily for the linear front end computations req... View full abstract»

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  • A general realization scheme for 2-D digital transfer functions

    Publication Year: 1977, Page(s):539 - 542
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (83 KB)

    Because of the inherent impossibility of factorizing arbitrary 2-D polynomials, all realization techniques for the general 2-D digital transfer function presented so far deal with the realization of such functions as a whole. This paper presents a new realization scheme which is based on realizing multi-input, multi-output subnetworks of multipliers and adders interconnecting 2-D digital subnetwor... View full abstract»

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  • A modular audio response system for computer output

    Publication Year: 1977, Page(s):579 - 582
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (89 KB)

    We describe a flexible, modular system for representation of a text-to-speech system which provides for ease of understanding, facile development, a variety of end-use systems, exportability, and ready implementation into novel technology to meet real-time needs. This system thus serves as a research base, a framework for experimentation, and the seat of structural formalisms which can imply speci... View full abstract»

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  • On two or more dimensional optimum quantizers

    Publication Year: 1977, Page(s):640 - 643
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (94 KB)

    It is hard to compute the performance of an N-level K-dimensional optimum quantizerhat{Q}_{N}directly. In this paper the performance ofhat{Q}_{N}is studied more closely from the performance of Q'N, a class of stationary quantizers. An analytical derivation of the algorithm for generating Q'Nquantizers and a computer experimental study on the performance ... View full abstract»

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  • [Back cover]

    Publication Year: 1977, Page(s): c4
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    Freely Available from IEEE
  • Automatic speaker recognition for use over communication channels

    Publication Year: 1977, Page(s):764 - 767
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    A system is described for the automatic comparison of speakers given short samples of their speech. The method does not depend on knowing what is being said, and is to a large extent independent of the degradations likely to be suffered by the speech during transmission. A small computer has been used to generate statistics on fundamental frequency and spectral shape information produced by a real... View full abstract»

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  • Word verification in the Hearsay II speech understanding system

    Publication Year: 1977, Page(s):795 - 798
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (184 KB)

    A key problem for speech understanding systems is the verification of word hypotheses generated by various knowledge sources in the system. In this paper we will discuss the general problem of word verification in speech understanding systems. A description of our matching algorithm for word verification which is based on that used in the HARPY system, a general connected speech recognition system... View full abstract»

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  • Grapheme-to-phoneme transcription for French

    Publication Year: 1977, Page(s):575 - 578
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (192 KB)

    A programming language designed for the simple description of letter-to-sound rules is described. Its use in an application of grapheme-to-phoneme transformation rules system for French is studied, and some results are given. The rules system is based on the application of a partially ordered set of phonological rules. The left-hand side of each rule indicates the graphemes involved by the rule. T... View full abstract»

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  • Speaker-identification with real time formant extraction

    Publication Year: 1977, Page(s):761 - 763
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (72 KB)

    The Westinghouse speaker identification system uses frequency-locked loops for the extraction of the first and second formant of voiced phonemes. The analog control voltage of the loop circuits (digitized at 8 bits/sample) represents characteristic formant features at a bit rate that is low enough for subsequent real time processing by an Intel 8080 microprocessor. The system determines for each f... View full abstract»

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  • Model errors in Kalman filter and simple correction techniques

    Publication Year: 1977, Page(s):708 - 712
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (144 KB)

    As techniques cf divergence reduction in Kalman filtering, the adaptive Jazwinski's plant noise covariance matching is compared with the non-adaptive Schmidt's gain matrix scaling. In the simple case studied, the two give comparable performances and both are effective. However, Jazwinski's algorithm is more suitable than the non-adaptive Schmidt's for the purpose of real time processing. View full abstract»

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  • Autocorrelation prediction

    Publication Year: 1977, Page(s):5 - 9
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (152 KB)

    Autocorrelation Prediction (AP) has been shown to be an effective technique for Pole-Zero modeling. This paper develops a new linear method for identifying a stable Pole-Zero model whose spectrum matches the envelope of a given spectrum. All the operations are performed in Autocorrelation domain, using no Fourier transformations. At one extreme, Autocorrelation Prediction reduces to a linear metho... View full abstract»

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  • High resolution sonar simulation techniques

    Publication Year: 1977, Page(s):840 - 844
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (432 KB)

    A mathematical model developed for a computer simulation of both side-looking and forward-looking high resolution sonar systems is described. The purpose of the simulation is to permit systematic examination of the effects of first-order parameter variations on sonar image quality and target detection capability. The model permits specification of the major parameters such as height above bottom, ... View full abstract»

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  • Design of instantaneously companded delta modulators with M-bit memory

    Publication Year: 1977, Page(s):196 - 199
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    In instantaneously companded delta modulators, the step-size is multiplied, at each sampling time, by an expansion-- compression factor depending on the sequence of the M most recent output bits. Some authors determined the values of the multipliers by optimizing the S/N ratio on a typical signal. This procedure can be effectively used only when a small number of bits is observed, but it becomes v... View full abstract»

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  • Rule synthesis of speech from dyadic units

    Publication Year: 1977, Page(s):568 - 570
    Cited by:  Papers (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (47 KB)

    Peterson, Wang, and Sivertsen[1] suggested the use of the units called "dyads" as the basic unit for speech synthesis. This paper describes an approach to speech synthesis by rule which uses a unit that is similar but smaller than the dyad as defined by Peterson et al. This new unit specifies only the transition between the two phones of the dyad, while the "steady state" portions are obtained by ... View full abstract»

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  • Speaker verification using composite reference

    Publication Year: 1977, Page(s):758 - 760
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    This paper examines the effectiveness of parametric representation of speech derived from the linear prediction model using the new concept of composite reference for speaker verification. In the proposed verification system the combined information of the identity of several speakers is given. That is, the existence of an overall reference contour is assumed which contains the information corresp... View full abstract»

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