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

9-11 May 1977

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Displaying Results 1 - 25 of 221
  • [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
  • A linguistic approach to speech recognition via a minicomputer net

    Publication Year: 1977, Page(s):819 - 822
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    This paper presents a voice recognition experiment for speech understanding. The approach is based on the fact that a voice recognition system can have a big improvement by exploiting the intrinsic redundancy of the spoken natural language, that is by delaying every decision to the highest available information level. Namely any decision taken at phoneme level (acoustic level) carries the loss of ... View full abstract»

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  • 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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  • A unique hardware and software approach for digital signal processing

    Publication Year: 1977, Page(s):151 - 154
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (136 KB)

    A novel concept for digital signal processing using universal hardware and flexible software is presented. A universal signal processor (USP) offers multiprocessing facilities with multiple function blocks for data and structured data processing. The basic components such as the CROSSBUS, the input/output processor, the data and structured data processor and the instruction processor are build wit... View full abstract»

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  • The effects of peak clipping on speech intelligibility in the presence of a competing message

    Publication Year: 1977, Page(s):216 - 218
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    First Page of the Article
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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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  • ICMF programming language (ICMF/L1)

    Publication Year: 1977, Page(s):179 - 183
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    ICMF/L1 is a poly-level language designed to facilitate microprogramming for the Integer Coefficients Microprogrammed Filter ICMF. It is based on APL syntactic rules and has its inherent facilities, independently from the APL level installed. Compilation, assembly, update and microprogram output aids are provided to enable a widespread use by hardware specialists, in a large variety of structural ... View full abstract»

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  • Use of segmentation and labeling in analysis-synthesis of speech

    Publication Year: 1977, Page(s):28 - 32
    Cited by:  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (232 KB)

    We have been attempting to produce further bandwidth reduction in LPC based analysts-synthesis techniques by used the segmentation and labeling algorithms used in the Harpy and Hearsay-II systems. Preliminary results indicate that a factor of 3 to 5 further reduction in bandwidth might be possible using segmentation and labeling in conjunction with LPC vocoders. View full abstract»

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  • Dynamic speaker adaptation in the Harpy speech recognition system

    Publication Year: 1977, Page(s):788 - 790
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (152 KB)

    The Harpy speech recognition system works optimally when it "knows" the speaker, i,e. when it has learned the speaker dependent characteristics (speaker dependent parameters) of the speaker. There are three methods of learning these parameters. One way is to generate them from a set of training data which covers all the allophones that occur in the task language. A second method is to use "speaker... View full abstract»

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  • Least mean square, finite length, predictive digital filters

    Publication Year: 1977, Page(s):747 - 749
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (59 KB)

    Useful analytic solutions of the discrete Wiener-Hopf (W-H) matrix equation (using the method of undetermined coefficients (UC)) are presented for general rational input spectra. For the specific case of N sinusoids in white noise, the W-H equation is transformed via the UC method into a set of N coupled linear equations which decouple as the filter length becomes large. For all pole input spectra... View full abstract»

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  • A functional description of the Hearsay-II speech understanding system

    Publication Year: 1977, Page(s):799 - 802
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (160 KB)

    A description of the September, 1976, version of the Hearsay-II system is given at the knowledge-source level, indicating the actions of each knowledge-source and their interactions. View full abstract»

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