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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)

    First Page of the Article
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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)

    First Page of the Article
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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
  • Microprocessor realization of a linear predictive vocoder.

    Publication Year: 1977, Page(s):421 - 424
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (136 KB)

    A microprocessor realization for a linear predictive vocoder is presented. The goal was a low power, low cost, compact special purpose realization of a narrowband speech terminal. The resultant design is a general purpose two bus structure running at a 150 ns cycle time using as the basic signal processing element four of the AMD 2901 CPE chips. This basic structure is augmented by a four cycle mu... View full abstract»

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  • Improving the performance of LPC-CVSD tandem connections by phase modification

    Publication Year: 1977, Page(s):433 - 436
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (144 KB)

    This paper describes an approach to modifying the waveshape of LPC synthetic speech, which involves a combination of spectral flattening prior to estimation of the LPC parameters and non-minimum phase de-emphasis following conventional LPC synthesis. Results indicate that significant modifications of the synthetic waveform can be effected. View full abstract»

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  • A real-time phrase recognizer using telephone bandwidth speech

    Publication Year: 1977, Page(s):464 - 467
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (144 KB)

    A system for the recognition of spoken phrases is described. The input speech is filtered to 3.3 KHz (approximating telephone bandwidth), and parameters are computed while the utterance is spoken. The remaining time necessary to produce a result is generally less than the duration of the input utterance. The recognizer assumes that the input utterance contains one of a known set of key phrases; th... 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 real-time floating point variable frame rate LPC vocoder

    Publication Year: 1977, Page(s):429 - 432
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (136 KB)

    As part of a Defense Advanced Research Projects Agency program investigating voice transmission over a packet-switched computer network, a real-time variable frame rate LPC vocoder has been implemented at USC-ISI. The vocoder is implemented on a Floating Point Systems AP-120B array processor. A likelihood ratio threshold is used to control when frames of data are transmitted. The average data rate... View full abstract»

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  • Evaluation of some distance measures for speaker independent isolated word recognition

    Publication Year: 1977, Page(s):460 - 463
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    In this paper a number of distance measures for speaker independent recognition of isolated words are proposed. These distance measures use auto-correlation coefficients alone or autocorrelation and linear predictor coefficients as feature paramaters of the speech samples. One measure used is the measure in discrete l2space of linear functionals. Some of the other distance measures used... View full abstract»

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  • An algorithm for the two dimensional FFT

    Publication Year: 1977, Page(s):552 - 555
    Cited by:  Papers (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (73 KB)

    Conventional two dimensional fast Fourier transforms become very slow if the size of the matrix becomes too large to be contained in memory. This is due to the transposition of the matrix that is required. This new algorithm is designed to remove the requirement for transposition, thereby, greatly increasing the speed of the process. This algorithm is extremely valuable on small disc based compute... 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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  • A pipeline fast Walsh-Fourier transform

    Publication Year: 1977, Page(s):515 - 518
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (63 KB)

    The structure for hardware realisation of Fast Walsh-Fourier Transform (FWFT) is presented in this paper. The development of the approach is based on Shanks' algorithm for FWFT and use is made in the realisation of serial storage and simple arithmetic. Two alternative structures are proposed one based on PCM encoding and serial two's complement arithmetic, whilst the other is based on delta-sigma ... View full abstract»

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  • Comparison of natural speech with glottal area waveform synthetic speech

    Publication Year: 1977, Page(s):37 - 40
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    We report on an investigation of the role of the vocal cord vibratory pattern in the production of natural phonation. The naturalness of synthetic phonation, generated by exciting a vocal tract model with the glottal area waveform; was evaluated and compared to the actual phonation via a formal listener evaluation. The glottal area was measured from ultra high-speed films obtained by photographing... View full abstract»

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