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ICASSP '80. IEEE International Conference on Acoustics, Speech, and Signal Processing

9-11 April 1980

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Displaying Results 1 - 25 of 258
  • [Front cover and table of contents]

    Publication Year: 1980, Page(s): 0
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    Freely Available from IEEE
  • Signal processing and complexity of computation

    Publication Year: 1980, Page(s):94 - 101
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (258 KB)

    First Page of the Article
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  • A compound random process for underwater ambient acoustic noise

    Publication Year: 1980, Page(s):124 - 127
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    A stationary compound process has been formulated and investigated as a model for underwater acoustic ambient noise pressure with long-term power fluctuations. The process is represented bysqrt{P(t)} G(t), whereP(t)is a non-negative stationary random process, andG(t)is a zero-mean, unit-variance stationary Gaussian process independent ofP(t). The process... View full abstract»

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  • Computation of the discrete cosine transform via the arcsine transform

    Publication Year: 1980, Page(s):231 - 234
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (56 KB)

    A procedure is described for the computation of the discrete cosine transform (DCT) via the use of the arcsine transform. The approach eliminates time-consuming multiplications, the DCT being accomplished with only additions and table lookups. While a fast Fourier transform (FFT) approach to computing the DCT involves on the order ofNlog_{2}2N"butterfly" computations to evaluate all View full abstract»

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  • Design of FIR digital phase networks

    Publication Year: 1980, Page(s):252 - 255
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (66 KB)

    The problem of the minimax design of FIR digital filters with prescribed phase characteristics and unit magnitude is a nonlinear optimization problem. In this paper it is approximated by a linear programming problem which is optimal to first order. That is, if δ0and ε0are optimal deviations of magnitude and phase characteristics, then the actual deviations obtained ... View full abstract»

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  • A new approach to causal filter design by Padé approximants

    Publication Year: 1980, Page(s):264 - 267
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (71 KB)

    In recursive digital filter design, the only linear technique available is probably the method of Padé approximants. Unfortunately, to obtain a Padé approximant, a formal power (Maclaurin) series must be given. If an ideal amplitude response|H(e^{jomega})|is given, the usual method is to approximate its truncated delayed Fourier series,H_{N}(e^{jomega}) = Sigmamin{0}max{2... View full abstract»

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  • Moving image restoration and registration

    Publication Year: 1980, Page(s):418 - 421
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (141 KB)

    A new method for moving image restoration and registration is established. The observations are sequences of low-resolution, possibly undersampled, discrete frames. The result is a restored high-resolution image. The restoration part is attractive for the purpose of real time implementation since the computation consists of only a few complex operations per pel of the resultant high resolution ima... View full abstract»

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  • Spectral distance measure applied to the optimum design of DPCM coders with L predictors

    Publication Year: 1980, Page(s):512 - 515
    Cited by:  Papers (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (115 KB)

    This paper explores the intermediate solutions between fixed prediction and forward adaptative prediction in ADPCM which consists of using a finite number, L, of preselected linear predictors of order M. The design problem of selecting the optimum set of predictors with respect to the overall prediction gain is formulated and an iterative procedure is described to obtain the solutions. The relativ... View full abstract»

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  • A phonetic dictionary for demisyllabic speech synthesis

    Publication Year: 1980, Page(s):565 - 567
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (77 KB)

    Demisyllables and affixes (1,2,3) have been found to be very promising units for concatenative speech synthesis View full abstract»

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  • An iterative procedure for moving average models estimation

    Publication Year: 1980, Page(s):614 - 617
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (74 KB)

    Using the formulation of Levinson's algorithm in terms of cross correlation estimates as presented by RISSANEN, it is shown that, in the case of a finite length autocorrelation serieR(z)(i.e. moving average models), this procedure generates a serie that converges towards a causal impulse responseE(z). This response has a finite length and its autocorrelationE(z)cdotE(z^{... View full abstract»

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  • A Wiener filter approach to coherence estimation

    Publication Year: 1980, Page(s):646 - 649
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (100 KB)

    This paper presents a new method of estimating the magnitude-squared-coherence (MSC) between two random processesxandy. An unrealizable Wiener filter transfer function is first computed in thexchannel, then another one in the y channel. The product of these transfer functions then gives the MSC. The unrealizable Wiener filter is approximated by a finite impulse fil... View full abstract»

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  • Reconstruction of discrete-time signals from a subset of weighted DFT outputs

    Publication Year: 1980, Page(s):654 - 657
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (104 KB)

    An algorithm is presented for combining the output ofrDFT bins to produce a single discrete-time signal, wherer < n, the DFT size. The technique described compensates for the effect of any weighting function used in the original DFT, as well as any DFT overlap fraction. The output sampling rate can be selected appropriate to the bandwidth of the resulting signal. The proce... View full abstract»

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  • A hardware realization of an NTT convolver using ROM arrays

    Publication Year: 1980, Page(s):788 - 791
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (102 KB)

    This paper describes the construction of a digital signal convolver. The convolution is performed using a Number Theoretic Transform computed over a ring which is isomorphic to three extension fields of second degree:R simeq GF(191^{2}) oplus GF(193^{2}) oplus GF(449^{2}). The transform is implemented using arrays of latched read-only-memories to provide a high throughput computational ... View full abstract»

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  • Large moduli multipliers

    Publication Year: 1980, Page(s):792 - 795
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (73 KB)

    In this work we present a new table look-up storage scheme and a class of table look-up multipliers capable of working with exact (modular) numbering systems. Memory savings associated with the new look-up multiplier, when compared to contemporary methods, are shown to be on the order of2/NwhereN=2^{n},n=input wordlength. Throughput is shown to be equal to that obt... View full abstract»

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  • Doubling algorithms for Toeplitz and related equations

    Publication Year: 1980, Page(s):954 - 959
    Cited by:  Papers (37)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (159 KB)

    A new class of doubling or halving algorithms for solving Toeplitz and related equations is presented. For scalar n by n Toeplitz matrices, they requireO(n log^{2}n)computations, similarly to the HGCD (half-greatest-common-divisor) based algorithm of Gustavson and Yun. However, these new algorithms are based on the notions of "shift" or displacement rank1 leq alpha leq n, an ... View full abstract»

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  • Signal processing techniques in error control systems

    Publication Year: 1980, Page(s):968 - 973
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (178 KB)

    Several standard signal processing techniques may be effectively applied in the design and mechanization of optimum encoders and decoders in error control systems that employ block error-correcting codes. Many of the approaches rely upon generalized transforms that resemble the Walsh functions. They use a fast algorithm that is developed over a subspace in the generalized frequency domain. The dec... View full abstract»

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  • Rapid detection of weak signals using adaptive recursive filter weights

    Publication Year: 1980, Page(s):993 - 996
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (157 KB)

    The adaptive weights of a frequency domain adaptive recursive filter in the line enhance mode are investigated for their ability to rapidly detect and identify a weak tonal in the presence of additive broadband noise. The investigation utilizes both Monte Carlo computer simulations and first order theory to obtain an approximate analytic solution. In the simulation, rapid identification meant with... View full abstract»

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  • [Back cover]

    Publication Year: 1980, Page(s): c4
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    Freely Available from IEEE
  • Frequency domain data transmission using reduced computational complexity algorithms

    Publication Year: 1980, Page(s):964 - 967
    Cited by:  Papers (238)  |  Patents (39)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    In this paper we describe a frequency domain data transmission method to be used for digital data transmission over analog telephone lines which exploits recently derived reduced computational complexity algorithms, such as the Winograd Fourier Transform, to achieve a significantly lower computational rate than comparable time domain QAM modems implemented digitally using signal processing techniq... View full abstract»

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  • An LSI digital signal processor

    Publication Year: 1980, Page(s):383 - 385
    Cited by:  Papers (12)  |  Patents (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (160 KB)

    An integrated circuit digital signal processor has been designed and built to realize a variety of important telecommunications applications. The flexibility of full stored program control and the processing speed of a pipelined architecture are essential to this general purpose device. Special emphasis has also been placed on circuit interface features which simplify the use of the processor as a... View full abstract»

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  • A two-threshold detector for pulses of unknown duration and doppler

    Publication Year: 1980, Page(s):630 - 633
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (168 KB)

    The problem of detecting pulses of unknown duration and doppler in gaussian noise is solved using a two threshold detector and averaging. The performance of the detector is estimated using a novel gaussian approximation. The detector has possible application in variable pulse length or multiple pulsed active sonars and radars and also in passive counter measure systems. It is shown that a passive ... View full abstract»

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  • A speech/Speaker recognition and response system

    Publication Year: 1980, Page(s):394 - 397
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (136 KB)

    The realization of a speech analyzer plus an LPC synthesizer in a single chip signal processing microprocessor is described. The chip is able to process both algorithms in real time to create an interactive voice analyzer/response system operating under the control of a microprocessor and with the LPC speech data stored in a ROM. The chip is a 16 bit microprocessor specially architectured for sign... View full abstract»

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  • An efficient method for formant to reflection coefficient conversion

    Publication Year: 1980, Page(s):552 - 556
    Cited by:  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    The exact transformation from formant frequencies and bandwidths to reflection coefficients is a complex and time consuming process. A simple approximate transformation may be performed using a table look-up procedure with linear interpolation. Experiments have been conducted using a four formant model, each formant being represented by a complex conjugate pole pair. The formant bandwidths and the... View full abstract»

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  • LSI signal processor development for communications equipment

    Publication Year: 1980, Page(s):386 - 389
    Cited by:  Papers (13)  |  Patents (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

    A single chip, versatile digital signal processor has been developed, which enables compact and low cost telecommunications equipment realization. The processor architecture has been optimized for real time voiceband signal processing by adopting efficient micro-program control and by integrating a 16 × 16 bit parallel multiplier on a single chip. The processor is fabricated with N-channel MO... View full abstract»

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  • Generalized Levinson algorithms and Ladder filters for nonstationary signal processing

    Publication Year: 1980, Page(s): 102
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (12 KB)

    The Levinson algorithm, implemented via a ladder filter, has been widely used in seismic signal processing and more recently in speech analysis. This algorithm was originally developed (in 1947) for prediction of stationary stochastic processes. It has been adapted to applications where only a single sample function (time-series) is available by the use of appropriate "window functions" to "simula... View full abstract»

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