ICASSP '78. IEEE International Conference on Acoustics, Speech, and Signal Processing

10-12 April 1978

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  • [Front cover and table of contents]

    Publication Year: 1978, Page(s): 0
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    Freely Available from IEEE
  • New stochastic realization algorithms for identification of ARMA models

    Publication Year: 1978, Page(s):208 - 213
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (155 KB)

    Autoregressive moving-average (ARMA) models are of great interest in speech processing. This paper presents new stochastic realization algorithms for identification of such models, by use of a special canonical filter form in the state space, directly and simply connected with ARMA models. We take advantage of certain matrix properties to develop algorithms, which eliminate a matrix inversion, usi... View full abstract»

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  • Design considerations for feedback amplifiers

    Publication Year: 1978, Page(s):252 - 254
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    Circuit design concepts for negative feedback amplifiers are discussed. Preferred circuit architecture for the minimization of transient and static distortions is presented. A discussion of the application of these concepts to audio power amplifiers and phonograph preamplifiers is given. View full abstract»

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  • Computer analysis of transient distortion and low transient distortion amplifier design

    Publication Year: 1978, Page(s):267 - 269
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    First Page of the Article
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  • High resolution spectral analysis of sinusoids in correlated noise

    Publication Year: 1978, Page(s):349 - 351
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (65 KB)

    A high resolution technique for estimating the frequencies, powers, and the number of sinusoids in noise is examined. Unlike the Pisarenko decomposition method which may only be applied to sinusoids in uncorrelated noise, the technique examined in this paper may be applied even when the noise is correlated. This technique consists of applying Prony's algorithm to a segment of the autocorrelation f... View full abstract»

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  • An error formula for iterative prefiltering frequency estimates

    Publication Year: 1978, Page(s):369 - 371
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  • Digital filters with prescribed zeros

    Publication Year: 1978, Page(s):487 - 490
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (68 KB)

    We discuss the design of digital filters with maximally flat or equiripple passband behavior and transfer functions of the formH(z)= K frac{(z+1)^{q}Pimin{i=1}max{r}(z-e^{jomega_{i}})(z-e^{-jomega_{i}})}{D_{n}(z)}, where q+2r, the number of finite-plane zeros, is allowed to vary from 0 to n, the filter order. Analytic expressions are given for the magnitude squared function. Identificat... View full abstract»

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  • A phoneme recognition system based on human audition

    Publication Year: 1978, Page(s):557 - 560
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (169 KB)

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  • Signal processing with number theoretic transforms and limited word lengths

    Publication Year: 1978, Page(s):619 - 623
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (85 KB)

    Number Theoretic Transforms (NTT's), unlike the Discrete Fourier Transform (DFT), are defined in finite rings and fields rather than in the field of complex numbers. Some NTT's have a transform structure like the Fast Fourier Transform (FFT) and can be used for fast digital signal processing. The computational effort and the signal-to-noise ratio (SNR) performance of linear filtering in finite rin... View full abstract»

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  • Number theoretic transforms with modulus 22q- 2q+ 1

    Publication Year: 1978, Page(s):624 - 627
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (87 KB)

    Number theoretic transforms using the modulus2^{2_{q}} - 2^{q} + 1, which require no multiplications, are introduced. These transforms have 50% greater sequence lengths for a given wordlength than Fermat transforms, and there is greater flexibility in the choice of wordlength. Furthermore, arithmetic modulo2^{2_{q}} - 2^{q} + 1is not significantly more complex than arithmetic... View full abstract»

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  • Real-factor FFT algorithms

    Publication Year: 1978, Page(s):634 - 637
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (61 KB)

    The FFT algorithms are described which, like those described by Rader and Brenner, use pure real (or imaginary) constant factors. Since these factors containcos(2pik/N)(orsin(2pik/N)), these processes are somewhat better conditioned than those of Rader/Brenner, and are comparably efficient. View full abstract»

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  • An algorithm for testing stability of two-dimensional digital recursive filters

    Publication Year: 1978, Page(s):769 - 772
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (78 KB)

    The Principle of the Argument is applied to test the stability of a 2-D recursive filter with denominator B(Z1,Z2). This approach to stability testing leads to an efficient numerical implementation requiring only the computation of a 2-D DFT, followed by phase unwrapping and a search for zeros ofB(e^{jomega_{1}},e^{jomega_{2}}). In the absence of zeros ofB(e^{... View full abstract»

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  • On an extension of A. N. Krylov's numerical method for determining the frequencies of small vibrations of systems with damping

    Publication Year: 1978, Page(s):846 - 847
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (22 KB)

    In many engineering problems we must find the roots of an equation of the formdet B(lambda)=0where B(λ) is a nonlinear matrix of λ. In this paper we show how this problem may be reduced to integrating a system of ordinary differential equations subject to known initial conditions. The method covers also the case of complex roots. It gives directly the roots i: without the need... View full abstract»

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  • [Back cover]

    Publication Year: 1978, Page(s): c4
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    Freely Available from IEEE
  • Adaptive data orthogonalization

    Publication Year: 1978, Page(s):109 - 112
    Cited by:  Papers (74)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (168 KB)

    The decomposition of vector time series data into orthogonal components can be applied in both temporal and spatial discrete frequency analysis. If the observed multidimensional data is non-stationary, then adaptive procedures can be used for estimation of the eigendata. This paper presents the relationship between multicomponent, spectral signals in noise and the corresponding eigendata. Two adap... View full abstract»

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  • Inherent errors in sonar range prediction

    Publication Year: 1978, Page(s):248 - 251
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (68 KB)

    It is common practice to make sonar range predictions by solving the sonar equation for the range at which a predicted value of propagation loss is equal to the system-target Figure-of-Merit. The inherent limitations of this method are derived in this paper. It is shown that small errors in estimates of Figure-of-Merit can cause large errors in predicted range. Results are given for both passive a... View full abstract»

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  • High quality adaptive predictive coding of speech

    Publication Year: 1978, Page(s):303 - 306
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (82 KB)

    We report on the results of research to code speech at 16 kbps under the condition that the quality of the transmitted speech be equal to that of the original. Some of the original speech had been corrupted by noise and distortions typical of long distance telephone lines. The rigorous requirements of this work led to a new outlook on adaptive predictive coding. We have found that the pitch predic... View full abstract»

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  • Automatic speaker identification for a large number of speakers

    Publication Year: 1978, Page(s):295 - 298
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (73 KB)

    Design of speaker identification systems for a small number of speakers (around 10) with a high degree of accuracy has evolved over the past few years. A sequential identification technique gives better results when the number of speakers is large. This scheme is implemented as a decision tree classifier in which the final decision is made only after a predetermined number of stages. The error rat... View full abstract»

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  • Simulation of large length filters using fermat number transforms

    Publication Year: 1978, Page(s):628 - 631
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (120 KB)

    A technique based on the partitioning of the impulse response is proposed to overcome the length restriction and the overflow problem that are encountered when Fermat number transforms are used to implement digital convolutions. Two different schemes are proposed to implement this method which require less hard-ware than that of two-dimensional technique. The method, when implemented on a general ... View full abstract»

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  • An all-digital phase-measurement techniques using clipped-quadrature correlation

    Publication Year: 1978, Page(s):674 - 677
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (72 KB)

    An all-digital method is presented for determining the phase of a noise-contaminated tone signal relative to an internal reference after the signal has been bandpass-filtered and hardlimited. The process involves the cross correlation of the input signal with a pair of quadrature-related one-bit binary reference phase signals, followed by an algorithm which computes the phase from these clipped-qu... View full abstract»

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  • Gaps in the technology of speech understanding

    Publication Year: 1978, Page(s):405 - 408
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (68 KB)

    We have determined how the state of the art in speech recognition has advanced in recent years. We have surveyed recent accomplishments in research and system design, and have determined a number of gaps that exist in speech understanding technology. There is a limited but growing market acceptance for available isolated-word recognizers, and some uncertainty about the impact of recent advances in... View full abstract»

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  • 2400/16,000 Bps Multirate voice processor

    Publication Year: 1978, Page(s):299 - 302
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (72 KB)

    This paper describes a voice digitizer whose 16,000 bps data rate contains an embedded 2400 bps linear predictive coder. This 2400/16,000 bps multirate coder eliminates the need for tandeming conversion when going from 2400 bps to 16,000 bps systems and when going from 16,000-bps to 2400-bps systems. Because the 16,000 bps rate uses the principles of adaptive predictive coding, this multirate proc... View full abstract»

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  • Text-independent speaker identification based on piecewise canonical discriminant analysis

    Publication Year: 1978, Page(s):291 - 294
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (96 KB)

    This paper describes a method for text-independent speaker identification. In this method, in order to utilize phoneme-dependent personal information in addition to personal information common to all phonemes, multiple personal factor spaces are constructed by applying canonical discriminant analysis to the predetermined subspaces in the observation space. The decision is based on a liklihood meas... View full abstract»

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  • An investigation of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noise

    Publication Year: 1978, Page(s):602 - 605
    Cited by:  Papers (9)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    This paper describes results of a study of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noise. Five categories of processing methods are explored. These include the INTEL technique, a technique based upon minimum mean square filtering, several techniques based upon subtraction of the estimated spectrum of the noise from the spectrum of ... View full abstract»

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  • Design of inherently stable two-dimensional recursive filters imitating the behaviour of one-dimensional analog filters

    Publication Year: 1978, Page(s):765 - 768
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (144 KB)

    A novel method is presented for the design of two-dimensional (2-D) recursive filters that are inherently stable, thereby avoiding computationally lengthy tests of stability. The new method is based on mapping the 1-D continuous complex frequency variable s by an appropriate function of the 2-D digital filter variables Z1and Z2. Thus, passive analog filters (Butterworth, Cheb... View full abstract»

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