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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1978

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Displaying Results 1 - 21 of 21
  • [Front cover and table of contents]

    Publication Year: 1978 , Page(s): 0
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    Freely Available from IEEE
  • Comments on "A prime factor FFT algorithm using high-speed convolution"

    Publication Year: 1978 , Page(s): 254
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (95 KB)  

    The purpose of this correspondence is to point out that in Table II of the above paper, the number of additions reported for the radix-2 FFT algorithm are highly erroneous. View full abstract»

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  • Comments on "An introduction to programming the winograd Fourier transform algorithm (WFTA)"

    Publication Year: 1978 , Page(s): 268 - 269
    Cited by:  Papers (2)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    This correspondence points out some inconsistencies between definitions and algorithms presented in the paper by H. F. Silverman. View full abstract»

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  • Corrections and an addendum to "An introduction to programming the winograd Fourier transform algorithm (WFTA)"

    Publication Year: 1978 , Page(s): 268
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (96 KB)  

    First Page of the Article
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  • Introduction to discrete-time signal processing

    Publication Year: 1978 , Page(s): 270 - 271
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1978 , Page(s): c4
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    Freely Available from IEEE
  • Adaptive enhancement of multiple sinusoids in uncorrelated noise

    Publication Year: 1978 , Page(s): 240 - 254
    Cited by:  Papers (116)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1336 KB)  

    The steady-state behavior of the adaptive line enhancer (ALE), a new implementation of adaptive filtering that has application in detecting and tracking narrow-band signals in broad-band noise, is analyzed for a stationary input consisting of multiple sinusoids in white noise. It is shown that the steady-state performance of an L-weight ALE for this case can be modeled by the L × L Wiener-Hopf matrix equation and that this matrix equation can be transformed into a set of 2N coupled linear equations, where N is the number of sinusoids. It is also shown that the expected values of the ALE weights in steady state can be written as a sum of sinusoids and that the amplitude of each sinusoid is coupled to that of all other sinusoids by coefficients that approach zero as the number of ALE weights becomes large. The analytical results are compared to experimental results obtained with a hardware implementation of the ALE of variable length (up to 256 weights) and show good agreement. Theoretical expressions for linear predictive spectral estimates are also derived for multiple sinusoids in white noise. Comparisons are made between the magnitude of the discrete Fourier transform of the ALE weights and the linear predictive spectral estimate for two sinusoids in white noise. View full abstract»

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  • All-pole modeling of degraded speech

    Publication Year: 1978 , Page(s): 197 - 210
    Cited by:  Papers (174)  |  Patents (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1520 KB)  

    This paper considers the estimation of speech parameters in an all-pole model when the speech has been degraded by additive background noise. The procedure, based on maximum a posteriori (MAP) estimation techniques is first developed in the absence of noise and related to linear prediction analysis of speech. The modification in the presence of background noise is shown to be nonlinear. Two suboptimal procedures are suggested which have linear iterative implementations. A preliminary illustration and discussion based both on a synthetic example and real speech data are given. View full abstract»

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  • Quasi-periodic instability in a linear prediction analysis of voiced speech

    Publication Year: 1978 , Page(s): 210 - 216
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (664 KB)  

    A significant semiperiodic fluctuation of the vocal tract area functions derived by linear prediction of the speech waveform has been noted during apparently stationary voiced segments of speech. In one example some values of the area function varied over a range of 9:1 over a few pitch periods. The phenomenon is attributed to "beating" of the pitch period and the time interval between successive computations which causes variations of the time relationship between glottal pulse and analysis window. This is supported by the fact that no fluctuations occur in the area function derived from natural or synthetic speech when the computation interval is equal to the pitch period. Any slight difference between the two leads to significant pulsations, however. A simple theoretical model is used to show how the positioning of the analysis window can influence area function estimates. The problem can be largely overcome by using longer time windows (greater than 2.5 pitch periods), or alternatively, by averaging the area functions over several adjacent intervals View full abstract»

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  • The least squares estimation of time delay and its use in signal detection

    Publication Year: 1978 , Page(s): 217 - 222
    Cited by:  Papers (47)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (624 KB)  

    This paper examines the use of two spatially separated receivers to determine the presence of a distant signal source and its relative bearing. Ideally, the phase shift between the receivers' output is proportional to the frequency with the time delay between outputs equal to the proportionality constant. Because of noise, the plot of phase against frequency is scattered along a straight line whose slope is the time delay. A least squares estimator of the slope turns out to be equivalent to the maximum likelihood estimator developed by Hamon and Hannan [1]. Since the goodness of fit of the least squares line is a function of the coherence between the receivers' output, the sum of the squared errors is used as a test statistic in detection. The proposed detector has a detection threshold that depends only on the probability of false alarm and not on the ambient noise level. It can also be simply extended to an array of receivers. View full abstract»

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  • Equal ripple approximation for envelope detection

    Publication Year: 1978 , Page(s): 254 - 256
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB)  

    We give a simplified derivation of the well-known equal ripple approximation to the modulus of a complex number as a linear combination of its real and imaginary parts. We also generalize the approximation to smaller sectors in phase, with corresponding smaller errors. We discuss the implementation of the generalized approximation, and indicate the number of computer operations required. View full abstract»

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  • A sampling theorem for dyadic stationary processes

    Publication Year: 1978 , Page(s): 265 - 267
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB)  

    This paper deals with some aspects of the derivation of sampling theorems for sequency (Walsh domain) bandlimited signals. First, the deterministic case is treated rather briefly to arrive at a useful interpolation (reconstruction) formula. Next, it is shown that a sampling theorem for the stochastic case can be proved in a straightforward manner via this interpolation formula. View full abstract»

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  • Solution of complex integrals using the Laurent expansion

    Publication Year: 1978 , Page(s): 263 - 265
    Cited by:  Papers (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    A simple, unified approach to solve in closed forms various complex integrals is presented. The basic idea behind this method is to identify the well-known infinite series solution of the integral as the constant term in a Laurent expansion of the integrand. The constant is then obtained by a finite algebra. View full abstract»

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  • On using the symmetry of FIR filters for digital interpolation

    Publication Year: 1978 , Page(s): 267 - 268
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (200 KB)  

    By implementing an interpolating linear phase FIR filter in a modified direct form, it is possible to take advantage of the symmetry conditions and reduce the number of multiplications. View full abstract»

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  • Digital filter design in the quefrency domain

    Publication Year: 1978 , Page(s): 226 - 235
    Cited by:  Papers (1)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (800 KB)  

    The desired log magnitude and phase responses are realized by approximating the log frequency response (the complex logarithm of the frequency response). Since the log frequency response is represented by its Fourier series in terms of the quefrency response (the cepstrum of the impulse response), the desired log frequency response is approximated by the filter with a finite length quefrency response. The elemental filter presented in this paper has a component of the quefrency response. Therefore, the quefrency response is realized by a cascade connection of the elemental filters for all its quefrency components. The cascade filter provides the best mean-square approximation to the desired log frequency response. By introducing the elemental filter, digital filters can be designed in the quefrency domain. By designing digital filters in the quefrency domain, the best approximation filter for the desired log frequency response can be easily realized. View full abstract»

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  • Discrete Fourier transform computation via the Walsh transform

    Publication Year: 1978 , Page(s): 236 - 240
    Cited by:  Papers (21)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (456 KB)  

    This paper presents a new computational algorithm for the discrete Fourier transform (DFT). In an algorithm proposed here, DFT coefficients are computed via the Walsh transform (WT). The number of multiplications required by the new algorithm is approximately NL/6, where N is the number of data points and L is the number of Fourier coefficients desired. As such, it is superior to the fast Fourier transform (FFT) approach in applications where L is relatively small compared with N. It is also useful in cases where the Walsh and Fourier coefficients are both desired. View full abstract»

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  • A local method for digitally restoring motion-blurred images

    Publication Year: 1978 , Page(s): 256 - 263
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (4136 KB)  

    A method is developed to restore photographs blurred by image motion using a nearest neighbor digital reconstruction method. The algorithm is simple to implement and is useful in restoring large images very rapidly. No knowledge of the statistics of the noise is assumed and noise amplification in most cases is minimal. View full abstract»

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  • A generalized hamming window

    Publication Year: 1978 , Page(s): 269 - 270
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (232 KB)  

    A formal generalization of the Hamming window is given in terms of cosines raised to an arbitrary power. View full abstract»

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  • A new algorithm for the radix-3 FFT

    Publication Year: 1978 , Page(s): 222 - 225
    Cited by:  Papers (18)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (416 KB)  

    A radix-3 FFT which has no multiplications in the three-point DFT's is introduced. It uses arithmetic with numbers of the form a + bμ, where μ is a complex cube root of unity. The application to fast convolution of real sequences is discussed. View full abstract»

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  • Digital filters

    Publication Year: 1978 , Page(s): 271
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    First Page of the Article
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  • Multidimensional pseudo-maximum-likelihood pitch estimation

    Publication Year: 1978 , Page(s): 185 - 196
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1272 KB)  

    An estimator algorithm for the pitch of voiced speech is presented, based on the following sequence of operations: 1) linear-prediction inverse filtering; 2) short-time spectral analysis by a bank of bandpass filters; 3) envelope extraction on the filter outputs; 4) period determination on the parallel envelopes considered as a multicomponent vector signal, using an algorithm described in a previous work. Results of a comparative evaluation indicate superior immunity to added noise and to bandlimiting with loss of the fundamental component. View full abstract»

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Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope