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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1981

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Displaying Results 1 - 25 of 71
  • [Front cover and table of contents]

    Publication Year: 1981 , Page(s): 0
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    Freely Available from IEEE
  • Comments on "FFT algorithm for both input and output pruning"

    Publication Year: 1981 , Page(s): 448 - 449
    Cited by:  Papers (2)
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    It is pointed out that the FFT and FW-AT algorithm for both input and output pruning have been described in general matrix form in Russian in 1977. View full abstract»

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  • Editorial

    Publication Year: 1981 , Page(s): 625 - 626
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  • [Back cover]

    Publication Year: 1981 , Page(s): c4
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  • Polyphase filter banks using wave digital filters

    Publication Year: 1981 , Page(s): 423 - 428
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (640 KB)  

    The branch filters in a digital polyphase network can be designed either as FIR filters by decomposing the impulse response of an FIR low-pass prototype filter, or as ordinary IIR filters by the synthesis method of Bellanger. The use of all-pass networks has hitherto been considered unfeasible because of the associated computational difficulties involved in the design of filter banks with many branches. The purpose of this paper is to demonstrate that it is indeed possible to design the branch filters as all-pass-low-pass sections without the need of a prototype filter. Moreover, these sections can be realized as wave digital filters, which give improved properties over the other designs with respect to hardware requirements, group delay, sensitivity, dynamics and limit cycles. Examples, including the design of the practically important 60-channel filter bank for the transmultiplexer, are given. View full abstract»

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  • Time delay estimation using the LMS adaptive filter--Static behavior

    Publication Year: 1981 , Page(s): 561 - 571
    Cited by:  Papers (75)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (968 KB)  

    A new application of the LMS adaptive filter, that of determining the time delay in a signal between two split-array outputs, is described. In a split array sonar, this time delay can be converted to the bearing of the target radiating the signal. The performance of such a tracker is analyzed for stationary broad-band targets. It is shown that a continuous adaptive tracker performs within 0.5 dB of the Cramér-Rao lower bound. Further, performance predictions are developed for a discrete adaptive tracker which demonstrates excellent agreement with simulations. It is shown that the adaptive tracker can have significantly less sensitivity to changing input spectra than a conventional tracker using a fixed input filter. View full abstract»

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  • Geophysical signal analysis

    Publication Year: 1981 , Page(s): 457
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  • Analysis of experimental curves using singular value decomposition

    Publication Year: 1981 , Page(s): 429 - 433
    Cited by:  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (560 KB)  

    This paper describes a singular value decomposition (SVD) technique for the analysis of experimental response curves. The SVD of a set of curves shows that curves can be decomposed to become a linear combination of some intrinsic-component patterns. The interpolation of curves can then be reduced to a simple interpolation of weighting coefficients. The SVD method is applied to an estimation problem of γ-ray response spectra, and a library of the response curves is constructed. View full abstract»

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  • Time-scale modification of speech based on short-time Fourier analysis

    Publication Year: 1981 , Page(s): 374 - 390
    Cited by:  Papers (58)  |  Patents (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1696 KB)  

    This paper develops the theoretical basis for time-scale modification of speech based on short-time Fourier analysis. The goal is the development of a high-quality system for changing the apparent rate of articulation of recorded speech, while at the same time preserving such qualities as naturalness, intelligibility, and speaker-dependent features. The results of the theoretical study were used as the framework for the design of a high-quality speech rate-change system that was simulated on a general-purpose minicomputer. View full abstract»

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  • DPCM-AQF using second-order adaptive predictors for speech signals

    Publication Year: 1981 , Page(s): 337 - 341
    Cited by:  Papers (7)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (568 KB)  

    A differential pulse code modulation (DPCM) system having adaptive quantization with forward (AQF) transmission of step-size, and second-order predictors that are adaptive and operate on the locally decoded speech signal, is proposed. For a transmission rate of 40 kbits/s, a block size of 256 speech samples, the DPCM-AQF system using the sequential gradient estimation predictor (SGEP) has segmental signal-to-noise ratio (SNR) gains of 3 and 9 dB compared to the stochastic approximation predictor (SAP) and the leaky integrator, respectively. The dynamic range of the DPCM-AQF using SGEP for an SNR of 35 dB is 30 dB, and it is insensitive to block size (<512). When transmission errors are introduced, it has a higher SNR than that achieved with the leaky integrator for bit error rates <0.08 percent. View full abstract»

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  • Adaptive estimation of time delays in sampled data systems

    Publication Year: 1981 , Page(s): 582 - 587
    Cited by:  Papers (72)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (560 KB)  

    An adaptive technique is developed which iteratively determines the time delay between two sampled signals that are highly correlated. Although the procedure does not require a priori information on the input signals, it does require that the signals have a unimodal or periodically unimodal cross-correlation function. The adaptive delay algorithm uses a gradient technique to find the value of the adaptive delay that minimizes the mean-squared (MS) error function. This iterative algorithm is similar to the adaptive filter coefficient algorithm developed by Widrow. However, the MS error function for the adaptive delay is not quadratic, as it is in the adaptive filter. A statistical analysis determines the value of the convergence parameter which effects rapid convergence of the adaptive delay. This convergence parameter is a function of the power of the input signal. Computer simulations are presented which verify that the adaptive delay correctly estimates the time delay difference between two sinusoids, including those in noisy environments. The adaptive delay is also shown to perform correctly in a time delay tracking application. View full abstract»

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  • Design of ARMA digital filters by pole-zero decomposition

    Publication Year: 1981 , Page(s): 433 - 439
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (752 KB)  

    A new technique for design of digital filters is presented in this paper. The technique exploits the spectral approximation property of autoregressive modeling to reduce ripple at the edges of the transition band in the filter response. An autoregressive model approximates a given spectrum better at the peaks than at the Valleys. Spectral information around the transition band is transformed into peaks by splitting the given squared-magnitude frequency response into two component spectra. This splitting is accomplished using a pole-zero decomposition technique, which in turn uses the properties of the derivative of phase spectrum of minimum phase filters. One of the component spectra corresponds nearly to the response of an all-pole filter, and the other component spectrum corresponds nearly to the response of an all-zero filter. Each of these corresponends spectra can be represented by a small number of coefficients using autoregressive modeling. The resulting two sets of autoregressive coefficients determine poles and zeros of the autoregressive moving average (ARMA) digital filter. Ripple characteristics in the response of the ARMA filter can be controlled by appropriately choosing the number of poles and zeros. It is shown that a wide variety of magnitude frequency response characteristics can be approximated by an ARMA filter of low order using this technique. The technique does not work well in cases where spectral approximation by autoregressive modeling is poor. Such cases arise when the component spectra have very large dynamic range. View full abstract»

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  • Two-dimensional spectral estimation

    Publication Year: 1981 , Page(s): 396 - 401
    Cited by:  Papers (35)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (720 KB)  

    In this paper, effective methods for generating two-dimensional quarter-plane causal autoregressive (AR) and autoregressive moving average (ARMA) spectral estimation models are developed. These procedures are found to provide super resolution capabilities when compared to other more classical methods such as the Fourier transform. The ARMA method involves manipulation of the model equation \sum \min{k = 0}\max {p_{1}} \sum \min{k = 0}\max {p_{2}} a_{km}x(n_{1} - k, n_{2} - m) = \sum \min{k = 0}\max {q_{1}} \sum \min{k = 0}\max {q_{2}} b_{km}\epsilon(n_{1} - k, n_{2} - m) and utilizes the given finite set of observations x(n_{1}, n_{2}) for 1 \leq n_{1} \leq N_{1},1 \leq n_{2} \leq N_{2} . In the above relationship, the random excitation {\epsilon(n_{1}, n_{2})} is taken to be white. This ARMA model's autoregressive akmcoefficients are selected to minimize a weighted least-squares criterion composed of error elements while the moving average bkmcoefficients are obtained using an alternative approach. The spectral estimation performance of the AR and ARMA methods will be empirically demonstrated by considering the problem of resolving two sinusoids embedded in noise. View full abstract»

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  • Intelligibility and quality of linear predictor and eigenparameter coded speech

    Publication Year: 1981 , Page(s): 391 - 395
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    Results are reported for two experiments on intelligibility and quality of linear predictor and eigenparameter coded speech. Speech intelligibility was not improved by eigenparameter coding and speech quality was dependent on the eigenparameter quantization schemes employed. View full abstract»

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  • Time delay estimation

    Publication Year: 1981 , Page(s): 461 - 462
    Cited by:  Papers (9)
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    First Page of the Article
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  • On the simulation of a class of time delay estimation algorithms

    Publication Year: 1981 , Page(s): 534 - 540
    Cited by:  Papers (40)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (736 KB)  

    The time delay between signals received at two (or more) sensors has proven to be a useful parameter in passive sonar for estimating the location of an acoustic source. This paper presents the results of a simulation comparing the smoothed coherence transform and maximum likelihood estimation methods to the basic cross correlation technique for time delay estimation. Band-limited random signals which are corrupted by white noise and received at two sensors are considered at various signal-to-noise ratios. The variance of the time delay estimates are compared to the minimum variance obtainable in theory as given by the Cramér-Rao lower bound. View full abstract»

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  • A distributed-µ implementation of the LMS algorithm

    Publication Year: 1981 , Page(s): 753 - 762
    Cited by:  Papers (2)
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    This paper describes the effects of component nonidealities upon the performance of analog and sampled data adaptive noise cancellers. It is shown that the effects of analog implementation differ significantly from those encountered in ideal cases or those in which the added circuit noise is assumed to have a zero-mean distribution. It is also shown that the effects of nonzero mean errors, such as offset voltages and nonlinearities of the input multipliers contribute an excess mean-square error (MSE) which is inversely proportional to the parameter which controls the stability and the rate of convergence of the algorithm. An implementation configuration is presented which minimizes these deleterious effects by distributing the gain of the adaptive system around the signal path. View full abstract»

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  • A note on the application of the Hilbert transform to time delay estimation

    Publication Year: 1981 , Page(s): 607 - 609
    Cited by:  Papers (29)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    It is shown that the Hilbert transform may be employed to convert the maximum finding task of a delay estimator to one of finding a zero crossing. The basic theory of the technique is shown and a few applications are described. Its primary application is in servo tracking and servo velocity measurement systems which employ correlation techniques. A review of prior work in this field is given. View full abstract»

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  • Adaptive frequency sampling filters

    Publication Year: 1981 , Page(s): 684 - 694
    Cited by:  Papers (18)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1240 KB)  

    We present two new structures for adaptive filters based on the idea of frequency sampling filters and gradient based estimation algorithms. These filters have a finite impulse response (FIR) and can be thought of as attempting to approximate a desired frequency response at given points on the unit circle. The filters operate in real time with no batch processing of signals as is the case when using the discrete Fourier transform. They result in a marked reduction in dimension of the time-domain problem of fitting an Nth-order FIR transversal filter to a collection of length 2 transversal filters and further to a collection of N scalar filters. The advantages of this are then discussed. View full abstract»

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  • A synthesis of frequency domain filters for time delay estimation

    Publication Year: 1981 , Page(s): 540 - 548
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (992 KB)  

    A simple model for two-channel delay estimation filtering is presented. The problem is subdivided into three classes based on initial assumptions. General filters described in the frequency domain are presented as solutions to these specific classes. It is shown that many of these filters, which include the "Wiener" least-squares estimation filter and classical, matched detection filter, can be derived as specific cases of a very general ideal filter form. We call this general ideal filter the weighted distortion balance filter. Relationships between a standard set of ideal filters and some filters previously proposed in the literature for delay estimation are discussed. An illustrative example is presented to compare the delay estimated from the use of various filters. View full abstract»

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  • Tracking properties of adaptive signal processing algorithms

    Publication Year: 1981 , Page(s): 439 - 446
    Cited by:  Papers (27)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (792 KB)  

    Adaptive signal processing algorithms are often used in order to "track" an unknown time-varying parameter vector. This work develops an upper bound on the mean of the norm-squared error between the unknown parameter vector being tracked and the value obtained by the algorithm. The results require very mild covariance decay rate conditions on the training data and a bounded algorithm. The upper bound illustrates the relationship between the algorithm step size and the maximum rate of variation in the parameter vector being tracked. View full abstract»

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  • A performance analysis of adaptive line enhancer-augmented spectral detectors

    Publication Year: 1981 , Page(s): 694 - 701
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (912 KB)  

    This paper discusses the receiver operating characteristic performance of Adaptive Line Enhancer (ALE) augmented spectral detectors for sinusoidal signals in both stationary white Gaussian noise and in nonstationary noise. The detectors considered are based on the discrete Fourier transform (DFT) and include both cases with and without incoherent integration. Analytical expressions are provided for the detector output probability density functions in the stationary noise case. Extensive Monte Carlo simulation results are used to verify these expressions, and to treat the nonstationary noise case. View full abstract»

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  • Modeling of time delay and its application to estimation of nonstationary delays

    Publication Year: 1981 , Page(s): 577 - 581
    Cited by:  Papers (64)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (480 KB)  

    A method to model a time delay by a finite impulse response filter is presented. It is useful in simulation work that involves time delays and transforms the time delay estimation problem into one of parameter estimation. The benefits of this approach are the elimination of spectral estimation, a choice of many parameter estimation algorithms, and the capability to track time-varying delays. Two examples of estimating nonstationary time delays are also given. View full abstract»

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  • A new algorithm for two-dimensional maximum entropy power spectrum estimation

    Publication Year: 1981 , Page(s): 401 - 413
    Cited by:  Papers (46)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1184 KB)  

    A new iterative algorithm for the maximum entropy power spectrum estimation is presented in this paper. The algorithm, which is applicable to two-dimensional signals as well as one-dimensional signals, utilizes the computational efficiency of the fast Fourier transform (FFT) algorithm and has been empirically observed to solve the maximum entropy power spectrum estimation problem. Examples are shown to illustrate the performance of the new algorithm. View full abstract»

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  • Delay estimation using narrow-band processes

    Publication Year: 1981 , Page(s): 478 - 484
    Cited by:  Papers (43)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (696 KB)  

    Array processing of narrow-band Gaussian signals is studied with emphasis on delay estimation. The Barankin bound is used to examine the effect of ambiguity on mean-square measurement error. When the bound is plotted as a function of signal-to-noise ratio one observes a distinct threshold. Above the critical signal-to-noise ratio the lower bound on mean-square error is given by the Cramér-Rao inequality, which is approached by the Barankin inequality under these conditions. Below the threshold the Barankin bound can exceed the Cramér-Rao bound by large factors. The relative magnitude of the bounds in that region depends critically on the ratio of signal center frequency to signal bandwidth. 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