By Topic

Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date Feb. 1988

Filter Results

Displaying Results 1 - 16 of 16
  • Perturbation analysis of TK method for harmonic retrieval problems

    Page(s): 228 - 240
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (785 KB)  

    The authors present a first-order perturbation analysis of the Tufts-Kumaresan (TK) (1980) method used to estimate frequencies of complex sinusoids in small additive noise. Several fundamental properties are presented and proved. Further illustrations are provided through numerical examples.<> View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • The complex cepstrum of higher order cumulants and nonminimum phase system identification

    Page(s): 186 - 205
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1148 KB)  

    A computationally efficient identification procedure is proposed for a nonGaussian white-noise-driven linear, time-invariant, nonminimum phase system. The method is based on the idea of computing the complex cepstrum of higher order cumulants of the system output. In particular, the differential cepstrum parameters of the nonminimum phase impulse response are estimated directly from higher-order cumulants by least-squares solution or two-dimensional FFT operations. The method reconstructs the minimum-phase and maximum-phase impulse response components separately. It is flexible enough to be applied on autoregressive (AR), moving average (MA), or ARMA system without a priori knowledge of the type of the system. Benchmark simulation examples demonstrate the effectiveness of the method even with short length data records View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Signal synthesis from modified discrete short-time transform

    Page(s): 168 - 181
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1076 KB)  

    The discrete short-time transform (DSTT) is a generalization of the discrete short-time Fourier transform (DSTFT). The necessary and sufficient conditions on the analysis filter, under which perfect reconstruction of the input signal is possible (when the DSTT is not modified), are presented. The class of linear modifications for which the original input can be reconstructed when the modification is applied is characterized. The synthesis of an optimal (in the minimum-mean-square-error sense) signal from a modified DSTT (MDSTT) of finite duration is presented. It is shown that for an analysis filter length that does not exceed a given value, the optimal synthesis scheme is independent of the duration of the given MDSTT and is an extension of the weighted overlap add (WOLA) synthesis method. For longer analysis filters, the optimal synthesis scheme becomes quite cumbersome, and therefore, a steady-state solution (as the duration of the MDSTT approaches infinity) is presented for this case. It is shown that this solution can be approximated with arbitrarily small reconstruction error View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Tracking improvements in fast RLS algorithms using a variable forgetting factor

    Page(s): 206 - 227
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1296 KB)  

    The concept of a variable forgetting factor (VFF) is incorporated into fast recursive least-squares (FRLS) algorithms. Compromises in the data matrix that are needed to do this are examined. Both prewindowed and growing memory covariance algorithms are presented in transversal and lattice structures. Forgetting-factor adaptation schemes, which improve tracking performance over conventional FRLS algorithms, are suggested. Finally, the bias introduced by the use of the VFF is analyzed View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An iterative algorithm for power spectrum estimation in the maximum entropy method

    Page(s): 294 - 296
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (256 KB)  

    The maximum entropy method is considered that uses the entropy model of the form given by B.R. Frieden (1973). An iterative algorithm is presented that does not require conditions such as causality, minimum-phase, etc., and can be used for any dimensionality. The implementation of the algorithm is discussed View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Bayesian estimation in seismic migration

    Page(s): 252 - 264
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (944 KB)  

    Seismic migration is a technique widely used in seismic oil exploration for wavefield reconstruction and for imaging the geometrical distribution of the reflection surfaces within the Earth from the seismic data recorded on the Earth surface. These data are usually corrupted by noise (white noise, surface waves, multiple reflections, etc.) that degrades the result of the migration. Another factor which influences the migration is the inadequate knowledge of the distribution of the acoustic wave propagation velocity in the subsurface of the Earth. The authors use estimation theory techniques to find the MAP (maximum a posterior) estimate of the wave propagation velocity and the geometrical distribution of the subsurface reflector points View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An adaptive IIR algorithm with unimodal performance surfaces

    Page(s): 286 - 287
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (152 KB)  

    The authors develop an adaptive IIR (infinite-impulse response) algorithm by means of least squares inverses; the algorithm also guarantees the resulting IIR system to be stable. This IIR algorithm consists of two subalgorithms, i.e. the all-pole QN( z) and all-zero PN(z) algorithms. Both are with quadratic performance surfaces and stable. The iterations for QN(z) and PN(z ) can be executed simultaneously or alternatively. The all-pole part is independent of PN(z), but the all-zero part depends on QN(z) at each iteration. In many cases, the all-pole model itself is a fairly good approximation of an IIR system View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Eigenstructure variability of the multiple-source multiple-sensor covariance matrix with contaminated Gaussian data

    Page(s): 153 - 167
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (812 KB)  

    Several methods of current interest for counting and locating signal sources using data from a passive array depend on the accuracy of estimating the eigenstructure of the covariance matrix of the array's data vectors. When errors in the measured data vectors are Gaussian conventional covariance estimation is optimal, but robust procedure are required for data with nonGaussian additive contamination. Two different robust covariance estimators are compared by simulation with the conventional one for different degrees of contamination. Even in relatively good signal-to-noise ratios, however, closeness of signal sources in the temporal and spatial frequency domains can cause inaccurate signal-related eigenvalue and eigenvector estimates. The degree of adversity for these problems is also shown by simulation View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Chebyshev design of filters for subband coders

    Page(s): 182 - 185
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    Linear phase and minimum phase quadrature mirror filters, with prescribed stopband attenuation and optimum reconstruction error, are designed by linear programming. The variation of the reconstruction error as a function of the filter is investigated on an example View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Adaptive eigensubspace algorithms for direction or frequency estimation and tracking

    Page(s): 241 - 251
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (956 KB)  

    The authors present an adaptive estimator of the complete noise or signal subspace of a sample covariance matrix as well as the estimator's practical implementations. The general formulation of the proposed estimator results from an asymptotic argument, which shows the signal or noise subspace computation to be equivalent to a constrained gradient search procedure. A highly parallel algorithm, denoted the inflation method, is introduced for the estimation of the noise subspace. The simulation results of these adaptive estimators show that the adaptive subspace algorithms perform substantially better than P.A. Thompson's (1980) adaptive version of V.F. Pisarenko's technique (1973) in estimating frequencies or directions of arrival (DOA) of plane waves. For tracking nonstationary parameters, the simulation results also show that the adaptive subspace algorithms are better than direct eigendecomposition methods for which computational complexity is much higher than the adaptive versions View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On computing the inverse DFT

    Page(s): 285 - 286
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (140 KB)  

    The authors indicate an apparently novel method for computing an inverse discrete Fourier transform (IDFT) through the use of a forward DFT program. They point out that, in many cases, this is obtained without any additional cost, either in terms of program length or in terms of computational time View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Inverse filtering of room acoustics

    Page(s): 145 - 152
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (540 KB)  

    A novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room. This method is based on the principle called the multiple-input/output inverse theorem (MINT). The inverse is constructed from multiple finite-impulse response (FIR) filters (transversal filters) by adding some extra acoustic signal-transmission channels produced by multiple loudspeakers or microphones. The coefficients of these FIR filters can be computed by the well-known rules of matrix algebra. Inverse filtering in a sound field is investigated experimentally. It is shown that the proposed method is greatly superior to previous methods that use only one acoustic signal-transmission channel. The results prove the possibility of sound reproduction and sound reception without any distortion caused by reflected sounds View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Estimation of noise variance from the noisy AR signal and its application in speech enhancement

    Page(s): 292 - 294
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (244 KB)  

    In a number of applications involving the processing of noisy signals, it is desirable to know a priori the noise variance. The author proposes a method of estimating the noise variance from the autoregressive (AR) signal corrupted by the additive white noise. This method first estimates the AR parameters from the high-order Yule-Walker equations, and then uses these AR parameters to estimate the noise variance from the low-order Yule-Walker equations. The method is used in a speech enhancement application where its performance is studied for stationary as well as nonstationary noise conditions. The results are found to be encouraging View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Architectural strategies for an application-specific synchronous multiprocessor environment

    Page(s): 265 - 284
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1956 KB)  

    A multiprocessor architecture is presented that is suited for the customized and automated VLSI realization of complex low-to-medium-speed digital signal processing applications. The proposed architecture is constructed from a set of flexible and parameterizable data paths, a selection of powerful control units (for decision-making tasks), and a number of protocols for fast interprocessor communication. The flexible nature of this system allows for an efficient hardware realization utilizing the inherent parallelism of a particular application. The effectiveness of the approach is substantiated with the synthesis of several test vehicles, such as a pitch-extraction algorithm for speech, in terms of the defined architecture View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An improved fast Fourier transform algorithm using mixed frequency and time decimations

    Page(s): 290 - 292
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (200 KB)  

    An improved FFT (fast Fourier transform) algorithm combining both decimations in frequency and in time is presented. Stress is placed on a derivation of general formulas for submatrices and multiplicands. Computational efficiency is briefly discussed View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Optimal pre- and postfilters for multichannel signal processing

    Page(s): 287 - 289
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (264 KB)  

    The authors present closed-form solutions to the problem of optimal pre-filtering and postfiltering for transmission of an N-dimensional signal though a communications channel composed of M identical subchannels with uncorrelated additive noise. The optimal system may include both Hadamard and Karhunen-Loeve transforms as factors of the prefilters and postfilters. The error improvement due to optimal prefiltering and postfiltering can be as high as 6 dB, as compared to optimal postfiltering only, for a first-order Gauss-Markov signal, for example View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.

Aims & Scope

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

Full Aims & Scope