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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date Feb 1990

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Displaying Results 1 - 23 of 23
  • Quantization noise, fixed-point multiplicative roundoff noise, and dithering

    Publication Year: 1990 , Page(s): 286 - 300
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (972 KB)  

    The author considers the characteristics of the error resulting when a continuous amplitude signal xn is quantized and then multiplied by a constant multiplier a under fixed-point roundoff arithmetic. It is shown that the overall error of such an operation can be decomposed into two terms: one being a scaled version of the error due to the quantization of xn and the other due to rounding off the product aQ(xn). Exact first- and second-order moments are derived for the quantization error, the roundoff error, and the overall error as a function of the multiplier a and the distribution of xn. Sufficient conditions are given for the quantization error and the roundoff error to be individually uniformly distributed and white up to the first- and second-order moments, and also for them to be mutually uncorrelated. It is also shown that regardless of the probability distribution of the input signal xn, it is always possible to add a suitable dither signal to the input of the system so that both the quantization error and the roundoff error are uniformly distributed, white, and mutually uncorrelated. For Gaussian inputs, the sufficient conditions given are not satisfied View full abstract»

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  • The isolation of undistorted sinusoids in real time

    Publication Year: 1990 , Page(s): 360 - 364
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (468 KB)  

    A single-input, multiple-output adaptive filter that takes a multisinusoidal input signal and produces the individual sinusoids at each of the outputs is described. The individual outputs have identical amplitudes and phases to the components of the original input signal. The filter can be adapted in real time using approximate Gauss-Newton updates without requiring any matrix operations or divisions. The filter is an extension of a cascade adaptive IIR (infinite impulse response) notch filter reported previously. The filter is very amenable to being realized using custom ICs or DSP (digital signal processing) ICs, as no divisions are required and most of the filter is composed of identical biquads (which are very insensitive to the errors caused by finite accuracy in fixed-point implementations). The structure is especially good at isolating small-amplitude sinusoids when in the proximity of much larger sinusoids of unknown frequencies View full abstract»

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  • A novel implementation of a chirp Z-transform using a CORDIC processor

    Publication Year: 1990 , Page(s): 352 - 354
    Cited by:  Papers (15)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (292 KB)  

    An efficient implementation of the chirp Z transform (CZT) using a CORDIC (coordinate rotation digital computer) processor is presented. In particular, it is shown that a scaling operation in the CZT algorithm can be conveniently implemented with a norm correction (normalization) computation, which is often considered as an overhead in the CORDIC algorithm. Furthermore, since the desired frequencies of CZT are specified before computation, it is possible to reduce the total number of CORDIC iterations by finding a most economical representation of the angle in terms of the elementary CORDIC rotation angles. A simple suboptimal solution is proposed to solve this difficult optimization problem. This implementation is most effective when very few complex frequencies on the Z plane are to be evaluated via CZT View full abstract»

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  • A new method for adaptive time delay estimation for non-Gaussian signals

    Publication Year: 1990 , Page(s): 209 - 219
    Cited by:  Papers (29)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (700 KB)  

    A novel adaptive scheme for time delay estimation is introduced for signal environments where the signal is non-Gaussian and the additive noise sources are spatially correlated Gaussian with unknown power spectrum characteristics. This scheme is based on parametric modeling between two sensor measurements and employs higher-order statistics (third- or fourth-order) of the data. It is demonstrated by means of extensive simulations that this scheme works well for both stationary and nonstationary cases. As expected, it outperforms the cross-correlation-based gradient method for time-delay adaptation in spatially correlated Gaussian noises. The proposed scheme is compared to the overdetermined recursive instrumental variable method and is shown to exhibit substantially less computational complexity View full abstract»

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  • Control-theoretic design of the LMS and the sign algorithms in nonstationary environments

    Publication Year: 1990 , Page(s): 253 - 259
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (500 KB)  

    The feedback structure of the LMS (least mean squared) algorithm proposed by B. Widrow et al. (Proc. IEEE, vol.64, p.1151-62, Aug. 1976) is reexamined from a control system design viewpoint. The minimization of the misadjustments due to gradient noise and lag can then be recast as the disturbance rejection and tracking problems in control. A frequency-domain approach to the latter problems that has the advantages of transparency, ease of computation, and generality compared with the time-domain approach previously used is presented. With the same set of assumptions of white input and a Markovian plant, it is shown that the optimum step size obtained by the present approach is identical to that obtained by Widrow. Applying the same approach, the optimum step size of a simplified version of the LMS algorithm-the sign algorithm-is derived for the case when the plant is slowly varying and the input signals are Gaussian View full abstract»

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  • Multidelay block frequency domain adaptive filter

    Publication Year: 1990 , Page(s): 373 - 376
    Cited by:  Papers (60)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (268 KB)  

    A flexible multidelay block frequency domain (MDF) adaptive filter is presented. The distinctive feature of the MDF adaptive filter is to allow one to choose the size of an FFT tailored to the efficient use of the hardware, rather than the requirements of a specific application. The MDF adaptive filter also requires less memory and thus reduces the hardware requirements and cost. In performance, the MDF adaptive filter introduces smaller block delay and is faster, making it ideal for a time-varying application such as modeling an acoustic path in a teleconference room. This is achieved by using a smaller block size, updating the weight vectors more often, and reducing the total execution time of the adaptive process. The MDF adaptive filter compares favorably to other frequency-domain adaptive filters when its adaptation speed and misadjustment are tested in computer simulations View full abstract»

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  • Generalized target description and wavelet decomposition [sonar]

    Publication Year: 1990 , Page(s): 350 - 352
    Cited by:  Papers (10)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB)  

    Generalized target description by means of colored bright spots is very attractive for recognition or classification tasks in active sonar applications. In the present work, it is shown how such a description can be achieved directly from the impulse response. It turns out that the resulting procedure is closely connected with the technique of wavelet decomposition. It is shown that these two techniques share important common features concerning both the structure of their privileged analysis tools (transmitted signal or analyzing wavelet) and the way in which they obtain relevant information on a system under investigation View full abstract»

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  • Direction-of-arrival estimation for wide-band signals using the ESPRIT algorithm

    Publication Year: 1990 , Page(s): 317 - 327
    Cited by:  Papers (28)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (888 KB)  

    A novel direction-of-arrival estimation algorithm is proposed that applies to wideband emitter signals. A sensor array with a translation invariance structure is assumed, and an extension of the ESPRIT algorithm for narrowband emitter signals is obtained. The emitter signals are modeled as the stationary output of a finite-dimensional linear system driven by white noise. The array response to a unit impulse from a given direction is represented as the impulse response of a linear system. The measured data from the sensor array can then be seen as the output of a multidimensional linear system driven by white noise sources and corrupted by additive noise. The emitter signals and the array output are characterized by the modes of the linear system. The ESPRIT algorithm is applied at the poles of the system, the power of the signals sharing the pole is captured, and the effect of noise is reduced. The algorithm requires no knowledge, storage, or search of the array manifold, as opposed to wideband extensions of the MUSIC algorithm. This results in a computationally efficient algorithm that is insensitive to array perturbations. Simulations are presented comparing the wideband and ESPRIT algorithm to the modal signal subspace method and the coherent signal subspace method View full abstract»

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  • Pipelined algorithm for LS FIR filters with symmetric impulse response

    Publication Year: 1990 , Page(s): 260 - 270
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (660 KB)  

    A novel highly parallel algorithm is derived for LS FIR (least-squares finite impulse response) multichannel filters with symmetry in their impulse response. The derived algorithm requires O (p)+O(N) computing time and can be performed on a linear array of O(p) processors, p being the order of the corresponding filter and N the number of data points. Therefore, the computational time is reduced by an order of magnitude compared to existing Levinson-type algorithms. The problem is treated for the unknown statistics case. A by-product of the derived algorithm in the single-channel case is a concurrent algorithm for MEM (maximum entropy method) power spectral estimation. The proposed algorithm features high modularity and localized communication requirements, and lends itself to VLSI integration. A scheme that realizes the power spectral estimation algorithm in a systolic/wavefront mode is also proposed View full abstract»

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  • Focused wide-band array processing by spatial resampling

    Publication Year: 1990 , Page(s): 356 - 360
    Cited by:  Papers (42)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (404 KB)  

    The authors present a novel approach for coherently focusing wideband data received by a linear array of sensors. The proposed preprocessing consists of adjusting the spatial sampling rate, or spatially resampling the array outputs as a function of temporal frequency so that broadband sources are aligned in the spatial frequency domain. Spatial resampling has the advantage that it can reduce each wideband source in multigroup multiple-source scenarios to essentially a rank-one representation without preliminary estimates or a priori knowledge of the spatial distribution of the sources. Preliminary simulations performed using a well-known digital interpolation method to implement spatial resampling with the MUSIC spectral estimator have yielded promising results View full abstract»

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  • A linear predictive HMM for vector-valued observations with applications to speech recognition

    Publication Year: 1990 , Page(s): 220 - 225
    Cited by:  Papers (39)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (512 KB)  

    The authors describe a new type of Markov model developed to account for the correlations between successive frames of a speech signal. The idea is to treat the sequence of frames as a nonstationary autoregressive process whose parameters are controlled by a hidden Markov chain. It is shown that this type of model performs better than the standard multivariate Gaussian HMM (hidden Markov model) when it is incorporated into a large-vocabulary isolated-word recognizer View full abstract»

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  • Derivation of new and existing discrete-time Kharitonov theorems based on discrete-time reactances [digital signal processing]

    Publication Year: 1990 , Page(s): 277 - 285
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (708 KB)  

    The author first uses a discrete-time reactance approach to give a second proof of existing discrete-time Kharitonov-type results (1979). He then uses the same reactance language to derive a new discrete-time Kharitonov-type theorem which, in some sense, is a very close analog to the continuous-time case. He also points out the relation between discrete-time reactances and the technique of line-spectral pairs (LSP) used in speech compression View full abstract»

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  • Regular sets and rank order processors

    Publication Year: 1990 , Page(s): 241 - 246
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (572 KB)  

    Rank order processors (ROPs) can be specified in terms of regular sets and consequently studied as finite-state automata or sequential machines. A necessary and sufficient condition for a regular set to represent an ROP is given. Examples are given to illustrate the advantages of such a specification in terms of a primary focus on root signals and input-output relations. One result is a practical equivalent of the recursive median smoother that is easier to compute. Another is a demonstration that some ROPs, although not implementable as finite autoregressive algorithms involving only the input-output signal values, are implementable as finite state sequential machines, e.g. those with a modified stack filter structure. The demonstration is constructive and coincidentally shows how the idea of a nondeterministic automaton relates to the subject matter View full abstract»

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  • Application-specific architectural methodologies for high-throughput digital signal and image processing

    Publication Year: 1990 , Page(s): 339 - 349
    Cited by:  Papers (18)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1108 KB)  

    An attempt has been made to classify the architectural styles suited for application-specific high-throughput DSP (digital signal processing) applications. In principle, they can be classified between the two extremes of cooperating multiplexed data paths and regular arrays which adopt hard-wired control flow with low hardware sharing. Three DSP subclasses have been identified according to the required sample rate in combination with the regularity and modularity of the flowgraph and the types of operation which are involved, such as straightforward arithmetic or complex decision making with loops and branches. Correspondingly, three architectural strategies have been identified as efficient candidates for realization, namely, microcoded multiprocessors, cooperating multiplexed data paths, and regular arrays. Architectures exploiting large hardware-sharing ratio's which are more suited for the back-end modules in the target DSP system are investigated. Realistic test vehicles are discussed in order to evaluate the alternatives. Moreover, guidelines have been extracted to select such an ASIC (application-specific integrated circuit) architecture. These are dependent on the required throughput, the structure of the algorithm, and the corresponding signal flowgraph View full abstract»

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  • Estimation of decay rates of transient signals

    Publication Year: 1990 , Page(s): 370 - 372
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (248 KB)  

    The problem of estimating time constants of multiple exponentials in additive white Gaussian noise is investigated. It is shown that existing linear-production-based techniques, which perform very well for decay-rate estimation of signals with different frequencies, fail when applied to the case of decaying exponentials at a single frequency. An alternate, minimum weighted-norm method is proposed. It is shown that this new technique achieves acceptable estimates of time constants at a signal-to-noise ratio of 20 dB, a significant improvement over existing techniques View full abstract»

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  • Design of IIR digital filters with arbitrary log magnitude function by WLS techniques

    Publication Year: 1990 , Page(s): 247 - 252
    Cited by:  Papers (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (452 KB)  

    The authors propose a technique for designing IIR (infinite impulse response) digital filters to have an arbitrary log magnitude frequency response. The technique is based on an iterative weighted least-squares (WLS) approach in the frequency domain. A weight updating procedure is introduced to obtain a nearly optimal approximation to the given log magnitude function in the least-squares sense. The weighting function is updated using the results of the previous iteration in such a way that the weighted error approximates the log magnitude error. Filter coefficients at each iteration are efficiently computed using a fast recursive algorithm for a set of linear equations derived from the WLS problem. Several design examples demonstrate the rapid convergence of the design algorithm. The algorithm is extended to equiripple approximation by means of a minor modification of the weight updating procedure View full abstract»

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  • Efficient, numerically stabilized rank-one eigenstructure updating [signal processing]

    Publication Year: 1990 , Page(s): 301 - 316
    Cited by:  Papers (39)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1144 KB)  

    Rank-one eigenvalue decomposition (EVD) updating is well suited to the problem of tracking time-varying subspaces. Previously published rank-one EVD updating algorithms suffer from a linear buildup of roundoff error, which makes them impractical for a large number of recursive updates. In the present work, the authors develop a numerical stabilization technique, which eliminates the error buildup problem in a computationally efficient manner and makes the rank-one EVD update a practical numerical tool for online computation. A simplified eigenvalue iteration is also given. It reduces the complexity of the algorithm somewhat as well as the computation time. Simulations are presented to illustrate numerical performance View full abstract»

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  • Coherent wide-band ESPRIT method for directions-of-arrival estimation of multiple wide-band sources

    Publication Year: 1990 , Page(s): 354 - 356
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (252 KB)  

    A strategy extending the coherent signal-subspace method (CSM) is proposed for estimating directions of arrival of multiple wideband sources. The proposed method, coherent wideband ESPRIT, extends the ESPRIT algorithm to a framework based on the CSM. A simulation example is provided to illustrate the effectiveness of the proposed method View full abstract»

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  • Computing discrete Fourier transform on a rectangular data array

    Publication Year: 1990 , Page(s): 271 - 276
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (400 KB)  

    The authors generalize the weighted redundancy transform (WRT) algorithm for computing the multidimensional discrete Fourier transform (DFT) in the case in which the sample size (blocklength) is not the same on every axis. The proposed algorithm, like the WRT algorithm, is based on the one-dimensional fast Fourier transform (FFT) and, compared to the traditional ways of computing the multidimensional DFT, offers substantial savings in the number of one-dimensional FFT procedure calls. While the algorithm is applicable to transforms of any dimensions, only the two-dimensional case is explored in detail View full abstract»

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  • Using a ring parallel processor for hidden Markov model training

    Publication Year: 1990 , Page(s): 366 - 369
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (372 KB)  

    The authors present a novel solution to the computationally intensive problem of training HMMs (hidden Markov models) by showing how a bidirectional ring multiprocessor can achieve potentially optimal speed in the training of left-to-right HMMs. The solution presented avoids interprocessor communications problems in the HMM training algorithm. This is achieved by having the ring multiprocessor calculate the α's (from the forward-backward training algorithm) in a clockwise direction around the ring, and the β's in a counterclockwise direction at the same time. The two sets of calculations are designed so that when this stage of the iteration is completed, each processor will have all of the data needed for the next stage of the iteration already stored locally View full abstract»

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  • Fast Fourier transforms over finite groups by multiprocessor systems

    Publication Year: 1990 , Page(s): 226 - 240
    Cited by:  Papers (9)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1168 KB)  

    The authors present a method for an optimal implementation of general discrete Fourier transform (GDFT) algorithms over finite groups (Abelian and non-Abelian) in a multiprocessor environment. Tradeoffs between hardware complexity/speed and computation time are investigated for different multiprocessor implementations with local nonshared memories (unibus, complete communication network). Formulas are presented for the number of arithmetic operations, for the number of interprocessor data transfers, and for the number of communication links among the processors View full abstract»

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  • Nonuniform image motion estimation from noisy data

    Publication Year: 1990 , Page(s): 364 - 366
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (260 KB)  

    Image motion estimation is viewed as a problem in nonlinear demodulation. The motion D(x) modulates the intensity of the previous frame s(x) to result in the intensity of the present frame s(x-D(x)). An iterative algorithm based on the generalized maximum likelihood criterion is developed and implemented to demodulate D(x ). An advantage of this scheme is the incorporation of the covariance function matrix, which is specifically useful in the stationary case to determine the integration region αx. Simulation demonstrated the ability of the algorithm to deal with cases such as Markov-2 motion View full abstract»

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  • Structural processing of waveforms as trees

    Publication Year: 1990 , Page(s): 328 - 338
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (892 KB)  

    Waveforms can be represented symbolically in such a manner that their underlying global structural composition is emphasized. The authors consider one such symbolic representation: a computer data structure, known as the relational tree, that describes the relative size and placement of peaks and valleys in a waveform. To analyze the relational tree, the authors examine various distance measures which serve as tree metrics. These metrics make it possible to cluster groups of trees by their proximity in tree space. In traditional cluster analysis, linear discriminants are used to reduce vector space dimensionality and to improve cluster performance. A tree transformation accomplishes this same goal operating on relational trees in a tree space. By combining these concepts, the authors have developed a waveform recognition system. This system recognizes waveforms even when they have undergone a monotonic transformation of the time axis. The system performs well with high signal-to-noise ratios, but further refinements are necessary for a working waveform interpretation system. The technique is illustrated by application to seismic and electrocardiographic data 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