The implementation of digital filtering algorithms using pipelined vector processors is investigated. Modeling of vector processors and vectorization methods are explained, and then the performances of several impleme...
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The implementation of digital filtering algorithms using pipelined vector processors is investigated. Modeling of vector processors and vectorization methods are explained, and then the performances of several implementation methods are evaluated based on the model. Vector processor implementation of FIR filtering algorithms using the outer product method and the indirect convolution method is evaluated. Recursive and adaptive filtering algorithms, which lead to dependency problems in direct vector processor implementations, are implemented very efficiently using a newly developed vectorization method. The proposed method computes multiple output samples at a time, making the vector length independent of the filter order. Illustrative examples comparing theoretical results with Cray X-MP simulation results are included.
The phase reconstruction problem in the trispectrum domain is addressed in this correspondence. In particular, three different reconstruction algorithms are discussed and their performance is evaluated. The first two ...
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The phase reconstruction problem in the trispectrum domain is addressed in this correspondence. In particular, three different reconstruction algorithms are discussed and their performance is evaluated. The first two algorithms are recursive in nature and are shown to be sensitive to initial condition estimates.
Necessary and sufficient conditions are developed for a pair of transfer functions to have magnitude responses which sum to a constant. If, in addition, the transfer function pair is constrained to be all-pass complem...
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Necessary and sufficient conditions are developed for a pair of transfer functions to have magnitude responses which sum to a constant. If, in addition, the transfer function pair is constrained to be all-pass complementary, the transfer function pair so obtained is well suited for loudspeaker crossover applications. Such crossover systems are characterized by a pair of transfer functions which exhibit the same phase angle at all frequencies. This property has twofold implications: i) in active crossover biamplified audio systems, whereby the low frequencies and high frequencies are reproduced using separate power amplifiers, in-phase crossover transfer functions require less power in subsequent amplifier stages than other designs to achieve a given acoustic sound pressure level; and ii) the summed acoustic magnitude response is least sensitive to noncoincidental mounting of the low- and high-frequency transducers when the crossover transfer functions exhibit the same phase angle at all frequencies. The class of transfer functions realizable is quite wide, and includes squared versions of Butterworth, Chebyshev, and elliptic transfer functions of all orders.
This tutorial paper describes the methods for constructing fast algorithms for the computation of the discrete Fourier transform (DFT) of a real-valued series. The application of these ideas to all the major fast Four...
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This tutorial paper describes the methods for constructing fast algorithms for the computation of the discrete Fourier transform (DFT) of a real-valued series. The application of these ideas to all the major fast Fourier transform (FFT) algorithms is discussed, and the various algorithms are compared. We present a new implementation of the real-valued split-radix FFT, an algorithm that uses fewer operations than any other real-valued power-of-2-length FFT. We also compare the performance of inherently real-valued transform algorithms such as the fast Hartley transform (FHT) and the fast cosine transform (FCT) to real-valued FFT algorithms for the computation of power spectra and cyclic convolutions. Comparisons of these techniques reveal that the alternative techniques always require more additions than a method based on a real-valued FFT algorithm and result in computer code of equal or greater length and complexity.
The three‐dimensional Schr?dinger equationinverse scattering problem is solved using an orthogonalization approach. The plane waves propagating in free space are orthogonalized with respect to an inner product define...
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The three‐dimensional Schr?dinger equationinverse scattering problem is solved using an orthogonalization approach. The plane waves propagating in free space are orthogonalized with respect to an inner product defined in terms of a Jost operator. The resulting integral equation is identical to the generalized Gel’fand–Levitan equation of Newton, although the present derivation is simpler and more physical than that of Newton. Newton’s generalized Marchenko equation is derived from the defining integral equation for the Jost operator. These integral equations are shown to be solved by fast algorithms derived directly from the properties of their solutions. This paper thus presents a simple interpretation of Newton’s two integral equations, two fast algorithms for solving these integral equations, and relations between the various approaches. This is a generalization of previously obtained results, which are also reviewed here, for the one‐dimensional inverse scattering problem.
A new computationally efficient identification procedure is proposed for a non-Gaussian white noise driven linear, time-invariant, non-minimum phase system. The method is based on the idea of computing the complex cep...
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A new computationally efficient identification procedure is proposed for a non-Gaussian white noise driven linear, time-invariant, non-minimum 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 system's impulse response are computed directly from higher-order cumulants via least-squares solution. The method is flexible enough to reconstruct the minimum and maximum phase components of the impulse response of MA, AR or ARMA systems without any prior knowledge of the type of the system. It does not require model order selection criteria and is shown to provide estimates with small bias and variance even with "short" length data records.
We consider multi-armed bandit problems with switching cost, and present asymptotically optimal adaptive allocation schemes. We illustrate an optimal allocation scheme for a large class of distributions which includes...
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We consider multi-armed bandit problems with switching cost, and present asymptotically optimal adaptive allocation schemes. We illustrate an optimal allocation scheme for a large class of distributions which includes members of the exponential family.
In this paper we propose a new method for non-minimum phase wavelet estimation and deconvolution based on non-causal autoregressive (AR) modeling of third moments of the output seismic trace. The magnitude and phase r...
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In this paper we propose a new method for non-minimum phase wavelet estimation and deconvolution based on non-causal autoregressive (AR) modeling of third moments of the output seismic trace. The magnitude and phase response of the wavelet are expressed in terms of the parameters of the noncausal AR model. The set of the AR parameters is obtained by solving an overdetermined linear system of equations which is formed using third moment estimates of the output data. A comparison of the non-causal AR bispectrum approach against more conventional approaches for wavelet magnitude and phase reconstruction is a major component of the paper. It is demonstrated by means of standard example that the proposed procedure achieves improved performance.
There are many significant applications of nonlinear adaptive digital filters such as the cancellation of echoes and intersymbol interference, the equalization of transmission channels, adaptive noise cancellation and...
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There are many significant applications of nonlinear adaptive digital filters such as the cancellation of echoes and intersymbol interference, the equalization of transmission channels, adaptive noise cancellation and design of optimal predictors in communication systems. In this paper, we introduce a class of efficient architectures for adaptive quadratic digital filters based on the LMS algorithm and on rank compressed lower-upper (LU) triangular decomposition method. The architectures exhibit high parallelism as well as modularity and regularity. They are mapped into parallel pipeline and systolic array implementations and are evaluated on hardware cost (in bits), and data throughput delay.
The theory of set membership (SM) identification is formulated, and applied to linear prediction (LP) analysis of speech. The LP parameters of a simulated vowel are identified as an illustration. The SM strategy resul...
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The theory of set membership (SM) identification is formulated, and applied to linear prediction (LP) analysis of speech. The LP parameters of a simulated vowel are identified as an illustration. The SM strategy results in a significant computational savings due to rejection of data which are informationless in the SM sense.
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