Browsing by Author "Parks, Thomas W."

Now showing 1 - 12 of 12
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    A prime factor FFT algorithm using high speed convolution
    (1977) Kolba, Dean Paul; Parks, Thomas W.
    Two recently developed ideas, the conversion of a DFT to convolution and the implementation of short convolutions with a minimum of multiplications, are combined to give efficient algorithms for long transforms. Three transform algorithms are compared in terms of number of multiplications and additions. Timing for a prime factor FFT algorithm using high speed convolution, which was programmed for an IBM 37 and an 88 microprocessor, is presented.
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    A ray model for head waves in a fluid-filled borehole
    (1982) Scheibner, David James; Parks, Thomas W.; Johnson, Don H.; Figueiredo, Rui J. P. de
    A model, suggested by tbe ray expansion of Roever et al., is constructed to rapidly generate the compressional and shear refracted arrivals, known as head waves, received from a point source on the axis of an ideal fluid-filled borehole. An impulse response is derived, and its frequency characteristics are investigated. The waves are compared to those obtained by the real axis integration method of Tsang and Rader, which results in a complete waveform, including the modes as well as the refracted arrivals. The ray model gives accurate results for the compressional head wave. The shear region of the complete waveform contains strong modal interference, making it difficult to evaluate the quality of the ray model shear wave. A useful filter results from insight provided by the basic structure of the model. This filter can be used to estimate the borehole diameter or formation compressional velocity. It can also remove the second and later compressional arrivals, thus providing an accurate estimate of the source pulse and and a relatively uncorrupted vie of the initial arrival in the shear region.
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    A study and comparison of digital speech systems
    (1976) Quilici, Timothy Jon; Parks, Thomas W.
    Several of the main digital communication systems in use today are surveyed. Particular attention is paid to the transmission of the speech signal. The autocorrelation and covariance formulations of linear predictive coding are compared for varying length segments of synthetically produced vowel sounds. The two formulations are also compared on vowel sounds contaminated with varying amounts of additive white noise. The two formulations were found to be roughly equivalent in their abilities to extract Z-plane pole locations from a signal. The autocorrelation method was shown to be computationally faster, however. A new pole extraction algorithm is tested in which the covariance method is used on several short over-lapping segments and a "cluster" decision scheme is used. This computationally expensive algorithm proves to yield better results than a straight forward application of the covariance method.
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    Computational methods in the design of linear control systems
    (1980) Kontos, Athanasios V.; Pearson, J. B.; Parks, Thomas W.; Figueiredo, Rui J. P. de
    This thesis considers the problem of computing controllers for multivariable systems. System representation is in terms of polynomial matrices and two algorithms are presented which are shown to be useful in the design of controllers for such systems. These algorithms are: i) Factorization of a polynomial matrix, and ii) Computation of a unimodular matrix U satisfying the relation [A B]U = I 1, where A and B are left coprime polynomial matrices. These algorithms do not involve numerically unsatisfactory Euclidean type operations. It is shown that the two algorithms can be used to compute solutions to the system stabilization problem and to the model matching problem. The Regulator Problem with Internal Stability (RPIS) is also discussed, and under certain assumptions it is shown how solutions can be computed.
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    Digital filters with thinned numerators
    (1980) Boudreaux-Bartels, Gloria Faye; Parks, Thomas W.; Johnson, Don H.; Burrus, C. S.
    An algorithm is described for designing digital filters that require few multiplies to produce good frequency response. The process of reducing the number of multiplies needed to implement a digital filter is called thinning. The thinning algorithm uses Dynamic Programming techniques to optimally approximate a desired Finite Impulse Response (FIR) filter with another FIR filter that requires significantly fewer non-zero coefficients to produce similar frequency response characteristics. The effects of coefficient quantization and finite-precision computer arithmetic upon the thinned filter structure are also described. Examples of thinned narrowband, broadband, lowpass, and bandpass filters are given. Several of these thinned filters require fewer than one-third the number of multiplies required for the corresponding unthinned filter while still retaining desirable frequency response characteristics.
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    Estimation of repetition rate from signal and texture features
    (1983) Tagare, Hemant D.; Figueiredo, Rui J. P. de; Parks, Thomas W.; Dufour, Reginald J.
    This thesis develops relevant definitions and a theoretical basis for estimating the repetition rate of a random repetitive signal. The repetition rate is estimated by looking for repetition amongst local features of the signals. These features have to satisfy a uniqueness condition, and we have shown that the derivatives of a signal constitute a set of such features. The estimator has been shown to be asymptotically unbiased. The estimation algorithm can not only be tuned to the waveshape information of the signal (by a proper choice of features), but also to the extent of non-stationarity expected in the input signal class. A set of features has been obtained for applying this algorithm to repetitive textured images and voiced speech signals. Vith these features, it has been possible to extract the repetition rate in both the above classes of signals. In the case of voiced speech this rate corresponds to its pitch.
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    Estimation of the parameters of all-pole sequences corrupted by additive observation noise
    (1983) McGinn, Darcy; Johnson, Don H.; Thompson, James R.; Parks, Thomas W.
    Ordinary Least Squares procedures and the equivalent Yule-Walker formulation result in biased estimates of all-pole model parameters when applied to noise corrupted all-pole sequences. This bias is shown to be proportional to the inverse of the signal-to-noise ratio. The algorithm investigated applies an autocorrelation-like operation to the noise corrupted all-pole sequence which increases the signal-to-noise ratio but preserves the pole locations. This operation is applied recursively until acceptable signal-to-noise ratio is obtained. The all-pole parameters are then estimated from the high signal-to-noise ratio sequence using an Ordinary Least Squares estimator. The improvement in signal-to-noise ratio varies for different modes in an allpole sequence with modes corresponding to pole locations close to the unit circle showing the most improvement. A signal-to-noise ratio cutoff exists below which no improvement in signal-to-noise ratio is possible for a given mode. This cutoff is dependent on the radius of the poles of the mode and goes to zero as the pole approaches the unit circle. The signal-to-noise ratio cutoff also corresponds to the point at which the mode’s peak spectral value just equals the level of the noise floor. Estimates of the poles from the high signal-to-noise ratio sequences show reduction in the noise induced bias concomitant with the increased signal-to-noise ratio. Correlations of up to four times are shown to be advantageous. The sensitivity of the successive autocorrelation algorithm to a white observation noise assumption is found to be small. With long correii lation length signals, such as sinusoids, unbiased low variance estimates of the parameters are possible at signal-to-noise ratios of as low as .1.
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    Estimation techniques in non-stationary renewal processes
    (1980) Swami, Ananthram; Johnson, Don H.; Parks, Thomas W.; Thompson, James R.
    The multiplicative intensity model for the intensity function u(t;N(t);w) = v(t)r(t - of a self-exciting point process is analyzed in terms of the distortion of v(t) by the channel r(x). A convenient and common method of presenting point process data, the Post Stimulus Histogram is shown to be related to the ensemble average of the intensity process and hence incorporates stimulus v() as well as refractory r() related effects. This quantity is not usually amenable to closed-form representation. We propose an approximation to the PST which is reasonably good under specified conditions. A maximum likelihood estimator of r(x), where v(t) is known, is derived. A maximum likelihood estimator of v(t), given r(x), is also derived. This estimator is meaningful only when the signal v(t) is known to be periodic. The M.L. Estimator compensates for relative dead-time effects. We propose an iterative dead-time processor, which operating on the histogram obtained from the M.L. Estimate, partially compensates for absolute dead-time effects. The performance of these estimators is compared with those of other procedures. Applications to spike trains recorded from auditory neurons are discussed.
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    Fast algorithms for DFT and convolution
    (1978) Merchant, Gulamabbas A.; Parks, Thomas W.
    In this thesis, a detailed analysis of sufficient conditions for existence of unique multidimensional linear and multidimensional non-linear Index map has been presented, along with a new Index representation. The recent Ideas of converting Discrete Fourier Transform to convolution and Implementing convolution efficiently, have been combined to give two algorithms viz. Nested Fourier Algorithm (NFA -- using linear multidimensional map) and Index Fourier Algorithm (IFA using a non-linear Index map). The two algorithms have been compared for the amount of arithmetic computations required. The algorithms have been Implemented In FORTRAN on IBM 37/155 and their execution timings have been compared.
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    Finite register effects in block digital filters
    (1979) Loeffler, Charles M.; Burrus, C. Sidney; Johnson, Don H.; Parks, Thomas W.
    An equivalence relationship between convolution block filters and state block filters is established. The single-output basis filter for each convolution block filter is found. It is shown that the average round-off noise variance at the output of all block filters is inversely proportional to the block length. A sufficient condition on the block length is found that guarantees complete suppression of the limit cycles in the block filter. A minimum output noise variance block structure is found that, when compared to its single variate counterpart, has a lower output noise variance and is not as computationally complex to implement.
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    Microprocessor control of a combined assist system for the profound support of the failing heart
    (1979) Philippe, Edouard A.; Clark, John W.; Walker, William F.; Parks, Thomas W.
    Conventional non-invasive mechanical circulatory assist methods such as intra-aortic balloon pumping are of little effectiveness in cases of severe ventricular failure. The proposed study is concerned with the development and testing, on a mock circulatory loop and in a series of dog experiments, of an automated combined assist system for the profound support of the failing heart. This system is mildly invasive and consists of the synergistic use of intra-aortic balloon pumping and partial veno-arterial bypass. Previous in vivo studies using this system with no automation have shown that it is quite effective in achieving its clinical objectives, and our purpose is to investigate the ability of the control systems developed herein to match the optimal manual setting of the assist system. The end result of the project is a portable compact automatically controlled ventricular assist system that can be rapidly and easily instituted, thus improving the chance of survival and recovery of acute myocardial infarction and cardiogenic shock victims.
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    The effect of coherant signals on the capability of array processing algorithms to resolve source bearings
    (1982) De Graaf, Stuart Randall; Johnson, Don H.; Parks, Thomas W.; Wilson, William L.
    In the ocean environment, where multipath propagation of sound is common, the acoustic signals incident on a passive sonar array are often coherent. Cox has demonstrated that the capability of minimum energy adaptive beamforming to resolve the source bearings of incoherent signals is superior to that of classical beamforming. Seligson observed that when the signal wavefronts deviate from their assumed shape, the adaptive beamformer can be inferior to the classical beamformer in this regard. We study the effect of coherent signals on the capability of classical, adaptive, and linear predictive array processing algorithms to resolve source bearings. For a linear array of equally spaced sensors, we demonstrate the superior resolution capability of the linear predictive algorithm, and the significant effect of signal coherence on all three processing algorithms. We demonstrate the value of utilizing prediction elements in the center of the array to resolve closely-spaced signal bearings. Finally, we investigate the sensitivity of the processing algorithms to imperfections introduced into the correlation matrix by finite averaging, and establish the trade-off that exists between this sensitivity and the capability to resolve closely-spaced signal bearings.