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Showing papers on "Impulse response published in 2004"


Book ChapterDOI
01 Aug 2004
TL;DR: In this article, a model for the DGP of a set of economic time series variables can be constructed, which can be used for analyzing the dynamic interactions between the variables when such a model is available, usually done by tracing the effect of an impulse in one of the variables through the system.
Abstract: Introduction In the previous chapter we have seen how a model for the DGP of a set of economic time series variables can be constructed. When such a model is available, it can be used for analyzing the dynamic interactions between the variables. This kind of analysis is usually done by tracing the effect of an impulse in one of the variables through the system. In other words, an impulse response analysis is performed. Although this is technically straightforward, some problems related to impulse response analysis exist that have been the subject of considerable discussion in the literature. As argued forcefully by Cooley & LeRoy (1985), vector autoregressions have the status of “reduced form” models and therefore are merely vehicles to summarize the dynamic properties of the data. Without reference to a specific economic structure, such reduced-formVAR models are difficult to understand. For example, it is often difficult to draw any conclusion from the large number of coefficient estimates in a VAR system. As long as such parameters are not related to “deep” structural parameters characterizing preferences, technologies, and optimization behavior, the parameters do not have an economic meaning and are subject to the so-called Lucas critique. Sims (1981, 1986), Bernanke (1986), and Shapiro & Watson (1988) put forward a new class of econometric models that is now known as structural vector autoregression (SVAR) or identified VAR . Instead of identifying the (autoregressive) coefficients, identification focuses on the errors of the system, which are interpreted as (linear combinations of) exogenous shocks. In the early applications of Sargent (1978) and Sims (1980), the innovations of the VAR were orthogonalized using a Choleski decomposition of the covariance matrix.

186 citations


Journal ArticleDOI
TL;DR: A reconstruction method is demonstrated that allows the optical absorption distribution of a sample to be reconstructed without knowing the impulse response of the ultrasonic transducer, and results demonstrated that the reconstructed images agreed well with the original phantom samples.
Abstract: The impulse response of the ultrasonic transducer used for detection is crucial for photoacoustic imaging with high resolution. We demonstrate a reconstruction method that allows the optical absorption distribution of a sample to be reconstructed without knowing the impulse response of the ultrasonic transducer. A convolution relationship between photoacoustic signals measured by an ultrasound transducer and optical absorption distribution is developed. Based on this theory, the projection of the optical absorption distribution of a sample can be obtained directly by deconvolving the recorded PA signal originating from a point source out of that from the sample. And a modified filtered back projection algorithm is used to reconstruct the optical absorption distribution. We constructed a photoacoustic imaging system to validate the reconstruction method and the experimental results demonstrated that the reconstructed images agreed well with the original phantom samples. The spatial resolution of the system reaches 0.3 mm.

182 citations


PatentDOI
TL;DR: In this article, a normalized wavefield is obtained for each trace of the input data set in the frequency domain, which is obtained with respect to the frequency response of a reference trace selected from the set of seismic trace data.
Abstract: A set of seismic trace data is collected in an input data set that is first Fourier transformed in its entirety into the frequency domain. A normalized wavefield is obtained for each trace of the input data set in the frequency domain. Normalization is done with respect to the frequency response of a reference trace selected from the set of seismic trace data. The normalized wavefield is source independent, complex, and dimensionless. The normalized wavefield is shown to be uniquely defined as the normalized impulse response, provided that a certain condition is met for the source. This property allows construction of the inversion algorithm disclosed herein, without any source or source coupling information. The algorithm minimizes the error between data normalized wavefield and the model normalized wavefield. The methodology is applicable to any 3-D seismic problem, and damping may be easily included in the process.

159 citations


Book
31 Jan 2004
TL;DR: In this paper, two-dimensional Fourier Transform of the Shah Function was used for line detection in two dimensions, one dimension and the other dimension of the Hartley Transform of a Shah Function.
Abstract: 1 Introduction.- Summary of the Chapters.- Notation.- Teaching a Course from This Book.- The Problems.- Aspects of Imaging.- Computer Code.- Literature References.- Recommendation.- 2 The Image Plane.- Modes of Representation.- Some Properties of a Function of Two Variables.- Projection of Solid Objects.- Image Distortion.- Operations in the Image Plane.- Binary Images.- Operations on Digital Images.- Reflectance Distribution.- Data Compression.- Summary.- Appendix: A Contour Plot Programt.- Literature Cited.- Further Reading.- Problems.- 3 Two-Dimensional Impulse Functions.- The Two-Dimensional Point Impulse.- Rules for Interpreting Delta Notation.- Generalized Functions.- The Shah Functions iii and 2III.- Line Impulses.- Regular Impulse Patterns.- Interpretation of Rectangle Function of f(x).- Interpretation of Rectangle Function of f(x,y).- General Rule for Line Deltas.- The Ring Impulse.- Impulse Function of f(x,y).- Sifting Property.- Derivatives of Impulses.- Summary.- Literature Cited.- Problems.- 4 The Two-Dimensional Fourier Transform.- One Dimension.- The Fourier Component in Two Dimensions.- Three or More Dimensions.- Vector Form of Transform.- The Corrugation Viewpoint.- Examples of Transform Pairs.- Theorems for Two-Dimensional Fourier Transforms.- The Two-Dimensional Hartley Transform.- Theorems for the Hartley Transform.- Discrete Transforms.- Summary.- Literature Cited.- Further Reading.- Problems.- 5 Two-Dimensional Convolution.- Convolution Defined.- Cross-Correlation Defined.- Feature Detection by Matched Filtering.- Autocorrelation Defined.- Understanding Autocorrelation.- Cross-Correlation Islands and Dilation.- Lazy Pyramid and Chinese Hat Function.- Central Value and Volume of Autocorrelation.- The Convolution Sum.- Computing the Convolution.- Digital Smoothing.- Matrix Product Notation.- Summary.- Literature Cited.- Problems.- 6 The Two-Dimensional Convolution Theorem.- Convolution Theorem.- An Instrumental Caution.- Point Response and Transfer Function.- Autocorrelation Theorem.- Cross-Correlation Theorem.- Factorization and Separation.- Convolution with the Hartley Transform.- Summary.- Problems.- 7 Sampling and Interpolation in Two Dimensions.- What is a Sample?.- Sampling at a Point.- Sampling on a Point Pattern, and the Associated Transfer Function.- Sampling Along a Line.- Curvilinear Sampling.- The Shah Function.- Fourier Transform of the Shah Function.- Other Patterns of Sampling.- Factoring.- The Two-Dimensional Sampling Theorem.- Undersampling.- Aliasing.- Circular Cutoff.- Double-Rectangle Pass Band.- Discrete Aspect of Sampling.- Interpolating Between Samples.- Interlaced Sampling.- Appendix: The Two-Dimensional Fourier Transform of the Shah Function.- Literature Cited.- Problems.- 8 Digital Operations.- Smoothing.- Nonconvolutional Smoothing.- Trend Reduction.- Sharpening.- What is a Digital Filter?.- Guard Zone.- Transform Aspect of Smoothing Operator.- Finite Impulse Response (FIR).- Special Filters.- Densifying.- The Arbitrary Operator.- Derivatives.- The Laplacian Operator.- Projection as a Digital Operation.- Moire Patterns.- Functions of an Image.- Digital Representation of Objects.- Filling a Polygon.- Edge Detection and Segmentation.- Discrete Binary Objects.- Operations on Discrete Binary Objects.- Union and Intersection.- Pixel Morphology.- Dilation.- Coding a Binary Matrix.- Granulometry.- Conclusion.- Literature Cited.- Problems.- 9 Rotational Symmetry.- What Is a Bessel Function?.- The Hankel Transform.- The jinc Function.- The Struve Function.- The Abel Transform.- Spin Averaging.- Angular Variation and Chebyshev Polynomials.- Summary.- Table of the jinc Function.- Problems.- 10 Imaging by Convolution.- Mapping by Antenna Beam.- Scanning the Spherical Sky.- Photography.- Microdensitometry.- Video Recording.- Eclipsometry.- The Scanning Acoustic Microscope.- Focusing Underwater Sound.- Literature Cited.- Problems.- 11 Diffraction Theory of Sensors and Radiators.- The Concept of Aperture Distribution.- Source Pair and Wave Pair.- Two-Dimensional Apertures.- Rectangular Aperture.- Example of Circular Aperture.- Duality.- The Thin Lens.- What Happens at a Focus?.- Shadow of a Straight Edge.- Fresnel Diffraction in General.- Literature Cited.- Problems.- 12 Aperture Synthesis and Interferometry.- Image Extraction from a Field.- Incoherent Radiation Source.- Field of Incoherent Source.- Correlation in the Field of an Incoherent Source.- Visibility.- Measurement of Coherence.- Notation.- Interferometers.- Radio Interferometers.- Rationale Behind Two-Element Interferometer.- Aperture Synthesis (Indirect Imaging).- Literature Cited.- Problems.- 13 Restoration.- Restoration by Successive Substitutions.- Running Means.- Eddington's Formula.- Finite Differences.- Finite Difference Formula.- Chord Construction.- The Principal Solution.- Finite Differencing in Two Dimensions.- Restoration in the Presence of Errors.- The Additive Noise Signal.- Determination of the Real Restoring Function.- Determination of the Complex Restoring Function.- Some Practical Remarks.- Artificial Sharpening.- Antidiffusion.- Nonlinear Methods.- Restoring Binary Images.- CLEAN.- Maximum Entropy.- Literature Cited.- Problems.- 14 The Projection-Slice Theorem.- Circular Symmetry Reviewed.- The Abel-Fourier-Hankel Cycle.- The Projection-Slice Theorem.- Literature Cited.- Problems.- 15 Computed Tomography.- Workingfrom Projections.- An X-Ray Scanner.- Fourier Approach to Computed Tomography.- Back-Projection Methods.- The Radon Transform.- The Impulse Response of the Radon Transformation.- Some Radon Transforms.- The Eigenfunctions.- Theorems for the Radon Transform.- The Radon Boundary.- Applications.- Literature Cited.- Problems.- 16 Synthetic-Aperture Radar.- Doppler Radar.- Some History of Radiofrequency Doppler.- Range-Doppler Radar.- Radargrarnmetry.- Literature Cited.- Problems.- 17 Two-Dimensional Noise Images.- Some Types of Random Image.- Gaussian Noise.- The Spatial Spectrum of a Random Scatter.- Autocorrelation of a Random Scatter.- Pseudorandom Scatter.- Random Orientation.- Nonuniform Random Scatter.- Spatially Correlated Noise.- The Familiar Maze.- The Drunkard's Walk.- Fractal Polygons.- Conclusion.- Literature Cited.- Problems.- Appendix A Solutions to Problems.

133 citations


Journal ArticleDOI
TL;DR: In this paper, the propagation physics leading to focusing are studied with both experimental data and a propagation model using surface wave profiles measured during the collection of the experimental data, demonstrating the stages of the focusing event and the impact of the high intensity arrivals and rapid fluctuations on the ability of an algorithm to accurately estimate the impulse response.
Abstract: The forward scattering of acoustic signals off of shoaling surface gravity waves in the surf zone results in a time-varying channel impulse response that is characterized by intense, rapidly fluctuating arrivals. In some cases, the acoustic focusing by the curvature of the wave crest results in the formation of caustics at or near a receiver location. This focusing and the resulting caustics present challenges to the reliable operation of phase coherent underwater acoustic communications systems that must implicitly or explicitly track the fluctuations in the impulse response. The propagation physics leading to focusing are studied with both experimental data and a propagation model using surface wave profiles measured during the collection of the experimental data. The deterministic experimental and modeled data show good agreement and demonstrate the stages of the focusing event and the impact of the high intensity arrivals and rapid fluctuations on the ability of an algorithm to accurately estimate the impulse response. The statistical characterization of experimental data shows that the focusing by surface gravity waves results in focused surface reflected arrivals whose intensity often exceeds that of the direct arrival and the focusing and caustic formation adversely impacts the performance of an impulse response estimation algorithm.

128 citations


Journal ArticleDOI
TL;DR: A rapid method for calculating the nearfield pressure distribution generated by a rectangular piston is derived for time-harmonic excitations with an equivalent integral expression that removes the numerical singularities caused by inverse trigonometric functions.
Abstract: A rapid method for calculating the nearfield pressure distribution generated by a rectangular piston is derived for time-harmonic excitations. This rapid approach improves the numerical performance relative to the impulse response with an equivalent integral expression that removes the numerical singularities caused by inverse trigonometric functions. The resulting errors are demonstrated in pressure field calculations using the time-harmonic impulse response solution for a rectangular source 5 wavelengths wide by 7.5 wavelengths high. Simulations using this source geometry show that the rapid method eliminates the singularities introduced by the impulse response. The results of pressure field computations are then evaluated in terms of relative errors and computational speeds. The results show that, when the same number of Gauss abscissas are applied to both approaches for time-harmonic pressure field calculations, the rapid method is consistently faster than the impulse response, and the rapid method consistently produces smaller maximum errors than the impulse response. For specified maximum error values of 10% and 1%, the rapid method is 2.6 times faster than the impulse response for pressure field calculations performed on a 61 by 101 point grid. The rapid approach achieves even greater reductions in the computation time for smaller errors and larger grids.

112 citations


Patent
07 Dec 2004
TL;DR: Pilot transmission and channel estimation techniques for an OFDM system with excess delay spread are described in this article, where the number of pilot subbands is greater than the cyclic prefix length.
Abstract: Pilot transmission and channel estimation techniques for an OFDM system with excess delay spread are described. To mitigate the deleterious effects of excess delay spread, the number of pilot subbands is greater than the cyclic prefix length. This 'oversampling' may be achieved by using more pilot subbands in each symbol period or different sets of pilot subbands in different symbol periods. In one channel estimation technique, first and second groups of received pilot symbols are obtained for first and second pilot subband sets, respectively, and used to derive first and second frequency response estimates, respectively. First and second impulse response estimates are derived based on the first and second frequency response estimates, respectively, and used to derive a third impulse response estimate having more taps than the number of pilot subbands in either set.

98 citations


Proceedings ArticleDOI
20 Jun 2004
TL;DR: A scheme based on sphere decoding appears to give the best performance while maintaining moderate complexity in the estimation of a channel whose impulse response is characterized by a large number of negligible tap coefficients.
Abstract: Algorithms for the estimation of a channel whose impulse response is characterized by a large number of negligible tap coefficients are developed and compared. Exploiting this sparsity, the estimation problem is transformed into an equivalent on-off keying detection problem, whose solution gives an indication on the position of the zero taps. The proposed schemes are compared to the standard least squares estimate via simulations in terms of mean square error and bit error rate. A scheme based on sphere decoding appears to give the best performance while maintaining moderate complexity.

81 citations


PatentDOI
TL;DR: In this paper, a method for reproducing spatial impression of existing spaces in multichannel or binaural listening is proposed, which consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multi-channel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room.
Abstract: The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

75 citations


Journal ArticleDOI
TL;DR: The relation between the MF and ER notch filter is presented in order to emphasize the superior performance of the ER narrow-band filters over their MF counterparts.
Abstract: Fast analytical design procedures for finite impulse response (FIR) maximally flat (MF) and optimal equiripple (ER) notch filters are introduced. The closed form solution provides recursive computation of the impulse response coefficients of the filter. The ER FIR filters are optimal in the Chebyshev sense. The relation between the MF and ER notch filter is presented in order to emphasize the superior performance of the ER narrow-band filters over their MF counterparts. The discrete nature of the notch frequency in both filter types is emphasized. Four design examples are included to demonstrate the efficiency of the presented approach.

74 citations


Journal ArticleDOI
TL;DR: This review describes impulse response techniques with a curve-fitting method to measure thermodynamic properties, such as binary diffusion coefficient, retention factor, and partial molar volume, under supercritical conditions.

Journal ArticleDOI
TL;DR: In this article, two identification algorithms, a least square and a correlation analysis based, are developed for dual-rate stochastic systems in which the output sampling period is an integer multiple of the input updating period.
Abstract: Two identification algorithms, a least squares and a correlation analysis based, are developed for dual-rate stochastic systems in which the output sampling period is an integer multiple of the input updating period. The basic idea is to use auxiliary FIR models to predict unmeasurable noise-free (true) outputs, and then use these and system inputs to identify parameters of underlying fast single-rate models. The simulation results indicate that the proposed algorithms are effective. Copyright © 2004 John Wiley & Sons, Ltd.

Journal ArticleDOI
TL;DR: Results presented in this paper provide a better understanding of the role of the coupling term in elastography and should be used to compensate diffraction and coupling effects observed in transientElastography.
Abstract: The transient radiation of low-frequency elastic waves through isotropic and homogeneous soft media is investigated using the Green's function approach. A careful analysis of the coupling term is performed and yields the introduction of a very near field region in which its amplitude behaves as 1/r. To address the calculation of impulse responses, a simplified Green's function is proposed for semi-infinite media and compared to exact solutions. Impulse response calculations are successfully compared with experimental measurements obtained for circular radiators of different diameters using transient elastography. Results presented in this paper provide a better understanding of the role of the coupling term in elastography and should be used to compensate diffraction and coupling effects observed in transient elastography.

Patent
Shoichi Okamura1
29 Jul 2004
TL;DR: In this article, a radiographic apparatus removes lag-back parts from radiation detection signals taken from an FPD as X rays are emitted from an X-ray tube, on the assumption that the lag-behind part included in each Xray detection signal is due to an impulse response formed of a plurality of exponential functions with different attenuation time constants.
Abstract: A radiographic apparatus removes lag-behind parts from radiation detection signals taken from an FPD as X rays are emitted from an X-ray tube, on an assumption that the lag-behind part included in each X-ray detection signal is due to an impulse response formed of a plurality of exponential functions with different attenuation time constants. When a single attenuation time constant and intensity are provisionally set, checking is made whether an attenuation to a noise level of X-ray detection signals occurs in an X-ray non-emission state following an X-ray emission state. When the set attenuation time constant and intensity are found appropriate (OK), the impulse response having the single exponential function is determined valid. Corrected radiation detection signals are obtained by removing the lag-behind parts using the impulse response determined.

Journal ArticleDOI
TL;DR: In this article, the phase of an incoherent layered sediment reflection coefficient is restored by spectral factorization, where the Fourier transform is the minimum phase impulse response at each angle.
Abstract: Spectral factorization is shown to restore the phase of an incoherent layered sediment reflection coefficient so that its Fourier transform is the minimum phase impulse response at each angle. The method requires the reflection coefficient to be known over a range of frequencies and the grazing angles in question to be above critical. It is developed here in the context of another recently established technique for extracting the seabed’s plane wave reflection coefficient from ambient noise data measured on a moored or drifting vertical array (VLA). Thus it offers the possibility of sub-bottom profiling from a single platform with no sound source. Limitations of the phase restoration method are discussed and, using modeled data, comparisons are made between the “true” impulse response derived from the known complex reflection coefficient and the result of applying spectral factorization to the absolute value of the reflection coefficient. The method is also demonstrated on experimental reflection loss inf...

Proceedings ArticleDOI
17 May 2004
TL;DR: The measurements were performed in February 2002 in an Airbus A319 in Hamburg (Germany), a single-aisle short-haul aircraft to determine the propagation characteristics of the indoor cabin environment for personal wireless communications at the downlink band of a UMTS FDD system.
Abstract: This work presents a part of the results of a test campaign for channel propagation in aircraft cabins. The measurements were performed in February 2002 in an Airbus A319 in Hamburg (Germany), a single-aisle short-haul aircraft. Wideband measurements were performed inside the passenger's area to determine the propagation characteristics of the indoor cabin environment for personal wireless communications at the downlink band of a UMTS FDD system. The paper describes the measurement site and technique, the scenario under scope, and discusses the results of the statistical analysis for the impulse response characterization. The results of the analysis show that the cabin can be divided into 5 areas of roughly 3 m long each. In each of these areas the parameters of the linear filter modelling the impulse response are governed by the same law. The main outputs of the study are: (i) the distribution of the number of multipath components follows a Nakagami distribution; (ii) the mean amplitude of the multipath components for a certain delay results in an exponential function, with amplitude decaying exponentially with the distance and a different decay factor for each of the five areas of the cabin; (iii) the variations of the echo amplitude with respect to its mean is described by a log-normal distribution in most of the cabin area and by a Nakagami distribution at the furthest positions; (iv) the rms delay spread over the cabin increases with increasing transmitter-receiver distance.

Journal Article
TL;DR: Using the simple idea of frequency-domain multiplexing, a technique for performing simultaneous multiple-channel impulse response measurements is proposed and a previous technique that uses time-domainmultiplexing is revisited.
Abstract: The time involved in measuring the linear characteristics of acoustic systems is a common problem in audio signal processing. Recent years have witnessed a major advance in multiple-channel sound reproduction systems. Using the simple idea of frequency-domain multiplexing, a technique for performing simultaneous multiple-channel impulse response measurements is proposed. A previous technique that uses time-domain multiplexing is also revisited. Several measurements are performed in order to compare the reliability of simultaneous and sequential methods. Experimental results show that both methods have similar accuracy, but the simultaneous measurement case provides measurement versatility and saves time.

Proceedings ArticleDOI
17 May 2004
TL;DR: An improved adaptive echo cancellation algorithm designed for use with sparse echo path impulse responses such as arise from packet-switched networks that outperforms the best existing technique and has lower complexity.
Abstract: We present an improved adaptive echo cancellation algorithm designed for use with sparse echo path impulse responses such as arise from packet-switched networks. The new approach implicitly segments the impulse response into 'active' and 'inactive' regions, and employs different proportionate updating in each region. An efficient partial updating scheme is then formulated for the new algorithm. Evaluation results are presented to compare the new algorithm against three existing methods in terms of convergence and computational complexity. The results show that the new algorithm outperforms the best existing technique and has lower complexity.

Journal ArticleDOI
TL;DR: In this article, a new method for constructing confidence intervals for impulse response functions and half-lives of nearly nonstationary processes is proposed based on inverting the acceptance region of the likelihood ratio statistic under a sequence of null hypotheses.
Abstract: Summary Many economic time series are characterized by high persistence which typically requires nonstandard limit theory for inference. This paper proposes a new method for constructing confidence intervals for impulse response functions and half-lives of nearly nonstationary processes. It is based on inverting the acceptance region of the likelihood ratio statistic under a sequence of null hypotheses of possible values for the impulse response or the half-life. This paper shows the consistency of the restricted estimator of the localizing constant which ensures the validity of the asymptotic inference. The proposed method is used to study the persistence of shocks to real exchange rates.

Journal ArticleDOI
TL;DR: A stochastic-ray model of wave propagation based on the theory of random walks is considered, deriving a closed form solution for the power delay profile of the received signal.
Abstract: We consider a stochastic-ray model of wave propagation based on the theory of random walks that accounts for only two parameters: the amount of clutter and the amount of absorption in the environment. This model applies to indoor and outdoor environments, characterized by a large number of (small) scattering objects. Examples are microcells for personal communication and networks of multihop wireless sensors or laptop computers. We extend our previous results on narrow-band signals to impulse waveforms, deriving a closed form solution for the power delay profile of the received signal.

Patent
21 Dec 2004
TL;DR: In this article, a speaker and microphone system for detecting the position of an obstacle and controlling speakers or microphones on the basis of the detected position of the obstacle is presented, where the acoustic apparatus has a plurality of speakers, pulse sound generating means for generating a pulse sound via at least any one of the speakers; one or a plurality-of-mics for acquiring sounds; impulse response detecting means for detecting impulse response, based on the pulse sound generated by the pulse-sound generating means from the sounds acquired by the one or the plurality of microphones; position calculating means for calculating
Abstract: PROBLEM TO BE SOLVED: To provide a speaker and microphone system for detecting the position of an obstacle and controlling speakers or microphones on the basis of the detected position of the obstacle. SOLUTION: The acoustic apparatus has a plurality of speakers 15; pulse sound generating means for generating a pulse sound via at least any one of the plurality of speakers; one or a plurality of microphones 12 for acquiring sounds; impulse response detecting means for detecting impulse response, based on the pulse sound generated by the pulse sound generating means from the sounds acquired by the one or the plurality of microphones; position calculating means for calculating the position of the obstacle from the impulse response detected by the impulse response detecting means; and sound field control means for controlling outputs of the plurality of speakers depending on the position of the obstacle calculated by the position calculating means, and controlling the sound field formed by these speakers. COPYRIGHT: (C)2006,JPO&NCIPI

Journal ArticleDOI
TL;DR: In this article, a two-step technique is used to estimate structural vector error correction models and provide the asymptotic distribution of the impulse response functions of such a system.
Abstract: This paper adopts a two-step technique to estimate structural vector error correction models and provides the asymptotic distribution of the impulse response functions of such a system. The method combines two popular tools in econometrics, namely, vector autoregressive cointegration analysis in the first step and structural vector autoregression analysis in the second. The proposed structural model structure is very general in the sense that all just-identifying or overidentifying schemes that can be expressed as linear restrictions on either the contemporaneous or long-run impact of the structural shocks are allowed for. The long-run restrictions complicate the derivation of the asymptotic distribution of the structural parameter estimates as these restrictions are a function of the reduced form parameters. Consequently, the asymptotic distribution involves an extra partial derivative. This paper adopts a two-step procedure to estimate structural vector error correction models and provides the asymptotic distribution of the impulse response functions of such a system even if those contain long-run restrictions. The method combines two popular tools in econometrics, namely, vector autoregressive cointegration analysis and structural vector autoregressions. Vector autoregression (VAR) analysis has become a popular tool in empirical macroeconomics and finance. An important element in these models is the identification of independent structural shocks. Originally (Sims, 1980) a Choleski decomposition of the covariance matrix of the residuals was used for this purpose, thereby implicitly assuming a recursive causal ordering. Later on, "structural" VAR models were developed, in which the identifying restrictions are explicitly derived from theory. These restrictions can involve either contemporaneous relationships (Blanchard and Watson, 1986; Bernanke, 1986;

Posted Content
TL;DR: In this paper, the authors proposed a method for constructing confidence bands for multivariate impulse response functions that is not pointwise and that is robust to the presence of highly persistent processes.
Abstract: Existing methods for constructing confidence bands for multivariate impulse response functions depend on auxiliary assumptions on the order of integration of the variables. Thus, they may have poor coverage at long lead times when variables are highly persistent. Solutions that have been proposed in the literature may be computationally challenging. The goal of this Paper is to propose a simple method for constructing confidence bands for impulse response functions that is not pointwise and that is robust to the presence of highly persistent processes. The method uses alternative approximations based on local-to-unity asymptotic theory and allows the lead time of the impulse response function to be a fixed fraction of the sample size. These devices provide better approximations in small samples. Monte Carlo simulations show that our method tends to have better coverage properties at long horizons than existing methods. We also investigate the properties of the various methods in terms of the length of their confidence bands. Finally, we show, with empirical applications, that our method may provide different economic interpretations of the data. Applications to real GDP and to nominal versus real sources of fluctuations in exchange rates are discussed.

Patent
20 Dec 2004
TL;DR: In this article, an initial frequency response estimate is obtained for a first set of P uniformly spaced subbands based on pilot symbols received on a second set of subbands used for pilot transmission, where P is a power of two.
Abstract: For channel estimation in a spectrally shaped wireless communication system, an initial frequency response estimate is obtained for a first set of P uniformly spaced subbands (1) based on pilot symbols received on a second set of subbands used for pilot transmission and (2) using extrapolation and/or interpolation, where P is a power of two. A channel impulse response estimate is obtained by performing a P-point IFFT on the initial frequency response estimate. A final frequency response estimate for N total subbands is derived by (1) setting low quality taps for the channel impulse response estimate to zero, (2) zero-padding the channel impulse response estimate to length N, and (3) performing an N-point FFT on the zero-padded channel impulse response estimate. The channel frequency/impulse response estimate may be filtered to obtain a higher quality channel estimate.

Patent
20 May 2004
TL;DR: In this article, a two-way cable system with a fixed CW signal injected into a downstream signal path, a swept signal transmitted from a network analyzer, a mixer for generating an up-converted swept signal, and a source of CPD in the two way cable system that mixes the fixed CW signals and the swept signals to create an upstream swept signal.
Abstract: A system and method to range a distance to a source of CPD on a two-way cable system, comprises a fixed CW signal injected into a downstream signal path, a swept signal transmitted from a network analyzer, a mixer for generating an up-converted swept signal, and a source of CPD in the two-way cable system that mixes the fixed CW signal and the swept signal to create an upstream swept signal. The network analyzer receives the upstream swept signal and determines a complex frequency response created by the source of CPD. An impulse response is determined from the complex frequency response, and the distance to the source of CPD is determined from the impulse response.

Journal ArticleDOI
TL;DR: It is argued that the phase obtained by the proposed method has a low susceptibility to measurement noise and a low rate of artificial phase slips.
Abstract: A method for measuring the phase of oscillations from noisy time series is proposed. To obtain the phase, the signal is filtered in such a way that the filter output has minimal relative variation in the amplitude over all filters with complex-valued impulse response. The argument of the filter output yields the phase. Implementation of the algorithm and interpretation of the result are discussed. We argue that the phase obtained by the proposed method has a low susceptibility to measurement noise and a low rate of artificial phase slips. The method is applied for the detection and classification of mode locking in vortex flow meters. A measure for the strength of mode locking is proposed.

Journal ArticleDOI
TL;DR: This letter proposes a new method for designing finite-impulse response (FIR) filters with variable characteristics that is simple and effective in designing FIR VDF with good frequency characteristics and allows for more complicated frequency characteristics or a larger tuning range to be approximated.
Abstract: This letter proposes a new method for designing finite-impulse response (FIR) filters with variable characteristics. The impulse response of the variable digital filter (VDF) is parameterized as a linear combination of functions in the spectral or tuning parameters. Using the least square objective function, the optimal solution is obtained by solving a system of linear equations. Design results show that this method is simple and effective in designing FIR VDF with good frequency characteristics. Furthermore, by using a piecewise polynomial, instead of an ordinary polynomial, more complicated frequency characteristics, or a larger tuning range can be approximated.

Journal ArticleDOI
TL;DR: It is demonstrated that by using the second-order statistics of the channel outputs, under mild conditions on the nonstationarity of sources, and under the condition that channel is column-wise coprime, the impulse response of the MIMO channel can be identified up to an inherent scaling and permutation ambiguity.
Abstract: This paper discusses a frequency domain method for blind identification of multiple-input multiple-output (MIMO) convolutive channels driven by white quasistationary sources. The sources can assume arbitrary probability distributions, and in some cases, they can even be all Gaussian distributed. We also show that under slightly more restrictive assumptions, the algorithm can be applied to the case when the sources are colored, nonstationary signals. We demonstrate that by using the second-order statistics of the channel outputs, under mild conditions on the nonstationarity of sources, and under the condition that channel is column-wise coprime, the impulse response of the MIMO channel can be identified up to an inherent scaling and permutation ambiguity. We prove that by using the new algorithm, under the stated assumptions, a uniform permutation across all frequency bins is guaranteed, and the inherent frequency-dependent scaling ambiguities can be resolved. Hence, no post processing is required, as is the case with previous frequency domain algorithms. We further present an efficient, two-step frequency domain algorithm for identifying the channel. Numerical simulations are presented to demonstrate the performance of the new algorithm.

Journal ArticleDOI
TL;DR: In this paper, a modified Single Station Time Domain (SSTD) method is proposed, which can be applied to include the effect of purely harmonic vibrations, assuming the harmonic frequencies are known a priori.

Proceedings ArticleDOI
17 May 2004
TL;DR: Two new sparse adaptive filtering algorithms using partial update achieve faster convergence speed with even less computational complexity and perform well in applications where identification of long sparse impulse responses is needed.
Abstract: In this paper, we propose two new sparse adaptive filtering algorithms using partial update. By taking advantage of both impulse response sparseness and partial update, we design different criteria to determine which coefficients to be updated in order to improve the performance of typical partial update algorithms. Compared with the normalized least mean square (NLMS), selective partial update NLMS (SPUNLMS) and proportionate NLMS (PNLMS++) algorithm, the proposed partial update sparse NLMS (PSNLMS) algorithms achieve faster convergence speed with even less computational complexity. Simulation results show that they perform well in applications where identification of long sparse impulse responses is needed. Network echo cancellation is a typical example.