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Showing papers on "Impulse response published in 1993"


Journal ArticleDOI
TL;DR: A recursive method for evaluating the impulse response of an indoor free-space optical channel with Lambertian reflectors, which enables accurate analysis of the effects of multipath dispersion on high-speed indoor optical communication systems.
Abstract: A recursive method for evaluating the impulse response of an indoor free-space optical channel with Lambertian reflectors is presented. The method, which accounts for multiple reflections of any order, enables accurate analysis of the effects of multipath dispersion on high-speed indoor optical communication systems. A simple algorithm for computer implementation of the technique and computer simulation results for both line-of-sight and diffuse transmitter configurations are also presented. In both cases, it is shown that reflections of multiple order are a significant source of intersymbol interference. Experimental measurements of optical multipath, which help verify the accuracy of the simulations, are discussed. >

867 citations


Journal ArticleDOI
Mark R. Bell1
01 Sep 1993
TL;DR: The use of information theory to design waveforms for the measurement of extended radar targets exhibiting resonance phenomena is investigated and the target impulse response is introduced to model target scattering behavior.
Abstract: The use of information theory to design waveforms for the measurement of extended radar targets exhibiting resonance phenomena is investigated. The target impulse response is introduced to model target scattering behavior. Two radar waveform design problems with constraints on waveform energy and duration are then solved. In the first, a deterministic target impulse response is used to design waveform/receiver-filter pairs for the optimal detection of extended targets in additive noise. In the second, a random target impulse response is used to design waveforms that maximize the mutual information between a target ensemble and the received signal in additive Gaussian noise. The two solutions are contrasted to show the difference between the characteristics of waveforms for extended target detection and information extraction. The optimal target detection solution places as much energy as possible in the largest target scattering mode under the imposed constraints on waveform duration and energy. The optimal information extraction solution distributes the energy among the target scattering modes in order to maximize the mutual information between the target ensemble and the received radar waveform. >

694 citations


Journal ArticleDOI
TL;DR: In this article, first-order autoregressive/unit root models with independent identically distributed normal errors are considered, including those without an intercept, those with an intercept and time trend.
Abstract: First-order autoregressive/unit root models with independent identically distributed normal errors are considered, including those without an intercept, those with an intercept, and those with an intercept and time trend. The autoregressive parameter is allowed to lie in the interval (-1, 1], which includes the unit root case. Exactly median-unbiased estimators and exact confidence intervals of the autoregressive parameter are introduced. Corresponding exactly median-unbiased estimators and exact confidence intervals are also provided for the impulse response function, the cumulative impulse response, and the half life of a unit shock. An unbiased model selection procedure is discussed. The introduced procedures are applied to several data series. Copyright 1993 by The Econometric Society.

507 citations


Journal ArticleDOI
TL;DR: It is shown that the number of multipath components in each impulse response estimate is a normally-distributed random variable with a mean value that increases with increasing antenna separations, and the RMS delay spread over large areas is normally distributed with mean values that increase with increasing antennas separation.
Abstract: If indoor radio propagation channels are modeled as linear filters, they can be characterized by reporting the parameters of their equivalent impulse response functions. The measurement and modeling of estimates for such functions in two different office buildings are reported. The resulting data base consists of 12000 impulse response estimates of the channel that were obtained by inverse Fourier transforming of the channel's transfer functions. It is shown that the number of multipath components in each impulse response estimate is a normally-distributed random variable with a mean value that increases with increasing antenna separations; a modified Poisson distribution shows a good fit to the arrival time of the multipath components; amplitudes are lognormally distributed over both local and global areas, with a log-mean value that decreases almost linearly with increasing excess delay; for small displacements of the receiving antenna, the amplitude of the multipath components are correlated; the amplitudes of adjacent multipath components of the same impulse response function show negligible correlations; and the RMS delay spread over large areas is normally distributed with mean values that increase with increasing antenna separation. >

460 citations


Journal ArticleDOI
TL;DR: In this paper, nonlinear impulse response analysis is introduced based on conditional moment profiles defined for a stationary time series and compared to baseline profiles is the nonlinear analog of conventional impulse-response analysis.
Abstract: Methods for nonlinear impulse response analysis are introduced. The methods are based on conditional moment profiles defined for a stationary time series. Comparing conditional moment profiles to baseline profiles is the nonlinear analog of conventional impulse-response analysis. The bootstrap may be used for statistical inference. Profile bundles may be examined for evidence of damping or persistence. Application to bivariate NYSE price and volume series from 1928 to 1987 finds evidence of a heavily damped 'leverage effect' and a differential response of trading volume to 'common-knowledge' price shocks. Copyright 1993 by The Econometric Society.

451 citations


Proceedings ArticleDOI
23 May 1993
TL;DR: Several algorithms for designing the time-domain equalizer, which use adaptation in the frequency domain and windowing in the time domain in order to minimize the mean squared error of the equalized response, are described.
Abstract: In a multicarrier data transmission system the length of a symbol is often limited by the maximum permissible delay of data from input to output. If the length of the impulse response of the channel is not negligible compared to the permitted symbol length, the impulse response can be shortened by passing the received signal through a time-domain equalizer before modulation. The best performance for a given computational complexity can then be achieved by appending to each block of samples of the transmit signal a cyclic prefix that has the same length as the shortened impulse response. Several algorithms for designing the time-domain equalizer, which use adaptation in the frequency domain and windowing in the time domain in order to minimize the mean squared error of the equalized response, are described. >

221 citations


Proceedings ArticleDOI
02 Jun 1993
TL;DR: In this paper, a new design technique for a robust model predictive controller using an uncertainty description expressed in the time-domain is proposed using a set of Finite Impulse Response (FIR) models, and necessary and sufficient conditions for asymptotic stability are stated.
Abstract: A new design technique for a robust model predictive controller is proposed using an uncertainty description expressed in the time-domain. Robust stability of the resulting closed-loop system is guaranteed for a set of Finite Impulse Response (FIR) models. Both necessary and sufficient conditions for asymptotic stability are stated. If the uncertainty is described as lower and upper bounds on impulse response coefficients, then the resulting optimization problem can be cast as a linear program of moderate size.

187 citations


Journal ArticleDOI
TL;DR: In this paper, an expression for the impulse response up-sampling ratio M, which will produce a minimum complexity design, is derived, and an optimum design relationship for the interpolated impulse response technique is also derived.
Abstract: An expression for the impulse response up-sampling ratio M, which will produce a minimum complexity design, is derived. It is shown that M approaches e (the base of the natural logarithm) as the number of frequency response masking stages increases; in a K-stage design the complexity of the filter is inversely proportional to the (K+1)th root of the transition width; the frequency response masking technique is effective if the normalized transition width is less than 1/16; and the frequency response masking technique is more efficient than the interpolated impulse response technique if the square root of the normalized transition width is less than the arithmetic mean of the normalized passband edge and stopband edge. An expression for the multistage frequency response ripple compensation is derived. An optimum design relationship for the interpolated impulse response technique is also derived. The design of narrow-band two-dimensional filters using the frequency response masking technique is also presented. >

183 citations


Journal ArticleDOI
TL;DR: Cases of first-order plants, second-order underdamped plants, and plants with multiunit delay are discussed and an optimum time scale such that an index is minimized is derived.
Abstract: Any discrete-time stable transfer function can be expressed by a discrete-time Laguerre series with a chosen time scale. An optimum time scale such that an index is minimized is derived. This index ensures that the coefficients of higher-order Laguerre functions go toward zero quickly. The solution derived requires the knowledge of the impulse response of the discrete plant. Cases of first-order plants, second-order underdamped plants, and plants with multiunit delay are also discussed. >

172 citations


Journal ArticleDOI
TL;DR: A normalized least-mean-squares (NLMS) adaptive algorithm with double the convergence speed, at the same computational load, of the conventional NLMS for an acoustic echo canceller is proposed and its fast convergence is demonstrated.
Abstract: A normalized least-mean-squares (NLMS) adaptive algorithm with double the convergence speed, at the same computational load, of the conventional NLMS for an acoustic echo canceller is proposed. This algorithm, called the ES (exponentially weighted stepsize) algorithm, uses a different stepsize (feedback constant) for each weight of an adaptive transversal filter. These stepsizes are time-invariant and weighted proportionally to the expected variation of a room impulse response. The algorithm adjusts coefficients with large errors in large steps, and coefficients with small errors in small steps. A transition formula is derived for the mean-squared coefficient error of the algorithm. The mean stepsize determines the convergence condition, the convergence speed, and the final excess mean-squared error. Modified for a practical multiple DSP structure, the algorithm requires only the same amount of computation as the conventional NLMS. The algorithm is implemented in a commercial acoustic echo canceller, and its fast convergence is demonstrated. >

148 citations


Patent
04 Oct 1993
TL;DR: In this article, the analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters.
Abstract: Synthetic head related transfer functions (HRTFs) for imposing reprogrammable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROMs) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed and fed to a pair of headphones.

Journal ArticleDOI
TL;DR: In this paper, the authors re-examine the issues that surround the identification of shocks to particular sectors in a multisectoral model, and propose "persistence profiles" of shocks as a method of carrying out impulse response analysis in multivariate systems.

Patent
13 Aug 1993
TL;DR: In this paper, the authors proposed a receiver that employs symbol timing adjustment and carrier tracking algorithms based on channel impulse response estimates, using variable coefficient least mean square tracking during estimation of the transmission channel's impulse response.
Abstract: The present invention provides for a receiver that employs integrated symbol timing adjustment and carrier tracking algorithms based on channel impulse response estimates. It uses variable coefficient least mean square tracking during estimation of the transmission channel's impulse response. Maximum likelihood sequence estimation equalization can also be integrated into the system using a non-real time operation that permits and exploits "time-reversed" equalization to significantly enhance bit-error rate performance and permits implementation of all the functions in a reasonably simple receiver.

Journal ArticleDOI
TL;DR: In this article, the authors considered the identification of linear systems in the presence of measurement noise which is unknown but bounded in magnitude by some δ > 0 and showed that the optimal identification error is related to the diameter of a so-called uncertainty set and that the latter diameter is upper-bounded by 2δ, if a sufficiently long identification experiment is performed.

Journal ArticleDOI
TL;DR: In this paper, a method for removing the azimuth ambiguities from synthetic aperture radar (SAR) images is proposed, which corresponds to an ideal filter concept, where an ideal impulse response function is obtained even in the presence of several phase and amplitude errors.
Abstract: A method for removing the azimuth ambiguities from synthetic aperture radar (SAR) images is proposed. The basic idea is to generate a two-dimensional reference function for SAR processing which provides, in addition to the matched filtering for the unaliased part of the received signal, the deconvolution of the azimuth ambiguities. This approach corresponds to an ideal filter concept, where an ideal impulse response function is obtained even in the presence of several phase and amplitude errors. Modeling the sampled azimuth signal shows that the absolute phase value of the ambiguities cannot easily be determined due to their undersampling. The concept of the ideal filter is then extended to accommodate the undefined phase of the ambiguities and also the fading of the azimuth signal. Raw data from the E-SAR system have been used to verify the improvement in image quality obtained by the new method. It has a substantial advantage in enabling the pulse-repetition frequency (PRF) constraints in the SAR system design to be relaxed and also for improving SAR image quality and interpretation. >

Journal ArticleDOI
Levent Onural1
TL;DR: The system impulse response representing the Fresnel diffraction is shown to form a wavelet family of functions that relates the diffraction-holography-related studies and the wavelet theory, and the presented wavelet structure is a tool for space–depth analysis.
Abstract: The system impulse response representing the Fresnel diffraction is shown to form a wavelet family of functions. The scale parameter of the wavelet family represents the depth (distance). This observation relates the diffraction-holography-related studies and the wavelet theory. The results may be used in various optical applications such as designing robust volume optical elements for optical signal processing and finding new formulations for optical inverse problems. The results also extend the wavelet concept to the nonbandpass family of functions with the implication of new applications in signal processing. The presented wavelet structure, for example, is a tool for space–depth analysis.

Journal ArticleDOI
TL;DR: In this article, the authors measured thresholds for detecting pairs of briefly flashed stimuli displayed successively at variable onset asynchronies, where the stimuli were 1 cycle/deg vertical sinusoidal gratings, modulated either in luminance (yellow-black) or in color (red-green).
Abstract: Thresholds were measured for detecting pairs of briefly flashed stimuli displayed successively at variable onset asynchronies. The stimuli were 1 cycle/deg vertical sinusoidal gratings, modulated either in luminance (yellow–black) or in color (red–green). The successive presentations were either of the same contrast (positive) or of opposite contrast (negative), yielding four separate summation curves: positive and negative summation for color and for luminance. Both the positive and the negative curves followed a shorter time course for luminance than for color, implying a faster response at threshold. To calculate impulse response functions from the summation data, we assumed that the neural impulse response from two successive stimuli sum linearly at threshold, that thresholds are determined by probability summation of the combined impulse response over time, and that the impulse response can be described by an exponentially damped frequency-modulated sinusoidal function with four free parameters. The predicted impulse responses for luminance and for color are quite different, being biphasic for luminance and monophasic for color. Fourier transform of these functions yielded estimates of the amplitude and the phase functions of hypothetical visual detectors: the amplitude functions predicted well the contrast sensitivity of counterphased gratings (as a function of temporal frequency) both for luminance and for chromatic stimuli.

PatentDOI
TL;DR: In this paper, a sound environment simulator including a sound field analyzing unit, sound field reproducing unit, and an output unit is presented, where the sound environment analyzing unit divides the solid surfaces of a space to be analyzed into a set of sections to compute the volume of reflected sounds with a sound absorption coefficient of walls and form factors.
Abstract: The present invention discloses a sound environment simulator including a sound field analyzing unit, a sound field reproducing unit, and an output unit. The sound environment analyzing unit divides the solid surfaces of a space to be analyzed into a set of sections to compute the volume of reflected sounds with a sound absorption coefficient of walls and form factors. It further computes time series data related to the arrival volume of sounds emanated from a certain sound source to the sound receiving point. An impulse response computing unit in the sound field reproducing unit transduces the time series data into an impulse response. Accordingly, the sound field reproducing unit convolutes the impulse response on a dry source in accordance with data related to a listener's position inputted from an associated virtual reality equipment to generate a reproduced sound over a headphone.

Journal Article
TL;DR: In this paper, the information contained in an energy histogram obtained by ray tracing can be converted into an equivalent impulse response which, after binaural processing, is ready for auralization sufficient for the auralisation process.
Abstract: It is shown how the information contained in an energy histogram obtained by ray tracing can be converted into an equivalent impulse response which, after binaural processing, is ready for auralization sufficient for the auralization process. Furthermore the steps are discussed which are needed to transform this information into a binaural impulse response ready for a convolution with music signals

Journal ArticleDOI
TL;DR: It is shown that, under some constraints, the impulse response of the system can be expressed as a linear combination of cumulant slices, which is used to obtain a well-conditioned linear method for estimating the MA parameters of a nonGaussian process.
Abstract: A linear approach to identifying the parameters of a moving-average (MA) model from the statistics of the output is presented. First, it is shown that, under some constraints, the impulse response of the system can be expressed as a linear combination of cumulant slices. Then, this result is used to obtain a well-conditioned linear method for estimating the MA parameters of a nonGaussian process. The linear combination of slices used to compute the MA parameters can be constructed from different sets of cumulants of different orders, provided a general framework in which all the statistics can be combined. It is not necessary to use second-order statistics (autocorrelation slice), and therefore the proposed algorithm still provides consistent estimates in the presence of colored Gaussian noise. Another advantage of the method is that while most linear methods give totally erroneous estimates if the order is overestimated, the proposed approach does not require a previous estimation of the filter order. The simulation results confirm the good numerical conditioning of the algorithm and its improvement in performance in comparison to existing methods. >

Journal ArticleDOI
TL;DR: In this paper, a linear theory that may be used to predict the primary responses (stress, strain displacement) of continuous pavements to moving dynamic loads is presented, derived from the well-known convolution integral and its formulations in moving and stationary frames of reference.
Abstract: A linear theory that may be used to predict the primary responses (stress, strain displacement) of continuous pavements to moving dynamic loads is presented. The theory is derived from the well‐known convolution integral and its formulations in moving and stationary frames of reference are derived. A detailed solution, in the frequency/wave‐number domain, is given for a quarter‐car vehicle‐model driving up a step on an elastic beam resting on a damped elastic foundation. The effects of speed and frequency on predicted stresses and displacements are observed to be consistent with those measured by other writers. The convolution's practical implementation is explained. The results of validation experiments, involving impulse response measurements to check the assumptions of the theory, and an instrumented vehicle driving over an instrumented test track, are described. The results demonstrate that the theory can be used to predict accurately the strains generated in a road as a vehicle drives over, including...

Journal ArticleDOI
TL;DR: Results show that the relationship between RMS delay spread and range on static indoor channels is nonmonotonic and has a maximum at a range that depends on the building dimensions and the electrical properties of reflecting surfaces.
Abstract: The authors report experimental and modeling studies that investigate the dependence of indoor radio channel multipath characteristics on a transmit/receive range. Specifically, a simple model for estimation of the complex baseband equivalent impulse response for indoor channels is explained. Using this model, the relationship between RMS delay spread and range on static indoor channels is estimated. Results show that this relationship is nonmonotonic and has a maximum at a range that depends on the building dimensions and the electrical properties of reflecting surfaces. The model is used in infinite, as well as finite, resolution modes to supplement the measurements of impulse response characteristics in different buildings using a limited resolution channel sounder. Experimental and modeling results are combined to derive conclusions that confirm the anticipated nonmonotonic relationship exists in empty buildings. Measurement results demonstrate that furniture has the effect of destroying this relationship, making RMS delay spread almost independent of range. >

Journal ArticleDOI
TL;DR: Analysis of the data collected from fourteen healthy male subjects in various conditions revealed that there are three components in the impulse response: fast positive, delayed slow negative, and oscillatory.
Abstract: A time domain technique for estimating transfer characteristics from fluctuations of instantaneous lung volume (ILV) to heart rate (HR) is presented. An effective procedure for estimating the impulse response of HR to ILV is proposed. Pre- and post-processing procedures, including prefiltering of the HR signal, preenhancement of the high frequency content of the ILV signal, and post-filtering of the estimated impulse response, together with a random breathing technique, are shown to effectively reduce spurious transfer gain so as to get a stable estimate of the impulse response. Analysis of the data collected from fourteen healthy male subjects in various conditions revealed that there are three components in the impulse response: fast positive, delayed slow negative, and oscillatory. The effects of the autonomic blocking agents propranolol and atropine on these transfer characteristics are also described. >

Patent
11 May 1993
TL;DR: In this paper, a common convolution computing element by an inverse filter means and a correction characteristic attaching means is used to simplify hardware constitution and to miniaturize a device by performing a convolution arithmetic operation.
Abstract: PURPOSE:To simplify hardware constitution and to miniaturize a device by performing a convolution arithmetic operation by using a common convolution computing element by an inverse filter means and a correction characteristic attaching means. CONSTITUTION:The application of the convolution computing element 34 can be switched by switching a switch 102. In other words, a filter coefficient read out from TSP inverse filter waveform memory 100 is set when a response characteristic arithmetic operation in a TSP method is performed, and time compression is applied to a collected TSP signal, and an impulse response is found. Also, an equalizer filter coefficient in accordance with correction characteristic found by an equalizer filter coefficient arithmetic part 144 is set at the time of confirmation of a correction effect and the reproduction of music, and it is operated as an equalizer. In this way, the computing element 34 can be used as the inverse filter in the TSP method when response characteristic is calculated, and also as the equalizer at the time of confirmation of the correction effect and the reproduction of the music, which simplifies the hardware constitution.

Journal ArticleDOI
TL;DR: In this paper, an approach to add a diffuse background signal to the result of an image source method, in which only the specular parts of the reflections are considered, is described.

Patent
28 Apr 1993
TL;DR: In this article, a system for measuring a multipath radio channel impulse response by cross-correlating digital samples of a received pseudo-random signal with a digital copy of the transmitted signal was proposed.
Abstract: A system for measuring a multipath radio channel impulse response by cross-correlating digital samples of a received pseudo-random signal with a digital copy of the transmitted signal. A transmitter within the system repeatedly transmits the pseudo-random signal across a multipath radio channel. A receiver within the system receives the transmitted signal, as well as delayed versions of the original signal, and digitally samples the received signals. A computer then cross-correlates the samples of the received signal with the digital stored version of the transmitted signal to produce a measure of the impulse response of the multipath radio channel.

Journal Article
TL;DR: In this paper, the effects of nonlinearity in maximum-length-sequence (MLS) measurement are studied, in particular the way in which impulse response errors due to nonlinearities are distributed across the measurement period.
Abstract: Maximum-length-sequence (MLS) measurement of system impulse responses offers a potential enhancement in error immunity over periodic impulse testing, although care must be exercised in setting the MLS excitation amplitude in order to realize this potential. The effects of nonlinearity in MLS measurement are studied, in particular the way in which impulse response errors due to nonlinearity are distributed across the measurement period. The consequences of such errors in cumulative spectral dispIay plots are also investigated. Finally inverse-repeat sequences (IRS) are shown to have complete immunity to even-order nonlinearity while maintaining many of the advantages of MLS.

Proceedings ArticleDOI
TL;DR: In this article, an intensity modulated optical spectrometer (IMOS) is described, which uses four different wavelengths, a wideband modulation frequency (1 MHz to 500 MHz), and simultaneous measurement of three parameters the dc intensity, ac amplitude, and the phase shift.
Abstract: Quantitative determination of chromophore concentrations by near infrared spectroscopy (NIRS) is possible by using the differential pathlength of the detected light, measured noninvasively as the mean time delay of the impulse response function of the tissue. Experimentally this information in the time domain is also available in the frequency domain by intensity modulating the input light and sweeping the modulation frequency between zero and infinity. We describe an intensity modulated optical spectrometer which differs from previously described systems in using four different wavelengths, a wideband modulation frequency (1 MHz to 500 MHz), and the simultaneous measurement of three parameters the dc intensity, ac amplitude, and the phase shift. The measured dc intensity in conjunction with the ac phase shift data allows changes in absorption coefficient (and hence chromophore concentration) to be determined more accurately by correcting for real time path length variations. The ac phase shift in combination with the ac modulation depth theoretically allows for the absolute measurement of tissue absorption and scattering coefficient. Preliminary performance figures for the system suggest values of rms noise of 0.0006 OD, 0.0011 rad and 0.0008% for the measured attenuation, ac phase shift and modulation depth. Using a phantom of fixed geometry with known scattering and absorption properties, the ability of the system to reproduce the information content of the impulse response function for a homogeneous phantom is investigated.

Journal ArticleDOI
TL;DR: In this article, a generalized beamformer is presented which coherently recombines the multipaths to provide enhanced detectability of broadband transients as well as range and depth localization of the source.
Abstract: Acoustic energy in the ocean propagates from a source to an array of sensors via multiple pathways. Conventional time‐delay‐and‐sum beamforming does not utilize energy contained in the multipath arrivals. In this paper, a generalized beamformer is presented which coherently recombines the multipaths to provide enhanced detectability of broadband transients as well as range and depth localization of the source. Modeling of the wave field at the array due to a broadband source is accomplished with a normal mode model using 1.0‐Hz increments across the 1‐ to 100‐Hz band. These results are combined to yield the impulse response from a specified source location to each of the array elements. The set of calculated impulse responses is used in the generalized beamformer to coherently recombine energy arriving along multipaths that exist from that source range and depth to the array. The location of a source is determined by examining the beamformer output at a number of candidate range/depth cells; a peak in the...

Journal ArticleDOI
TL;DR: In this article, a model-based matched filter is generated by correlating the received signal with a reference channel that consists of the transmitted signal convolved with the impulse response of the medium.
Abstract: It is shown that the performance of a conventional matched filter can be improved if the reference (replica) channel compensates for the distortion by the ocean medium. A model-based matched filter is generated by correlating the received signal with a reference channel that consists of the transmitted signal convolved with the impulse response of the medium. The channel impulse responses are predicted with a broadband propagation model using in situ sound speed measured data and archival bottom loss data. The relative performance of conventional and model-based matched filter processing is compared for large time-bandwidth-product linear-frequency-modulated signals propagating in a dispersive waveguide. From ducted propagation measurements conducted in an area west of Sardinia, the model-based matched filter localizes the depths of both the source and receiving array and the range between them. The peak signal-to-noise ratio for the model-based matched filter is always larger than that of the conventional filter. >