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The fractional Fourier transform (FrFT) provides a valuable tool for the analysis of linear chirp signals. This paper develops two short-time FrFT variants which are suited to the analysis of multicomponent and nonlinear chirp signals. Outputs have similar properties to the short-time Fourier transform (STFT) but show improved time-frequency resolution. The FrFT is a parameterized transform with parameter, α, related to chirp rate. The two short-time implementations differ in how the value of α is chosen. In the first, a global optimization procedure selects one value of α with reference to the entire signal. In the second, α values are selected independently for each windowed section. Comparative variance measures based on the Gaussian function are given and are shown to be consistent with the uncertainty principle in fractional domains. For appropriately chosen FrFT orders, the derived fractional domain uncertainty relationship is minimized for Gaussian windowed linear chirp signals. The two short-time FrFT algorithms have complementary strengths demonstrated by time-frequency representations for a multicomponent bat chirp, a highly nonlinear quadratic chirp, and an output pulse from a finite-difference sonar model with dispersive change. These representations illustrate the improvements obtained in using FrFT based algorithms compared to the STFT. [ABSTRACT FROM AUTHOR]

Adaptive matched field processing such as the minimum variance distortionless processor (MV) provides excellent sidelobe (or ambiguity) suppression capability in source localization given a perfect knowledge of the ocean environment. Unfortunately, this processing is very sensitive to sources of mismatch and robust adaptive algorithms are then employed such as a white noise constraint (WNC) often at the expense of insufficient sidelobe control. The CLEAN algorithm was introduced in radio astronomy [Astron. Astrophys. Suppl. Ser. 15, 417–426 (1974)] to produce a high quality image of the sky by reducing sidelobe-induced artifacts. In this paper, the CLEAN concept is extended to matched field processing. Numerical simulations and experimental data demonstrate that matched field processing combined with the CLEAN algorithm can improve performance, especially when a weak source is masked by sidelobes from a much stronger source. © 2003 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

This paper deals with the analytic continuation of a coherent pressure field specified on a finite sheet located close to and conformal to the surface of a vibrator. This analytic continuation is an extension or extrapolation of the given (measured) field into a region outside and tangential to the original finite sheet, and is based on the Green’s function (the transfer function) relating acoustic quantities on the two conformal surfaces. The continuation of the measured pressure field is an inverse problem that requires the use or regularization theory, especially when noise is present in the data. An iteration algorithm is presented that is successful in continuing the pressure field into the tangential sheet. The results are accurate close to the original boundary and taper (decay) toward zero with distance away from it. The algorithm is tested on numerical and experimental data from a point-driven rectangular plate. Results show the successful extrapolation (continuation) of this data into an area nearly double that of the original pressure field. This algorithm is not limited to planar surfaces and can be applied to arbitrarily shaped surfaces. © 2003 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

In some situations of active noise control, infinite impulse response (IIR) filters are more suitable than finite impulse response (FIR) filters owing to the poles in the transfer function. A number of algorithms have been derived for applying IIR filters in active noise control; however, most of them use the direct form IIR filter structure, which faces the difficulties of checking stability and relatively slow convergence speed for noise composed of narrow-band components with large power disparity. To overcome these difficulties along with using the direct form IIR filters, a new adaptive algorithm is proposed in this paper, which uses and updates the lattice form adaptive IIR filter in an active noise control system. Full mathematical derivations of the proposed algorithm are presented, and the comparison between the proposed algorithm and the commonly used filtered-u LMS and filtered-v LMS algorithms shows the superiority of the proposed algorithm. [ABSTRACT FROM AUTHOR]

This paper addresses the efficient solution of acoustic problems in which the primary interest is obtaining the solution only on restricted portions of the domain but over a wide range of frequencies. The exterior acoustics boundary value problem is approximated using the finite element method in combination with the Dirichlet-to-Neumann (DtN) map. The restriction domain problem is formally posed in transfer function form based on the finite element solution. In order to obtain the solution over a range of frequencies, a matrix-valued Padé approximation of the transfer function is employed, using a two-sided block Lanczos algorithm. This approach provides a stable and efficient representation of the Padé approximation. In order to apply the algorithm, it is necessary to reformulate the transfer function due to the frequency dependency in the nonreflecting boundary condition. This is illustrated for the case of the DtN boundary condition, but there is no restriction on the approach which can also be applied to other radiation boundary conditions. Numerical tests confirm that the approach offers significant computational speed-up. [ABSTRACT FROM AUTHOR]

Estimating the direction of arrival of sound in three-dimensional space is typically performed by generalized time-delay processing on a set of signals from a fixed array of omnidirectional microphones. This requires specialized multichannel A/D hardware, and careful arrangement of the microphones into an array. This work is motivated by the desire to instead only use standard two-channel audio A/D hardware and portable equipment. To estimate direction of arrival of persistent sound, the position of the microphones is made variable by mounting them on one or more computer-controlled pan-and-tilt units. In this paper, we describe the signal processing and control algorithm of a device with two omnidirectional microphones on a fixed baseline and two rotational degrees of freedom. Experimental results with real data are reported with both impulsive and speech sounds in an untreated, normally reverberant indoor environment. [ABSTRACT FROM AUTHOR]

This letter deals with the problem of differential beamforming with microphone arrays of arbitrary planar geometry. By approximating the beampattern with the Jacobi-Anger expansion, it develops an algorithm that can form any specified frequency-invariant beampattern with a microphone array of any planar geometry as long as the sensors' coordinates are given and the spacing between neighboring sensors is smaller than the smallest wavelength. This method is rather general and it can be used to design differential beamformers with linear, circular, and concentric circular differential microphone arrays as well as differential arrays of arbitrary planar geometry where sensors are placed in any specified positions. [ABSTRACT FROM AUTHOR]

Acoustic wave propagation (up to 50 kHz) within a water-filled high-density polyethylene (HDPE) pipeline is studied using laboratory experiments and theoretical analysis. Experiments were carried out in a 15m length of cylindrical HDPE pipeline using acoustic transducers to acquire signals uniformly spaced along the axis of the pipe. By proposing the use of the iterative quadratic maximum likelihood algorithm to this experimental configuration, wavenumbers, attenuations, and mode amplitudes could be accurately extracted from the measurement data. To allow comparisons with theoretical analysis, dispersion curves of the wavenumbers, attenuations, and acoustic power characteristics of the axisymmetric and nonaxisymmetric modes are predicted by extending an existing waveguide model. The model extensions included the introduction of a monopole acoustic source into the water medium so that amplitude variations with respect to individual modes and frequencies could be investigated in detail. In addition, stiffness coefficients of HDPE material are carefully used to account for viscoelastic effects. The comparisons between the theoretical predictions and experimental results demonstrate a very good match and are a validation of the theoretical model. [ABSTRACT FROM AUTHOR]

In marine magnetic detection, in order to improve the accuracy of the measurement data of the field, to consider the impact of the storm interference environment, and to design a weakly driven USV tracking control system, it is necessary to drive the USV in a complex marine environment to determine the accuracy and stability of tracking. First, a 3-degrees of freedom mathematical model is built and decoupled by virtue of coordinate transformation to rely only on the absolute geographic velocity. Second, the variable forward-looking distance integral line-of-sight method is used to design the guidance law and stabilize the course tracking error. Moreover, sliding mode control is adopted to design the heading and velocity virtual control law and stabilize the heading angle and longitudinal velocity errors. Third, an adaptive observer is designed by introducing tracking errors to estimate the interference. The algorithm effectively improves the accuracy and reduces the oscillation during convergence. In the end, theoretical analysis and simulation experiments are carried out to confirm whether the proposed method is effective and advanced. [ABSTRACT FROM AUTHOR]

Indoor mobile robots are widely used in modern industry. Traditional motion control methods for robots suffer from discontinuous path curvature, low planning efficiency, and insufficient verification of theoretical algorithms. Therefore, a motion control system for an intelligent indoor robot was designed. By optimizing the radar map detecting and positioning, path planning, and chassis motion control, the performance of the system has been improved. First, a map of the warehouse environment is established, and the number of resampling particles interval is set for the Gmapping building process to improve the efficiency of map construction. Second, an improved A* algorithm is proposed, which converts the path solution with obstacles between two points into the path solution without obstacles between multiple points based on the Rapidly expanding Random Trees and Jump Point Search algorithms and further improves the pathfinding speed and efficiency of the A* algorithm by screening the necessary expansion nodes. The Dynamic Window Approach (DWA) algorithm based on the dynamic window is used to smooth the path, and the target velocity is reasonably assigned according to the kinematic model of the robot to ensure the smooth motion of the chassis. By establishing raster map models of different sizes, the traditional and improved A* pathfinding algorithms are compared and validated. The results illustrate that the improved pathfinding algorithm reduces the computing time by 67% and increases the pathfinding speed by 47% compared with the A* algorithm. Compared with the traditional method, the speed and effect are greatly improved, and the motion control system can meet the requirements of autonomous operation of mobile robots in indoor storage. [ABSTRACT FROM AUTHOR]

I describe a simple algorithm for numerically finding the ground state and low-lying excited states of a quantum system. The algorithm is an adaptation of the relaxation method for solving Poisson's equation, and is fundamentally based on the variational principle. It is especially useful for two-dimensional systems with nonseparable potentials, for which simpler techniques are inapplicable yet the computation time is minimal. [ABSTRACT FROM AUTHOR]

A modified generalized sidelobe canceller (IGSC) algorithm is proposed to enhance the resolution and robustness against the noise of the traditional generalized sidelobe canceller (GSC) and coherence factor combined method (GSC-CF). In the GSC algorithm, weighting vector is divided into adaptive and non-adaptive parts, while the non-adaptive part does not block all the desired signal. A modified steer vector of the IGSC algorithm is generated by the projection of the non-adaptive vector on the signal space constructed by the covariance matrix of received data. The blocking matrix is generated based on the orthogonal complementary space of the modified steer vector and the weighting vector is updated subsequently. The performance of IGSC was investigated by simulations and experiments. Through simulations, IGSC outperformed GSC-CF in terms of spatial resolution by 0.1◻ mm regardless there is noise or not, as well as the contrast ratio respect. The proposed IGSC can be further improved by combining with CF. The experimental results also validated the effectiveness of the proposed algorithm with dataset provided by the University of Michigan. [ABSTRACT FROM AUTHOR]

We present a new method, the solving fractional order dynamical systems using reservoir computing (RC-FODS) algorithm, for solving fractional order nonlinear dynamical systems using deep learning. The method is shown to have advantages over traditional methods, such as less calculation time and higher accuracy. This study also compares the RC-FODS algorithm with the traditional recurrent neural network and echo state network algorithms and finds that it has a higher accuracy and shorter computation time. The accuracy of the method is validated using the largest Lyapunov exponent, and the study also analyzes the advantages and disadvantages of different deep learning models. Our study concludes that the RC-FODS algorithm is a promising method for solving fractional order nonlinear dynamical systems with a high accuracy and low error rate. [ABSTRACT FROM AUTHOR]

Textbook treatments of classical mechanics typically assume that the Lagrangian is nonsingular; that is, the matrix of second derivatives of the Lagrangian with respect to the velocities is invertible. This assumption ensures that (i) Lagrange's equations can be solved for the accelerations as functions of coordinates and velocities, and (ii) the definitions of the conjugate momenta can be inverted to solve for the velocities as functions of coordinates and momenta. This assumption, however, is unnecessarily restrictive—there are interesting classical dynamical systems with singular Lagrangians. The algorithm for analyzing such systems was developed by Dirac and Bergmann in the 1950s. After a brief review of the Dirac–Bergmann algorithm, several examples are presented using familiar components: point masses connected by massless springs, rods, cords, and pulleys. [ABSTRACT FROM AUTHOR]

Reliable fundamental frequency (f0) extraction algorithms are crucial in many fields of speech research. The current bulk of studies testing the robustness of different algorithms have focused on healthy speech and/or measurements of sustained vowels. Few studies have tested f0 estimations in the context of pathological speech, and even fewer on continuous speech. The present study evaluated 12 available pitch detection algorithms on a corpus of read speech by 24 speakers (8 healthy speakers, 8 speakers with Parkinson's disease, and 8 with head and neck cancer). Two fusion methods' algorithms have been tested: one based on the median of algorithms and one based on the fusion between the best algorithm for voicing detection and the algorithm that generates the most accurate f0 estimations on voiced parts. Our results show that time-domain algorithms, like REAPER, are best for voicing detection while deep neural network algorithms, like FCN- f0, yield better accuracy for the f0 values on voiced parts. The combination of REAPER and FCN- f0 yields the best ratio performance/implementation complexity, since it generates less than 4% errors on voicing detection and less than 5% of gross errors in the estimation of the f0 values for all speaker groups. [ABSTRACT FROM AUTHOR]

Biochemical reaction networks are often modelled using discrete-state, continuoustime Markov chains. System statistics of these Markov chains usually cannot be calculated analytically and therefore estimates must be generated via simulation techniques. There is a well documented class of simulation techniques known as exact stochastic simulation algorithms, an example of which is Gillespie's direct method. These algorithms often come with high computational costs, therefore approximate stochastic simulation algorithms such as the tau-leap method are used. However, in order to minimise the bias in the estimates generated using them, a relatively small value of tau is needed, rendering the computational costs comparable to Gillespie's direct method. The multi-level Monte Carlo method (Anderson and Higham, Multiscale Model. Simul. 10:146-179, 2012) provides a reduction in computational costs whilst minimising or even eliminating the bias in the estimates of system statistics. This is achieved by first crudely approximating required statistics with many sample paths of low accuracy. Then correction terms are added until a required level of accuracy is reached. Recent literature has primarily focussed on implementing the multilevel method efficiently to estimate a single system statistic. However, it is clearly also of interest to be able to approximate entire probability distributions of species counts. We present two novel methods that combine known techniques for distribution reconstruction with the multi-level method. We demonstrate the potential of our methods using a number of examples. [ABSTRACT FROM AUTHOR]

Classical beamforming algorithms are based on rectilinear propagation of signals that are corrupted by additive noise. However, sound waves propagating in the atmosphere are refracted by temperature and wind gradients and distorted by turbulence and scattering from the ground. These effects can reduce the performance of acoustic systems. To investigate these effects, a simulation capability is used to propagate signals from elevated sources to an array of microphones near the ground. Detection algorithms based on four standard beamforming algorithms, implemented with either no model mismatch error or model mismatch errors generated by atmospheric refraction, are tested. The results are evaluated using the area under the receiver operating characteristic curve statistic. The analysis shows no significant improvement in detection performance when refraction effects are included in the beamforming algorithms over a wide range of frequencies. Any small performance improvements achieved by incorporating refraction effects into the beamforming algorithms are overwhelmed by the negative effects of atmospheric turbulence and rough surface scattering. For the selected scenario, incorporating refraction effects into the beamformers is over-engineering. These results demonstrate some of the limitations of developing new signal processing algorithms or array designs for atmospheric acoustic systems. [ABSTRACT FROM AUTHOR]

A major challenge in critical dimension atomic force microscope width metrology is accounting for the effects of the tip on the apparent features in an image. The overall effect of the tip is to broaden the apparent width of lines and narrow the apparent width of trenches as a result of the geometrical interaction of the tip with the surface. To a first approximation, this effect can be well-modeled as a constant bias in width measurements, independent of the specific feature characteristics, by the width of the tip. Beyond this approximation, there are a number of smaller tip effects related to the measurement details. Some of these result from the details and interactions of the tip shape and feature shape, resulting in small variations of the tip bias from feature to feature and measurement to measurement. These effects are sometimes called shape effects, secondary effects, or higher order tip effects. One source of higher order effects is due to the lateral dithering of the tip, which increases the effective tip width. However, in addition to this known effect, the tip dither may also affect the apparent edge heights of the tip flare. Although the apparent tip width is expected to depend on lateral dither, the apparent edge height, at least in principle, would not be expected to depend on lateral dither. Our analysis suggests that this apparent dependence results from the interaction of the fine details of the tip flare shape and the dither envelope with the specific algorithm used to estimate the edge height. Consequently, both the sign and the magnitude of the dependence are specific to every tip. This is one example of the inter-dependencies that can be present when evaluating higher-order tip effects. Although these are often small, accurate metrology in some applications may require consideration of possible interactions between such smaller, secondary tip effects. The authors illustrate this with examples from photomask metrology, where measurement of chrome features on binary masks requires some attention to secondary tip effects. [ABSTRACT FROM AUTHOR]

Traditional loudspeaker equalization algorithms cannot decide the order of an equalizer before the whole equalization procedure has been completed. Designers have to try many times before they determine a proper order of the equalization filter. A method which solves this drawback is presented for loudspeaker equalization using balanced model truncation. The order of the equalizer can be easily decided using this algorithm and the error between the model and the loudspeaker can also be readily controlled. Examples are presented and the performance of the proposed method is discussed with comparative experiments. [ABSTRACT FROM AUTHOR]

Ray tracing is a simple and efficient three-dimensional method which reduces the problem of infrasound propagation to a series of one-dimensional cases along acoustical rays. However, in relatively frequent cases, infrasound stations are located in geometrical shadow zones, where only diffracted waves are recorded. The corresponding arrivals cannot be predicted by ray theory. To simulate infrasound propagation in these zones, the ray tracing method is generalized to complex ray theory. The source, media, and ground parameters are all considered as complex numbers. For applications with realistic atmospheric data, including stratified temperature and wind as well as the range dependency of atmospheric profiles, an efficient algorithm determining complex eigenrays in the shadow zones is presented. It is illustrated by a two-dimensional case of a point source. [ABSTRACT FROM AUTHOR]

Near-field acoustical holography (NAH) is a useful tool for sound field reconstruction and sound source identification. In NAH, a basis model is first selected to represent the physical sound field, and then a near-field measurement is made with a microphone array. Next, the parameters in the selected model can be estimated based on the measurements by using an inverse approach, resulting in the sound field near the source being reconstructed so that the sound source location can be identified. But, in addition to being able to reconstruct the near-field of a source, the far-field can also be predicted with the identified sound source model. A significant amount of work has been performed to study the near-field reconstruction capability of different NAH algorithms, but there has been a limited number of publications in which the far-field prediction accuracy, based on the near-field measurement constructed model, has been considered. In the present experimental work, two multi-transducer loudspeakers were placed side-by-side to create a multi-component sound source, and two sets of measurements were conducted: an intensity probe scanned the sound field generated by the loudspeakers in both the near-field (0.05 m) and far-field (0.48 m) such that the true near- and far-field intensity spatial distributions and total sound power could be identified. Then, based on the near-field pressure measurements, four acoustical holography algorithms, statistically optimized near-field acoustical holography, wideband acoustical holography, l1-norm minimization, and a hybrid compressive sampling method were used to predict the near- and far-field sound intensity distributions. The near- and far-field prediction results were compared with the direct measurement, and the sound field reconstruction accuracy was studied. It was found that all of the algorithms were able to reconstruct the near-field well when the near-field measurements were used to construct the model. It was found that with the abovementioned models, far-field reconstructions could correctly predict the spatial sound field distribution, but in all of the cases, the total sound power was underestimated. [ABSTRACT FROM AUTHOR]

Memristor-based computer architectures are becoming more attractive as a possible choice of hardware for the implementation of neural networks. However, at present, memristor technologies are susceptible to a variety of failure modes, a serious concern in any application where regular access to the hardware may not be expected or even possible. In this study, we investigate whether certain training algorithms may be more resilient to particular hardware failure modes and, therefore, more suitable for use in those applications. We implement two training algorithms—a local update scheme and a genetic algorithm—in a simulated memristor crossbar and compare their ability to train for a simple image classification task as an increasing number of memristors fail to adjust their conductance. We demonstrate that there is a clear distinction between the two algorithms in several measures of the rate of failure to train. [ABSTRACT FROM AUTHOR]

A method is presented for the reconstruction of rotating monopole source distributions using acoustic pressures measured on a sideline parallel to the source axis. The method requires no a priori assumptions about the source other than that its strength at the frequency of interest varies sinusoidally in azimuth on the source disk so that the radiated acoustic field is composed of a single circumferential mode. When multiple azimuthal modes are present, the acoustic field can be decomposed into azimuthal modes and the method applied to each mode in sequence. The method proceeds in two stages, first finding an intermediate line source derived from the source distribution and then inverting this line source to find the radial variation in source strength. A far-field form of the radiation integrals is derived, showing that the far-field pressure is a band-limited Fourier transform of the line source, establishing a limit on the quality of source reconstruction, which can be achieved using far-field measurements. The method is applied to simulated data representing wind-tunnel testing of a ducted rotor system (tip Mach number of 0.74) and to control of noise from an automotive cooling fan (tip Mach number of 0.14), studies which have appeared in the literature of source identification. [ABSTRACT FROM AUTHOR]

A technique for the recording of large sets of room impulse responses or head-related transfer functions is presented. The technique uses a microphone moving with constant speed. Given a setup (e.g., length of the room impulse response), a careful choice of the recording parameters (excitation signal, speed of movement) leads to the reconstruction of all impulse responses along the trajectory. In the case of a moving microphone along a circle, the maximal angular speed is given as a function of the length of the impulse response, its maximal temporal frequency, the speed of sound propagation, and the radius of the circle. As a result of the presented algorithm, head-related transfer functions sampled at 44.1 kHz can be measured at all angular positions along the horizontal plane in less than 1 s. The presented theory is compared with a real system implementation using a precision moving microphone holder. The practical setup is discussed together with its limitations. [ABSTRACT FROM AUTHOR]

Broadband array transducers are widely used in medical imaging systems, and their beam patterns determine the imaging quality. The characteristics of the beam patterns such as mainlobe beamwidth and sidelobe levels are the major criteria for designing and optimizing sparse array systems. The spatial impulse response approach can be used to calculate the beam patterns rigorously in both near field and far field. However, it is time consuming due to the high sampling frequency required, and thus not suitable for array optimization. On the other hand, assuming the array elements to be omnidirectional point sources the field calculation can be much simplified and fast but at the expense of poor accuracy. This paper presents a modified model that has the accuracy comparable to that from the impulse response model while keeping the efficiency close to that of the simple point source model. The developed algorithm is particularly suitable for stochastic methods for two-dimensional (2D) sparse arrays design and optimization such as genetic algorithms (GA), in which the iterative beam pattern calculation dominates the optimization program running time. [ABSTRACT FROM AUTHOR]

This paper shows an accurate speech detection algorithm for improving the performance of speech recognition systems working in noisy environments. The proposed method is based on a hard decision clustering approach where a set of prototypes is used to characterize the noisy channel. Detecting the presence of speech is enabled by a decision rule formulated in terms of an averaged distance between the observation vector and a cluster-based noise model. The algorithm benefits from using contextual information, a strategy that considers not only a single speech frame but also a neighborhood of data in order to smooth the decision function and improve speech detection robustness. The proposed scheme exhibits reduced computational cost making it adequate for real time applications, i.e., automated speech recognition systems. An exhaustive analysis is conducted on the AURORA 2 and AURORA 3 databases in order to assess the performance of the algorithm and to compare it to existing standard voice activity detection (VAD) methods. The results show significant improvements in detection accuracy and speech recognition rate over standard VADs such as ITU-T G.729, ETSI GSM AMR, and ETSI AFE for distributed speech recognition and a representative set of recently reported VAD algorithms. [ABSTRACT FROM AUTHOR]

The decomposition of the time reversal operator (DORT) is a detection and focusing technique using an array of transmit-receive transducers. In the absence of noise and under certain conditions, the eigenvectors of the time reversal operator contain the focal laws to focus ideally on well-resolved scatterers even in the presence of strong aberration. This paper describes a new algorithm, FDORT, which uses focused transmission schemes to acquire the operator. It can be performed from medical scanner data. A mathematical derivation of this algorithm is given and it is compared with the conventional algorithm, both theoretically and with numerical experiments. In the presence of strong speckle signals, the DORT method usually fails. The influence of the speckle noise is explained and a solution based on FDORT is presented, that enables detection of targets in complex media. Finally, an algorithm for microcalcification detection is proposed. In-vivo results show the potential of these techniques. [ABSTRACT FROM AUTHOR]

This paper describes results of geoacoustic inversion using broadband signals from an experiment carried out at a site near the South Florida Ocean Measurement Centre in the Florida Straits. M-sequence coded pulse trains at different center carrier frequencies from 100 to 3200 Hz were recorded at a vertical line array at a distance around 10 km. Geoacoustic inversion was carried out to determine the feasibility of inverting the environmental parameters from this long-range propagation experiment. The received signal at lower frequencies below 400 Hz consisted of a dominant water column signal and a secondary arrival delayed by 0.4 s. The secondary signal was spatially filtered by beamforming the array data, and the beam data were inverted by matched beam processing in the time domain, combined with an adaptive simplex simulated annealing algorithm. The estimated values of compressional wave speed and density were in good agreement with ground truth values from sediment cores. The inverted shear wave speed appears to be a sensitive parameter and consistent with compressional wave speed. As a cross check, range and water depth were also included as inversion parameters, and the inversion results were close to the known values within small uncertainties. [ABSTRACT FROM AUTHOR]

Recently, development of a numerical calculation of the head-related transfer function (HRTF) has been conducted using a computer model of a human head and the boundary element method. The reciprocity theorem is incorporated into the computational process in order to shorten the computational time, which is otherwise very long. On the other hand, another fast HRTF calculation method for any source position, which is realized by calculating factors independent of the source position in advance, has been suggested by the authors. Using this algorithm, the HRTF for any source position can be obtained in a few seconds with a common PC. The resulting HRTFs are more precise and are calculated faster than those by using the reciprocity theorem. However, speeding the process up even further is required in order to respond to a head movement and rotation or to moving sources during binaural sound reproduction. In this paper, a faster calculation method by incorporating a time domain operation into the authors’ previous algorithm is proposed. Additionally, the new formulation, which eliminates the extra computational time in the preprocess, is proposed. This method is shown to be faster than the previous ones, but there are some discrepancies at higher frequencies. [ABSTRACT FROM AUTHOR]

Noise degrades the accuracy of sonar systems. We demonstrate a practical method for increasing the effective signal-to-noise ratio (SNR) by fusing time delay information from a burst of multiple sonar pings. This approach can be useful when there is no relative motion between the sonar and the target during the burst of sonar pinging. Otherwise, the relative motion degrades the fusion and therefore, has to be addressed before fusion can be used. In this paper, we present a robust motion estimation algorithm which uses information from multiple receivers to estimate the relative motion between pings in the burst. We then compensate for motion, and show that the fusion of information from the burst of motion compensated pings improves both the resilience to noise and sonar accuracy, consequently increasing the operating range of the sonar system. [ABSTRACT FROM AUTHOR]

Platelike structures, made of composites, are being increasingly used for fabricating aircraft wings and other aircraft substructures. Continuous monitoring of the health of these structures would aid the reliable operation of aircrafts. This paper considers the use of a Lamb wave based structural health monitoring (SHM) system to identify and locate defects in large multilayered composite plates. The SHM system comprises of a single transmitter and multiple receivers, coupled to one side of the plate that send and receive Lamb waves. The proposed algorithm processes the data collected from the receivers and generates a reconstructed image of the material state of the composite plate. The algorithm is based on phased addition in the frequency domain to compensate for the dispersion of Lamb waves. In addition, small deviations from circularity of the slowness curves of Lamb wave modes, due to anisotropy, are corrected for by assuming that the phase and group velocity directions coincide locally. Experiments were performed on an anisotropic multilayered composite plate containing a single defect. Reconstruction of the defect is carried out using data for a weakly anisotropic Lamb wave mode as a proof of concept of the proposed algorithm. [ABSTRACT FROM AUTHOR]

In this paper, a gain function for noise cancellation with a two-channel microphone array is presented. This gain function combines ideas from one- and multichannel algorithms. It is developed using a minimum mean square error estimator for the amplitude of the speech signal from the cross spectrum between two microphone signals. To consider speech pauses and the absence of spectral components of the speech, an extension of this gain function is presented. The performance of the overall gain function is shown in terms of the cancellation of (diffuse) driving noise as well as the cancellation of an interfering speech signal, both recorded in a car. [ABSTRACT FROM AUTHOR]

This paper presents the hybrid compliance-stiffness matrix method for stable analysis of elastic wave propagation in multilayered anisotropic media. The method utilizes the hybrid matrix of each layer in a recursive algorithm to deduce the stack hybrid matrix for a multilayered structure. Like the stiffness matrix method, the hybrid matrix method is able to eliminate the numerical instability of transfer matrix method. By operating with total stresses and displacements, it also preserves the convenience for incorporating imperfect or perfect interfaces. However, unlike the stiffness matrix, the hybrid matrix remains to be well-conditioned and accurate even for zero or small thicknesses. The stability of hybrid matrix method has been demonstrated by the numerical results of reflection and transmission coefficients. These results have been determined efficiently based on the surface hybrid matrix method involving only a subset of hybrid submatrices. In conjunction with the recursive asymptotic method, the hybrid matrix method is self-sufficient without hybrid asymptotic method and may achieve low error level over a wide range of sublayer thickness or the number of recursive operations. [ABSTRACT FROM AUTHOR]

A modification of the spectrogram (log magnitude of the short-time Fourier transform) to more accurately show the instantaneous frequencies of signal components was first proposed in 1976 [Kodera et al., Phys. Earth Planet. Inter. 12, 142–150 (1976)], and has been considered or reinvented a few times since but never widely adopted. This paper presents a unified theoretical picture of this time-frequency analysis method, the time-corrected instantaneous frequency spectrogram, together with detailed implementable algorithms comparing three published techniques for its computation. The new representation is evaluated against the conventional spectrogram for its superior ability to track signal components. The lack of a uniform framework for either mathematics or implementation details which has characterized the disparate literature on the schemes has been remedied here. Fruitful application of the method is shown in the realms of speech phonation analysis, whale song pitch tracking, and additive sound modeling. [ABSTRACT FROM AUTHOR]

This paper presents the recursive algorithm of stiffness matrix method with improved efficiency for computing the total and surface stiffness matrices for a general multilayered anisotropic media. Based on the eigensolutions commonly available for analysis of such media, the recursive algorithm deals with eigen-submatrices directly and bypasses all intermediate layer stiffness submatrices. The improved algorithm obviates the need to compute certain inverse of the original scheme and makes the stiffness matrix recursion more robust. In situation where transfer matrix is numerically stable and easily accessible, an improved recursive algorithm is also given directly in terms of transfer submatrices without involving their explicit inverse. [ABSTRACT FROM AUTHOR]

This paper focuses on increasing the accuracy of low-order (four-node quadrilateral) finite elements for the transient analysis of wave propagation. Modified integration rules, originally proposed for time-harmonic problems, provide the basis for the proposed technique. The modified integration rules shift the integration points to locations away from the conventional Gauss or Gauss-Lobatto integration points with the goal of reducing the discretization errors, specifically the dispersion error. Presented here is an extension of the idea to time-dependent analysis using implicit as well as explicit time-stepping schemes. The locations of the stiffness integration points remain unchanged from those in time-harmonic case. On the other hand, the locations of the integration points for the mass matrix depend on the time-stepping scheme and the step size. Furthermore, the central difference method needs to be modified from its conventional form to facilitate fully explicit computation. The superior performance of the proposed algorithms is illustrated with the help of several numerical examples. [ABSTRACT FROM AUTHOR]

In this paper we demonstrate the possibility of designing a radiator using structural-acoustic interaction by predicting the pressure distribution and radiation pattern of a structural-acoustic coupling system that is composed by a wall and two spaces. If a wall separates spaces, then the walls role in transporting the acoustic characteristics of the spaces is important. The spaces can be categorized as bounded finite space and unbounded infinite space. The wall considered in this study composes two plates and an opening, and the wall separates one space that is highly reverberant and the other that is unbounded without any reflection. This rather hypothetical circumstance is selected to study the general coupling problem between the finite and infinite acoustic domains. We developed an equation that predicts the energy distribution and energy flow in the two spaces separated by a wall, and its computational examples are presented. Three typical radiation patterns that include steered, focused, and omnidirected are presented. A designed radiation pattern is also presented by using the optimal design algorithm. [ABSTRACT FROM AUTHOR]

In this paper we present an algorithm for building an empirical model of facial biomechanics from a set of displacement records of markers located on the face of a subject producing speech. Markers are grouped into clusters, which have a unique primary marker and a number of secondary markers with an associated weight. Motion of the secondary markers is computed as the weighted sum of the primary markers of the clusters to which they belong. This model may be used to produce facial animations, by driving the primary markers with measured kinematic signals. [ABSTRACT FROM AUTHOR]

Geoacoustic inversion using a matched-field inversion algorithm is a well-established technique for estimating the geoacoustic parameters of the seabed. This paper demonstrates how parameter estimation can be affected by unknown or wishfully ignored random range dependence of the true environment when the inversion model is—for practicality—assumed to be range independent. Simulations with controlled statistics were carried out using a simple shallow water model: an isospeed water column over a homogeneous elastic halfspace. The inversion parameters included water depth, compressional speed in the seabed, seabed density, and compressional wave attenuation. On average the environment is range independent: some parameters are constant while other parameters are random with range-independent means and variances. A Parabolic Equation underwater acoustic propagation model is used to calculate the simulated data fields for the range-dependent environment as well as to calculate the model fields for the range-independent inversion model. The Adaptive Simplex Simulated Annealing inversion algorithm is used to estimate the best-fit solution. It is found that ignoring the variability of even a single geoacoustic parameter leads to significant and correlated uncertainty (bias and variance) in the estimation of all inverted parameters. Results are presented for range variation of compressional sound speed and water depth. © 2005 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

The security of messages encoded via the widely used RSA public key encryption system rests on the enormous computational effort required to find the prime factors of a large number N using classical (conventional) computers. In 1994 Peter Shor showed that for sufficiently large N, a quantum computer could perform the factoring with much less computational effort. This paper endeavors to explain, in a fashion comprehensible to the nonexpert, the RSA encryption protocol; the various quantum computer manipulations constituting the Shor algorithm; how the Shor algorithm performs the factoring; and the precise sense in which a quantum computer employing Shor's algorithm can be said to accomplish the factoring of very large numbers with less computational effort than a classical computer. It is made apparent that factoring N generally requires many successive runs of the algorithm. Our analysis reveals that the probability of achieving a successful factorization on a single run is about twice as large as commonly quoted in the literature. [ABSTRACT FROM AUTHOR]

The seismo-acoustic method is one of the most promising emerging techniques for the detection of landmines. Numerous field tests have demonstrated that buried landmines manifest themselves at the surface through linear and nonlinear responses to acoustic/seismic excitation. The present paper describes modeling of the nonlinear response in the framework of the massspring model of the soilmine system. The perturbation method used in the model allows for the derivation of an analytical solution describing both quadratic and cubic acoustic interactions at the soilmine interface. This solution has been compared with actual field measurements to obtain nonlinear parameters of the buried mines. These parameters have been analyzed with respect to mine types and burial depths. It was found that the cubic nonlinearity could be a significant contributor to the nonlinear response. This effect has led to the development of a new intermodulation detection algorithm based on dual-frequency excitation. Both quadratic and intermodulation nonlinear algorithms were evaluated at the U.S. Army outdoor testing facilities. The algorithms appear to complement each other in improving the overall detection performance. 2005 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Multichannel auralizations based on spatial room impulse responses often employ sample-wise assignment of an omnidirectional response to form loudspeaker responses. This leads to sparse impulse responses in each reproduction loudspeaker and the auralization of transient signals can sound rough. Based on this observation, we conducted a listening test to examine the general phenomenon of roughness due to spatial assignment. First, participants assessed the roughness of both Gaussian noise and velvet noise, assigned sample-wise to up to 36 loudspeakers by two algorithms. The first algorithm assigns channels merely by selecting random indices, while the second one constrains the time between two peaks on each channel. The results show that roughness already occurs when few channels are used and that the assignment algorithm influences it. In a second experiment, virtualizations of the test were used to examine the factors contributing to increased roughness. We systematically show the effect of spatial assignment on noise and conclude that besides time-differences, level-differences caused by head-shadowing are the principal cause for the perceived roughness. The results have significance in spatial room impulse response rendering and spatial reverberator design. [ABSTRACT FROM AUTHOR]

Active noise control algorithms aimed at reducing Gaussian noise have been studied. However, the performance of these algorithms can deteriorate if impulsive noise or other non-Gaussian noise such as piling noise exists. To solve this problem, an algorithm named the filtered-x affine projection sign algorithm with a post-adaptive filter is proposed to reduce impulsive noise. The proposed algorithm is based on the filtered-x affine projection sign algorithm and the post-adaptive filter structure. In addition, an algorithm with a variable step size is proposed to improve the performance of the filtered-x affine projection sign algorithm with a post-adaptive filter by using a convex combination strategy. The fixed step size in the convex combination is replaced with a linear function of the ratio between the estimated error signal and the estimated desired signal. The detailed derivation processes of the proposed algorithms are given. The computational complexities are analyzed. Numerical simulations are carried out to demonstrate the performance of the proposed algorithms. The results illustrate that the proposed algorithms can effectively reduce impulsive noise. At the same time, the algorithm with a variable step size has better convergence performance than the filtered-x affine projection sign algorithm with a post-adaptive filter. [ABSTRACT FROM AUTHOR]

This work addresses the shear response of entangled linear polymers by assessing the current state-of-the-art model and proposing alternative directions. In particular, we examine the performance of the Graham–Likhtman–McLeish–Milner (GLaMM) nonlinear tube model in fast shear flows. Predictions are compared against experimental data for well-characterized, monodisperse, entangled linear polystyrene chains. Unlike previous works using the GLaMM model, finite extensibility is accounted for. Comparison of model predictions and data not only reveals an overall reasonable performance but also highlights limitations. For example, the predictions significantly depend on whether the contribution of contour length fluctuations to the retraction rate is accounted for or not. This specific sensitivity of the model is enhanced as the Rouse–Weissenberg number increases. Possible improvement of the model by modification of the existing mechanisms/model assumptions and/or by incorporation of overlooked mechanisms such as chain tumbling and CCR-driven disentanglement (CCR-D) is discussed. As a possible way to overcome the existing limitations, we propose an alternative approach based on the time marching algorithm (TMA) and focus on the description of the steady-state regime. Compared to the GLaMM model, this approach allows us to keep track of the relaxation state of each entanglement segment in a simple way, while ensuring consistency between chain stretch and constraint release mechanisms. Moreover, it accounts for CCR-D in an indirect manner. A key ingredient is the use of the recently advanced concept of the shear blob and its dependence on the shear rate. Our simple modification of TMA reproduces the experimental steady-state viscosities accurately for all samples and for all shear rates examined. [ABSTRACT FROM AUTHOR]

Automated generation of reasonable atomic-level interface models, for example, at a grain boundary, is generally computationally intensive partly because of the three degrees of freedom in a rigid-body translation (RBT) of one side of the interface against the other. We propose an algorithm to obtain reasonable interface models using as few first-principles calculations as possible. The valence charge densities of two surface slabs constituting the interface are calculated using first-principles calculations. The surface charge densities are filtered with an exponential function using a parameter λ to obtain the reaction front. Models where the overlap of filtered charge densities between the two slabs takes a local maximum are adopted as initial models with desirable RBTs, which are then relaxed using first-principles calculations to obtain a reasonable interface model. The proposed algorithm successfully generated reasonable initial models for three out of three orientations in 75% of homointerfaces of body-centered cubic, face-centered cubic, and hexagonal close-packed non-magnetic elementary metals. For the Al {001} Σ5 twist grain boundary, the present algorithm also reproduced γ-surface features of RBTs showing correct displacement shift complete lattice periodicity. Further modifications and improvements to this method are expected to accelerate automated interface model generation from a previously unexplored approach. [ABSTRACT FROM AUTHOR]

The application of the Sliding Frank-Wolfe algorithm to gridless compressive beamforming is investigated for single and multi-snapshot measurements and the estimation of the three-dimensional (3D) position of the sources and their amplitudes. Sources are recovered by solving an infinite dimensional optimization problem, promoting sparsity of the solutions, and avoiding the basis mismatch issue. The algorithm does not impose constraints on the source model or array geometry. A variant of the algorithm is proposed for greedy identification of the sources. The experimental results and Monte Carlo simulations in 3D settings demonstrate the performances of the method and its numerical efficiency compared to the state of the art. [ABSTRACT FROM AUTHOR]

The quantum Monte Carlo algorithm can provide significant speedup compared to its classical counterpart. So far, most reported works have utilized Grover's state preparation algorithm. However, this algorithm relies on costly controlled Y rotations to apply the correct amplitudes onto the superposition states. Recently, a comparison-based state preparation method was proposed to reduce computational complexity by avoiding rotation operations. One critical aspect of this method is the generation of the comparison threshold associated with the amplitude of the quantum superposition states. The direct computation of the comparison threshold is often very costly. An alternative is to estimate the threshold with a Taylor approximation. However, Taylor approximations do not work well with heavy-tailed distribution functions such as the Cauchy distribution, which is widely used in applications such as financial modeling. Therefore, a new state preparation method needs to be developed. In this study, an efficient comparison-based state preparation method is proposed for the heavy-tailed Cauchy distribution. Instead of a single Taylor approximation for the entire function domain, this study uses quantum piecewise arithmetic to increase accuracy and reduce computational cost. The proposed piecewise function is in the simplest form to estimate the comparison threshold associated with the amplitudes. Numerical analysis shows that the number of required subdomains increases linearly as the maximum tolerated approximation error decreases exponentially. 197 subdomains are required to keep the error below 1 8192 of the maximum amplitude. Quantum parallelism ensures that the computational complexity of estimating the amplitudes is independent from the number of subdomains. [ABSTRACT FROM AUTHOR]

Quantum algorithms offer efficient solutions to computational problems that are expensive to solve classically. Publicly available quantum computers, such as those provided by IBM, can now be used to run small quantum circuits that execute quantum algorithms. However, these quantum computers are highly prone to noise. Here, we introduce important concepts of quantum circuit noise and connectivity that must be addressed to obtain reliable results on quantum computers. We utilize several examples to show how noise scales with circuit depth. We present Simon's algorithm, a quantum algorithm for solving a computational problem of the same name, explain how to implement it in IBM's Qiskit platform, and compare the results of running it both on a noiseless simulator and on physical hardware subject to noise. We discuss the impact of Qiskit's transpiler, which adapts ideal quantum circuits for physical hardware with limited connectivity between qubits. We show that even circuits of only a few qubits can have their success rate significantly reduced by quantum noise unless specific measures are taken to minimize its impact. [ABSTRACT FROM AUTHOR]

How many ways can we explore the Sun? We have images in many wavelengths and squiggly lines of many parameters that we can use to characterize the Sun. We know that while the Sun is blindingly bright to the naked eye, it also has regions that are dark in some wavelengths of light. All of those classifications are based on vision. Hearing is another sense that can be used to explore solar data. Some data, such as the sunspot number or the extreme ultraviolet spectral irradiance, can be readily sonified by converting the data values to musical pitches. Images are more difficult. Using a raster scan algorithm to convert a full-disk image of the Sun to a stream of pixel values creates variations that are dominated by the pattern of moving on and off the limb of the Sun. A sonification of such a raster scan will contain discontinuities at the limbs that mask the information contained in the image. As an alternative, Hilbert curves are continuous space-filling curves that map a linear variable onto the two-dimensional coordinates of an image. We have investigated using Hilbert curves as a way to sample and analyze solar images. Reading the image along a Hilbert curve keeps most neighborhoods close together as the resolution (i.e., the order of the Hilbert curve) increases. It also removes most of the detector size periodicities and may reveal larger-scale features. We present several examples of sonified solar data, including sunspot number, extreme ultraviolet (EUV) spectral irradiances, an EUV image, and a sequence of EUV images during a filament eruption. [ABSTRACT FROM AUTHOR]

This work presents a single-channel speech enhancement (SE) framework based on the super-Gaussian extension of the joint maximum a posteriori (SGJMAP) estimation rule. The developed SE algorithm is an open-source research smartphone-based application for hearing improvement studies. In this algorithm, the SGJMAP-based estimation for noisy speech mixture is smoothed along the frequency axis by a Mel filter-bank, resulting in a Mel-warped frequency-domain SGJMAP estimation. The impulse response of this Mel-warped estimation is obtained by applying a Mel-warped inverse discrete cosine transform (Mel-IDCT). This helps in filtering out the background noise and enhancing the speech signal. The proposed application is implemented on an iPhone (Apple, Cupertino, CA) to operate in real time and tested with normal-hearing (NH) and hearing-impaired (HI) listeners with different types of hearing aids through wireless connectivity. The objective speech quality and intelligibility test results are used to compare the performance of the proposed algorithm to existing conventional single-channel SE methods. Additionally, test results from NH and HI listeners show substantial improvement in speech recognition with the developed method in simulated real-world noisy conditions at different signal-to-noise ratio levels. [ABSTRACT FROM AUTHOR]