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Sound field separation techniques are an advancing tool for extracting a target sound field from a mixed sound field. However, the methods bear a high measurement cost due to the restriction of the sampling theorem. In this study, a sound field separation method based on sparse sampling is established. The method initially utilizes dictionary learning to generate a sparse basis of the sound field. Then, a mixed sound field can be precisely recovered from sparse sampling of sound pressure and the target sound field can be extracted based on the recovered sound field by means of the theory of equivalent source method. The method is validated by numerical simulations. Compared to sound field separation based on the equivalent source method, the proposed method has advantage in terms of both the accuracy and the stability for sparse sampling. [ABSTRACT FROM AUTHOR]

The quantum-mechanical expression for the polarization of a crystalline solid does not bear any resemblance to the (trivial) expression for the dipole of a bounded crystallite; in fact, it has been proved via a conceptually different path. Here, I show how to alternatively define the dipole of a bounded sample in a somewhat unconventional way; from such a formula, the crystalline polarization formula—as routinely implemented in electronic-structure codes—follows almost seamlessly. [ABSTRACT FROM AUTHOR]

Frequency measurement is one of the key techniques in high-precision data acquisition technology of broadband signals. Generally, frequency measurement not only needs to deal with a large amount of data processing but also requires a high precision, but these two aspects are sometimes difficult to reconcile. Some algorithms are overly dependent on the accuracy of the to-be-measured data, which might not be the desired option for real projects since it is almost impossible to get ideal error-free data. This article adopts a frequency measurement method based on the coordinate rotation digital computer algorithm, differential algorithm, and Kalman filter. The use of these algorithms for the frequency measurement process would not only simplify the calculation but also reduce the effect of the measurement error. This method can measure all signals that satisfy the sampling theorem and can also measure multi-channel parallel signals. The experimental results of data simulation and actual measurement on the hardware platform show that the accurate frequency measurement algorithm has a strong data processing ability, stable measurement, and steady improvement in the accuracy of measurement results, which can meet the needs of most instruments for accurate frequency measurement. The measurement error could be reduced to the percentile by the Kalman filter and could be reduced to below the thousandth by the combining the algorithms. [ABSTRACT FROM AUTHOR]

The goal of this paper is to present the behavior of a jet shear layer in response to acoustic excitation from a signal processing perspective. The main idea is that the vortices that roll-up in the jet shear layer are similar to the discrete samples of a digital control system, and, hence, that the Nyquist–Shannon sampling theorem should apply. We further hypothesize that the strength of a vortex is determined by the mean amplitude of the excitation waveform during its creation. We also argue that, at least in some cases, demodulation occurs as a result of the vorticity signal generated by the convection of discrete vortices past a point in the shear layer. This vorticity signal is related to the amplitude modulation (AM) excitation waveform by a half-wave rectification operation, a common implementation of an AM demodulator. To investigate these ideas, a free, round jet that is excited upstream of the nozzle is studied using particle image velocimetry. Experiments are conducted that confirm that the sampling theorem applies, and an aliased response is observed when the Nyquist limit is exceeded. Previous authors have attributed demodulation to a vortex merging mechanism, but we demonstrate that merging is not always required for demodulation and suggest that it is one of two mechanisms at play. [ABSTRACT FROM AUTHOR]

We derive a compressive sampling method for acoustic field reconstruction using field measurements on a pre-defined spherical grid that has theoretically guaranteed relations between signal sparsity, measurement number, and reconstruction accuracy. This method can be used to reconstruct band limited spherical harmonic or Wigner D-function series (spherical harmonic series are a special case) with sparse coefficients. Contrasting typical compressive sampling methods for Wigner D-function series that use arbitrary random measurements, the new method samples randomly on an equiangular grid, a practical and commonly used sampling pattern. Using its periodic extension, we transform the reconstruction of a Wigner D-function series into a multi-dimensional Fourier domain reconstruction problem. We establish that this transformation has a bounded effect on sparsity level and provide numerical studies of this effect. We also compare the reconstruction performance of the new approach to classical Nyquist sampling and existing compressive sampling methods. In our tests, the new compressive sampling approach performs comparably to other guaranteed compressive sampling approaches and needs a fraction of the measurements dictated by the Nyquist sampling theorem. Moreover, using one-third of the measurements or less, the new compressive sampling method can provide over 20 dB better de-noising capability than oversampling with classical Fourier theory. [ABSTRACT FROM AUTHOR]

Current standards for the measurement of impulse noise (e.g., MIL-STD-1474E) recommend using a sampling rate of at least 200 kHz in order to accurately estimate the risk of hearing damage. The given motivation for this high sampling rate is to ensure a temporal resolution in the impulse waveform fine enough to accurately capture the peak pressure. However, the Nyquist-Shannon sampling theorem specifies that a sampled signal can accurately reconstruct both the amplitude and phase information of a signal given the sampling rate is at least twice the highest frequency present in the original signal. Thus, it is possible to reconstruct a band limited signal with the same temporal resolution as one captured at a higher sampling rate if the contributions of energy above the Nyquist rate can be ignored. In this study, resampling techniques are applied to a signal sampled at 48 kHz to extract A-weighted sound pressure energy estimates within 0.1 dB of those obtained at a higher sampling rate. Our results suggest sampling rates for impulsive noise should be based on the range of frequencies expected to make a contribution to injury risk rather than on concerns about temporal resolution. [ABSTRACT FROM AUTHOR]

The past decades have witnessed great efforts in the on-demand ultrasonic field design in which the time reversal technology was widely used in the whole-space acoustic hologram. In practice, the acoustic field of interest is usually bounded in a finite region with flexible distribution. Here, the use of prolate spheroidal wave functions to generate an arbitrary ultrasonic field in a finite region is proposed. The prolate spheroidal functions, which form a complete set of band limited functions and are orthogonal in the infinite and finite regions, can be efficiently reconstructed by the sampling theorem. To display the validation of the proposed method, two types of functional ultrasonic fields are numerically simulated. One type is the ultrasound standing wave field for which six nodes and two nodes are separately realized for two different types of standing waves in the limited range of (− 2 λ , 2 λ). In addition, a composite standing wave field is stimulated with more complicated nodal distributions. The other type is the ultrasound focusing field, where three focal spots with the mainlobe sizes of λ , 0.5 λ , and 0.35 λ are demonstrated. It is worth noting that the nontrivial side lobes for super-oscillation focusing are designed to be about 3 λ away from the central focal spot (the mainlobe size 0.35 λ). This work has much significance in the applications of acoustic tweezing, ultrasonic imaging, and treatment. [ABSTRACT FROM AUTHOR]

Organic magnetoresistance (OMAR) can be caused by either single carrier (bipolaron) or double carriers (electron-hole)-based mechanisms. In order to consider applications for OMAR, it is important to control the mechanism present in the device. In this paper, we report the effect of traps on OMAR resulting of disorder at the interface between the organic active layer with the hole injection layer [poly(3,4-ethylenedioxythiophene): poly(styrenesulfonate): PEDOT:PSS]. It has been found that while the single carriers OMAR is enhanced by the presence of traps, the double carriers OMAR is totally removed in a sample with a high interface trap density. The reasons for these results are discussed based on the impedance spectroscopy measurements. First, the mechanism (single or double carriers) responsible of the OMAR was determined with the support of the capacitance measurement. Then, the influence of traps was discussed with the Nyquist diagrams and phase angle-frequency plots of the samples. The results suggested that with a rough interface and thus high disorder, the presence of traps enhanced the bipolaron formation. Traps also acted as recombination centers for electron-hole pairs, which prevented the double carriers OMAR in devices with a rough interface. On the other hand, with a low trap density, i.e., with a smooth surface, the single carrier OMAR decreased, and double carriers OMAR appeared. The sign of the OMAR could then be controlled by simply sweeping the bias voltage. This work demonstrated that the roughness at the interface is important for controlling OMAR and its reproducibility, and that the combination of OMAR measurement and impedance spectroscopy is helpful for clarifying the processes at the interface. [ABSTRACT FROM AUTHOR]

A semi-coprime array (SCA) interleaves two undersampled uniform linear arrays (ULAs) and a Q-element standard ULA. The undersampling factors of the first two arrays are QM and QN, respectively, where M and N are coprime. The resulting non-ULA is highly sparse. Taking the minimum of the absolute values of the conventional beampatterns of the three arrays results in a beampattern that is free of grating lobes. A SCA requires fewer sensors than other popular sparse arrays such as coprime arrays, nested arrays, and minimum redundant arrays for a given aperture. Also, a SCA exhibits better side lobe patterns than other sparse arrays. This means that a SCA is better able to mask signals away from the look direction and detect weak sources in the presence of strong interferers. In this paper, the author explores the direction of arrival estimation with the SCA. The results illustrate the SCA's ability to reduce the number of sensors and decrease system cost and complexity in various signal processing applications. [ABSTRACT FROM AUTHOR]

Suitable data for spatial wave field analyses in concert halls need to satisfy the sampling theorem and hence requires densely spaced measurement positions over extended regions. The described measurement apparatus is capable of automatically sampling the sound field in auditoria over a surface of 5.30 m × 8.00 m to any appointed resolutions. In addition to discussing design features, a case study based on measured impulse responses is presented. The experimental data allow wave field animations demonstrating how sound propagating at grazing incidence over theater seating is scattered from rows of chairs (seat-dip effect). The visualized data of reflections and scattering from an auditorium's boundaries give insights and opportunities for advanced analyses. [ABSTRACT FROM AUTHOR]

Coprime linear microphone arrays allow for narrower beams with fewer sensors. A coprime microphone array consists of two staggered uniform linear subarrays with M and N microphones, where M and N are coprime with each other. By applying spatial filtering to both subarrays and combining their outputs, M + N - 1 microphones yield M N directional bands. In this work, the coprime sampling theory is implemented in the form of a linear microphone array of 16 elements with coprime numbers of 9 and 8. This coprime microphone array is experimentally tested to validate the coprime array theory. Both predicted and measured results are discussed. Experimental results confirm that narrow beampatterns as predicted by the coprime sampling theory can be obtained by the coprime microphone array. [ABSTRACT FROM AUTHOR]

Ground reflections cause problems when estimating the direction of arrival of aircraft noise. In traditional methods, based on the time differences between the microphones of a compact array, they may cause a significant loss of accuracy in the vertical direction. This study evaluates the use of first-order directional microphones, instead of omnidirectional, with the aim of reducing the amplitude of the reflected sound. Such a modification allows the problem to be treated as in free field conditions. Although further tests are needed for a complete evaluation of the method, the experimental results presented here show that under the particular conditions tested the vertical angle error is reduced ~10° for both jet and propeller aircraft by selecting an appropriate directivity pattern. It is also shown that the final level of error depends on the vertical angle of arrival of the sound, and that the estimates of the horizontal angle of arrival are not influenced by the directivity pattern of the microphones nor by the reflective properties of the ground. [ABSTRACT FROM AUTHOR]

The configurations of pixilated detectors for x-ray imaging crystal spectrometers and x-ray pinhole cameras can be optimized based on the Shannon-Nyquist sampling and interpolation theory, since the observed spectra are typically oversampled. The degree of oversampling is quantified and alternative uses for redundant pixels are proposed. [ABSTRACT FROM AUTHOR]