Air Ultrasonic Signal Localization with a Beamforming Microphone Array

This paper addresses the issue of estimating underwater noise mapping of a moving ship using a static linear hydrophone array. In the context of pass-by experiments, moving-source mapping is classically performed using an extension of conventional beamforming to moving sources: beamforming-MS. The main issue of moving-source mapping using beamforming-MS is the poor spatial resolution of two close sources in a noise map at low frequencies. To improve the spatial resolution of beamforming-MS, a passive synthetic array is built in the time domain as a pre-processing of the beamforming-MS, using the a-priori knowledge of the trajectory of the mapped vehicle. The short time synchronization strategy of the extended towed-array processing method ( [1]) is proposed in order to be robust to nonstationary propagation conditions. The efficiency of this strategy is validated using an aerial experiment with two loudspeakers in a semi-anechoic chamber, and an underwater experiment with a towed-ship model in a mountain lake. Two criteria are suggested to objectively and experimentally assessed the quality of the construction of the passive synthetic array.

Compressed air energy is expensive, but common in industrial manufacturing plant. However, a significant part of the generated compressed air energy is lost due to leakage. Best practice requires ongoing leak detection and repair. Leak detection in the ultrasonic frequency range using handheld devices is possible only over short distances as associated high-frequency sound is rapidly attenuated by atmospheric absorption. Pressurized air escaping to ambience also generates frequencies below 20 kHz. In this paper beamforming—a well known method for generating noise maps—is tested as a tool for localization of compressed air leaks at larger distances in the audible frequency range. Advanced beamforming methods in both time domain (broadband) and frequency domain (narrowband) have been implemented in a variety of situations on a laboratory experimental rig with several open blows representing leakage in a noisy environment similar to a factory setting. Based on the results achieved it is concluded that the microphone array approach has the potential to be a robust leak identification tool. The experience gained here can also provide useful guidance to the practitioner.

Conventional phased arrays require a large number of sources in forming a complex wave front, resulting in complexity and a high cost to operate the individual sources. We present a passive phased array using an acoustic metascreen that transmits sound energy from a single source and steers the transmitted wave front to form the desired fields. The metascreen is composed of elements that have a discrete resolution along the screen at an order smaller than the wavelength, allowing for fine wave-front shaping beyond the paraxial approximation. The performance is verified in experiment by forming a self-bending beam. Our metascreen-based passive array with its simplicity and capability has applications in places where conventional active arrays are complex and have limitations.

A passive synthetic array (PSA) using a single moving sensor is formed to estimate direction of arrival (DOA). And co-prime array is used to form a virtual L-array. The impact of the passive synthetic aperture on the DOA estimation under different conditions is simulated and analyzed. Numerical simulations show that the initial time error has no obvious effect on the DOA estimation in the condition of Tr≤Tc. Nevertheless, the DOA estimation becomes more inaccurate with the initial time error increasing, in the condition of Tr≤Tc.

In this paper, synthetic aperture techniques are proposed to improve the accuracy of direction-of-arrival (DOA) estimator in passive synthetic array (PASA). First, we establish signal model for PASA, and then we derive the Cramer-Rao Bound (CRB) of DOA estimator under different statistical assumptions for the incident signals. In addition, several factors that would affect the DOA estimation accuracy are discussed. Results state that the size of synthetic aperture depends on the platform velocity and the sample rate. Therefore, besides signal-to-noise ratio (SNR) and the number of independent snapshots, the platform velocity and sample rate have important impact on the angle estimation accuracy. Finally, the theoretical analysis is verified by numerical simulations.

This article develops the concept of the passive synthetic array, including its design and implementation, its response to signal and noise, and its overall performance. The only case considered is the ideal case of a constant velocity receiver, a constant bearing narrow-band source, stationary signal and noise statistics, and single-path propagation in the plane of the source and receiving synthetic array. It is shown that, in this ideal case, the passive synthetic array is identical to the narrow-band spectrum analyzer in both implementation and performance, with the total gain determined to a close approximation by the bandwidth of the signal. Considering the processor to be a synthetic array beamformer, however, yields additional information about a detected narrow-band source because the bearing of the source can be determined if a course or speed change is made by the sensor.

A passive synthetic array (PSA) using a single moving sensor is formed to estimate direction of arrival (DOA). And the signal model with velocity errors is constructed. By using MUSIC algorithm, the angle estimation performance is analyzed through numerical simulations under different velocity errors. With the increase of velocity error, resolution and accuracy of multi-target angle estimation decline gradually. When errors get 10 m/s, angle estimation can't be distinguished clearly. According to the simulation result, the velocity error should not be more than 3m/s, otherwise it will cause great negative effects on the performance of multi-target angle estimation.

A novel headphone system with microphone arrays was proposed whose objective is to boost forward sound signal, by doing so people can communicate well with a speaker during headphone playback. The main purpose of this research is to boost the sound signal from the front, particularly for a speech signal (300Hz~3kHz). A delay-and-sum beamforming method is applied to an end-fire microphone array to make a proposed headphone system. Three design parameters, i.e. an aperture size, the number of microphones, and a microphone arrangement of microphone arrays are decided based on performance of beamformer with respect to each parameter. To compare the performance regarding to each design parameter, a performance measure, efficiency, was defined as ratio of two resultant sound pressures between a forward and a backward direction. A sound scattering from a head of listener is modeled by using spherical head-related transfer function (HRTF). By changing an aperture size of a microphone array, 8 cm microphone array is thought to be proper when the efficiency in speech frequency range was considered. The use of four microphones is expected as the optimum by considering both good efficiency and reducing the number of microphones. By fixing the number of microphones and an aperture size of microphone array, the efficiency was calculated in various arrangements of microphones. As a result, microphone arrangement with inter-microphone spacing of decreasing interval is selected. The design parameters such as an aperture size, the number, and an arrangement of microphones can be changed regarding to a target frequency range or a direction wanted to be boosted, and the proposed methodology can be applied similarly.

The present paper introduces a microphone array design and describes the results of an evaluation of the developed microphone array. We propose an evaluation index of the directional characteristics of beamforming to optimize microphone array design. Using beamforming simulation, we obtain a microphone arrangement that minimizes sidelobes and improves the basic performance of beamforming. This microphone arrangement has 64 microphones arranged in a 350-mm-diameter sphere designed to be mounted on a mobile robot and omni-directional directivity in azimuth and elevation. The performance of the proposed microphone array is verified in different real environments. The experimental results of sound localization demonstrate the effectiveness of the array in challenging environments and the robustness of the proposed microphone array for different pressure sound sources to cover larger areas.

This paper gives a brief review, based on the concept of the inverse radiation problem, on the formulation of a synthetic virtual receiving array formed by scanning a single sensor following a designed locus in a stationary acoustic field. Both the beamforming method and the signal decomposition analysis are used to examine the parameters needed for determining the radiation sources’ locations. Studies show that considerable angular resolution can be achieved for observing distant sound sources through adaptive signal processing techniques. Applications in ocean acoustics for directional acoustic field measurement are discussed for two special types of synthetic arrays, linear and circular. Problems involved with the directionality ambiguity due to Doppler frequency shift, angular resolution degradation originated from array deformation, effects caused by instability of ocean conditions, and deployment feasibility are also addressed. The concluding results may be used to establish the guidelines for engineering design and deployment of passive synthetic arrays for ocean acoustics applications.

In the applications of microphone array, large steering direction error is often unavoidable because of the motion of target speaker. Although many robust beamformers possess robustness against arbitrary array steering vector (ASV) error within a presumed uncertainty set, facing a large steering direction error, their performance degrades significantly. Meanwhile, microphone array always request a controlled frequency response to target signal. In this paper, we present a new adaptive microphone array implemented in frequency domain with controlled mainlobe and frequency response. With the robust constraints on magnitude response, the proposed microphone array not only produces large controlled robust response region and robust frequency response, but also achieves high performance in SINR enhancement.

A new method of near-field radiation source localization, which is based on passive synthetic array (PSA) using single channel receiver, is proposed in this paper. Based on the concept of PSA in combination with the spatial spectrum estimation, the paper uses a revised multiple signal classification (MUSIC) algorithm to estimate the distance and angle of interfering sources, and also proposes a method of Hilbert phase extraction based on the EMI receiver to locate near-field radiation sources. The derived array model is structurally simpler than the conventional array model, and solves the problem of multi-channel amplitude-phase inconsistency. It also avoids the problem that the model size is too big to use in the near-field situation. The feasibility of this method is demonstrated with simulation and experiment.

The purpose of this letter is to formally comment on the accuracy of the ‘‘Background and Introduction’’ and the ‘‘Conclusions’’ sections of the article ‘‘Passive synthetic arrays’’ [S. W. Autrey, J. Acoust. Soc. Am. 84, 592–598 (1988)].

This is a reply to the Letter to the Editor by Carey [J. Acoust. Soc. Am. 85, 1372 (1989)] that takes issue with ‘‘Passive synthetic arrays’’ [S. W. Autrey, J. Acoust. Soc. Am. 84, 592–598 (1988)]. The assertions in that letter are shown to be incorrect.

Acoustic cameras visualize the origin and intensity of sound waves using an array of microphones and sophisticated signal-processing algorithms. Due to the high memory-bandwidth and signal-processing complexity required by such algorithms, current available devices either compute the corresponding intensity-images off-line or require large and expensive hardware equipment. In this paper, we describe a low-complexity field-programmable gate array (FPGA)-based prototype, which computes and visualizes acoustic intensity-images in real-time. The system consists of 32 microphones and performs all signal processing tasks on a low-cost Xilinx Spartan 3E FPGA. The prototype computes intensity-images with a resolution of 320×240 pixels at 10 frames-per-second.