Fixed Microphone Array Research Articles

Signal reconstruction of moving sound sources with a fixed microphone array

Abstract The time-domain signals of noise sources are of great importance to acoustic design of machinery. One available way to obtain source signals is to reconstruct them from the acoustic signals measured by a microphone array. However, when sources are moving, the acoustic signals sampled by a fixed microphone array are distorted in amplitude and frequency due to the Doppler effect, leading to that the reconstructed source signals may not reflect the real information of sources. This paper first presents a time-wavenumber-domain method to eliminate the Doppler effect in the measured acoustic signals, in which no priori information on the number, frequencies and locations of sources is needed. Then the acoustic signals after eliminating the Doppler effect are used as the input of the passive time reversal method to reconstruct source signals. Numerical simulation and experimental results show that the proposed method can effectively eliminate the Doppler effect and accurately reconstruct the source signals even when the sources emit transient signals. It is also found that the proposed method can work stably when the sources are moving at high speeds if increasing the array aperture and in the situation of low signal-to-noise ratio.

Doppler Correction Using Short-Time MUSIC and Angle Interpolation Resampling for Wayside Acoustic Defective Bearing Diagnosis

Wayside acoustic defective bearing detector (ADBD) system plays an important role in monitoring the condition of high-speed train bearings and ensuring their normal operation. However, as the system acquires acoustic signals from a passing train by a fixed microphone array, the acquired signals are severely distorted by the Doppler effect, which is an obstacle to defective bearings detection. This paper proposes a novel scheme by combining a short-time multiple signal classification (MUSIC) and angle interpolation resampling (AIR) to remove the Doppler distortion embedded in the microphone array signals. Two basic steps are included in the proposed scheme. First, the proposed short-time MUSIC method is performed on the array signals to calculate out the time-varying receiving angle information. Then, the proposed AIR method is conducted to correct the distorted signal. Compared with the traditional schemes, the proposed scheme is simple and hardly needs prior knowledge, which is expected to be applied in the ADBD system. The proposed scheme is verified by means of simulation and practical studies. Results indicate that it has superior performance to remove the Doppler distortion and has obvious advantages for the ADBD system.

Optimization of Fixed Microphone Array in High Speed Train Noises Identification Based on Far-Field Acoustic Holography

Acoustical holography has been widely applied for noise sources location and sound field measurement. Performance of the microphones array directly determines the sound source recognition method. Therefore, research is very important to the performance of the microphone array, its array of applications, selection, and how to design instructive. In this paper, based on acoustic holography moving sound source identification theory, the optimization method is applied in design of the microphone array, we select the main side lobe ratio and the main lobe area as the optimization objective function and then put the optimization method use in the sound source identification based on holography, and finally we designed this paper to optimize microphone array and compare the original array of equally spaced array with optimization results; by analyzing the optimization results and objectives, we get that the array can be achieved which is optimized not only to reduce the microphone but also to change objective function results, while improving the far-field acoustic holography resolving effect. Validation experiments have showed that the optimization method is suitable for high speed trains sound source identification microphone array optimization.

Sparse tomographic acoustic imaging for environmental noise-source mapping

Environmental noise has become a major problem in large cities, increasing health risks for the urban population. Current methods are overly expensive for city-wide noise-monitoring as they generally require dense deployment of fixed microphones for noise-level measurements. We present here alternative sparse tomographic acoustic imaging techniques using arrays with relatively few number of microphones for large-region acoustic noise mapping. We first demonstrate that the locations and sound pressure levels of fixed noise sources sparsely located in a large field are recovered by collecting acoustic data at multiple locations with a portable microphone array for tomographic reconstruction. We then introduce a nonstationary tomographic imaging approach using fixed microphone arrays, which can also capture the intermittent changes in the acoustic field due to transient and/or moving noise sources. We test both the sparse static and dynamic imaging models with acoustic measurements collected with a circular microphone array mounted on an electric vehicle and two prepositioned circular arrays, respectively. The experimental results show the promise of the proposed methodology for low-cost and time-efficient large-scale noise-monitoring.

A general framework for animal density estimation from acoustic detections across a fixed microphone array

SummaryAcoustic monitoring can be an efficient, cheap, non‐invasive alternative to physical trapping of individuals. Spatially explicit capture–recapture (SECR) methods have been proposed to estimate calling animal abundance and density from data collected by a fixed array of microphones. However, these methods make some assumptions that are unlikely to hold in many situations, and the consequences of violating these are yet to be investigated.We generalize existing acoustic SECR methodology, enabling these methods to be used in a much wider variety of situations. We incorporate time‐of‐arrival (TOA) data collected by the microphone array, increasing the precision of calling animal density estimates. We use our method to estimate calling male density of the Cape Peninsula Moss FrogArthroleptella lightfooti.Our method gives rise to an estimator of calling animal density that has negligible bias, and 95% confidence intervals with appropriate coverage. We show that using TOA information can substantially improve estimate precision.Our analysis of theA. lightfootidata provides the first statistically rigorous estimate of calling male density for an anuran population using a microphone array. This method fills a methodological gap in the monitoring of frog populations and is applicable to acoustic monitoring of other species that call or vocalize.

Broadband nearfield beamformer with compact size by using Nash genetic algorithm

Though the acoustical array is effective in the reverberant field, its usage has been limited at low frequencies since the array with the length of more than 1 meter is difficult to install inside a vehicle. Furthermore, it is impractical to use circular coordinates especially inside vehicle since the sound source, a human mouth, stays at a fixed distance from and moves along the parallel line to the surface of the headliner. For the delay-and-sum beamformer in near-field, it is difficult to get the complex coefficients, amplitude weight and time delay, by analytical methods. In this paper, one numerical optimization method for a fixed microphone array is investigated which uses the Nash Genetic Algorithm (Nash GA) which was originally used in electromagnetism problems. By using Nash GA, first the optimal geometry of array element alignment inside a vehicle is found, then the complex coefficients are obtained. The results measured in an anechoic chamber correspond to that of computer simulation and satisfy the requirement of hands-free mobile telephony where the microphone array is installed in a vehicle’s headliner and the data is also measured in a reverberant field to investigate the signal to noise ratio.

Application of near-field optimum microphone arrays to hands-free mobile telephony

This paper discusses the application of fixed microphone arrays to speech pickup in mobile telephone applications. Array optimization techniques are used to design two broad-band beamformers for speech pickup in the near field. The first beamformer provides optimum gain for spatially incoherent noise while the second beamformer provides optimum gain in spherically isotropic noise. Array performance was measured using vehicular noise recorded under realistic driving conditions. Results obtained are in agreement with theoretical predictions for a spherically isotropic noise field and are comparable to previously reported results obtained using adaptive beamforming algorithms.

Application of near-field optimum microphone arrays to hands-free mobile telephony

This paper discusses the application of fixed microphone arrays to speech pickup in mobile telephone applications. Array optimization techniques are used to design two broad-band beamformers for speech pickup in the near field. The first beamformer provides optimum gain for spatially incoherent noise while the second beamformer provides optimum gain in spherically isotropic noise. Array performance was measured using vehicular noise recorded under realistic driving conditions. Results obtained are in agreement with theoretical predictions for a spherically isotropic noise field and are comparable to previously reported results obtained using adaptive beamforming algorithms.

Simulation of fixed microphone arrays for directional hearing aids.

Microphone arrays with fixed optimum weights are known to suppress the background noise and reverberation that severely reduce the effectiveness of conventional hearing aids. By means of a general technique for digital frequency-domain implementation of optimum broadband arrays that was developed recently [C. Liu and S. Sideman, J. Acoust. Soc. Am. 98, 241-247 (1995)], a practically promising system is proposed to realize the arrays with the well-known sensitivity-constrained superdirective beamforming weights, and with five identical omnidirectional, cardioid, hypercardioid, or dipole microphones, respectively, in the endfire or broadside configurations, which were theoretically proposed by Stadler and Rabinowitz [J. Acoust. Soc. Am. 94, 1332-1342 (1993)]. The digital broadband frequency-domain beamforming system allows the broadband superdirective beamforming weights to be faithfully and independently applied to each frequency component of the signal. The practical application of this technique is demonstrated through computer simulation of the system in anechoic situations. Furthermore, its performance in simulated reverberant environments is evaluated.

Wideband microphone array for hearing aid preprocessing

Speech communication in environments with low signal/noise ratios (SNRs) is a primary complaint of the hearing impaired. Microphone beam formation techniques provide an effective approach to improving SNR in these environments. A novel, fixed microphone array is being developed with user-controlled mainlobe spatial look direction and attenuation band(s), and with a flat frequency response over the speech bandwidth. The array of R microphones and L taps per microphone maximizes energy concentration over a spatial look region and frequency band, subject to spatial and frequency constraints. Constrained maximization of w*Aw/w*Bw is required, where A and B are matrices specifying spatial and frequency factors, and w is the RL dimensional weight vector. The constraining subspace is specified by the array values, derivative values, and spatial directional constraints; w is obtained as the solution of a tractable unconstrained full-rank lower dimensional generalized eigenvalue problem. Numerical and simulation results for different values of R and L and for different bandwidths will be reported, as well as results of preliminary listening tests with normally hearing and hearing impaired individuals. The feasibility of real-time acoustic beamformers with arrays for hearing aids, and the advantages of this scheme over conventional adaptive schemes will also be discussed.

Noise reduction processing arrangement for microphone arrays

A sound processing arrangement couples sound from a prescribed environment through a fixed microphone array to a signal processing arrangement having a specifiable preferred sound source location. The microphone pickup signals are combined with a set of weighting signals to adjust the directional response pattern in successive analysis time intervals. The weighting signals are modified in each analysis time interval so that the total acoustic signal power of the signal processing arrangement output signal is decreased toward a minimum while substantially unity power transfer of sound signals from said preferred location is maintained at all frequencies over a prescribed frequency range. In this way, the preferred source location is in the main beam while unwanted sound source locations are at the null points of the adjusted directional response pattern.