processing of microphone arrays of various configurations involves mathematical models of the surrounding sound fields. These models are based on different space and time conventions found throughout the literature. I...
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processing of microphone arrays of various configurations involves mathematical models of the surrounding sound fields. These models are based on different space and time conventions found throughout the literature. In traditional open array processing, interchanging different space and time conventions can lead to confusion between the arrival and propagation directions. This is relatively easy to observe and adjustments can be made as necessary. However, in the case of spherical arrays, which are a more recent subject of research, the consequences of interchanging between the conventions are not so straightforward. In this paper, an analysis of the misuse of space and time conventions in array processing is presented, with special emphasis on the new results for spherical array configurations. It is shown that, in addition to the reversal of propagation direction, a misuse of notation has the potential of introducing a significant error in the array directivity pattern.
A high resolution iterative algorithm for estimating the direction-of-arrival (DOA) of multiple wideband sources is proposed. For equally spaced array structures, two unitary transform based approaches are proposed in...
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Microphone arrays can improve speech recognition in noise for hearing-impaired listeners by suppressing interference coming from other than the desired signal direction, In a previous paper [J. M. Kates and M. R. Weis...
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Microphone arrays can improve speech recognition in noise for hearing-impaired listeners by suppressing interference coming from other than the desired signal direction, In a previous paper [J. M. Kates and M. R. Weiss, J. Acoust. Sec. Am. 99, 3138-3148 (1996)], several array-processing techniques were evaluated in two rooms using the AI-weighted array gain as the performance metric. The array consisted of five omnidirectional microphones having uniform 2.5-cm spacing, oriented in the endfire direction. In this paper, the speech intelligibility for two of the array processing techniques, delay-and-sum beamforming and superdirective processing, is evaluated for a group of hearing-impaired subjects. Speech intelligibility was measured using the speech reception threshold (SRT) for spondees and speech intelligibility rating (SIR) for sentence materials. The array performance is compared with that for a single omnidirectional microphone and a single directional microphone having a cardioid response pattern. The SRT and SIR results show that the superdirective array processing was the most effective, followed by the cardioid microphone, the array using delay-and-sum beamforming, and the single omnidirectional microphone. The relative processing ratings do not appear to be strongly affected by the size of the room, and the SRT values determined using isolated spondees are similar to the SIR values produced from continuous discourse. (C) 1997 Acoustical Society of America.
A polynomial approach for maximum-likelihood (ML) estimation of superimposed signals in time-series problems and array processing was recently proposed. This technique was applied successfully to linear uniform arrays...
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A polynomial approach for maximum-likelihood (ML) estimation of superimposed signals in time-series problems and array processing was recently proposed. This technique was applied successfully to linear uniform arrays and to uniformly sampled complex exponential signals. However, uniformly spaced arrays are not optimal for minimum variance estimation of bearing, range or position, and uniform sampling of signals is not always possible in practice. The authors make use of the expectation-maximization algorithm to apply the polynomial approach to sublattice arrays and to missing samples in time-series problems.
The parameter estimation problem from noisy signal measurements plays a key role in several practical applications in the array signal processing area ranging from telecommunications and radar to biomedical and acoust...
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The parameter estimation problem from noisy signal measurements plays a key role in several practical applications in the array signal processing area ranging from telecommunications and radar to biomedical and acoustics. The performance of parameter estimation techniques is sensitive to the signal-tonoise ratio (SNR) and severely degrades in noisy scenarios. Classical denoising using SVD low-rank approximation and its tensor counterpart known as higher order SVD (HOSVD) have been widely applied as a preprocessing step to improve the SNR of the received signal. In this paper, we propose the tensor-based multiple denoising (MuDe) approach that successively applies spatial smoothing, denoising and reconstruction to the noisy data. By taking into account the knowledge of the model order and by exploiting subarrays created by the spatial smoothing, we can successively denoise the data by means of HOSVD-based and SVD-based low-rank approximation for tensor and matrix data, respectively. We show that our proposed approach significantly reduces the noise level, allowing a more accurate estimation of parameters compared to state-of-the-art matrix-based and tensor-based techniques without decreasing the sensor array aperture. (C) 2019 Elsevier Inc. All rights reserved.
We present a subspace-based direction-finding method for coherent signal environments using an antenna array. Our method, which uses fourth-order statistics, is capable of resolving more signals than a comparable seco...
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We present a subspace-based direction-finding method for coherent signal environments using an antenna array. Our method, which uses fourth-order statistics, is capable of resolving more signals than a comparable second-order statistics-based subspace method and is applicable to a larger class of arrays. The maximum number of signals resolvable with our method may exceed the number of sensors in the array, Only a uniform linear subarray is needed;the rest of the array may have arbitrary and unknown response and does not require calibration, On the other hand, the comparable second-order statistics-based method is limited to uniform linear arrays only, No search procedure is needed in our method. Simulation experiments supporting our conclusions are provided.
The information capacity of wireless communication systems may be increased dramatically by employing multiple transmit and receive antennas. The goal of system design is to exploit this capacity in a practical way, A...
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The information capacity of wireless communication systems may be increased dramatically by employing multiple transmit and receive antennas. The goal of system design is to exploit this capacity in a practical way, An effective approach to increasing data rate over wireless channels is to employ space-time coding techniques appropriate to multiple transmit antennas. These space-time codes introduce temporal and spatial correlation into signals transmitted from different antennas, so as to provide diversity at the receiver, and coding gain over an uncoded system. For large number of transmit antennas and at high bandwidth efficiencies, the receiver may become too complex whenever correlation across transmit antennas is introduced. This paper dramatically reduces encoding and decoding complexity by partitioning antennas at the transmitter into small groups, and using individual space-time codes, called the component codes, to transmit information from each group of antennas. At the receiver, an individual space-time code is decoded by a novel linear processing technique that suppresses signals transmitted by other groups of antennas by treating them as interference. A simple receiver structure is derived that provides diversity and coding gain over uncoded systems. This combination of array processing at the receiver and coding techniques for multiple transmit antennas can provide reliable and very high data rate communication over narrowband wireless channels. A refinement of this basic structure gives rise to a multilayered space-time architecture that both generalizes and improves upon the layered space-time architecture proposed by Foschini.
This paper attempts to provide a state-of-the-art of sound source localization in robotics. Noticeably, this context raises original constraints-e.g. embeddability, real time, broadband environments, noise and reverbe...
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This paper attempts to provide a state-of-the-art of sound source localization in robotics. Noticeably, this context raises original constraints-e.g. embeddability, real time, broadband environments, noise and reverberation-which are seldom simultaneously taken into account in acoustics or signal processing. A comprehensive review is proposed of recent robotics achievements, be they binaural or rooted in array processing techniques. The connections are highlighted with the underlying theory as well as with elements of physiology and neurology of human hearing. (C) 2015 Elsevier Ltd. All rights reserved.
This paper presents a novel SParse Iterative Covariance-based Estimation approach, abbreviated as SPICE, to array processing. The proposed approach is obtained by the minimization of a covariance matrix fitting criter...
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This paper presents a novel SParse Iterative Covariance-based Estimation approach, abbreviated as SPICE, to array processing. The proposed approach is obtained by the minimization of a covariance matrix fitting criterion and is particularly useful in many-snapshot cases but can be used even in single-snapshot situations. SPICE has several unique features not shared by other sparse estimation methods: it has a simple and sound statistical foundation, it takes account of the noise in the data in a natural manner, it does not require the user to make any difficult selection of hyperparameters, and yet it has global convergence properties.
The problem of passive geographical localization is commonly solved by independently measuring intermediate parameters such as angles of arrival (AOA), times of arrival (TOA) or frequencies of arrival (FOA) on differe...
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The problem of passive geographical localization is commonly solved by independently measuring intermediate parameters such as angles of arrival (AOA), times of arrival (TOA) or frequencies of arrival (FOA) on different separated multiple-sensors base stations on a first step. These intermediate parameters are then used to estimate the emitter's coordinates on a second step. Such approach is suboptimal. This is why, recently, under line-of-sight conditions, some approaches (sometimes called one-step procedures) exploiting directly all signals collected on all base stations were investigated. Based on a space time observation vector we investigate here a new one-step localization technique called LOST that is spectrum adaptive contrary to the existing one-step approach. Simulations shows that the proposed technique outperforms existing techniques and lies close to the corresponding Cramer-Rao bound.
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