This paper provides array signal processing super-resolution fast *** algorithm bases on the eigenvalue shift of covariance matrix and the power iteration of the shift covariance *** not needs the inverse *** converge...
详细信息
This paper provides array signal processing super-resolution fast *** algorithm bases on the eigenvalue shift of covariance matrix and the power iteration of the shift covariance *** not needs the inverse *** converges quickly and only needs several iteration to converge to *** algorithm architecture is very simple and easily implemented.
The problem of symbol synchronization for semi-blind antenna array signal processing in digital communications is considered. It is relevant to situations where the location of known training symbols within a slot of ...
详细信息
The problem of symbol synchronization for semi-blind antenna array signal processing in digital communications is considered. It is relevant to situations where the location of known training symbols within a slot of received data is not known. Training based and blind techniques have been employed to address this problem but they become ineffective when the length of the training sequence and/or the data received under quasi-stationary conditions is not sufficiently long. A more effective semi-blind solution is proposed in this paper. A synchronization algorithm is developed and compared with the known training-based algorithm by means of computer simulations. The testing and evaluation conditions correspond to those of short burst multiuser spatial division multiple access (SDMA) systems.
This paper discusses the problem of direction-of-arrival (DOA) estimation for Gaussian sources that have arbitrary correlation: from independent to fully correlated. For independent sources, the antenna array design i...
详细信息
This paper discusses the problem of direction-of-arrival (DOA) estimation for Gaussian sources that have arbitrary correlation: from independent to fully correlated. For independent sources, the antenna array design is governed by two competing considerations: maximum aperture, which inclines toward increasing sparsity for a given number of array sensors, and identifiability, which tends to exclude extreme sparsity. For fully correlated sources, these two competing criteria are augmented by a third which allows for the initialization of DOA estimation by the generalized spatial smoothing (GSS) technique. The maximum number of fully correlated sources is in turn an important factor in the GSS algorithm and subsequent array geometry design. We present a geometry optimization technique that permits accurate DOA estimation of arbitrarily correlated sources. (C) 2000 Academic Press.
Traditionally optimum-adaptive beamforming algorithms have been developed assuming fully coherent plane wavefronts, i.e., assuming a data model of point sources. In most applications this assumption is inappropriate, ...
详细信息
Traditionally optimum-adaptive beamforming algorithms have been developed assuming fully coherent plane wavefronts, i.e., assuming a data model of point sources. In most applications this assumption is inappropriate, since the channel model has to account for different kinds of dispersion phenomena due to both the propagation environment and the array itself. Significant examples are sonar and underwater communication systems. Indeed, in such circumstances, the resulting wavefronts can be randomly distorted, usually suffering a loss of spatial coherence. Here, assuming a more realistic stochastic channel model, we analyze the performance of a traditional optimum adaptive beamformer for point sources, when the signal or the interference undergo a spatial coherence degradation. It is shown, with analytical details, that the same coherence loss, for the interference results in larger performance degradation than for the signal, Furthermore, we provide a theoretical comparison among different beamforming algorithms, based on the estimate of the channel parameters and on spatial smoothing methods.
Quaternion-valued wireless communication systems have been studied in the past. Although progress has been made in this promising area, a crucial missing link is lack of effective and efficient quaternion-valued signa...
详细信息
Quaternion-valued wireless communication systems have been studied in the past. Although progress has been made in this promising area, a crucial missing link is lack of effective and efficient quaternion-valued signalprocessing algorithms for channel equalisation and beamforming. With most recent developments in quaternion-valued signalprocessing, in this work, we fill the gap to solve the problem and further derive the quaternion-valued Wiener solution for block-based calculation.
Acoustic signals which propagate through the ocean have wavefronts which can differ significantly from the "planar wavefronts" assumed in array signal processing. In this paper we investigate the performance...
详细信息
Acoustic signals which propagate through the ocean have wavefronts which can differ significantly from the "planar wavefronts" assumed in array signal processing. In this paper we investigate the performance of the the minimum variance distortionless response (MVDR) beamformer and the previously introduced Fourier integral method (FIM), when applied to real sonar data from a sparse linear array. It is shown that as long as the array is calibrated, MVDR will perform very well, but when sensor gain/phase errors are present the performance of the MVDR beamformer is degraded. FIM however seems to always perform very well, and a modified version of FIM is shown to be at worst comparable to a well calibrated MVDR beamformer.
This paper presents a combined microphone array and model adaptation algorithm for distant speech recognition. We aim at resolving the inconvenience of using a head-mounted/hand-holding microphone in a conventional sp...
详细信息
ISBN:
(纸本)0780367200
This paper presents a combined microphone array and model adaptation algorithm for distant speech recognition. We aim at resolving the inconvenience of using a head-mounted/hand-holding microphone in a conventional speech recognizer. To improve the distant speech quality, a linear microphone array is applied and acts as a robust acquisition system. We develop a time-domain coherence measure (TDCM) to precisely detect the time delay of speech signals collected by different microphones. The estimated delay is adopted in a delay-and-sum beamformer for speech enhancement. Further, we adapt the speech hidden Markov models to get close to the acoustic condition of enhanced test speech for robust speech recognition. In acquisition and recognition experiments on connected Chinese digits, we find that TDCM can estimate the time delay as precisely as that calculated assuming the speech source direction is known. Increase of speech sampling rate is helpful to determine time delay. Also, the incorporation of the model adaptation scheme can significantly reduce the recognition errors with moderate computation overhead.
Subspace-based algorithms for array signal processing typically begin with an eigenvalue decomposition of a sample covariance matrix. The eigenvectors are partitioned into two sets to get bases for signal and noise su...
详细信息
ISBN:
(数字)9781728143002
ISBN:
(纸本)9781728143019
Subspace-based algorithms for array signal processing typically begin with an eigenvalue decomposition of a sample covariance matrix. The eigenvectors are partitioned into two sets to get bases for signal and noise subspaces. However, the eigenvector subspace estimates are not the most accurate estimates obtainable from the data. Accuracy is defined here in terms of an intrinsic Cramer-Rao (CR) bound. A closed-form (non-iterative) algorithm that achieves the CR bound on subspace accuracy is derived, and examples are given for the applications of adaptive beamforming with a line array and DOA estimation with a planar array. The new algorithm requires far fewer snapshots (e.g. 10 to 100 times fewer) than the typical eigenvector approach to achieve a given level of performance.
This paper proposes a novel wideband structure for array signal processing. A new wideband model is formed where the observations are linear functions of the source amplitudes, but nonlinear in the direction of arriva...
详细信息
This paper proposes a novel wideband structure for array signal processing. A new wideband model is formed where the observations are linear functions of the source amplitudes, but nonlinear in the direction of arrival (DOA) parameters. The method lends itself well to a Bayesian approach for jointly estimating the model order and the DOAs through a reversible jump Markov chain Monte Carlo (MCMC) procedure. The source amplitudes are estimated through a maximum a posteriori (MAP) procedure. The DOA estimation performance of the proposed method is compared with the theoretical Cramer-Rao lower bound (CRLB) for this problem. Simulation results demonstrate the effectiveness and robustness of the method.
A new algorithm, DFT beamspace TLS-ESPRIT with structure weighting is presented. The proposed algorithm provides closed-form estimates of the directions-of-arrival (DOAs) of signals impinging on a uniform linear array...
详细信息
A new algorithm, DFT beamspace TLS-ESPRIT with structure weighting is presented. The proposed algorithm provides closed-form estimates of the directions-of-arrival (DOAs) of signals impinging on a uniform linear array (ULA). DFT beamspace TLS-ESPRIT with structure weighting outperforms DFT beamspace TLS-ESPRIT in terms of estimation accuracy, mainly at small and mean values of SNR, and conserves the low computational complexity of the latter algorithm. Improvement of performance is conditioned by exploiting structure (row) weighting method. Distinctions in realization of structure weighting in beamspace as compared to element space are noted. The performance of the proposed algorithm is evaluated by computer simulation, verifying its statistical efficiency.
暂无评论