Impulsive interference mitigation capabilities of robust array algorithms are considered in delay estimation of a direct sequence spread spectrum (DS/SS) system. The simulations are done in the environment where the d...
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ISBN:
(纸本)0780385454
Impulsive interference mitigation capabilities of robust array algorithms are considered in delay estimation of a direct sequence spread spectrum (DS/SS) system. The simulations are done in the environment where the direction of arrival (DOA) of individual impulses is either random or fixed. It is shown that the minimum variance distortionless response (MVDR) beamforming algorithm without any interference suppression algorithm is sufficient if impulses are arriving from a fixed DOA. In the random DOA case the performance of the MVDR beamformer algorithm alone is very poor and impulse suppression methods are needed. The paper compares four different impulse suppression methods. It is shown that the best algorithm allow over 60% of snapshots to be corrupted.
We consider the problem of estimating the parameters of linear chirp signals when received at a multi-sensor antenna. Exploiting the negligible variation of the parameters of interest over the coherence time of the ch...
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ISBN:
(纸本)0780385454
We consider the problem of estimating the parameters of linear chirp signals when received at a multi-sensor antenna. Exploiting the negligible variation of the parameters of interest over the coherence time of the channel, we derive a new data model that Outlines the space-time relation between the parameters. In the case of multiple chirps and single-path channels, the derived model allows the estimation of more signals than antennas. The model is directly employed in the case of a single chirp in multipath, where angles of arrival can be retrieved if more antennas than paths are available. We use a MUSIC-based estimator to recover the unknown parameters and compare its performance through simulations against the Cramer-Rao Bound.
In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constr...
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ISBN:
(纸本)0780385454
In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constraint of signal passage at one position (and closing in other positions corresponding to known disturbing, sources). By performing the directional opening towards the desired location in the fixed filter bank structure, the beamformer is left with the task of tracking and suppressing the continuously emerging noise sources. This algorithm has been implemented in MATLAB and tested on real speech recordings conducted in a car hands-free communication situation. Results show that a reduction of the total complexity can be achieved while maintaining the noise suppression performance and reducing the speech distorsion.
This paper presents a technique for real time multi-channel signalprocessing of biological neural data. The major objective is the extraction of action potentials from neural data streams continuously acquired from a...
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ISBN:
(纸本)0780385454
This paper presents a technique for real time multi-channel signalprocessing of biological neural data. The major objective is the extraction of action potentials from neural data streams continuously acquired from a multi-electrode sensorarray. For the actual detection process a wavelet-based method will be proposed, which allows efficient filtering as well as the definition of a robust threshold. In order to match the real time constraints for the wavelet transform an efficient and fast algorithm (lifting scheme) was used. An exemplary 8-channel implementation will be described exploiting the parallel potentials of a Xilinx Virtex field programmable gate array (FPGA). This work is part of a project concerned with the development of a microsensorchip system for neural data analysis at the University of Rostock, Germany.
This paper proposes an iterative compensation method to deal with relative change of sound source location caused by rapid movements of a microphone array and a sound source. This method introduces a delay filter that...
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ISBN:
(纸本)0780385454
This paper proposes an iterative compensation method to deal with relative change of sound source location caused by rapid movements of a microphone array and a sound source. This method introduces a delay filter that has shifted and sampled sinc functions. This paper presents a concept that applies both the error function of the adaptive algorithms to estimate two-dimensional direction-of-arrival and the coordinate system of a moving microphone array. This method directly estimates the relative directions of the microphone array to the sound source directions by minimizing the relative differences of arrival time among the observed signals, not by estimating the time difference of arrival (TDOA) between two observed signals. This method compensates the time delay of the observed signals simultaneously, and it has a feature to ensure that the output signals are in phase. Simulation results support effectiveness of the method.
In many signalprocessing applications of linear algebra tools, the signal part of a postulated model lies in a so-called signal sub-space, while the parameters of interest are in one-to-one correspondence with a cert...
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ISBN:
(纸本)0780385454
In many signalprocessing applications of linear algebra tools, the signal part of a postulated model lies in a so-called signal sub-space, while the parameters of interest are in one-to-one correspondence with a certain basis of this subspace. The signal subspace can often be reliably estimated from measured data, but the particular basis of interest cannot be identified without additional problem-specific structure. This is a manifestation of rotational indeterminacy, i.e., non-Uniqueness of low-rank matrix decomposition. The situation is very different for three- or higher-way arrays, i.e., arrays indexed by three or more independent variables, for which low-rank decomposition is unique under mild conditions. This has fundamental implications for DSP problems which deal with such data. This paper provides a brief tour of the basic elements of this theory, along with many examples of application in problems of current interest in the signalprocessing community.
We consider the problem of target location estimation in the context of large scale, dense sensor networks. We model the probability of detection in each sensor, p(d), as a function of the distance between the sensor ...
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ISBN:
(纸本)0780385454
We consider the problem of target location estimation in the context of large scale, dense sensor networks. We model the probability of detection in each sensor, p(d), as a function of the distance between the sensor and the target. Based on a binary (detection vs. no detection) information from each sensor and the model Of p(d), we propose two different fusion rules for estimating the target location: a maximum likelihood estimate and an empirical risk minimization method. Moreover, we also consider the case where only sensors with a positive detection transmit their reading. This can be helpful to economize the power of sensor units. By employing gaussian like p(d) models, we develop versions of both methods based on simple initialization procedures and a gradient search. We compare and discuss both algorithms in terms of complexity and accuracy.
Low-rank MVDR beamformers such as Conjugate Gradients (CG) and Principal Components Inverse (PCI) can yield a higher SINR than full-rank MVDR beamformer when the sample support is inadequate. However, a large drop in ...
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ISBN:
(纸本)0780385454
Low-rank MVDR beamformers such as Conjugate Gradients (CG) and Principal Components Inverse (PCI) can yield a higher SINR than full-rank MVDR beamformer when the sample support is inadequate. However, a large drop in output SINR occurs if the low-rank beamformer operates at an improper rank. Indirect Dominant Mode Rejection (IDMR) is proposed wherein one first employs a high-resolution spatial spectrum estimation technique to estimate the directions and powers of the dominant interferers. Subsequently, this information is used to construct an estimate of the signal-free (interference plus noise only) autocorrelation matrix (for a given look-direction.) In this process, any residual correlations between the interferers and the signal arriving from the look-direction due to finite sample averaging is effectively removed. Simulations reveal that IDMR yields a dramatic improvement in output SINR relative to CG and PCI/DMR, even when the latter operate at the optimal rank.
Electronic noses and tongues are two recent examples in chemical sensing that employ statistical array techniques in order to overcome the intrinsic limitations of current solid-state chemical sensors like ion-selecti...
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ISBN:
(纸本)0780385454
Electronic noses and tongues are two recent examples in chemical sensing that employ statistical array techniques in order to overcome the intrinsic limitations of current solid-state chemical sensors like ion-selective field transistors (ISFET). In particular, ISFETs are sensitive to the concentration of a particular ion in a solution to be measured, but it can be also strongly affected by several interfering ions found in the solution. Hence, they must be employed in regions in which the effect of interferences is negligible thus limiting their range of operation. However, since ISFETs behave as non-linear mixers of main ion activities and interfering ones, an attempt to separate the original main ion activity and interferences from the mixed response is suitable with blind source separation (BSS) techniques and related methods like independent component analysis (ICA) methods. In this direction, several experiments with real ISFET measurements demonstrate the interest of employing BSS for dealing with the separation in ISFET responses, and further reconstruction, of ion activities in those operating regions in which interferences notably affect their response.
This paper describes sound spot generation by 128-channel surrounded speaker array. A sound spot is a limited area within which an emitted sound is amplified. Sound spot generation is achieved by the Sum and Delay Bea...
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ISBN:
(纸本)0780385454
This paper describes sound spot generation by 128-channel surrounded speaker array. A sound spot is a limited area within which an emitted sound is amplified. Sound spot generation is achieved by the Sum and Delay Beam Former (SDBF). A dedicated PCI 128-channel simultaneous input digital-to-analog (D/A) board is developed for a 128 channel speaker array with a maximum sampling rate of 22.7 us/sample. The 128-channel square speaker array can generate steerable a sound spot towards a specific point. Sound spots can be moved in approximately 10 ms. The speaker array can also generate different sound spots at specific points simultaneously. The performance of sound spot generation is evaluated by using sound pressure distribution maps. The speaker array can be used to that transmit necessary sound information to only a specific person in quiet places, such as a museum and a hospital.
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