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  • About
  • The Global ETD Search service is a free service for researchers to find electronic theses and dissertations. This service is provided by the Networked Digital Library of Theses and Dissertations.
    Our metadata is collected from universities around the world. If you manage a university/consortium/country archive and want to be added, details can be found on the NDLTD website.
1

The measurement of underwater acoustic noise radiated by a vessel using the vessel's own towed array

Duncan, Alexander John January 2003 (has links)
The work described in this thesis tested the feasibility of using a towed array of hydrophones to: 1. localise sources of underwater acoustic noise radiated by the towvessel, 2. determine the absolute amplitudes of these sources, and 3. determine the resulting far-field acoustic signature of the tow-vessel. The concept was for the towvessel to carry out a U-turn manoeuvre so as to bring the acoustic section of the array into a location suitable for beamforming along the length of the tow-vessel. All three of the above were shown to be feasible using both simulated and field data, although no independent field measurements were available to fully evaluate the accuracy of the far-field acoustic signature determinations. A computer program was written to simulate the acoustic signals received by moving hydrophones. This program had the ability to model a variety of acoustic sources and to deal with realistic acoustic propagation conditions, including shallow water propagation with significant bottom interactions. The latter was accomplished using both ray and wave methods and it was found that, for simple fluid half-space seabeds, a modified ray method gave results that were virtually identical to those obtained with a full wave method, even at very low frequencies, and with a substantial saving in execution time. A field experiment was carried out during which a tug towing a 60-hydrophone array carried out a series of U-turn manoeuvres. The signals received by the array included noise radiated by the tow-vessel, signals from acoustic tracking beacons mounted on the tow-vessel, and transient signals generated by imploding sources deployed from a second vessel. / Algorithms were developed to obtain snapshots of the vertical plane and horizontal plane shapes of the array from the transient data and to use range data derived from the tracking beacon signals to track the hydrophones in the horizontal plane. The latter was complicated by a high proportion of dropouts and outliers in the range data caused by the directionality of the hydrophones at the high frequencies emitted by the beacons. Despite this, excellent tracking performance was obtained. Matched field inversion was used to determine the vertical plane array shapes at times when no transient signals were available, and to provide information about the geoacoustic properties of the seabed. There was very good agreement between the inversion results and array shapes determined using transient signals. During trial manoeuvres the array was moving rapidly relative to the vessel and changing shape. A number of different array-processing algorithms were developed to provide source localisation and amplitude estimates in this situation: a timedomain beamformer; two frequency-domain, data independent beamformers; an adaptive frequency-domain beamformer; and an array processor based on a regularised least-squares inversion. The relative performance of each of these algorithms was assessed using simulated and field data. Data from three different manoeuvres were processed and in each case a calibrated source was localised to within 1 m of its known position at the source's fundamental frequency of 112 Hz. / Localisation was also successful in most instances at 336 Hz, 560 Hz and 784 Hz, although with somewhat reduced accuracy due to lower signal to noise ratios. Localisation results for vessel noise sources were also consistent with the positions of the corresponding items of machinery. The estimated levels of the calibrated source obtained during the three manoeuvres were all within 4.1 dB of the calibrated value, and varied by only 1.3 dB between manoeuvres. Results at the higher frequencies had larger errors, with a maximum variation of 3.8 dB between serials, and a maximum deviation from the calibrated value of 6.8 dB. An algorithm was also developed to predict the far-field signature of the tow-vessel from the measured data and results were produced. This algorithm performed well with simulated data but no independent measurements were available to compare with the field results.
2

FleXilicon: a New Coarse-grained Reconfigurable Architecture for Multimedia and Wireless Communications

Lee, Jong-Suk Mark 23 March 2010 (has links)
High computing power and flexibility are important design factors for multimedia and wireless communication applications due to the demand for high quality services and frequent evolution of standards. The ASIC (Application Specific Integrated Circuit) approach provides an area efficient, high performance solution, but is inflexible. In contrast, the general purpose processor approach is flexible, but often fails to provide sufficient computing power. Reconfigurable architectures, which have been introduced as a compromise between the two extreme solutions, have been applied successfully for multimedia and wireless communication applications. In this thesis, we investigated a new coarse-grained reconfigurable architecture called FleXilicon which is designed to execute critical loops efficiently, and is embedded in an SOC with a host processor. FleXilicon improves resource utilization and achieves a high degree of loop level parallelism (LLP). The proposed architecture aims to mitigate major shortcomings with existing architectures through adoption of three schemes, (i) wider memory bandwidth, (ii) adoption of a reconfigurable controller, and (iii) flexible wordlength support. Increased memory bandwidth satisfies memory access requirement in LLP execution. New design of reconfigurable controller minimizes overhead in reconfiguration and improves area efficiency and reconfiguration overhead. Flexible word-length support improves LLP by increasing the number of processing elements executable. The simulation results indicate that FleXilicon reduces the number of clock cycles and increases the speed for all five applications simulated. The speedup ratios compared with conventional architectures are as large as two orders of magnitude for some applications. VLSI implementation of FleXilicon in 65 nm CMOS process indicates that the proposed architecture can operate at a high frequency up to 1 GHz with moderate silicon area. / Ph. D.
3

Localization of Subsurface Targets using Optimal Maneuvers of Seismic Sensors

Alam, Mubashir 10 May 2006 (has links)
The use of seismic waves to detect subsurface targets such as landmines is a very promising technology compared to existing methods like Ground Penetrating Radar (GPR) and Electromagnetic Induction (EMI) sensing. The fact that seismic waves induce resonance in man-made targets, and hence more scattering, gives this method a natural ability to discriminate landmines from common types of clutter like rocks, wood, etc. Reflection and resonance from the targets can be used in imaging to detect the location of targets. However, existing methods require a large number of measurements for imaging and detection, which are expensive and time consuming. To reduce the number of measurements and enable faster detections, a new sensing strategy is proposed based on optimally maneuvering sensors. The system would operate in two main modes. In search mode, the goal would be to move on top of a target using the minimum number of measurements. Once the target is found, the system would switch to a detection mode to make its final decision. The seismic sensor system is an active system, where a seismic source generates the probing pulse. The waves reflected from buried targets are collected by an array of sensors placed on the surface, and then an imaging algorithm is used to estimate the target position. The performance bounds for this position estimate are derived in terms of the Fisher information matrix (FIM). This matrix gives the dependence of the target position estimate on the array position. Based on the FIM, the next optimal array position is determined by using the theory of optimal experiments. The next array position will be the one that reduces the uncertainty of the target position estimate the most. The whole array is moved to this new position, where the same steps are repeated. In this way, the target can be localized in a few iterations.
4

The Design of A Matrix Completion Signal Recovery Method for Array Processing

January 2016 (has links)
abstract: For a sensor array, part of its elements may fail to work due to hardware failures. Then the missing data may distort in the beam pattern or decrease the accuracy of direction-of-arrival (DOA) estimation. Therefore, considerable research has been conducted to develop algorithms that can estimate the missing signal information. On the other hand, through those algorithms, array elements can also be selectively turned off while the missed information can be successfully recovered, which will save power consumption and hardware cost. Conventional approaches focusing on array element failures are mainly based on interpolation or sequential learning algorithm. Both of them rely heavily on some prior knowledge such as the information of the failures or a training dataset without missing data. In addition, since most of the existing approaches are developed for DOA estimation, their recovery target is usually the co-variance matrix but not the signal matrix. In this thesis, a new signal recovery method based on matrix completion (MC) theory is introduced. It aims to directly refill the absent entries in the signal matrix without any prior knowledge. We proposed a novel overlapping reshaping method to satisfy the applying conditions of MC algorithms. Compared to other existing MC based approaches, our proposed method can provide us higher probability of successful recovery. The thesis describes the principle of the algorithms and analyzes the performance of this method. A few application examples with simulation results are also provided. / Dissertation/Thesis / Masters Thesis Electrical Engineering 2016
5

Improving Signal Clarity through Interference Suppression and Emergent Signal Detection

Hoppe, Elizabeth A. 28 September 2009 (has links)
Microphone arrays have seen wide usage in a variety of fields; especially in sonar, acoustic source monitoring and localization, telecommunications, and diagnostic medicine. The goal of most of these applications is to detect or extract a signal of interest. This task is complicated by the presence of interferers and noise, which corrupts the recorded array signals. This dissertation explores two new techniques that increase signal clarity: interferer suppression and emergent signal detection. Spatial processing is often used to suppress interferers that are spatially distinct from the signal of interest. If the signal of interest and the interferer are statistically independent, blind source separation can be used to statistically extract the signal of interest. The first new method to improve signal clarity presented in this work combines spatial processing with blind source separation to suppress interferers. This technique allows for the separation of independent sources that are not necessarily simultaneously mixed or spatially distinct. Simulations and experiments are used to show the capability of the new algorithm for a variety of conditions. The major contributions in this dissertation under this topic are to use independent component analysis to extract the signal of interest from a set of array signals, and to improve existing independent component analysis algorithms to allow for time delayed mixing. This dissertation presents a novel method of improving signal clarity through emergent signal detection. By determining which time frames contain the signal of interest, frames that contain only interferers and noise can be eliminated. When a new signal of interest emerges in a measurement of a mixed set of sources, the principal component subspace is altered. By examining the change in the subspace, the emergent signal can be robustly detected. This technique is highly effective for signals that have a near constant sample variance, but is successful at detecting a wide variety of signals, including voice signals. To improve performance, the algorithm uses a feed-forward processing technique. This is helpful for the VAD application because voice does not have a constant sample variance. Experiments and simulations are used to demonstrate the performance of the new technique. / Ph. D.
6

Analytical Framework for the Performance Analysis of Multiple Antenna Systems

Bae, Kyung Kyoon 04 November 2005 (has links)
There has been great interest in antenna array processing (diversity, beamforming, null steering, and spatial multiplexing) to enhance the received signal quality and the capacity of wireless communications systems. However, in order to properly exploit the characteristics of different array processing techniques, understanding trade-offs among different techniques and parametric investigation, which offers an insight as to what parameters determine system performance under different situations is necessary. In this study, we present analytical framework which can facilitate the performance analysis of systems with antenna array. Five original contributions to the performance analysis of antenna array processing are presented in this study. First, we present theoretical outage probability of a system equipped with an array which suppresses a few dominant interering signals in TDMA cellular networks when the fading statistics of interfering signals are independent but non-identically distributed. Most of the related previous works assumed either independent and identically distributed fading statistics among cochannel interferences (CCI) or Rayleigh fading when CCI signals are subject to i.n.d. fading statistics. Secondly, the performance of multi-branch predetection equal gain combiner for different modulation techniques in equally correlated Nakagami-m fading is presented through analytical analysis. Specifically, the characteristic function (CHF) and the moment generating function (MGF) of EGC output with correlated inputs are derived and used to evaluate the average symbol error probability (ASEP) and the outage probability performance, respectively. Thirdly, we derived analytical expression which can be used to analyze the performance of different types of diversity techniques in equally correlated Nakagami-m or Rice fading channels. Fourthly, asymptotic analysis on different types of diversity combiners in generalized fading channels is presented in a unifying way. Finally, we investigate and present the impact of transmit diversity at handsets on the reverse link DS/CDMA systems in terms of capacity and coverage over generalized fading channels through analytical approaches. Then, we validate the analytical results with simulation results and investigate practical issues which are hard to capture through analytical analysis using system level simulator we developed. Although we have mainly focused on applying the analytical framework we have derived in this work to the performance analysis of physical layer algorithms such as spatial diversity and adaptive null steering, the framework can be extended to assist the analysis and design of wireless communication systems such as, to name a few, distributed multiple input multiple output (MIMO) system in cooperative wireless networks, multipath routing protocol analysis in wireless fading channels, and antenna selection problems in MIMO system. / Ph. D.
7

Multi-dimensional Signal Processing And Circuits For Advanced Electronically Scanned Antenna Arrays

Abewardana Wijenayake, Chamith K. January 2014 (has links)
No description available.
8

DIGITAL DIRECTION FINDING SYSTEM DESIGN AND ANALYSIS

LIU, HUAZHOU 02 September 2003 (has links)
No description available.
9

Traitement d’antenne adaptatif pour l’imagerie ultrasonore passive de la cavitation / Adaptive array processing for passive ultrasound imaging of cavitation

Polichetti, Maxime 01 October 2019 (has links)
Ce travail s'intéresse au suivi spatio-temporel par imagerie ultrasonore de la cavitation acoustique. Celle-ci est un phénomène physique complexe utilisé au cours de certaines techniques de thérapie par ultrasons, correspondant à la formation de bulles de gaz qui oscillent et éclatent. Initialement, la méthode TD-PAM (Time Domain Passive Acoustic Mapping, en anglais), a été développée pour cartographier l’activité de cavitation à partir des signaux acoustiques émis par les bulles, enregistrés passivement par une sonde linéaire d'imagerie ultrasonore. Toutefois, le TD-PAM souffre d’une trop faible résolution et de nombreux artefacts de reconstruction. De plus, il est lourd en temps de calcul car il est formalisé dans le domaine temporel (TD). Pour pallier ces deux limitations, il est proposé d'étudier, de comparer et de développer des méthodes avancées d'imagerie ultrasonore passive. Ce manuscrit s'articule autour de trois contributions principales : Une méthode adaptative originale a été formalisée dans le domaine temporel, reposant sur la compression d'amplitude des signaux ultrasonores par racine pième : le TD-pPAM. Cette approche améliore la résolution et le contraste des cartes de cavitation pour un temps de calcul équivalent au TD-PAM. La notion de matrice de densité inter-spectrale a été introduite pour l'imagerie de la cavitation. Dès lors, quatre méthodes dans le domaine de Fourier (FD) ont été étudiées et comparées : le FD-PAM (non-adaptatif), la méthode Robuste de Capon FD-RCB (adaptatif, par optimisation), le Functional Beamforming FD-FB (adaptatif, par compression non-linéaire) et la méthode MUltiple Signal Classification FD-MUSIC (adaptatif, par projection en sous-espaces). Les performances de ces méthodes FD ont été étudiées expérimentalement in vitro cuve d’eau avec une comparaison par imagerie optique. Les méthodes adaptatives FD proposées ont démontré leur potentiel à améliorer le suivi spatio-temporel des bulles. Le FD-RCB offre une localisation supérieure au FD-PAM mais souffre d'une importante complexité algorithmique. Les performances du FD-FB sont intermédiaires à celles du FD-PAM et du FD-RCB, pour une complexité de calcul équivalente au FD-PAM. Le FD-MUSIC a le potentiel de mettre en évidence de faibles sources acoustiques, mais ne conserve pas leurs quantifications relatives / This work focuses on the spatio-temporal monitoring of acoustic cavitation by ultrasonic imaging. This is a complex physical phenomenon used in some ultrasound therapy techniques, corresponding to the formation of gas bubbles that oscillate and implode. Initially, the TD-PAM (Time Domain Passive Acoustic Mapping) method was developed to map cavitation activity from acoustic signals emitted by bubbles, passively recorded by a linear ultrasonic imaging probe. However, the TD-PAM suffers from too low resolution and many reconstruction artifacts. In addition, it is time-consuming because it is formalized in the time domain (TD). To overcome these two limitations, it is proposed to study, compare and develop advanced methods of passive ultrasound imaging. This manuscript is structured around three main contributions: An original adaptive method has been formalised in the time domain, based on the amplitude compression of ultrasonic signals by root pth: TD-pPAM. This approach improves the resolution and contrast of cavitation maps for a computing time equivalent to the TD-PAM. The notion of cross-spectral density matrix has been introduced for cavitation imaging. Four Fourier domain (FD) methods were therefore studied and compared: FD-PAM (non-adaptive), Capon Robuste FD-RCB (adaptive, by optimization), Functional Beamforming FD-FB (adaptive, by non-linear compression) and MUltiple Signal Classification FD-MUSIC (adaptive, by subspaces projection). The performance of these FD methods was studied experimentally in vitro in water tank with a comparison by optical imaging. The proposed adaptive FD methods have demonstrated their potential to improve the spatial and temporal tracking of bubbles. The FD-RCB offers a superior localization to the FD-PAM but suffers from a high algorithmic complexity. The performance of the FD-FB is intermediate to that of the FD-PAM and the FD-RCB, for a calculation complexity equivalent to the FD-PAM. The FD-MUSIC has the potential to highlight weak acoustic sources, but does not keep their relative quantifications
10

Arrays de microfones para medida de campos acústicos. / Microphone arrays for acoustic field measurements.

Ribeiro, Flávio Protásio 23 January 2012 (has links)
Imageamento acústico é um problema computacionalmente caro e mal-condicionado, que envolve estimar distribuições de fontes com grandes arranjos de microfones. O método clássico para imageamento acústico utiliza beamforming, e produz a distribuição de fontes de interesse convoluída com a função de espalhamento do arranjo. Esta convolução borra a imagem ideal, significativamente diminuindo sua resolução. Convoluções podem ser evitadas com técnicas de ajuste de covariância, que produzem estimativas de alta resolução. Porém, estas têm sido evitadas devido ao seu alto custo computacional. Nesta tese, admitimos um arranjo bidimensional com geometria separável, e desenvolvemos transformadas rápidas para acelerar imagens acústicas em várias ordens de grandeza. Estas transformadas são genéricas, e podem ser aplicadas para acelerar beamforming, algoritmos de deconvolução e métodos de mínimos quadrados regularizados. Assim, obtemos imagens de alta resolução com algoritmos estado-da-arte, mantendo baixo custo computacional. Mostramos que arranjos separáveis produzem estimativas competitivas com as de geometrias espirais logaritmicas, mas com enormes vantagens computacionais. Finalmente, mostramos como estender este método para incorporar calibração, um modelo para propagação em campo próximo e superfícies focais arbitrárias, abrindo novas possibilidades para imagens acústicas. / Acoustic imaging is a computationally intensive and ill-conditioned inverse problem, which involves estimating high resolution source distributions with large microphone arrays. The classical method for acoustic imaging consists of beamforming, and produces the source distribution of interest convolved with the array point spread function. This convolution smears the image of interest, significantly reducing its effective resolution. Convolutions can be avoided with covariance fitting methods, which have been known to produce robust high-resolution estimates. However, these have been avoided due to prohibitive computational costs. In this thesis, we assume a 2D separable array geometry, and develop fast transforms to accelerate acoustic imaging by several orders of magnitude with respect to previous methods. These transforms are very generic, and can be applied to accelerate beamforming, deconvolution algorithms and regularized least-squares solvers. Thus, one can obtain high-resolution images with state-of-the-art algorithms, while maintaining low computational cost. We show that separable arrays deliver accuracy competitive with multi-arm spiral geometries, while producing huge computational benefits. Finally, we show how to extend this approach with array calibration, a near-field propagation model and arbitrary focal surfaces, opening new and exciting possibilities for acoustic imaging.

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