Sound propagation can be greatly affected by seabed, especially in shallow water, therefore by understanding the geoacoustic parameters of sea bottom can help to improve the performance of sonar systems. In this study, ship noise collected by the vertical line array (VLA) in South China Sea experiment of the Asian Seas International Acoustics Experiment (ASIAEX SCS) in 2001 was used as a sound source to invert the geoacoustic parameters. The nearest horizontal distance between VLA and the passing ship was estimated by beamforming the receiving sounds on the array, and this distance was used in the sound propagation modal. In the modal, two layers structure were assumed for the bottom, so the sound speed (C1) and density (£l1) of sediment layer, sound speed (C2 ) and density (£l2) of subbottom layer, and total absorption coefficient (£\) need to be inverted. Matched field processing is used to solve this inverse problem, and computing the minimum cost function between the measured and modeled sound field, the best matched bottom parameters are C1¡×1600 m/s¡BC2¡×1650 m/s¡B£l1=1.6 g/cm3¡B£l2=2.1 g/cm3¡B£\=0.6 dB/£f. These results were compared with chirp sonar survey in this area, and the agreement is satisfactory.

Abstract Because of the rapid development of mobile communications, the bandwidth efficiency and the communication quality have become interesting issues for researchers. Multipath fading and multiple-access interference (MAI) are the main barriers of performance improvement. Fortunately, spread spectrum techniques have an inherent ability of anti-interference and anti-multipath. Thus, the commercialized CDMA systems are developed on the basis of spread spectrum techniques. Recently, smart antennas using adaptive array techniques have been applied in CDMA communications. There are two different approaches to realize smart antenna systems: one is named as ¡¥switching-beam arrays¡¦; the other is called ¡¥tracking-beam arrays¡¦. Switching-beam arrays use a number of fixed beams at an antenna site. The mobiles select the beam that offers the best signal enhancement and interference reduction. On the other hand, tracking-beam arrays can adjust their pattern to track desired signals, reduce interference, and collect correlated multipath power. In this thesis, several smart antenna systems based on tracking-beam arrays in DS-CDMA systems are developed. The proposed smart antennas comprise two main structures, i.e., the direction-of-arrival (DOA) tracking and the dynamic beamforming. The improved recursively updated method and the Kalman filter algorithm are employed to track the mobiles¡¦ DOAs. By referring to the estimated DOA, the minimum- variance-distortionless-response (MVDR) dynamic beamformer produces a main beam to target on a certain mobile and simultaneously, interferences from other mobiles are suppressed. Therefore, the MAI in DS-CDMA systems can be suppressed properly at all times. Besides, the inherent processing gain in DS-CDMA systems is utilized to break through the limit that the number of mobiles must be smaller than that of sensors. The proposed smart antennas work properly even the number of mobiles exceeds that of sensor. Simulation results show that the advantages of the proposed technique over the conventional tracking-beam arrays.

Ocean ambient noise is one of factors that can affect the performance of sonar and underwater communication system, it can degrade the performance of sonar system on listening or active detection, and also can affect the quality of underwater communication. Due to the variation of temperature and density in the ocean which make ambient noise has directionality. Beamforming can analyze the directionality of noise energy. Conventional beamforming is based on the assumption of plane wave sound field, so the energy from each angle is obtained by linear accumulation of every element. However plane wave assumption may not be satisfied because of the boundary interactions of sound propagation and energy attenuation of water column, therefore conventional beamforming may have poor beam resolution and SNR in applications. This research is to study of the influence of spatial coherence of ambient noise on beam resolution, and to improve the beam resolution by using the adaptive algorithm from the communication system theory. Firstly, simulations were performed to study spatial coherence between plane wave and non-plane wave in ambient noise, and the results were compared with beam resolution. This research also analyzes the influence of different conditions of noise spatial coherence on beamforming with ASIAEX data. The results showed that ambient noise has lower spatial coherence at high frequency, and the beamforming has poor beam resolution because of the lower spatial coherence in noise. Therefore, the adaptive beamforming were performed to improve the beam resolution, and compared with the conventional beamforming. The results showed that the highest improvement on beam resolution is 42.9 %, and increased SNR by 6 dB. On the other hand, the application of ASIAEX data show that, the highest improvement on beam resolution is 40.0 %, and increased SNR by 8 dB. The noise notch of ambient noise became more significant by increasing in beam resolution, and it also promoted the accuracy of analysis on noise directionality.

In single-input multiple-output (SIMO) systems based on orthogonal frequency division multiplexing (OFDM), adaptive beamforming at the receiver side can be used to combat the effect of directional co-channel interference (CCI). Since pilot-aided beamforming suffers from consuming precious channel bandwidth, there has been much interest in blind beamforming approaches that can adapt their weights by restoring certain properties of the transmitted signals. Within this class of blind algorithms, the recursive least squares constant modulus algorithm (RLS-CMA) is of particular interest due to its good overall CCI cancelation performance and fast convergence. Nevertheless, the direct use of RSL-CMA within a SIMO-OFDM receiver induces considerable computational complexity, since a distinct copy of the RLS-CMA must be run on each individual sub-carriers. In this thesis, we present two approaches to reduce the computational complexity of SIMO-OFDM beamforming based on the RLS-CMA, namely: frequency interpolation and distributed processing. The former approach, which exploits the coherence bandwidth of the broadband wireless channels, divides the sub-carriers into several contiguous groups and applies the RLS-CMA to a selected sub-carrier in each group. The weight vectors at other frequencies are then obtained by interpolation. The distributed processing approach relies on the partitioning of the receiving array into sub-arrays and the use of a special approximation in the RLS-CMA. This allows a partial decoupling of the algorithm which can then be run on multiple processors with reduced overall complexity. This approach is well-suited to collaborative beamforming i~ multi-node distributed relaying. Through numerical simulation experiments of a SIMO-OFDM system, it is demonstrated that the proposed modifications to the RLS-CMA scheme can lead to substantial computational savings with minimal losses in adaptive cancelation performance.

Beamforming is a signal processing technique used in sensor arrays to direct signal transmission or reception. Beamformer combines input signals in the array to achieve constructive interference at particular angles (beams) and destructive interference for other angles. According to the following facts: 1- Beamforming can be computationally intensive, so real-time sonar beamforming algorithms in sonar devices is important. 2- Parallel computing has become a critical component of computing technology of the 1990s, and it is likely to have as much impact over the next 20 years as microprocessors have had over the past 20 [5]. 3- The high-performance computing community has been developing parallel programs for decades. These programs run on large scale, expensive computers. Only a few elite applications can justify the use of these expensive computers [2]. 4- GPU computing has the ability of parallel computing and it could be available on the personal computers. The objective of this thesis is to use Graphics Processing Unit (GPU) as real-time digital beamformer to accelerate the intensive signal processing.

In this thesis, we consider a wireless network consisting of d source-destination pairs and R relaying nodes. Each source wishes to communicate to its corresponding destination. By exploiting the spatial multiplexing capability of the wireless medium, we develop two cooperative beamforming schemes in order to establish wireless connections between multiple source-destination pairs through a collaborative relay network. Our first communication scheme consists of two steps. In the first step, all sources transmit their signals simultaneously to the relay network. As a result, each relay receives a noisy faded mixture of all source signals. In the second step, each relay transmits an amplitude- and phase-adjusted version of its received signal, i.e., the relay received signals are multiplied by a set of complex coefficients and are retransmitted. Our goal is to obtain these complex coefficients (beamforming weights) through minimization of the total relay transmit power while the signal-to-interference-plus-noise ratio at the destinations are guaranteed to be above certain pre-defined thresholds. Our second scheme is a distributed downlink beamforming technique which is performed in d + 1 successive time slots. In the first d time slots, the d sources transmit their data to the relay network successively. The relay nodes receive and store the noisy faded versions of the source signals. In the (d + 1)th time slot, the relays aim to collectively provide downlink connections to all d destinations. To do so, each relay transmits a linear combination of the stored signals received during the first d time slots. Again, our goal is to determine the complex weights (used at the relaying nodes to linearly combine the source signals) by minimizing the total relay transmit power while satisfying certain quality of services at the destinations. We use semi-definite relaxation to turn both problems into semi-definite programming (SDP) problems. Therefore, they can be efficiently solved using interior point methods. We showed that our proposed schemes significantly outperform orthogonal multiplexing schemes, such as time-division multiple access schemes, in a large range of network data rates. / Thesis (Master, Electrical & Computer Engineering) -- Queen's University, 2008-09-17 13:07:21.505

A new method, Computational Acoustic Beamforming, is proposed in this thesis. This novel numerical sound source localization methodology combines the advantages of the Computational Fluid Dynamics (CFD) simulation and experimental acoustic beamforming, which enable this method to take directivity of sound source emission into account while maintaining a relatively low cost. This method can also aid the optimization of beamforming algorithm and microphone array design. In addition, it makes sound source prediction of large structures in the low frequency range possible. Three modules, CFD, Computational Aeroacoustics (CAA) and acoustic beamforming, are incorporated in this proposed method. This thesis adopts an open source commercial software OpenFOAM for the flow field simulation with the Improved Delayed Detached Eddy Simulation (IDDES) turbulence model. The CAA calculation is conducted by an in-house code using impermeable Ffowcs-Williams and Hawkings (FW-H) equation for static sound source. The acoustic beamforming is performed by an in-house Delay and Sum (DAS) beamformer code with several different microphone array designs. Each module has been validated with currently available experimental data and numerical results. A flow over NACA 0012 airfoil case was chosen as a demonstration case for the new method. The aerodynamics and aeroacoustics results are shown and compared with the experimental measurements. A relatively good agreement has been achieved which gives the confidence of using this newly proposed method in sound source localization applications.

La reproduction réaliste d’environnements sonores est à ce jour un domaine actif de recherche. Ici, le cas traité est celui de la reproduction de l’environnement sonore d’une cabine d’avion en vol dans une maquette de celle-ci. Dans cette optique une connaissance la plus complète possible du milieu de reproduction ainsi que du champ de pression acoustique à reproduire est alors nécessaire. Pour ce faire la technique de résolution du problème inverse constitue un outil des plus adaptés. Le gros défaut de cette technique est sa sensibilité aux bruits de mesures inévitables dans des conditions réalistes d’expérimentation. Pour pallier à ce problème de nombreuses techniques pour le « régulariser » existent. La méthode plus particulièrement étudiée ici est celle combinant la régularisation classique de Tykhonov avec la technique du beamforming. Au travers des simulations et expériences en chambre anéchoique cette méthode est alors apparue comme étant plus efficace que la simple régularisation de Tikhonov. Pour compléter cette validation des mesures en situations moins « idéales » ont été faites comme en chambre réverbérante ou dans la cabine taille réelle d’avion du laboratoire GAUS. Ces expériences ont alors confirmé les observations faites préalablement. Parallèlement à cela, le développement d’un système de mesure a été fait ainsi que des outils nécessaires à son exploitation.

Swath bathymetry sidescan (SBS) sonar and the angle-of-arrival processing that underlies these systems has the capability to produce much higher resolution three dimensional imagery and bathymetry than traditional beamformed approaches. However, the performance of these high resolution systems is limited by signal-to-noise ratio (SNR) and they are also susceptible to multipath interference. This thesis explores two methods for increasing SNR and mitigating multipath interference for SBS systems. The first, binary coded pulse transmission and pulse compression is shown to increase the SNR and in turn provide reduced angle variance in SBS systems. The second, transmit beamforming, and more specifically steering and shading, is shown to increase both acoustic power in the water and directivity of the transmitted acoustic radiation. The transmit beamforming benefits are achieved by making use of the 8-element linear angle-of-arrival array typical in SBS sonars, but previously not utilized for transmit. Both simulations and real world SBS experiments are devised and conducted and it is shown that in practice pulse compression increases the SNR, and that transmit beamforming increases backscatter intensity and reduces the intensity of interfering multipaths. The improvement in achievable SNR and the reduction in multipath interference provided by the contributions in this thesis further strengthens the importance of SBS systems and angle-of-arrival based processing, as an alternative to beamforming, in underwater three dimensional imaging and mapping. / Graduate / 2015-04-15 / 0544 / 0547 / mark.butowski@gmail.com

In this thesis, we consider joint beamforming, power, and channel allocation in a multi-user and multi-channel underlay cognitive radio network (CRN). In this system, beamforming is implemented at each SU-TX to minimize the co-channel interference. The formulated joint optimization problem is a non-convex, mixed integer nonlinear programming (MINLP) problem. We propose a solution which consists of two stages. At first, given a channel allocation, a feasible solutions for power and beamforming vectors are derived by converting the problem into a convex form with an introduced optimal auxiliary variable and semidefinite relaxation (SDR) approach. Next, two explicit searching algorithms, i.e., genetic algorithm (GA) and simulated annealing (SA)-based algorithm are proposed to determine optimal channel allocations. Simulation results show that beamforming, power and channel allocation with SA (BPCA-SA) algorithm achieves a close optimal sum-rate with a lower computational complexity compared with beamforming, power and channel allocation with GA (BPCA-GA) algorithm. Furthermore, our proposed allocation scheme shows significant improvement than zero-forcing beamforming (ZFBF).