Lamb: a simulation tool for air-coupled lamb wave based ultrasonic NDE systems

La técnica de las ondas de Lamb acopladas por aire representa un importante avance en el área de los Ensayos No Destructivos (END) de materiales laminares.Sin embargo la compleja naturaleza de las vibraciones mecánicas encontradas en acústica, hacen que el análisis y el estudio de esta área del conocimiento sea un tema muy complejo. De allí que la posibilidad de contar con una herramienta de simulación de software que permita la evaluación y prueba de diferentes configuraciones de excitación y recepción acústica utilizando la flexibilidad de un modelo de computadora sea de una gran utilidad y ayuda.El objetivo de la presente tesis es proveer al área de los END con un software de simulación gratuito: The LAMB Matlab® toolbox basado en el modelo del software libre de la GNU.El software es capaz de simular el comportamiento de sistemas de END basados en ondas de Lamb acopladas por aire en láminas isótropas simples utilizando transductores tipo array.El programa se basa en un arreglo tipo C-scan de un sistema de END y está compuesto por tres bloques principales: 1) Excitación, 2) Propagación y 3) Recepción.La verificación individual del funcionamiento de dichos módulos se presenta a lo largo de la tesis mediante una serie de comparaciones entre simulaciones y datos experimentales provenientes de diferentes pruebas. Por otro lado, la validación del programa completo se llevo a cabo por medio de experimentos en láminas de cobre y aluminio; utilizando un sistema real de END por ondas de Lamb acopladas en aire mediante arrays cóncavos.La influencia negativa en el desempeño general de dicho sistema de END real basado en este tipo de transductores se comprobó efectivamente mediante el simulador desarrollado. Esto se debió fundamentalmente al efecto de directividad de los sensores individuales en los transductores y a la simetría cóncava de los arrays.Para emular este comportamiento la tesis presenta un modelo geométrico bidimensional simple de un filtro espacial, junto a las simulaciones de un nuevo tipo de array plano propuesto.El programa desarrollado comprobó así mismo la naturaleza coherente de los campos acústicos emitidos en aire por las láminas sujetas a vibraciones de Lamb. Esto se realizó mediante la implementación de un conformador de haz simple de suma y demora; constituyéndose así la etapa inicial de procesamiento de señal del bloque de recepción del programa.El objetivo principal del presente trabajo fue contribuir con un modelo operativo de simulación y prueba de nuevos diseños de arrays e implementación de estrategias de procesado de señal útiles en sistemas de END basados en ondas de Lamb acopladas por aire.Finalmente, si bien el objetivo de la calibración del programa no se pudo conseguir; si se logró efectivamente un notable grado de similitud con un sistema de END real.Air-coupled ultrasonic Lamb waves represent an important advance in Non- Destructive Testing and Evaluation (NDT & NDE) techniques of plate materials and structures. Examples of these advances are the characterization and quality assessment of laminate materials in manufacturing processes, the location of damaged parts in aircrafts and structure monitoring in the aerospace industry.However the rich and complex nature of mechanical vibrations encountered in acoustics make the subject of analysis and study of these systems a very complex task. Therefore a simulation tool that permits the evaluation and testing of different configuration scenarios using the flexibility of a computer model is an invaluable aid and advantage.The objective of this thesis is to provide the field of NDT with free open source software i.e. the LAMB Matlabrtoolbox. The toolbox is capable of simulating the behaviour of Lamb wave based NDE systems for single ideal isotropic laminates using air-coupled ultrasonic arrays. The programme usesa pitch-catch type of a Cscan NDE arrangement and is composed of three integrated sections each individually modelling a feature in the system: 1) Excitation, 2) Propagation, and 3) Reception.For assessment of the individual modules of the toolbox the thesis presents comparisons between each section simulations and the data obtained from different acoustic experiments. The validation of the complete simulator was carried out by evaluation tests on the copper and aluminium plates by use of a real hardware prototype of a Lamb wave based NDE system with aircoupled concave arrays.The negative impact on the performance of the real air-coupled NDE systembased on concave arrays was effectively confirmed by the programme. This was produced by the inherent directivity of the individual sensors as well as their concave arrangement. To emulate this behaviour the thesis introduces a simple two-dimensional geometric model for the inclusion of the spatial filtering effect of the sensors plus a group of simulations for a new proposed air-coupled plane array transducer.The software also verified the spatial coherent nature of the Lamb wave fields emitted by a plate in air. This was demonstrated by the implementation of a delay and sum beamformer to constitute an initial signal processing stage in the reception section

Beamforming design and power control for spectrum sharing systems

In order to provide wireless services for the current demand of high data rate mobile applications, more spectrally efficient systems are needed. As a matter of fact, the current wireless systems are limited by a frequency splitting spectrum management which in one hand minimizes the multiuser interference but; on the other hand, it precludes the use of wider bandwidth signals. As a more aggressive frequency reuse is targeted (ideally, all transmitters might eventually share the same frequency band), the use of multiple antennas for interference reliving, jointly with a smart power allocation is compulsory. In addition, novel spectrum management regulatory policies are required for ensuring a peaceful coexistence between adjacent spectrum sharing networks and for promoting their development. The aim of this dissertation is provide a beamforming and power allocation design for these novel spectrum sharing systems which are meant to exponentially increase the spectral efficiency of the systems. A mathematical framework based on multicriteria optimization for analyzing the beamforming design is provided which serves as a fundamental tool for describing the state-of-the-art studies in multiantenna interference networks. Indeed, the achievable rates are described and several ways of computing the Pareto rate region of MISO interference channel (i.e. the communication model that represents the spectrum sharing network when the transmitters use multiple antennas) are studied. Nevertheless, as the system designer aims to work in a single efficient rate point, the sum-rate optimal beamforming design is studied. Curiously, it results that under some realistic assumptions on both the desired and interference power levels, the obtained beamformer is the reciprocal version of a known receiving one and it optimizes a notion of antenna directivity for multiuser communications. Neverthelss, it is important to remark that the higher transmit power is used, the more interference dominated is the medium, not only within the wireless network, but also to eventually adjacent networks that might suffer from inter-network interference. In order to cope with this problem, a spectrum licensing system is revisited, namely time-area-spectrum license. Under this spectrum management mechanism, a license holder is able to radiate signals under a certain portion of time, within a concrete area and in a given band. Moreover, the amount of signal strength within the area is constraint by a certain value. Since controlling the signal power levels in a given area is cumbersome, we propose to restrict the receive power as an estimation of the overall accumulated signal strength. Therefore, the optimal transmit beamformers and power allocations are studied. Concretely, the achievable rates are derived and an operational working point is envisaged. In addition, a suboptimal yet low computationally complex and decentralized beamforming design is presented and it shows a good performance in front of other decentralized designs

Spatial dissection of a soundfield using spherical harmonic decomposition

A real-world soundfield is often contributed by multiple desired and undesired sound sources. The performance of many acoustic systems such as automatic speech recognition, audio surveillance, and teleconference relies on its ability to extract the desired sound components in such a mixed environment. The existing solutions to the above problem are constrained by various fundamental limitations and require to enforce different priors depending on the acoustic condition such as reverberation and spatial distribution of sound sources. With the growing emphasis and integration of audio applications in diverse technologies such as smart home and virtual reality appliances, it is imperative to advance the source separation technology in order to overcome the limitations of the traditional approaches. To that end, we exploit the harmonic decomposition model to dissect a mixed soundfield into its underlying desired and undesired components based on source and signal characteristics. By analysing the spatial projection of a soundfield, we achieve multiple outcomes such as (i) soundfield separation with respect to distinct source regions, (ii) source separation in a mixed soundfield using modal coherence model, and (iii) direction of arrival (DOA) estimation of multiple overlapping sound sources through pattern recognition of the modal coherence of a soundfield. We first employ an array of higher order microphones for soundfield separation in order to reduce hardware requirement and implementation complexity. Subsequently, we develop novel mathematical models for modal coherence of noisy and reverberant soundfields that facilitate convenient ways for estimating DOA and power spectral densities leading to robust source separation algorithms. The modal domain approach to the soundfield/source separation allows us to circumvent several practical limitations of the existing techniques and enhance the performance and robustness of the system. The proposed methods are presented with several practical applications and performance evaluations using simulated and real-life dataset

From Algorithmic to Neural Beamforming

Human interaction increasingly relies on telecommunication as an addition to or replacement for immediate contact. The direct interaction with smart devices, beyond the use of classical input devices such as the keyboard, has become common practice. Remote participation in conferences, sporting events, or concerts is more common than ever, and with current global restrictions on in-person contact, this has become an inevitable part of many people's reality. The work presented here aims at improving these encounters by enhancing the auditory experience. Augmenting fidelity and intelligibility can increase the perceived quality and enjoyability of such actions and potentially raise acceptance for modern forms of remote experiences. Two approaches to automatic source localization and multichannel signal enhancement are investigated for applications ranging from small conferences to large arenas. Three first-order microphones of fixed relative position and orientation are used to create a compact, reactive tracking and beamforming algorithm, capable of producing pristine audio signals in small and mid-sized acoustic environments. With inaudible beam steering and a highly linear frequency response, this system aims at providing an alternative to manually operated shotgun microphones or sets of individual spot microphones, applicable in broadcast, live events, and teleconferencing or for human-computer interaction. The array design and choice of capsules are discussed, as well as the challenges of preventing coloration for moving signals. The developed algorithm, based on Energy-Based Source Localization, is discussed and the performance is analyzed. Objective results on synthesized audio, as well as on real recordings, are presented. Results of multiple listening tests are presented and real-time considerations are highlighted. Multiple microphones with unknown spatial distribution are combined to create a large-aperture array using an end-to-end Deep-Learning approach. This method combines state-of-the-art single-channel signal separation networks with adaptive, domain-specific channel alignment. The Neural Beamformer is capable of learning to extract detailed spatial relations of channels with respect to a learned signal type, such as speech, and to apply appropriate corrections in order to align the signals. This creates an adaptive beamformer for microphones spaced on the order of up to 100m. The developed modules are analyzed in detail and multiple configurations are considered for different use cases. Signal processing inside the Neural Network is interpreted and objective results are presented on simulated and semi-simulated datasets

Circuit Design Techniques For Wideband Phased Arrays

University of Minnesota Ph.D. dissertation.June 2015. Major: Electrical Engineering. Advisor: Ramesh Harjani. 1 computer file (PDF); xii, 143 pages.This dissertation focuses on beam steering in wideband phased arrays and phase noise modeling in injection locked oscillators. Two different solutions, one in frequency and one in time, have been proposed to minimize beam squinting in phased arrays. Additionally, a differential current reuse frequency doubler for area and power savings has been proposed. Silicon measurement results are provided for the frequency domain solution (IBM 65nm RF CMOS), injection locked oscillator model verification (IBM 130nm RF-CMOS) and frequency doubler (IBM 65nm RF CMOS), while post extraction simulation results are provided for the time domain phased array solution (the chip is currently under fabrication, TSMC 65nm RF CMOS). In the frequency domain solution, a 4-point passive analog FFT based frequency tunable filter is used to channelize an incoming wideband signal into multiple narrowband signals, which are then processed through independent phase shifters. A two channel prototype has been developed at 8GHz RF frequency. Three discrete phase shifts (0 & +/- 90 degrees) are implemented through differential I-Q swapping with appropriate polarity. A minimum null-depth of 19dB while a maximum null-depth of 27dB is measured. In the time domain solution, a discrete time approach is undertaken with signals getting sampled in order of their arrival times. A two-channel prototype for a 2GHz instantaneous RF bandwidth (7GHz-9GHz) has been designed. A QVCO generates quadrature LO signals at 8GHz which are phase shifted through a 5-bit (2 extra bits from differential I-Q swapping with appropriate polarity) cartesian combiner. Baseband sampling clocks are generated from phase shifted LOs through a CMOS divide by 4 with independent resets. The design achieves an average time delay of 4.53ps with 31.5mW of power consumption (per channel, buffers excluded). An injection locked oscillator has been analyzed in s-domain using Paciorek's time domain transient equations. The simplified analysis leads to a phase noise model identical to that of a type-I PLL. The model is equally applicable to injection locked dividers and multipliers and has been extended to cover all injection locking scenarios. The model has been verified against a discrete 57MHz Colpitt's ILO, a 6.5GHz ILFD and a 24GHz ILFM with excellent matching between the model and measurements. Additionally, a differential current reuse frequency doubler, for frequency outputs between 7GHz to 14GHz, design has been developed to reduce passive area and dc power dissipation. A 3-bit capacitive tuning along with a tail current source is used to better conversion efficiency. The doubler shows FOM$_{T}$ values between 191dBc/Hz to 209dBc/Hz when driven by a 0.7GHz to 5.8GHz wide tuning VCO with a phase noise that ranges from -114dBc/Hz to -112dBc/Hz over the same bandwidth

Requirements for a mobile communications satellite system. Volume 2: Technical report

Three types of satellite aided mobile communications are considered for users in areas not served by (terrestrial) cellular radio systems. In system 1, mobile units are provided a direct satellite link to a gateway station, which serves as the interface to the terrestrial toll network. In system 2, a terrestrial radio link similar to those in cellular systems connects the mobile unit to a translator station; each translator relays the traffic from mobile units in its vicinity, via satellite, to the regional gateway. It is not feasible for system 2 to provide obiquitous coverage. Therefore, system 3 is introduced, in which the small percentage of users not within range of a translator are provided a direct satellite link as in system 1

Acceleration Techniques for Sparse Recovery Based Plane-wave Decomposition of a Sound Field

Plane-wave decomposition by sparse recovery is a reliable and accurate technique for plane-wave decomposition which can be used for source localization, beamforming, etc. In this work, we introduce techniques to accelerate the plane-wave decomposition by sparse recovery. The method consists of two main algorithms which are spherical Fourier transformation (SFT) and sparse recovery. Comparing the two algorithms, the sparse recovery is the most computationally intensive. We implement the SFT on an FPGA and the sparse recovery on a multithreaded computing platform. Then the multithreaded computing platform could be fully utilized for the sparse recovery. On the other hand, implementing the SFT on an FPGA helps to flexibly integrate the microphones and improve the portability of the microphone array. For implementing the SFT on an FPGA, we develop a scalable FPGA design model that enables the quick design of the SFT architecture on FPGAs. The model considers the number of microphones, the number of SFT channels and the cost of the FPGA and provides the design of a resource optimized and cost-effective FPGA architecture as the output. Then we investigate the performance of the sparse recovery algorithm executed on various multithreaded computing platforms (i.e., chip-multiprocessor, multiprocessor, GPU, manycore). Finally, we investigate the influence of modifying the dictionary size on the computational performance and the accuracy of the sparse recovery algorithms. We introduce novel sparse-recovery techniques which use non-uniform dictionaries to improve the performance of the sparse recovery on a parallel architecture