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Dissertations / Theses on the topic 'Adaptive computing systems Signal processing'
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Myjak, Mitchell John. "A medium-grain reconfigurable architecture for digital signal processing." Online access for everyone, 2006. http://www.dissertations.wsu.edu/Dissertations/Spring2006/m%5Fmyjak%5F042706.pdf.
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Skarpas, Daniel. "CAD tool emulation for a two-level reconfigurable DSP architecture." Online access for everyone, 2007. http://www.dissertations.wsu.edu/Thesis/Spring2007/D_Skarpas_050407.pdf.
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Larson, Jonathan Karl. "CAD tool emulation for a two-level reconfigurable cell array for digital signal processing." Online access for everyone, 2005. http://www.dissertations.wsu.edu/Thesis/Fall2005/j%5Flarson%5F120805.pdf.
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King, Timothy L. "Hardware interface to connect an AN/SPS-65 radar to an SRC-6E reconfigurable computer." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Mar%5FKing.pdf.
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Yoo, Heejong. "Low-Power Audio Input Enhancement for Portable Devices." Diss., Georgia Institute of Technology, 2005. http://hdl.handle.net/1853/6821.
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With the development of VLSI and wireless communication
technology, portable devices such as personal digital assistants
(PDAs), pocket PCs, and mobile phones have gained a lot of
popularity. Many such devices incorporate a speech recognition
engine, enabling users to interact with the devices using
voice-driven commands and text-to-speech synthesis.
The power consumption of DSP microprocessors has been
consistently decreasing by half about every 18 months, following
Gene's law. The capacity of signal processing, however, is still
significantly constrained by the limited power budget of these
portable devices. In addition, analog-to-digital (A/D) converters
can also limit the signal processing of portable devices. Many
systems require very high-resolution and high-performance A/D
converters, which often consume a large fraction of the limited
power budget of portable devices.
The proposed research develops a low-power audio signal
enhancement system that combines programmable analog signal
processing and traditional digital signal processing. By
utilizing analog signal processing based on floating-gate
transistor technology, the power consumption of the overall
system as well as the complexity of the A/D converters can be
reduced significantly. The system can be used as a front end of
portable devices in which enhancement of audio signal quality
plays a critical role in automatic speech recognition systems on
portable devices. The proposed system performs background audio
noise suppression in a continuous-time domain using analog
computing elements and acoustic echo cancellation in a
discrete-time domain using an FPGA.
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King, Graham A. "High performance computing systems for signal processing." Thesis, Southampton Solent University, 1996. http://ssudl.solent.ac.uk/2424/.
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The submission begins by demonstrating that the conditions required for consideration under the University's research degrees regulations have been met in full. There then follows a commentary which starts by explaining the origin of the research theme concerned and which continues by discussing the nature and significance of the work. This has been an extensive programme to devise new methods of improving the computational speed and efficiency required for effective implementation of FIR and IIR digital filters and transforms. The problems are analysed and initial experimental work is described which sought to quantify the performance to be derived from peripheral vector processors. For some classes of computation, especially in real time, it was necessary to tum to pure systolic array hardware engines and a large number of innovations are suggested, both in array architecture and in the creation of a new hybrid opto-electronic adder capable of improving the performance ofprocessing elements for the array. This significant and original research is extended further by including a class of computation involving a bit sliced co-processor. A means of measuring the performance of this system is developed and discussed. The contribution of the work has been evident in: software innovation for horizontal architecture microprocessors; improved multi-dimensional systolic array designs; the development of completely new implementations of processing elements in such arrays; and in the details of co-processing architectures for bit sliced microprocessors. The use of Read Only Memory in creating n-dimensional FIR or IIR filters, and in executing the discrete cosine transform is a further innovative contribution that has enabled researchers to re-examine the case for pre-calculated systems previously using stored squares. The Read Only Memory work has suggested that Read Only Memory chips may be combined in a way architecturally similar to systolic array processing elements. This led to original concepts of pipelining for memory devices. The work is entirely coherent in that it covers the application of these contributions to a set of common processes, producing a set of performance graded and scaleable solutions. In order that effective solutions are proposed it was necessary to demonstrate a solid underlying appreciation of the computational mechanics involved. Whilst the published papers within this submission assume such an understanding , two appendices are provided to demonstrate the essential groundwork necessary to underpin the work resulting in these publications. The improved results obtained from the programme were threefold: execution time; theoretical clocking speeds and circuit areas; and speed up ratios. In the case of the investigations involving vector signal processors the issue was one of quantifying the performance bounds of the architecture in performing specific combinations of signal processing functions. An important aspect of this work was the optimisation achieved in the programming of the device. The use of innovative techniques reduced the execution time for the complex combinational algorithms involved to sub 10 milliseconds. Given the real time constraints for typical applications and the aims for this research the work evolved toward dedicated hardware solutions. Systolic arrays were thus a significant area of investigation. In such systems meritorious criteria are concerned with achieving: a higher regularity in architectural structure; data exchanges only with nearest neighbour processing elements; minimised global distribution functions such as power supplies and clock lines; minimised latency; minimisation in the use of latches; the elimination of output adders; and the design of higher speed processing elements. The programme has made original and significant contributions to the art of effective array design culminating in systems calculated to clock at 100MHz when using 1 micron CMOS technology, whilst creating reductions in transistor count when compared with contemporary implementations. The improvements vary by specific design but are ofthe order of30-l00% speed advantage and 20-30% less real estate usage. The third type of result was obtained when considering operations best executed by dedicated microcode running on bit sliced engines. The main issues for this part of the work were the development of effective interactions between host processors and the bit sliced processors used for computationally intensive and repetitive functions together with the evaluation of the relative performance of new bit sliced microcode solutions. The speed up obtained relative to a range of state of the art microprocessors (68040, 80386, 32032) ranged from 2: 1 to 8: 1. The programme of research is represented by sixteen papers divided into three groups corresponding to the following stages in the work: problem definition and initial responses involving vector processors; the synthesis of higher performance solutions using dedicated hardware; and bit sliced solutions
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Hong, John Hyunchul Psaltis Demetri. "Optical computing for adaptive signal processing and associative memories /." Diss., Pasadena, Calif. : California Institute of Technology, 1987. http://resolver.caltech.edu/CaltechETD:etd-06142006-094757.
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Lorente, Giner Jorge. "Adaptive signal processing for multichannel sound using high performance computing." Doctoral thesis, Universitat Politècnica de València, 2015. http://hdl.handle.net/10251/58427.
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[EN] The field of audio signal processing has undergone a major development in recent years. Both the consumer and professional marketplaces continue to show growth in audio applications such as immersive audio schemes that offer optimal listening experience, intelligent noise reduction in cars or improvements in audio teleconferencing or hearing aids. The development of these applications has a common interest in increasing or improving the number of discrete audio channels, the quality of the audio or the sophistication of the algorithms. This often gives rise to problems of high computational cost, even when using common signal processing algorithms, mainly due to the application of these algorithms to multiple signals with real-time requirements. The field of High Performance Computing (HPC) based on low cost hardware elements is the bridge needed between the computing problems and the real multimedia signals and systems that lead to user's applications. In this sense, the present thesis goes a step further in the development of these systems by using the computational power of General Purpose Graphics Processing Units (GPGPUs) to exploit the inherent parallelism of signal processing for multichannel audio applications.
The increase of the computational capacity of the processing devices has been historically linked to the number of transistors in a chip. However, nowadays the improvements in the computational capacity are mainly given by increasing the number of processing units and using parallel processing. The Graphics Processing Units (GPUs), which have now thousands of computing cores, are a representative example. The GPUs were traditionally used to graphic or image processing, but new releases in the GPU programming environments such as CUDA have allowed the use of GPUS for general processing applications. Hence, the use of GPUs is being extended to a wide variety of intensive-computation applications among which audio processing is included. However, the data transactions between the CPU and the GPU and viceversa have questioned the viability of the use of GPUs for audio applications in which real-time interaction between microphones and loudspeakers is required. This is the case of the adaptive filtering applications, where an efficient use of parallel computation in not straightforward. For these reasons, up to the beginning of this thesis, very few publications had dealt with the GPU implementation of real-time acoustic applications based on adaptive filtering. Therefore, this thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications based on adaptive filtering that require high computational resources. To this end, different adaptive applications in the field of audio processing are studied and performed using GPUs. This manuscript also analyzes and solves possible limitations in each GPU-based implementation both from the acoustic point of view as from the computational point of view.[ES] El campo de procesado de señales de audio ha experimentado un desarrollo importante en los últimos años. Tanto el mercado de consumo como el profesional siguen mostrando un crecimiento en aplicaciones de audio, tales como: los sistemas de audio inmersivo que ofrecen una experiencia de sonido óptima, los sistemas inteligentes de reducción de ruido en coches o las mejoras en sistemas de teleconferencia o en audífonos. El desarrollo de estas aplicaciones tiene un propósito común de aumentar o mejorar el número de canales de audio, la propia calidad del audio o la sofisticación de los algoritmos. Estas mejoras suelen dar lugar a sistemas de alto coste computacional, incluso usando algoritmos comunes de procesado de señal. Esto se debe principalmente a que los algoritmos se suelen aplicar a sistemas multicanales con requerimientos de procesamiento en tiempo real. El campo de la Computación de Alto Rendimiento basado en elementos hardware de bajo coste es el puente necesario entre los problemas de computación y los sistemas multimedia que dan lugar a aplicaciones de usuario. En este sentido, la presente tesis va un paso más allá en el desarrollo de estos sistemas mediante el uso de la potencia de cálculo de las Unidades de Procesamiento Gráfico (GPU) en aplicaciones de propósito general. Con ello, aprovechamos la inherente capacidad de paralelización que poseen las GPU para procesar señales de audio y obtener aplicaciones de audio multicanal. El aumento de la capacidad computacional de los dispositivos de procesado ha estado vinculado históricamente al número de transistores que había en un chip. Sin embargo, hoy en día, las mejoras en la capacidad computacional se dan principalmente por el aumento del número de unidades de procesado y su uso para el procesado en paralelo. Las GPUs son un ejemplo muy representativo. Hoy en día, las GPUs poseen hasta miles de núcleos de computación. Tradicionalmente, las GPUs se han utilizado para el procesado de gráficos o imágenes. Sin embargo, la aparición de entornos sencillos de programación GPU, como por ejemplo CUDA, han permitido el uso de las GPU para aplicaciones de procesado general. De ese modo, el uso de las GPU se ha extendido a una amplia variedad de aplicaciones que requieren cálculo intensivo. Entre esta gama de aplicaciones, se incluye el procesado de señales de audio. No obstante, las transferencias de datos entre la CPU y la GPU y viceversa pusieron en duda la viabilidad de las GPUs para aplicaciones de audio en las que se requiere una interacción en tiempo real entre micrófonos y altavoces. Este es el caso de las aplicaciones basadas en filtrado adaptativo, donde el uso eficiente de la computación en paralelo no es sencillo. Por estas razones, hasta el comienzo de esta tesis, había muy pocas publicaciones que utilizaran la GPU para implementaciones en tiempo real de aplicaciones acústicas basadas en filtrado adaptativo. A pesar de todo, esta tesis pretende demostrar que las GPU son herramientas totalmente válidas para llevar a cabo aplicaciones de audio basadas en filtrado adaptativo que requieran elevados recursos computacionales. Con este fin, la presente tesis ha estudiado y desarrollado varias aplicaciones adaptativas de procesado de audio utilizando una GPU como procesador. Además, también analiza y resuelve las posibles limitaciones de cada aplicación tanto desde el punto de vista acústico como desde el punto de vista computacional.
[CAT] El camp del processament de senyals d'àudio ha experimentat un desenvolupament important als últims anys. Tant el mercat de consum com el professional segueixen mostrant un creixement en aplicacions d'àudio, com ara: els sistemes d'àudio immersiu que ofereixen una experiència de so òptima, els sistemes intel·ligents de reducció de soroll en els cotxes o les millores en sistemes de teleconferència o en audiòfons. El desenvolupament d'aquestes aplicacions té un propòsit comú d'augmentar o millorar el nombre de canals d'àudio, la pròpia qualitat de l'àudio o la sofisticació dels algorismes que s'utilitzen. Això, sovint dóna lloc a sistemes d'alt cost computacional, fins i tot quan es fan servir algorismes comuns de processat de senyal. Això es deu principalment al fet que els algorismes se solen aplicar a sistemes multicanals amb requeriments de processat en temps real. El camp de la Computació d'Alt Rendiment basat en elements hardware de baix cost és el pont necessari entre els problemes de computació i els sistemes multimèdia que donen lloc a aplicacions d'usuari. En aquest sentit, aquesta tesi va un pas més enllà en el desenvolupament d'aquests sistemes mitjançant l'ús de la potència de càlcul de les Unitats de Processament Gràfic (GPU) en aplicacions de propòsit general. Amb això, s'aprofita la inherent capacitat de paral·lelització que posseeixen les GPUs per processar senyals d'àudio i obtenir aplicacions d'àudio multicanal. L'augment de la capacitat computacional dels dispositius de processat ha estat històricament vinculada al nombre de transistors que hi havia en un xip. No obstant, avui en dia, les millores en la capacitat computacional es donen principalment per l'augment del nombre d'unitats de processat i el seu ús per al processament en paral·lel. Un exemple molt representatiu són les GPU, que avui en dia posseeixen milers de nuclis de computació. Tradicionalment, les GPUs s'han utilitzat per al processat de gràfics o imatges. No obstant, l'aparició d'entorns senzills de programació de la GPU com és CUDA, han permès l'ús de les GPUs per a aplicacions de processat general. D'aquesta manera, l'ús de les GPUs s'ha estès a una àmplia varietat d'aplicacions que requereixen càlcul intensiu. Entre aquesta gamma d'aplicacions, s'inclou el processat de senyals d'àudio. No obstant, les transferències de dades entre la CPU i la GPU i viceversa van posar en dubte la viabilitat de les GPUs per a aplicacions d'àudio en què es requereix la interacció en temps real de micròfons i altaveus. Aquest és el cas de les aplicacions basades en filtrat adaptatiu, on l'ús eficient de la computació en paral·lel no és senzilla. Per aquestes raons, fins al començament d'aquesta tesi, hi havia molt poques publicacions que utilitzessin la GPU per implementar en temps real aplicacions acústiques basades en filtrat adaptatiu. Malgrat tot, aquesta tesi pretén demostrar que les GPU són eines totalment vàlides per dur a terme aplicacions d'àudio basades en filtrat adaptatiu que requereixen alts recursos computacionals. Amb aquesta finalitat, en la present tesi s'han estudiat i desenvolupat diverses aplicacions adaptatives de processament d'àudio utilitzant una GPU com a processador. A més, aquest manuscrit també analitza i resol les possibles limitacions de cada aplicació, tant des del punt de vista acústic, com des del punt de vista computacional.
Lorente Giner, J. (2015). Adaptive signal processing for multichannel sound using high performance computing [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/58427
TESIS
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Chan, M. K. "Adaptive signal processing algorithms for non-Gaussian signals." Thesis, Queen's University Belfast, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.269023.
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Gerbracht, Sabrina, Eduard A. Jorswieck, Gan Zheng, and Björn Ottersten. "Non-regenerative Two-Hop Wiretap Channels using Interference Neutralization." Saechsische Landesbibliothek- Staats- und Universitaetsbibliothek Dresden, 2013. http://nbn-resolving.de/urn:nbn:de:bsz:14-qucosa-113245.
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In this paper, we analyze the achievable secrecy rates in the two-hop wiretap channel with four nodes, where the transmitter and the receiver have multiple antennas while the relay and the eavesdropper have only a single antenna each. The relay is operating in amplify-and-forward mode and all the channels between the nodes are known perfectly by the transmitter. We discuss different transmission and protection schemes like artificial noise (AN). Furthermore, we introduce interference neutralization (IN) as a new protection scheme. We compare the different schemes regarding the high-SNR slope and the high-SNR power offset and illustrate the performance by simulation results. It is shown analytically as well as by numerical simulations that the high SNR performance of the proposed IN scheme is better than the one of AN.
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Famorzadeh, Shahram. "BEEHIVE : an adaptive, distributed, embedded signal processing environment." Diss., Georgia Institute of Technology, 1997. http://hdl.handle.net/1853/14803.
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White, Paul Robert. "Adaptive signal processing and its application to infrared detector systems." Thesis, University of Southampton, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.316442.
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Wang, Cheng. "Adaptive downlink multi-user MIMO wireless systems /." View abstract or full-text, 2007. http://library.ust.hk/cgi/db/thesis.pl?ECED%202007%20WANG.
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Ranganathan, Raghuram. "Novel complex adaptive signal processing techniques employing optimally derived time-varying convergence factors with applications in digital signal processing and wireless communications." Orlando, Fla. : University of Central Florida, 2008. http://purl.fcla.edu/fcla/etd/CFE0002431.
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Hossain, Mohammed Alamgir. "Digital signal processing and parallel processing for real-time adaptive noise and vibration control." Thesis, University of Sheffield, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.484164.
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Lynch, Michael Richard. "Adaptive techniques in signal processing and connectionist models." Thesis, University of Cambridge, 1990. https://www.repository.cam.ac.uk/handle/1810/244884.
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This thesis covers the development of a series of new methods and the application of adaptive filter theory which are combined to produce a generalised adaptive filter system which may be used to perform such tasks as pattern recognition. Firstly, the relevant background adaptive filter theory is discussed in Chapter 1 and methods and results which are important to the rest of the thesis are derived or referenced. Chapter 2 of this thesis covers the development of a new adaptive algorithm which is designed to give faster convergence than the LMS algorithm but unlike the Recursive Least Squares family of algorithms it does not require storage of a matrix with n2 elements, where n is the number of filter taps. In Chapter 3 a new extension of the LMS adaptive notch filter is derived and applied which gives an adaptive notch filter the ability to lock and track signals of varying pitch without sacrificing notch depth. This application of the LMS filter is of interest as it demonstrates a time varying filter solution to a stationary problem. The LMS filter is next extended to the multidimensional case which allows the application of LMS filters to image processing. The multidimensional filter is then applied to the problem of image registration and this new application of the LMS filter is shown to have significant advantages over current image registration methods. A consideration of the multidimensional LMS filter as a template matcher and pattern recogniser is given. In Chapter 5 a brief review of statistical pattern recognition is given, and in Chapter 6 a review of relevant connectionist models. In Chapter 7 the generalised adaptive filter is derived. This is an adaptive filter with the ability to model non-linear input-output relationships. The Volterra functional analysis of non-linear systems is given and this is combined with adaptive filter methods to give a generalised non-linear adaptive digital filter. This filter is then considered as a linear adaptive filter operating in a non-linearly extended vector space. This new filter is shown to have desirable properties as a pattern recognition system. The performance and properties of the new filter is compared with current connectionist models and results demonstrated in Chapter 8. In Chapter 9 further mathematical analysis of the networks leads to suggested methods to greatly reduce network complexity for a given problem by choosing suitable pattern classification indices and allowing it to define its own internal structure. In Chapter 10 robustness of the network to imperfections in its implementation is considered. Chapter 11 finishes the thesis with some conclusions and suggestions for future work.
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Zhang, Ziming. "Adaptive Power Management for Autonomic Resource Configuration in Large-scale Computer Systems." Thesis, University of North Texas, 2015. https://digital.library.unt.edu/ark:/67531/metadc804939/.
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In order to run and manage resource-intensive high-performance applications, large-scale computing and storage platforms have been evolving rapidly in various domains in both academia and industry. The energy expenditure consumed to operate and maintain these cloud computing infrastructures is a major factor to influence the overall profit and efficiency for most cloud service providers. Moreover, considering the mitigation of environmental damage from excessive carbon dioxide emission, the amount of power consumed by enterprise-scale data centers should be constrained for protection of the environment.Generally speaking, there exists a trade-off between power consumption and application performance in large-scale computing systems and how to balance these two factors has become an important topic for researchers and engineers in cloud and HPC communities. Therefore, minimizing the power usage while satisfying the Service Level Agreements have become one of the most desirable objectives in cloud computing research and implementation. Since the fundamental feature of the cloud computing platform is hosting workloads with a variety of characteristics in a consolidated and on-demand manner, it is demanding to explore the inherent relationship between power usage and machine configurations. Subsequently, with an understanding of these inherent relationships, researchers are able to develop effective power management policies to optimize productivity by balancing power usage and system performance. In this dissertation, we develop an autonomic power-aware system management framework for large-scale computer systems. We propose a series of techniques including coarse-grain power profiling, VM power modelling, power-aware resource auto-configuration and full-system power usage simulator. These techniques help us to understand the characteristics of power consumption of various system components. Based on these techniques, we are able to test various job scheduling strategies and develop resource management approaches to enhance the systems' power efficiency.
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Gerbracht, Sabrina, Eduard A. Jorswieck, Gan Zheng, and Björn Ottersten. "Non-regenerative Two-Hop Wiretap Channels using Interference Neutralization." Technische Universität Dresden, 2012. https://tud.qucosa.de/id/qucosa%3A26894.
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In this paper, we analyze the achievable secrecy rates in the two-hop wiretap channel with four nodes, where the transmitter and the receiver have multiple antennas while the relay and the eavesdropper have only a single antenna each. The relay is operating in amplify-and-forward mode and all the channels between the nodes are known perfectly by the transmitter. We discuss different transmission and protection schemes like artificial noise (AN). Furthermore, we introduce interference neutralization (IN) as a new protection scheme. We compare the different schemes regarding the high-SNR slope and the high-SNR power offset and illustrate the performance by simulation results. It is shown analytically as well as by numerical simulations that the high SNR performance of the proposed IN scheme is better than the one of AN.
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Baines, Steven John. "Linear multi-user detection in DS-CDMA cellular systems." Thesis, University of York, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263690.
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Eriksson, Anton. "Robust Echo-Cancellation for Simple VoIP-Applications in Embedded Systems." Thesis, Linköpings universitet, Kommunikationssystem, 2015. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-121862.
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Voice over IP (VoIP) is the group of techniques for delivering voice communications over Internet Protocol (IP) networks. It has mainly served as the possible substitution for regular PSTN over the last decades, but has recently gained an increased interest in various areas such as alarm applications and customer service. Acoustic echo is the situation were a distorted version of the sent signal is transmitted back to the sender, due to acoustic feedback between loudspeaker and microphone. There already exists several algorithms to solve this problem, and this thesis provides a study of the performance in relation to the computational complexity of the algorithms. This is in order to indicate which approaches are better suited for implementation in an embedded system, where resources are limited. During the thesis a number of algorithms were tested, including variations of the LMS algorithm, some other approaches utilizing the correlation between echo and signal, and the RLS algorithm. They were first tested in MATLAB, on speech signals recorded at Syntronic and distorted by adding echo, then tested by implementation in C, and run on speech signals recorded in a simulated VoIP system at Syntronic. The results were then evaluated in terms of efficiency and computational complexity.
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Kwok, Tai-on Tyrone. "High performance embedded reconfigurable computing data security and media processing applications /." Click to view the E-thesis via HKUTO, 2005. http://sunzi.lib.hku.hk/hkuto/record/B3204043X.
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Kwok, Tai-on Tyrone, and 郭泰安. "High performance embedded reconfigurable computing: data security and media processing applications." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2005. http://hub.hku.hk/bib/B3204043X.
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Blunt, Shannon D. "Novel adaptive signal processing algorithms for wireless communications : echo cancellation and multiuser detection /." free to MU campus, to others for purchase, 2002. http://wwwlib.umi.com/cr/mo/fullcit?p3074375.
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Liu, Ying. "Complex-valued adaptive digital signal enhancement for applications in wireless communication systems." Doctoral diss., University of Central Florida, 2012. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/5405.
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In recent decades, the wireless communication industry has attracted a great deal of research efforts to satisfy rigorous performance requirements and preserve high spectral efficiency. Along with this trend, I/Q modulation is frequently applied in modern wireless communications to develop high performance and high data rate systems. This has necessitated the need for applying efficient complex-valued signal processing techniques to highly-integrated, multi-standard receiver devices. In this dissertation, novel techniques for complex-valued digital signal enhancement are presented and analyzed for various applications in wireless communications. The first technique is a unified block processing approach to generate the complex-valued conjugate gradient Least Mean Square (LMS) techniques with optimal adaptations. The proposed algorithms exploit the concept of the complex conjugate gradients to find the orthogonal directions for updating the adaptive filter coefficients at each iteration. Along each orthogonal direction, the presented algorithms employ the complex Taylor series expansion to calculate time-varying convergence factors tailored for the adaptive filter coefficients. The performance of the developed technique is tested in the applications of channel estimation, channel equalization, and adaptive array beamforming. Comparing with the state of the art methods, the proposed techniques demonstrate improved performance and exhibit desirable characteristics for practical use. The second complex-valued signal processing technique is a novel Optimal Block Adaptive algorithm based on Circularity, OBA-C. The proposed OBA-C method compensates for a complex imbalanced signal by restoring its circularity. In addition, by utilizing the complex Taylor series expansion, the OBA-C method optimally updates the adaptive filter coefficients at each iteration. This algorithm can be applied to mitigate the frequency-dependent I/Q mismatch effects in analog front-end. Simulation results indicate that comparing with the existing methods, OBA-C exhibits superior convergence speed while maintaining excellent accuracy. The third technique is regarding interference rejection in communication systems. The research on both LMS and Independent Component Analysis (ICA) based techniques continues to receive significant attention in the area of interference cancellation. The performance of the LMS and ICA based approaches is studied for signals with different probabilistic distributions. Our research indicates that the ICA-based approach works better for super-Gaussian signals, while the LMS-based method is preferable for sub-Gaussian signals. Therefore, an appropriate choice of interference suppression algorithms can be made to satisfy the ever-increasing demand for better performance in modern receiver design.Ph.D.
Doctorate
Electrical Engineering and Computer Science
Engineering and Computer Science
Electrical Engineering
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Kulakcherla, Sudheer. "Non [sic] linear adaptive filters for echo cancellation of speech coded signals /." free to MU campus, to others for purchase, 2004. http://wwwlib.umi.com/cr/mo/fullcit?p1426079.
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Lim, GuBong. "H-MIMO a hybrid of spatial multiplexing and adaptive beamforming /." Access to citation, abstract and download form provided by ProQuest Information and Learning Company; downloadable PDF file 1.65 Mb., 69 p, 2005. http://gateway.proquest.com/openurl?url_ver=Z39.88-2004&res_dat=xri:pqdiss&rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&rft_dat=xri:pqdiss:1428176.
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Perry, Stuart William. "Adaptive image restoration perception based neural network models and algorithms /." Connect to full text, 1998. http://hdl.handle.net/2123/389.
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Thesis (Ph. D.)--University of Sydney, 1999.Title from title screen (viewed Apr. 16, 2008). Submitted in fulfilment of the requirements for the degree of Doctor of Philosophy to the School of Electrical and Information Engineering, Faculty of Engineering. Degree awarded 1999; thesis submitted 1998. Includes bibliography. Also available in print form.
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Doheny, David A. "Real Time Digital Signal Processing Adaptive Filters for Correlated Noise Reduction in Ring Laser Gyro Inertial Systems." [Tampa, Fla.] : University of South Florida, 2004. http://purl.fcla.edu/fcla/etd/SFE0000306.
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Choi, Hyung Keun. "Blind source separation of the audio signals in a real world." Thesis, Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/14986.
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Lee, Bong-Woon. "Applications of signal processing techniques in direct-sequence spread spectrum communication systems." Ohio : Ohio University, 1990. http://www.ohiolink.edu/etd/view.cgi?ohiou1173208101.
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Abusultan, Monther Younis. "Digital implementation of direction-of-arrival estimation techniques for smart antenna systems." Thesis, Montana State University, 2010. http://etd.lib.montana.edu/etd/2010/abusultan/AbusultanM0510.pdf.
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Adaptive antenna arrays use multiple antenna elements to form directional patterns in order to improve the performance of wireless communication systems. The antenna arrays also have the ability to detect the direction of incoming signals. These two capabilities allow a smart antenna system to adaptively beamform to more efficiently communicate between nodes. The direction-of-arrival estimation is a crucial component of the smart antenna system that uses open-loop adaptive approach. Historically this estimation has been accomplished using a personal computer. Implementing the estimation in the digital domain has the potential to provide a low cost and light weight solution due to recent advances in digital integrated circuit fabrication processes. Furthermore, digital circuitry allows for more sophisticated estimation algorithms to be implemented using the computational power of modern digital devices. This thesis presents the design and prototyping of direction-of-arrival (DOA) estimation for a smart antenna system implemented on a reconfigurable digital hardware fabric. Two DOA estimation algorithms are implemented and the performance tradeoffs between a custom hardware approach and a microprocessor-based system are compared. The algorithms were implemented for a 5.8 GHz, 8-element circular antenna array and their functionality was verified using a testbed platform. The implementation and analysis presented in this work will aid system designers to understand the tradeoffs between implementing algorithms in custom hardware versus an embedded system and when a hybrid approach is more advantageous.
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Huang, Kuan Lun Electrical Engineering & Telecommunications Faculty of Engineering UNSW. "Precoder design and adaptive modulation for MIMO broadcast channels." Awarded by:University of New South Wales. Electrical Engineering & Telecommunications, 2007. http://handle.unsw.edu.au/1959.4/40735.
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Multiple-input multiple-output (MIMO) technology, originated in the 1990s, is an emerging and fast growing area of communication research due to the ability to provide diversity as well as transmission degrees-of-freedom. Recent research focus on MIMO systems has shifted from the point-to-point link to the one-to-many multiuser links due to the ever increasing demand for multimedia-intensive services from users. The downlink of a multiuser transmission is called the broadcast channel (BC) and the reverse many-to-one uplink is termed the multiple access channel (MAC). Early studies in the MIMO BC and the MIMO MAC were mostly information-theoretic in nature. In particular, the characterizations of the capacity regions of the two systems were of primary concerns. The information-theoretic results suggest the optimal uplink detection scheme involves successive interference cancellation while successive application of dirty paper coding at the transmitter is optimal in the downlink channels. Over the past few years, after the full characterizations of the capacity regions, several practical precoders had been suggested to realize the benefits of MIMO multiuser transmission. However, linear precoders such as the zero-forcing (ZF) and the MMSE precoders fall short on the achievable capacity despite their simple structure. Nonlinear precoders such as the ZF dirty paper (ZF-DP) and the the MMSE generalized decision feedback equalizer-type (MMSE-GDFE) precoders demonstrated promising performance but suffered from either restriction on the number of antennas at users, i.e. ZF-DP, or high computational load for the transmit filter, i.e. MMSE-GDFE. An novice MMSE feedback precoder (MMSE-FBP) with low computational requirement was proposed and its performance was shown to come very close to the bound suggested by information theory. In this thesis, we undertake investigation of the causes of the capacity inferiority and come to the conclusion that power control is necessary in a multiuser environment. New schemes that address the power control issue are proposed and their performances are evaluated and compared. Adaptive modulation is an effective and powerful technique that can increase the spectral efficiency in a fading environment remarkably. It works by observing the channel variations and adapts the transmission power and/or rate to counteract the instabilities of the channel. This thesis extends the pioneering study of adaptive modulation on single-input single-output (8180) Gaussian channel to the MIMO BC. We explore various combinations of power and rate adaptions and observe their impact on the system performance. In particular, we present analytical and simulation results on the successiveness of adaptive modulation in maximizing multiuser spectral efficiency. Furthermore, empirical research is conducted to validate its effectiveness in optimizing the overall system reliability.
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Ghafoor, Sheikh Khaled. "Modeling of an adaptive parallel system with malleable applications in a distributed computing environment." Diss., Mississippi State : Mississippi State University, 2007. http://sun.library.msstate.edu/ETD-db/theses/available/etd-11092007-145420.
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Gao, Ying. "A Digital Signal Processing Approach for Affective Sensing of a Computer User through Pupil Diameter Monitoring." FIU Digital Commons, 2009. http://digitalcommons.fiu.edu/etd/132.
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Recent research has indicated that the pupil diameter (PD) in humans varies with their affective states. However, this signal has not been fully investigated for affective sensing purposes in human-computer interaction systems. This may be due to the dominant separate effect of the pupillary light reflex (PLR), which shrinks the pupil when light intensity increases. In this dissertation, an adaptive interference canceller (AIC) system using the H∞ time-varying (HITV) adaptive algorithm was developed to minimize the impact of the PLR on the measured pupil diameter signal. The modified pupil diameter (MPD) signal, obtained from the AIC was expected to reflect primarily the pupillary affective responses (PAR) of the subject. Additional manipulations of the AIC output resulted in a processed MPD (PMPD) signal, from which a classification feature, PMPDmean, was extracted. This feature was used to train and test a support vector machine (SVM), for the identification of stress states in the subject from whom the pupil diameter signal was recorded, achieving an accuracy rate of 77.78%. The advantages of affective recognition through the PD signal were verified by comparatively investigating the classification of stress and relaxation states through features derived from the simultaneously recorded galvanic skin response (GSR) and blood volume pulse (BVP) signals, with and without the PD feature. The discriminating potential of each individual feature extracted from GSR, BVP and PD was studied by analysis of its receiver operating characteristic (ROC) curve. The ROC curve found for the PMPDmean feature encompassed the largest area (0.8546) of all the single-feature ROCs investigated. The encouraging results seen in affective sensing based on pupil diameter monitoring were obtained in spite of intermittent illumination increases purposely introduced during the experiments. Therefore, these results confirmed the benefits of using the AIC implementation with the HITV adaptive algorithm to isolate the PAR and the potential of using PD monitoring to sense the evolving affective states of a computer user.
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Li, Jonathan Chi Fai. "Eye closure penalty based signal quality metric for intelligent all-optical networks /." Connect to thesis, 2009. http://repository.unimelb.edu.au/10187/7047.
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Terry, John D. "Blind adaptive array techniques for mobile satellite communications." Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/13425.
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Bansal, Mayur. "DIGITAL CONTROL BOARD FOR PHASED ARRAY ANTENNA BEAM STEERING IN ADAPTIVE COMMUNICATION APPLICATIONS." DigitalCommons@CalPoly, 2013. https://digitalcommons.calpoly.edu/theses/1113.
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The application of adaptive communication techniques for mobile communications has attracted considerable interest in the last decade. One example of these techniques is spatial filtering through planar antenna array beam forming.
This thesis describes the development of a digital system that adaptively controls a phased array antenna. The radiating structure of the phased antenna array is tetrahedral-shaped and contains four antenna elements on each of its three faces. The overall system comprises of a digital control board with an external computer interface, an RF control board, and the phased antenna array. The RF controls the main lobe direction on the phased array antenna. This thesis describes the design and implementation of the digital control board.
The digital control board`s primary responsibilities are implementing inter- faces between the external computer and the RF board, which results in two operational modes: the MATLAB graphical user interface (GUI) mode and the adaptive receive mode. The GUI mode allows users to input parameters that provide interactive control of the phased antenna array by interfacing with an external computer and the RF control board. The adaptive receive mode im- plements an algorithm for an adaptive receive station. This algorithm uses a 58-point scanning technique that locates the maximum receive power direction.
Test results show that the digital control board successfully manages the RF board control voltage with an nominal error of less than 1%, which subsequently allows for precise control of the antenna`s active face. Additionally, testing of the GUI demonstrated the successful interactive application of various system control parameters.
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Pozzebon, Marlei. "The implementation of configurable technologies : negotiations between global principles and local contexts." Thesis, McGill University, 2003. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=84540.
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This investigation focuses on configurable technologies, a term which refers to technologies that are highly parameterizable and are built from a range of components to meet the very specific requirements of a particular organization. They cannot be seen independently of their representations through external intermediaries who "speak" for the technology by providing images, descriptions, demonstrations, policies, templates and "solutions". I use the term technology-configuring mediation to refer to the process characterized by a socially constructed relationship between clients and consultants, where visions of how the technology should operate are negotiated. Configurable tools are well illustrated by ERP projects and represent an important trend in IS, drawing its popularity from the hope of benefiting from increased economies of scale and access to cumulative knowledge supposedly "embedded" into these technological artifacts.From a critical interpretive perspective that combines ideas from structuration theory, social shaping views of technology and critical discourse analysis, this dissertation is based on an empirical investigation that spanned one year and is primarily organized in three papers. The first paper investigates the use of structuration theory in the IS field, asking: How can we successfully apply structuration theory in IS empirical research? Paper 1 contributes to the advancement of interpretive research methods by describing, analyzing and illustrating the ways IS scholars have used Giddens' theory in their research. In addition, it presents a repertoire of research strategies that may help overcome barriers to the empirical application of structurationist theory by dealing with three core elements: time, context and duality of technology.
The second paper discusses the rhetorical closure that often dominates discourses about IT, arguing that configurable technologies are social constructions and, to different degrees, are always open to change. Taking ERP projects as a typical illustration of configurable IT, Paper 2 describes a multilevel framework that identifies occasions for ERP package negotiation and change at three levels---segment, organization and individual---thereby breaking down the rhetorical closure that seems to dominate public debate. Paper 2 draws on structurationist and political streams of thinking about technology to set out a theoretical framework that contributes to advancing our knowledge of configurable IS phenomena.
The third paper addresses the question: How does the mediation process influence the negotiation between global principles and local contexts during the implementation of configurable IS, and how does such a negotiation influence the success of the implemented technology? Paper 3 provides a new understanding of configurable technology implementation. The structuring of a new configuration is seen as a mediation process where knowledge and power dependencies are created and recreated over time by consultants and clients, the entire process being bordered by internal and external constraints. Paper 3 recognizes different patterns of mediation and explains how these patterns affect the negotiation of global principles and local contexts as well as the project results. The study ends by identifying a collection of mediating strategies that are likely to improve the implementation of configurable IS.
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Jalali, Sammuel. "Wireless Channel Equalization in Digital Communication Systems." Scholarship @ Claremont, 2012. http://scholarship.claremont.edu/cgu_etd/42.
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Our modern society has transformed to an information-demanding system, seeking voice, video, and data in quantities that could not be imagined even a decade ago. The mobility of communicators has added more challenges. One of the new challenges is to conceive highly reliable and fast communication system unaffected by the problems caused in the multipath fading wireless channels. Our quest is to remove one of the obstacles in the way of achieving ultimately fast and reliable wireless digital communication, namely Inter-Symbol Interference (ISI), the intensity of which makes the channel noise inconsequential.
The theoretical background for wireless channels modeling and adaptive signal processing are covered in first two chapters of dissertation.
The approach of this thesis is not based on one methodology but several algorithms and configurations that are proposed and examined to fight the ISI problem. There are two main categories of channel equalization techniques, supervised (training) and blind unsupervised (blind) modes. We have studied the application of a new and specially modified neural network requiring very short training period for the proper channel equalization in supervised mode. The promising performance in the graphs for this network is presented in chapter 4.
For blind modes two distinctive methodologies are presented and studied. Chapter 3 covers the concept of multiple "cooperative" algorithms for the cases of two and three cooperative algorithms. The "select absolutely larger equalized signal" and "majority vote" methods have been used in 2-and 3-algoirithm systems respectively. Many of the demonstrated results are encouraging for further research.
Chapter 5 involves the application of general concept of simulated annealing in blind mode equalization. A limited strategy of constant annealing noise is experimented for testing the simple algorithms used in multiple systems. Convergence to local stationary points of the cost function in parameter space is clearly demonstrated and that justifies the use of additional noise. The capability of the adding the random noise to release the algorithm from the local traps is established in several cases.
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Wang, Zhaohong. "Information-Theoretic Secure Outsourced Computation in Distributed Systems." UKnowledge, 2016. http://uknowledge.uky.edu/ece_etds/88.
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Secure multi-party computation (secure MPC) has been established as the de facto paradigm for protecting privacy in distributed computation. One of the earliest secure MPC primitives is the Shamir's secret sharing (SSS) scheme. SSS has many advantages over other popular secure MPC primitives like garbled circuits (GC) -- it provides information-theoretic security guarantee, requires no complex long-integer operations, and often leads to more efficient protocols. Nonetheless, SSS receives less attention in the signal processing community because SSS requires a larger number of honest participants, making it prone to collusion attacks. In this dissertation, I propose an agent-based computing framework using SSS to protect privacy in distributed signal processing. There are three main contributions to this dissertation. First, the proposed computing framework is shown to be significantly more efficient than GC. Second, a novel game-theoretical framework is proposed to analyze different types of collusion attacks. Third, using the proposed game-theoretical framework, specific mechanism designs are developed to deter collusion attacks in a fully distributed manner. Specifically, for a collusion attack with known detectors, I analyze it as games between secret owners and show that the attack can be effectively deterred by an explicit retaliation mechanism. For a general attack without detectors, I expand the scope of the game to include the computing agents and provide deterrence through deceptive collusion requests. The correctness and privacy of the protocols are proved under a covert adversarial model. Our experimental results demonstrate the efficiency of SSS-based protocols and the validity of our mechanism design.
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Marr, Bo. "Learning, probabilistic, and asynchronous technologies for an ultra efficient datapath." Diss., Atlanta, Ga. : Georgia Institute of Technology, 2009. http://hdl.handle.net/1853/31724.
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Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2010.Committee Chair: Paul Hasler; Committee Co-Chair: David V. Anderson. Part of the SMARTech Electronic Thesis and Dissertation Collection.
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Kim, Ngan Trieu, and Olumide Ajiboye. "PERFORMANCE ANALYSIS OF ADAPTIVE ARRAY SYSTEM AND SPACE-TIME BLOCK CODING IN MOBILE WIMAX (802.16e) SYSTEMS." Thesis, Blekinge Tekniska Högskola, Avdelningen för telekommunikationssystem, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-4743.
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We live in an information hungry age, we generate and process information at a rate never before recorded in the history of mankind. Today’s computing platforms are run on Gigahertz multi-core processors churning out Gigabits streams of data that need to be transmitted as quickly as possible. Often times the source and the destination are mobile which means wired connections are not a choice. This has led to an ever increasing need to develop wireless access technologies that support high throughput regardless of the transmission environment. Till date, many proprietary solutions exist that seek to bridge this gap with little or no support for interoperability. For the sheer scale of development that is required, a standard based solution is the key. The IEEE 802.1x committee oversees the development of standards for wireless systems, it formed the 802.16 working group to develop a standards-based Wireless Metropolitan Area Network (MAN) solution. One of the fruits of this effort is the 802.16e standard fondly referred to as mobile WiMAX and it is the subject of study in this thesis. This thesis seeks to analyze the transmission characteristics of two of the antenna systems defined in the standard i.e. Adaptive Beamforming Systems and Multiple-Input Multiple-Output Systems. Multiple-Input Multiple-Output (MIMO): utilizes multiple antennas at the transmitter and receiver to provide diversity gain, multiplexing gain or both. Adaptive Antenna Systems (AAS): Adaptive array system uses an antenna array to generate in real-time radiation patterns with the main lobes and/or nulls dynamically tuned to specific directions in order to increase or suppress signal power in that direction.Worldwide Interoperability for Microwave Access (WiMAX) is the acronym for Institute of Electrical and Electronics Engineers (IEEE) 802.16 set of standards governing Air Interface for Fixed Broadband Wireless Access Systems. In the history of wireless systems, WiMAX is revolutionary technology as affords its users the Wi-Fi grade throughput and cellular system level of mobility. With WiMAX, broadband technology (traditionally ADSL and Fiber) goes wireless and WiMAX users can basically enjoy triple-play application, and split-second download and upload rates. WIMAX also offers full mobility much as traditional cellular systems do with features like seamless hand-over and roaming at vehicular speed; this is made possible because the system design covers the access network to core network. For the operator, WiMAX is a welcome development because it merges traditional cellular networks with broadband technology thus opening them to more business offerings and a larger client base and all this at a reduced cost of deployment. Base stations are comparatively cheaper and do not require extensive planning typical of other cellular systems thus WiMAX is aptly suited for emerging markets where infrastructure cost is a major issue; little wonder a lot of 3rd world countries have signified interest in the technology.
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Neville, Katrina Lee, and katrina neville@rmit edu au. "Channel Compensation for Speaker Recognition Systems." RMIT University. Electrical and Computer Engineering, 2007. http://adt.lib.rmit.edu.au/adt/public/adt-VIT20080514.093453.
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This thesis attempts to address the problem of how best to remedy different types of channel distortions on speech when that speech is to be used in automatic speaker recognition and verification systems. Automatic speaker recognition is when a person's voice is analysed by a machine and the person's identity is worked out by the comparison of speech features to a known set of speech features. Automatic speaker verification is when a person claims an identity and the machine determines if that claimed identity is correct or whether that person is an impostor. Channel distortion occurs whenever information is sent electronically through any type of channel whether that channel is a basic wired telephone channel or a wireless channel. The types of distortion that can corrupt the information include time-variant or time-invariant filtering of the information or the addition of 'thermal noise' to the information, both of these types of distortion can cause varying degrees of error in information being received and analysed. The experiments presented in this thesis investigate the effects of channel distortion on the average speaker recognition rates and testing the effectiveness of various channel compensation algorithms designed to mitigate the effects of channel distortion. The speaker recognition system was represented by a basic recognition algorithm consisting of: speech analysis, extraction of feature vectors in the form of the Mel-Cepstral Coefficients, and a classification part based on the minimum distance rule. Two types of channel distortion were investigated: Convolutional (or lowpass filtering) effects Addition of white Gaussian noise Three different methods of channel compensation were tested: Cepstral Mean Subtraction (CMS) RelAtive SpecTrAl (RASTA) Processing Constant Modulus Algorithm (CMA) The results from the experiments showed that for both CMS and RASTA processing that filtering at low cutoff frequencies, (3 or 4 kHz), produced improvements in the average speaker recognition rates compared to speech with no compensation. The levels of improvement due to RASTA processing were higher than the levels achieved due to the CMS method. Neither the CMS or RASTA methods were able to improve accuracy of the speaker recognition system for cutoff frequencies of 5 kHz, 6 kHz or 7 kHz. In the case of noisy speech all methods analysed were able to compensate for high SNR of 40 dB and 30 dB and only RASTA processing was able to compensate and improve the average recognition rate for speech corrupted with a high level of noise (SNR of 20 dB and 10 dB).
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Karunanidhi, Karthikeyan. "ARROS; distributed adaptive real-time network intrusion response." Ohio : Ohio University, 2006. http://www.ohiolink.edu/etd/view.cgi?ohiou1141074467.
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Glass, Robert B. "Mobile Indoor Positioning for Augmented Reality Systems." VCU Scholars Compass, 2014. http://scholarscompass.vcu.edu/etd/3643.
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This thesis explores the creation and setup of a prototype that allows users of the device to interact within an indoor real world environment and a virtual environment simultaneously using high-tech common technology. The prototype is comprised of a small mobile device such as a cellular mobile phone, Raspberry Pi computer, a battery powered handheld Pico projector, and software developed for the Android OS. The software can easily be ported to other mobile and non-mobile operating systems. The mobile device must contain accelerometer, magnetometer, and gyroscope embedded sensors as well as 802.11 wireless network chip. The prototype software implements an indoor positioning system to track the current location and orientation of the prototype device in real time. It also displays a virtual world projection upon the surfaces of the real world in relation to the prototype’s physical location and orientation.
Three different orientation estimation methods were tested and compared in this thesis. Accelerometer and magnetometer based method, gyroscope based method, and a combined method using a technique called sensor fusion were implemented. A multilateration approach was used for location estimation. Location estimates were calculated from the measured received signal strength of multiple 802.11 wireless network access points. The location of all wireless access points were known and fixed. Received signal strength data was converted to meters using a log distance propagation model, and tests were conducted to compare actual distance with converted distance. Tests were also conducted to compare multilateration estimates from unfiltered or raw RSS and filtered RSS data using a Kalman filter.
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Yang, Can. "Adaptive Sensor : Exploring the use of dynamic role allocation based on interesting to detect blood and tumors in a smart pill." Thesis, Högskolan i Halmstad, Akademin för informationsteknologi, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-42626.
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For intelligent systems, the ability to adapt a sensor's sensing capabilities offers promise for reducing numbers, weight, and volume of sensors required. This basic idea is in line with a recent assertion by the well-known roboticist Rodney Brooks, that versatile robots could be used to perform various tasks instead of requiring a large number of specialized robots.In the current work, we consider the concept of a "smart" sensor which could dynamically adapt itself to replace multiple static sensors--within the application area of ingestible smart pills, where small sensors might be required to detect problems such as bleeding or tumours.\\ Simulations were used to evaluate some basic strategies for how to adapt the sensor and their effectiveness was compared; as well, a hardware prototype using LEDs to indicate system switching was prepared.
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Li, Shen Carmen C. Duren Russell Walker. "Evaluating Impulse C and multiple parallelism partitions for a low-cost reconfigurable computing system." Waco, Tex. : Baylor University, 2008. http://hdl.handle.net/2104/5280.
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Sadeghi, Parastoo School of Electrical Engineering And Telecommunications UNSW. "Modelling, information capacity, and estimation of time-varying channels in mobile communication systems." Awarded by:University of New South Wales. School of Electrical Engineering And Telecommunications, 2006. http://handle.unsw.edu.au/1959.4/32310.
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In the first part of this thesis, the information capacity of time-varying fading channels is analysed using finite-state Markov channel (FSMC) models. Both fading channel amplitude and fading channel phase are modelled as finite-state Markov processes. The effect of the number of fading channel gain partitions on the capacity is studied (from 2 to 128 partitions). It is observed that the FSMC capacity is saturated when the number of fading channel gain partitions is larger than 4 to 8 times the number of channel input levels. The rapid FSMC capacity saturation with a small number of fading channel gain partitions can be used for the design of computationally simple receivers, with a negligible loss in the capacity. Furthermore, the effect of fading channel memory order on the capacity is studied (from first- to fourth-order). It is observed that low-order FSMC models can provide higher capacity estimates for fading channels than high-order FSMC models, especially when channel states are poorly observable in the presence of channel noise. To explain the effect of memory order on the FSMC capacity, the capacities of high-order and low-order FSMC models are analytically compared. It is shown that the capacity difference is caused by two factors: 1) the channel entropy difference, and 2) the channel observability difference between the high-order and low-order FSMC models. Due to the existence of the second factor, the capacity of high-order FSMC models can be lower than the capacity of low-order FSMC models. Two sufficient conditions are proven to predict when the low-order FSMC capacity is higher or lower than the high-order FSMC capacity. In the second part of this thesis, a new implicit (blind) channel estimation method in time- varying fading channels is proposed. The information source emits bits ???0??? and ???1??? with unequal probabilities. The unbalanced source distribution is used as a priori known signal structure at the receiver for channel estimation. Compared to pilot-symbol-assisted channel estimation, the proposed channel estimation technique can achieve a superior receiver bit error rate performance, especially at low signal to noise ratio conditions.
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Menezes, Alam Silva. "Avaliação de desempenho de radios cognitivos e proposta de estrutura de equalização temporal em sistemas OFDM." [s.n.], 2007. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259367.
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Orientadores: Jose Marcos Travassos Romano, Cristiano Magalhães PanazioDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
Made available in DSpace on 2018-08-09T08:52:54Z (GMT). No. of bitstreams: 1 Menezes_AlamSilva_M.pdf: 3586178 bytes, checksum: 0b475a352c1a0c8fc6c72a6a3c846f45 (MD5) Previous issue date: 2007
Resumo: A presente dissertação apresenta um estudo da tecnologia dos rádios cognitivos bem como uma avaliação dos possíveis ganhos desta promissora técnica em relação aos atuais meios de acesso ao espectro eletromagnético. Numa segunda frente de trabalho, tratamos do problema de equalização cega no contexto de canais SIMO, com uma única entrada e múltiplas saídas. Propomos a predição linear multicanal como estrutura de equalização em sistemas OFDM e avaliamos por meio de simulações a viabilidade da técnica proposta
Abstract: In this work, we provide a study of the cognitive radio technology and the potential gains that this technique may provide with regard to current electromagnetic spectrum access techniques. In a second workphase, we deal with the problem of blind equalization in the context of single input multiple output channels (SIMO). We propose the use of a multichannel linear prediction structure to equalize OFDM systems and we assess its performance through numerical simulations
Mestrado
Telecomunicações e Telemática
Mestre em Engenharia Elétrica
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Sridharan, M. K. "Subband Adaptive Filtering Algorithms And Applications." Thesis, Indian Institute of Science, 2000. http://hdl.handle.net/2005/266.
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In system identification scenario, the linear approximation of the system modelled by its impulse response, is estimated in real time by gradient type Least Mean Square (LMS) or Recursive Least Squares (RLS) algorithms. In recent applications like acoustic echo cancellation, the order of the impulse response to be estimated is very high, and these traditional approaches are inefficient and real time implementation becomes difficult. Alternatively, the system is modelled by a set of shorter adaptive filters operating in parallel on subsampled signals. This approach, referred to as subband adaptive filtering, is expected to reduce not only the computational complexity but also to improve the convergence rate of the adaptive algorithm. But in practice, different subband adaptive algorithms have to be used to enhance the performance with respect to complexity, convergence rate and processing delay. A single subband adaptive filtering algorithm which outperforms the full band scheme in all applications is yet to be realized.
This thesis is intended to study the subband adaptive filtering techniques and explore the possibilities of better algorithms for performance improvement. Three different subband adaptive algorithms have been proposed and their performance have been verified through simulations. These algorithms have been applied to acoustic echo cancellation and EEG artefact minimization problems.
Details of the work
To start with, the fast FIR filtering scheme introduced by Mou and Duhamel has been generalized. The Perfect Reconstruction Filter Bank (PRFB) is used to model the linear FIR system. The structure offers efficient implementation with reduced arithmetic complexity. By using a PRFB with non adjacent filters non overlapping, many channel filters can be eliminated from the structure. This helps in reducing the complexity of the structure further, but introduces approximation in the model. The modelling error depends on the stop band attenuation of the filters of the PRFB. The error introduced due to approximation is tolerable
for applications like acoustic echo cancellation.
The filtered output of the modified generalized fast filtering structure is given by
(formula)
where, Pk(z) is the main channel output, Pk,, k+1 (z) is the output of auxiliary channel filters at the reduced rate, Gk (z) is the kth synthesis filter and M the number of channels in the PRFB. An adaptation scheme is developed for adapting the main channel filters. Auxiliary channel filters are derived from main channel filters.
Secondly, the aliasing problem of the classical structure is reduced without using the cross filters. Aliasing components in the estimated signal results in very poor steady state performance in the classical structure. Attempts to eliminate the aliasing have reduced the computation gain margin and the convergence rate. Any attempt to estimate the subband reference signals from the aliased subband input signals results in aliasing. The analysis filter Hk(z) having the following antialiasing property
(formula)
can avoid aliasing in the input subband signal. The asymmetry of the frequency response prevents the use of real analysis filters. In the investigation presented in this thesis, complex analysis filters and real'synthesis filters are used in the classical structure, to reduce the aliasing errors and to achieve superior convergence rate.
PRFB is traditionally used in implementing Interpolated FIR (IFIR) structure. These filters may not be ideal for processing an input signal for an adaptive algorithm. As third contribution, the IFIR structure is modified using discrete finite frames. The model of an FIR filter s is given by Fc, with c = Hs. The columns of the matrix F forms a frame with rows of H as its dual frame. The matrix elements can be arbitrary except that the transformation should be implementable as a filter bank. This freedom is used to optimize the filter bank, with the knowledge of the input statistics, for initial convergence rate enhancement .
Next, the proposed subband adaptive algorithms are applied to acoustic echo cancellation problem with realistic parameters. Speech input and sufficiently long Room Impulse Response (RIR) are used in the simulations. The Echo Return Loss Enhancement (ERLE)and the steady state error spectrum are used as performance measures to compare these algorithms with the full band scheme and other representative subband implementations.
Finally, Subband adaptive algorithm is used in minimization of EOG (Electrooculogram) artefacts from measured EEG (Electroencephalogram) signal. An IIR filterbank providing sufficient isolation between the frequency bands is used in the modified IFIR structure and this structure has been employed in the artefact minimization scheme. The estimation error in the high frequency range has been reduced and the output signal to noise ratio has been increased by a couple of dB over that of the fullband scheme.
Conclusions
Efforts to find elegant Subband adaptive filtering algorithms will continue in the future. However, in this thesis, the generalized filtering algorithm could offer gain in filtering complexity of the order of M/2 and reduced misadjustment . The complex classical scheme offered improved convergence rate, reduced misadjustment and computational gains of the order of M/4 . The modifications of the IFIR structure using discrete finite frames made it possible to eliminate the processing delay and enhance the convergence rate. Typical performance of the complex classical case for speech input in a realistic scenario (8 channel case), offers ERLE of more than 45dB. The subband approach to EOG artefact minimization in EEG signal was found to be superior to their fullband counterpart.
(Refer PDF file for Formulas)