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

Space Time Processing for Third Generation CDMA Systems

Alam, Fakhrul 25 November 2002 (has links)
The capacity of a cellular system is limited by two different phenomena, namely multipath fading and multiple access interference (MAI). A Two Dimensional (2-D) receiver combats both of these by processing the signal both in the spatial and temporal domain. An ideal 2-D receiver would perform joint space-time processing, but at the price of high computational complexity. In this dissertation we investigate computationally simpler technique termed as a Beamformer-Rake. In a Beamformer-Rake, the output of a beamformer is fed into a succeeding temporal processor to take advantage of both the beamformer and Rake receiver. Wireless service providers throughout the world are working to introduce the third generation (3G) cellular service that will provide higher data rates and better spectral efficiency. Wideband CDMA (WCDMA) has been widely accepted as one of the air interfaces for 3G. A Beamformer-Rake receiver can be an effective solution to provide the receivers enhanced capabilities needed to achieve the required performance of a WCDMA system. This dissertation investigates different Beamformer-Rake receiver structures suitable for the WCDMA system and compares their performance under different operating conditions. This work develops Beamformer-Rake receivers for WCDMA uplink that employ Eigen-Beamforming techniques based on the Maximum Signal to Noise Ratio (MSNR) and Maximum Signal to Interference and Noise Ratio (MSINR) criteria. Both the structures employ Maximal Ratio Combining (MRC) to exploit temporal diversity. MSNR based Eigen-Beamforming leads to a Simple Eigenvalue problem (SE). This work investigates several algorithms that can be employed to solve the SE and compare the algorithms in terms of their computational complexity and their performance. MSINR based Eigen-Beamforming results in a Generalized Eigenvalue problem (GE). The dissertation describes several techniques to form the GE and algorithms to solve it. We propose a new low-complexity algorithm, termed as the Adaptive Matrix Inversion (AMI), to solve the GE. We compare the performance of the AMI to other existing algorithms. Comparison between different techniques to form the GE is also compared. The MSINR based beamforming is demonstrated to be superior to the MSNR based beamforming in the presence of strong interference. There are Pilot Symbol Assisted (PSA) beamforming techniques that exploit the Minimum Mean Squared Error (MMSE) criterion. We compare the MSINR based Beamformer-Rake with the same that utilizes Direct Matrix Inversion (DMI) to perform MMSE based beamforming in terms of Bit Error Rate (BER). In a wireless system where the number of co-channel interferers is larger than the number of elements of a practical antenna array, we can not perform explicit null-steering. As a result the advantage of beamforming is partially lost. In this scenario it is better to attain diversity gain at the cost of spatial aliasing. We demonstrate this with the aid of simulation. Orthogonal Frequency Division Multiplexing (OFDM) is a multi-carrier technique that has recently received considerable attention for high speed wireless communication. OFDM has been accepted as the standard for Digital Audio Broadcast (DAB) and Digital Video Broadcast (DVB) in Europe. It has also been established as one of the modulation formats for the IEEE 802.11a wireless LAN standard. OFDM has emerged as one of the primary candidates for the Fourth Generation (4G) wireless communication systems and high speed ad hoc wireless networks. We propose a simple pilot symbol assisted frequency domain beamforming technique for OFDM receiver and demonstrate the concept of sub-band beamforming. Vector channel models measured with the MPRG Viper test-bed is also employed to investigate the performance of the beamforming scheme. / Ph. D.
2

Algoritmos array para filtragem de sistemas lineares / Array algorithms for filtering of linear systems

Jesus, Gildson Queiroz de 06 June 2007 (has links)
Esta dissertação desenvolve filtro de informação, algoritmos array para estimador do erro médio mínimo quadrático para sistemas lineares sujeitos a saltos Markovianos e algoritmos array rápidos para filtragem de sistemas singulares convencionais. Exemplos numéricos serão apresentados para mostrarem as vantagens dos algoritmos array deduzidos. Parte dos resultados obtidos nesta pesquisa serão publicados no seguinte artigo: Terra et al. (2007). Terra, M. H., Ishihara, J. Y. and Jesus, G. Q. (2007). Information filtering and array algorithms for discrete-time Markovian jump linear systems. Proceedings of the American Control Conference ACC07. / This dissertation develops information filter and array algorithms for linear minimum mean square error estimator (LMMSE) of discrete-time Markovian jump linear systems (MJLSs) and fast array algorithms for filtering of standard singular systems. Numerical examples to show the advantage of the array algorithms are presented. Some results obtained in this research are published in the following paper: Terra et al. (2007). Terra, M. H., Ishihara, J. Y. and Jesus, G. Q. (2007). Information filtering and array algorithms for discrete-time Markovian jump linear systems. Proceedings of the American Control Conference ACC07.
3

Algoritmos array para filtragem de sistemas lineares / Array algorithms for filtering of linear systems

Gildson Queiroz de Jesus 06 June 2007 (has links)
Esta dissertação desenvolve filtro de informação, algoritmos array para estimador do erro médio mínimo quadrático para sistemas lineares sujeitos a saltos Markovianos e algoritmos array rápidos para filtragem de sistemas singulares convencionais. Exemplos numéricos serão apresentados para mostrarem as vantagens dos algoritmos array deduzidos. Parte dos resultados obtidos nesta pesquisa serão publicados no seguinte artigo: Terra et al. (2007). Terra, M. H., Ishihara, J. Y. and Jesus, G. Q. (2007). Information filtering and array algorithms for discrete-time Markovian jump linear systems. Proceedings of the American Control Conference ACC07. / This dissertation develops information filter and array algorithms for linear minimum mean square error estimator (LMMSE) of discrete-time Markovian jump linear systems (MJLSs) and fast array algorithms for filtering of standard singular systems. Numerical examples to show the advantage of the array algorithms are presented. Some results obtained in this research are published in the following paper: Terra et al. (2007). Terra, M. H., Ishihara, J. Y. and Jesus, G. Q. (2007). Information filtering and array algorithms for discrete-time Markovian jump linear systems. Proceedings of the American Control Conference ACC07.
4

Acoustic Beamforming : Design and Development of Steered Response Power With Phase Transformation (SRP-PHAT). / Acoustic Beamforming : Design and Development of Steered Response Power With Phase Transformation (SRP-PHAT).

Dey, Ajoy Kumar, Saha, Susmita January 2011 (has links)
Acoustic Sound Source localization using signal processing is required in order to estimate the direction from where a particular acoustic source signal is coming and it is also important in order to find a soluation for hands free communication. Video conferencing, hand free communications are different applications requiring acoustic sound source localization. This applications need a robust algorithm which can reliably localize and position the acoustic sound sources. The Steered Response Power Phase Transform (SRP-PHAT) is an important and roubst algorithm to localilze acoustic sound sources. However, the algorithm has a high computational complexity thus making the algorithm unsuitable for real time applications. This thesis focuses on describe the implementation of the SRP-PHAT algorithm as a function of source type, reverberation levels and ambient noise. The main objective of this thesis is to present different approaches of the SRP-PHAT to verify the algorithm in terms of acoustic enviroment, microphone array configuration, acoustic source position and levels of reverberation and noise.

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