Orientadores: Dalton Soares Arantes, Fabbryccio Akkazzha Chaves Machado Cardoso / Tese (doutorado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação / Made available in DSpace on 2018-08-22T22:53:39Z (GMT). No. of bitstreams: 1 Lopes_EstevanMarcelo_D.pdf: 5306029 bytes, checksum: 54cf032312d831b7efe9975406e1a28d (MD5) Previous issue date: 2013 / Resumo: Esta tese propõe o emprego da equalização concorrente, pós- e pré-FFT, em sistemas OFDM. O objetivo é minimizar o uso das subportadoras pilotos e do prefixo cíclico sem que ocorra prejuízo no desempenho do sistema OFDM. A meta com esta iniciativa é aumentar a vazão dos dados e a área de cobertura de transmissão, quando comparado ao sistema convencional de estimação do canal assistido por pilotos e prefixo cíclico suficiente para evitar a interferência entre os símbolos. Duas propostas empregando a equalização concorrente são abordadas. A primeira proposta aplica o conceito da equalização concorrente a um banco de equalizadores no domínio da frequência, denominado de pós-FFT. O algoritmo proposto pode ser considerado semi-cego, porque utiliza a informação das subportadoras pilotos na inicialização, mantendo-se cego durante sua operação. Para suportar a equalização concorrente, o sistema deve prover pilotos apenas no primeiro símbolo de um super quadro, de modo a permitir que o algoritmo se inicialize quando o receptor for ligado ou quando a equalização for perdida. Nos demais símbolos do super quadro as subportadoras pilotos são destinadas ao transporte de informação visando o aumento da vazão do sistema. A outra proposta, realizada no domínio temporal e denominada de pré-FFT, é responsável pela mitigação da interferência entre os símbolos. A vantagem de realizar a equalização pré-FFT é melhorar o desempenho do sistema OFDM quando o espalhamento do canal é maior que o prefixo cíclico. O objetivo é empregar um algoritmo de adaptação para os coeficientes do equalizador pré-FFT, que retropropague o gradiente estocástico do erro medido na frequência para o tempo. Essa estratégia possibilita empregar informações previamente conhecidas sobre a constelação do sinal nas subportadoras, para se minimizar funções de custo associada aos parâmetros do equalizador. Além disso, é proposto um esquema para que os algoritmos CMA e SDD atuem de modo concorrente no processo de equalização / Abstract: This thesis proposes the use of concurrent post- and pre-FFT equalization in OFDM systems. The objective is to improve the OFDM system design by increasing data throughput without performance loss, when compared with pilot based conventional channel estimation techniques and sufficient cyclic prefix system. Two approaches of concurrent equalization are studied. The first exploits the concurrent equalization concept to develop an efficient post-FFT equalizer bank. The algorithm can be considered semi-blind because it uses channel information, obtained from pilot subcarriers, to initialize and to supervise the equalizer bank when pilots are presented, otherwise remaining blind during the equalization process. To support such concurrent equalization, the system should provide pilot subcarriers only in the first symbol of each OFDM super-frame, al-lowing algorithm initialization when the receiver is turned on. In the remaining super-frame symbols, pilot subcarriers can be suppressed to increase the overall system throughput. The second approach proposes a concurrent pre-FFT algorithm to be used with time domain equalizers. This strategy has the capability to mitigate the inter symbol interference and to minimize the cyclic prefix. The objective is to realize this task using an LMS-like algorithm to adapt the pre-FFT equalizer coefficients. The stochastic gradient based on the error signal for each subcarrier in the frequency domain is back-propagated to adapt the filter coefficients in the time domain. This strategy allows the use of a priori information about the subcarrier signal constellation to minimize cost functions such as the CMA and SDD. Moreover, we propose a scheme to use CMA- and SDD-like algorithms in a concurrent mode / Doutorado / Telecomunicações e Telemática / Doutor em Engenharia Elétrica

Orientador: Luís Geraldo Pedroso Meloni / Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação / Made available in DSpace on 2018-08-21T21:07:48Z (GMT). No. of bitstreams: 1 Chanquini_JuliaJacobsenDornelles_M.pdf: 2607975 bytes, checksum: f9b57a1325c9977a5bfd0cdb69a56661 (MD5) Previous issue date: 2012 / Resumo: Este trabalho apresenta a adequação de um codificador de áudio padrão MPEG-4 AAC para aderência à norma brasileira do SBTVD. Também apresenta um estudo dos conceitos envolvidos em codificadores de áudio perceptuais com enfoque no codificador MPEG-4 AAC e também inclui a parte de multiplexação e sincronia do MPEG-4. Para o desenvolvimento do projeto foram estudados alguns códigos abertos de codificadores AAC: FAAD, 3GPP e o código de referência do padrão MPEG-4, especialmente a parte referente ao LATM/LOAS. O decodificador de áudio padrão MPEG-4 AAC que foi modificado para suportar a camada LATM/ LOAS foi o FAAD. Foi calculado o tempo adicional que o decodificador modificado leva para decodificar o áudio com a camada LATM/LOAS, sem ser notado um aumento significativo que não permite a decodificação em tempo real do áudio / Abstract: This work presents an adaptation of a standard MPEG-4 AAC audio coder to conform to the Brazilian digital TV standard SBTVD. It also presents a study of the concepts involved in perceptual audio coders focusing on MPEG-4 AAC and also including the multiplexing and synchronization part of the MPEG-4 standard. To develop this project, open source AAC coders were studied: FAAD, 3GPP and the MPEG-4 reference software code specially the part concerning LATM/LOAS. The AAC audio decoder which was modified to support the LATM / LOAS layer was FAAD. The additional time that the modified decoder needs to decode a sample audio with LATM / LOAS was calculated, and it did not introduce a large enough delay that would restrict real time audio decoding / Mestrado / Telecomunicações e Telemática / Mestra em Engenharia Elétrica

M.Ing. / Please refer to full text to view abstract

Placement is one of the most important steps in physical design for VLSI circuits. For field programmable gate arrays (FPGAs), the placement step determines the location of each logic block. I present novel timing and congestion driven placement algorithms for FPGAs with minimal runtime overhead. By predicting the post-routing timing-critical edges and estimating congestion accurately, this algorithm is able to simultaneously reduce the critical path delay and the minimum number of routing tracks. The core of the algorithm consists of a criticality-history record of connection edges and a congestion map. This approach is applied to the 20 largest Microelectronics Center of North Carolina (MCNC) benchmark circuits. Experimental results show that compared with the state-of-the-art FPGA place and route package, the Versatile Place and Route (VPR) suite, this algorithm yields an average of 8.1% reduction (maximum 30.5%) in the critical path delay and 5% reduction in channel width. Meanwhile, the average runtime of the algorithm is only 2.3X as of VPR.

This thesis investigates the design and implementation of turbo codes over the GNU radio. The turbo codes is a class of iterative channel codes which demonstrates strong capability for error correction. A software defined radio (SDR) is a communication system which can implement different modulation schemes and tune to any frequency band by means of software that can control the programmable hardware. SDR utilizes the general purpose computer to perform certain signal processing techniques. We implement a turbo coding system using the Universal Software Radio Peripheral (USRP), a widely used SDR platform from Ettus. Detail configuration and performance comparison are also provided in this research.

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A digital beacon receiver has been developed to monitor the OLYMPUS satellite beacons. The receiver accepts a nominal 10 MHz IF input and processes the signal using digital signal processing techniques. Fast Fourier transforms are used to locate the carrier within 0.5 Hz. The outputs of the receiver include the frequency and the power of the carrier. / Master of Science

Today digital signal processing techniques are employed in a variety of applications. Two factors contributing to the growth in the use of digital signal processing (DSP) are the advent of custom VLSI that has made using digital signal processing techniques to solve real time problems more attractive and powerful; and the ease and flexibility of application of digital signal processing technique both in hardware and software. The purpose of this paper is to present the design of a digital signal processor chip based on a consideration of VLSI technology and signal processing requirement for radar applications. The paper reviews basic signal processing tasks , giving emphasis to the digital filters and spectral analysis which are generally the required functions in radar signal processing. That leads to the discussion of two DSP algorithms Discret Fourier Transform and Fast Fourier Transform. The basic hardware components required are described along with the software to implement the DSP algorithms. Finally, an example demonstrates the use of processor chip to perform transversal filter function.

Thesis (M. Phil. (Music Technology)) -- University of Stellenbosch, 2006. / ENGLISH ABSTRACT: Modern digital signal processing, allowing a much greater degree of flexibility in audio processing and therefore greater potential for noise removal, pitch correction, filtering and editing, has allowed transfer and audio restoration engineers a diversity of ways in which to “improve” or “reinterpret” (in some cases even drastically altering) the original sound of recordings. This has lead to contrasting views regarding the role of the remastering engineer, the nature and purpose of audio restoration and the ethical implications of the restoration process. The influence of audio restoration on the recorded legacy of a performing artist is clearly illustrated in the case of Maria Callas (1923 - 1977), widely regarded not only as one of the most influential and prolific of opera singers, but also one of the greatest classical musicians of all time. EMI, for whom Callas recorded almost exclusively from 1953 - 1969, has reissued her recordings repeatedly, continually adapting their sound “to the perceived preferences of the record-buying public” (Seletsky 2000: 240). Their attempts at improving the sound of Callas’s recordings to meet with the sonic quality expected of modern recordings, as reissued in the latest releases that form part of EMI’s Callas Edition, Great Recordings of the Century (GROTC) and Historical Series, have resulted in often staggeringly different reinterpretations of the same audio material that bear no resemblance to previous CD or LP incarnations or “evince no consolidated conviction about exactly how Callas’s voice should sound.” In essence, some commentators argue that the “Callas sound” we hear on recent CD releases is not necessarily exactly as the great diva might have sounded. The purpose of this study is to consider the influence of audio restoration and remastering techniques on the recorded legacy of Callas, by illustrating the sometimes startlingly different ways in which her voice has been made to sound, examining and comparing the way in which different remasterings of the same audio material can vary in quality, as well as demonstrating how vastly different sonic reinterpretations of a single recording can affect our perception of an artist’s “true” sound. To this end, various reissues of six different complete opera recordings, including four studio recordings: Tosca (1953), Lucia di Lammermoor (1953), Norma (1954), Madama Butterfly (1955), as well as two “live” performances of Macbeth (1953) and La Traviata (1958), have been evaluated and compared, using the “true” sound of Callas’s voice as reference in comparing the different remasterings. Pitch and frequency spectrum analysis was used to confirm or support any subjective claims and observations and further analysis performed with the aid of a specialised Matlab algorithm. / AFRIKAANSE OPSOMMING: Moderne digitale seinprossesering bied kragtige en veelsydige moontlikhede vir die verwerking van klankseine. Die groter potensiaal vir ruisverwydering, toonhoogte verstelling, filtrering en redigering van opnames bied klankingenieurs ‘n wye verskeidenheid van maniere om die oorspronklike klank van opnames te verbeter, te interpreteer en soms ingrypend te verander. Dit het aanleiding gegee tot teenstrydige en uiteenlopende menings oor die funksie van die klankrestourasie-ingenieur, die aard en doel van klankrestourasie en die etiese gevolge van die restourasieproses. Die invloed van klankrestourasie op die klanknalatenskap van ‘n uitvoerende kunstenaar kan duidelik bestudeer word in die geval van Maria Callas (1923 – 1977), algemeen aanvaar as een van die mees invloedryke en grootse klassieke musici van alle tye. Die platemaatskappy EMI, vir wie Callas feitlik uitsluitlik vanaf 1953 tot 1969 opgeneem het, het haar klankopnames reeds verskeie kere heruitgereik en die klank daarvan deurlopend aangepas om aanklank te vind by die “veronderstelde voorkeure van die publiek” (Seletsky 2000: 240). EMI se pogings om die klank van Callas se opnames te verbeter om aan die klankvereistes van moderne opnames te voldoen, het ontaard in dikwels aangrypend verskillende interpretasies van dieselfde audio materiaal wat geen ooreenkomste toon met vorige laserskyf of langspeelplaat uitgawes nie, asook “geen vasgestelde oortuigings openbaar oor hoe Callas se stem presies moet klink nie.” Sommige critici argumenteer dat die “Callas klank” wat ons op hedendaagse CD uitgawes hoor, nie noodwendig klink soos wat Callas werklik geklink het nie. Die doel van hierdie studie is om die invloed van klankrestourasie op die klanknalatenskap van Callas te bestudeer deur die verskillende wyses waarop die klank van haar stem aangepas is te illustreer, die verskille in klankkwaliteit tussen verskillende uitgawes van dieselfde materiaal te ondersoek en te vergelyk, asook te demonstreer hoe uiteenlopend verskillende interpretasies van ‘n enkele opname die persepsie van ‘n kunstenaar se “ware” klank kan affekteer. Vir hierdie doel is verkeie uitgawes van ses verskillende volledige opera opnames, insluitend vier studio opnames van onderskeidelik Tosca (1953), Lucia di Lammermoor (1953), Norma (1954) en Madama Butterfly (1955), asook twee “lewendige” opnames van Macbeth (1952) en La Traviata (1958) bestudeer deur Callas se “ware” klank as maatstaf te gebruik om die onderskeie opnames te vergelyk. Toonhoogte- en frekwensie spektrum analise, asook analise deur middel van ‘n gespesialiseerde Matlab algoritme, is deurlopend gebruik om enige subjektiewe gevolgtrekkings en waarnemings te staaf.