846 resultados para Bit error rate


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El vertiginoso avance de la informática y las telecomunicaciones en las últimas décadas ha incidido invariablemente en la producción y la prestación de servicios, en la educación, en la industria, en la medicina, en las comunicaciones e inclusive en las relaciones interpersonales. No obstante estos avances, y a pesar de la creciente aportación del software al mundo actual, durante su desarrollo continuamente se incurre en el mismo tipo de problemas que provocan un retraso sistemático en los plazos de entrega, se exceda en presupuesto, se entregue con una alta tasa de errores y su utilidad sea inferior a la esperada. En gran medida, esta problemática es atribuible a defectos en los procesos utilizados para recoger, documentar, acordar y modificar los requisitos del sistema. Los requisitos son los cimientos sobre los cuáles se construye un producto software, y sin embargo, la incapacidad de gestionar sus cambios es una de las principales causas por las que un producto software se entrega fuera de tiempo, se exceda en coste y no cumpla con la calidad esperada por el cliente. El presente trabajo de investigación ha identificado la necesidad de contar con metodologías que ayuden a desplegar un proceso de Gestión de Requisitos en pequeños grupos y entornos de trabajo o en pequeñas y medianas empresas. Para efectos de esta tesis llamaremos Small-Settings a este tipo de organizaciones. El objetivo de este trabajo de tesis doctoral es desarrollar un metamodelo que permita, por un lado, la implementación y despliegue del proceso de Gestión de Requisitos de forma natural y a bajo coste y, por otro lado, el desarrollo de mecanismos para la mejora continua del mismo. Este metamodelo esta soportado por el desarrollo herramientas que permiten mantener una biblioteca de activos de proceso para la Gestión de Requisitos y a su vez contar con plantillas para implementar el proceso partiendo del uso de activos previamente definidos. El metamodelo contempla el desarrollo de prácticas y actividades para guiar, paso a paso, la implementación del proceso de Gestión de Requisitos para una Small-Setting utilizando un modelo de procesos como referencia y una biblioteca de activos de proceso como principal herramienta de apoyo. El mantener los activos de proceso bien organizados, indexados, y fácilmente asequibles, facilita la introducción de las mejores prácticas al interior de una organización. ABSTRACT The fast growth of computer science and telecommunication in recent decades has invariably affected the provision of products and services in education, industry, healthcare, communications and also interpersonal relationships. In spite of such progress and the active role of the software in the world, its development and production continually incurs in the same type of problems that cause systematic delivery delays, over budget, a high error rate and consequently its use is lower than expected. These problems are largely attributed to defects in the processes used to identify, document, organize, and track all system's requirements. It is generally accepted that requirements are the foundation upon which the software process is built, however, the inability to manage changes in requirements is one of the principal factors that contribute to delays on the software development process, which in turn, may cause customer dissatisfaction. The aim of the present research work has identified the need for appropriate methodologies to help on the requirement management process for those organizations that are categorised as small and medium size enterprises, small groups within large companies, or small projects. For the purposes of this work, these organizations are named Small-Settings. The main goal of this research work is to develop a metamodel to manage the requirement process using a Process Asset Library (PAL) and to provide predefined tools and actives to help on the implementation process. The metamodel includes the development of practices and activities to guide step by step the deployment of the requirement management process in Small-Settings. Keeping assets organized, indexed, and readily available are a main factor to the success of the organization process improvement effort and facilitate the introduction of best practices within the organization. The Process Asset Library (PAL) will become a repository of information used to keep and make available all process assets that are useful to those who are defining, implementing, and managing processes in the organization.

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La presente Tesis analiza las posibilidades que ofrecen en la actualidad las tecnologías del habla para la detección de patologías clínicas asociadas a la vía aérea superior. El estudio del habla que tradicionalmente cubre tanto la producción como el proceso de transformación del mensaje y las señales involucradas, desde el emisor hasta alcanzar al receptor, ofrece una vía de estudio alternativa para estas patologías. El hecho de que la señal emitida no solo contiene este mensaje, sino también información acerca del locutor, ha motivado el desarrollo de sistemas orientados a la identificación y verificación de la identidad de los locutores. Estos trabajos han recibido recientemente un nuevo impulso, orientándose tanto hacia la caracterización de rasgos que son comunes a varios locutores, como a las diferencias existentes entre grabaciones de un mismo locutor. Los primeros resultan especialmente relevantes para esta Tesis dado que estos rasgos podrían evidenciar la presencia de características relacionadas con una cierta condición común a varios locutores, independiente de su identidad. Tal es el caso que se enfrenta en esta Tesis, donde los rasgos identificados se relacionarían con una de la patología particular y directamente vinculada con el sistema de físico de conformación del habla. El caso del Síndrome de Apneas Hipopneas durante el Sueno (SAHS) resulta paradigmático. Se trata de una patología con una elevada prevalencia mundo, que aumenta con la edad. Los pacientes de esta patología experimentan episodios de cese involuntario de la respiración durante el sueño, que se prolongan durante varios segundos y que se reproducen a lo largo de la noche impidiendo el correcto descanso. En el caso de la apnea obstructiva, estos episodios se deben a la imposibilidad de mantener un camino abierto a través de la vía aérea, de forma que el flujo de aire se ve interrumpido. En la actualidad, el diagnostico de estos pacientes se realiza a través de un estudio polisomnográfico, que se centra en el análisis de los episodios de apnea durante el sueño, requiriendo que el paciente permanezca en el hospital durante una noche. La complejidad y el elevado coste de estos procedimientos, unidos a las crecientes listas de espera, han evidenciado la necesidad de contar con técnicas rápidas de detección, que si bien podrían no obtener tasas tan elevadas, permitirían reorganizar las listas de espera en función del grado de severidad de la patología en cada paciente. Entre otros, los sistemas de diagnostico por imagen, así como la caracterización antropométrica de los pacientes, han evidenciado la existencia de patrones anatómicos que tendrían influencia directa sobre el habla. Los trabajos dedicados al estudio del SAHS en lo relativo a como esta afecta al habla han sido escasos y algunos de ellos incluso contradictorios. Sin embargo, desde finales de la década de 1980 se conoce la existencia de patrones específicos relativos a la articulación, la fonación y la resonancia. Sin embargo, su descripción resultaba difícilmente aprovechable a través de un sistema de reconocimiento automático, pero apuntaba la existencia de un nexo entre voz y SAHS. En los últimos anos las técnicas de procesado automático han permitido el desarrollo de sistemas automáticos que ya son capaces de identificar diferencias significativas en el habla de los pacientes del SAHS, y que los distinguen de los locutores sanos. Por contra, poco se conoce acerca de la conexión entre estos nuevos resultados, los sé que habían obtenido en el pasado y la patogénesis del SAHS. Esta Tesis continua la labor desarrollada en este ámbito considerando específicamente: el estudio de la forma en que el SAHS afecta el habla de los pacientes, la mejora en las tasas de clasificación automática y la combinación de la información obtenida con los predictores utilizados por los especialistas clínicos en sus evaluaciones preliminares. Las dos primeras tareas plantean problemas simbióticos, pero diferentes. Mientras el estudio de la conexión entre el SAHS y el habla requiere de modelos acotados que puedan ser interpretados con facilidad, los sistemas de reconocimiento se sirven de un elevado número de dimensiones para la caracterización y posterior identificación de patrones. Así, la primera tarea debe permitirnos avanzar en la segunda, al igual que la incorporación de los predictores utilizados por los especialistas clínicos. La Tesis aborda el estudio tanto del habla continua como del habla sostenida, con el fin de aprovechar las sinergias y diferencias existentes entre ambas. En el análisis del habla continua se tomo como punto de partida un esquema que ya fue evaluado con anterioridad, y sobre el cual se ha tratado la evaluación y optimización de la representación del habla, así como la caracterización de los patrones específicos asociados al SAHS. Ello ha evidenciado la conexión entre el SAHS y los elementos fundamentales de la señal de voz: los formantes. Los resultados obtenidos demuestran que el éxito de estos sistemas se debe, fundamentalmente, a la capacidad de estas representaciones para describir dichas componentes, obviando las dimensiones ruidosas o con poca capacidad discriminativa. El esquema resultante ofrece una tasa de error por debajo del 18%, sirviéndose de clasificadores notablemente menos complejos que los descritos en el estado del arte y de una única grabación de voz de corta duración. En relación a la conexión entre el SAHS y los patrones observados, fue necesario considerar las diferencias inter- e intra-grupo, centrándonos en la articulación característica del locutor, sustituyendo los complejos modelos de clasificación por el estudio de los promedios espectrales. El resultado apunta con claridad hacia ciertas regiones del eje de frecuencias, sugiriendo la existencia de un estrechamiento sistemático en la sección del tracto en la región de la orofaringe, ya prevista en la patogénesis de este síndrome. En cuanto al habla sostenida, se han reproducido los estudios realizados sobre el habla continua en grabaciones de la vocal /a/ sostenida. Los resultados son cualitativamente análogos a los anteriores, si bien en este caso las tasas de clasificación resultan ser más bajas. Con el objetivo de identificar el sentido de este resultado se reprodujo el estudio de los promedios espectrales y de la variabilidad inter e intra-grupo. Ambos estudios mostraron importantes diferencias con los anteriores que podrían explicar estos resultados. Sin embargo, el habla sostenida ofrece otras oportunidades al establecer un entorno controlado para el estudio de la fonación, que también había sido identificada como una fuente de información para la detección del SAHS. De su estudio se pudo observar que, en el conjunto de datos disponibles, no existen variaciones que pudieran asociarse fácilmente con la fonación. Únicamente aquellas dimensiones que describen la distribución de energía a lo largo del eje de frecuencia evidenciaron diferencias significativas, apuntando, una vez más, en la dirección de las resonancias espectrales. Analizados los resultados anteriores, la Tesis afronta la fusión de ambas fuentes de información en un único sistema de clasificación. Con ello es posible mejorar las tasas de clasificación, bajo la hipótesis de que la información presente en el habla continua y el habla sostenida es fundamentalmente distinta. Esta tarea se realizo a través de un sencillo esquema de fusión que obtuvo un 88.6% de aciertos en clasificación (tasa de error del 11.4%), lo que representa una mejora significativa respecto al estado del arte. Finalmente, la combinación de este clasificador con los predictores utilizados por los especialistas clínicos ofreció una tasa del 91.3% (tasa de error de 8.7%), que se encuentra dentro del margen ofrecido por esquemas más costosos e intrusivos, y que a diferencia del propuesto, no pueden ser utilizados en la evaluación previa de los pacientes. Con todo, la Tesis ofrece una visión clara sobre la relación entre el SAHS y el habla, evidenciando el grado de madurez alcanzado por la tecnología del habla en la caracterización y detección del SAHS, poniendo de manifiesto que su uso para la evaluación de los pacientes ya sería posible, y dejando la puerta abierta a futuras investigaciones que continúen el trabajo aquí iniciado. ABSTRACT This Thesis explores the potential of speech technologies for the detection of clinical disorders connected to the upper airway. The study of speech traditionally covers both the production process and post processing of the signals involved, from the speaker up to the listener, offering an alternative path to study these pathologies. The fact that utterances embed not just the encoded message but also information about the speaker, has motivated the development of automatic systems oriented to the identification and verificaton the speaker’s identity. These have recently been boosted and reoriented either towards the characterization of traits that are common to several speakers, or to the differences between records of the same speaker collected under different conditions. The first are particularly relevant to this Thesis as these patterns could reveal the presence of features that are related to a common condition shared among different speakers, regardless of their identity. Such is the case faced in this Thesis, where the traits identified would relate to a particular pathology, directly connected to the speech production system. The Obstructive Sleep Apnea syndrome (OSA) is a paradigmatic case for analysis. It is a disorder with high prevalence among adults and affecting a larger number of them as they grow older. Patients suffering from this disorder experience episodes of involuntary cessation of breath during sleep that may last a few seconds and reproduce throughout the night, preventing proper rest. In the case of obstructive apnea, these episodes are related to the collapse of the pharynx, which interrupts the air flow. Currently, OSA diagnosis is done through a polysomnographic study, which focuses on the analysis of apnea episodes during sleep, requiring the patient to stay at the hospital for the whole night. The complexity and high cost of the procedures involved, combined with the waiting lists, have evidenced the need for screening techniques, which perhaps would not achieve outstanding performance rates but would allow clinicians to reorganize these lists ranking patients according to the severity of their condition. Among others, imaging diagnosis and anthropometric characterization of patients have evidenced the existence of anatomical patterns related to OSA that have direct influence on speech. Contributions devoted to the study of how this disorder affects scpeech are scarce and somehow contradictory. However, since the late 1980s the existence of specific patterns related to articulation, phonation and resonance is known. By that time these descriptions were virtually useless when coming to the development of an automatic system, but pointed out the existence of a link between speech and OSA. In recent years automatic processing techniques have evolved and are now able to identify significant differences in the speech of OSAS patients when compared to records from healthy subjects. Nevertheless, little is known about the connection between these new results with those published in the past and the pathogenesis of the OSA syndrome. This Thesis is aimed to progress beyond the previous research done in this area by addressing: the study of how OSA affects patients’ speech, the enhancement of automatic OSA classification based on speech analysis, and its integration with the information embedded in the predictors generally used by clinicians in preliminary patients’ examination. The first two tasks, though may appear symbiotic at first, are quite different. While studying the connection between speech and OSA requires simple narrow models that can be easily interpreted, classification requires larger models including a large number dimensions for the characterization and posterior identification of the observed patterns. Anyhow, it is clear that any progress made in the first task should allow us to improve our performance on the second one, and that the incorporation of the predictors used by clinicians shall contribute in this same direction. The Thesis considers both continuous and sustained speech analysis, to exploit the synergies and differences between them. On continuous speech analysis, a conventional speech processing scheme, designed and evaluated before this Thesis, was taken as a baseline. Over this initial system several alternative representations of the speech information were proposed, optimized and tested to select those more suitable for the characterization of OSA-specific patterns. Evidences were found on the existence of a connection between OSA and the fundamental constituents of the speech: the formants. Experimental results proved that the success of the proposed solution is well explained by the ability of speech representations to describe these specific OSA-related components, ignoring the noisy ones as well those presenting low discrimination capabilities. The resulting scheme obtained a 18% error rate, on a classification scheme significantly less complex than those described in the literature and operating on a single speech record. Regarding the connection between OSA and the observed patterns, it was necessary to consider inter-and intra-group differences for this analysis, and to focus on the articulation, replacing the complex classification models by the long-term average spectra. Results clearly point to certain regions on the frequency axis, suggesting the existence of a systematic narrowing in the vocal tract section at the oropharynx. This was already described in the pathogenesis of this syndrome. Regarding sustained speech, similar experiments as those conducted on continuous speech were reproduced on sustained phonations of vowel / a /. Results were qualitatively similar to the previous ones, though in this case perfomance rates were found to be noticeably lower. Trying to derive further knowledge from this result, experiments on the long-term average spectra and intraand inter-group variability ratios were also reproduced on sustained speech records. Results on both experiments showed significant differences from the previous ones obtained from continuous speech which could explain the differences observed on peformance. However, sustained speech also provided the opportunity to study phonation within the controlled framework it provides. This was also identified in the literature as a source of information for the detection of OSA. In this study it was found that, for the available dataset, no sistematic differences related to phonation could be found between the two groups of speakers. Only those dimensions which relate energy distribution along the frequency axis provided significant differences, pointing once again towards the direction of resonant components. Once classification schemes on both continuous and sustained speech were developed, the Thesis addressed their combination into a single classification system. Under the assumption that the information in continuous and sustained speech is fundamentally different, it should be possible to successfully merge the two of them. This was tested through a simple fusion scheme which obtained a 88.6% correct classification (11.4% error rate), which represents a significant improvement over the state of the art. Finally, the combination of this classifier with the variables used by clinicians obtained a 91.3% accuracy (8.7% error rate). This is within the range of alternative, but costly and intrusive schemes, which unlike the one proposed can not be used in the preliminary assessment of patients’ condition. In the end, this Thesis has shed new light on the underlying connection between OSA and speech, and evidenced the degree of maturity reached by speech technology on OSA characterization and detection, leaving the door open for future research which shall continue in the multiple directions that have been pointed out and left as future work.

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Los sistemas basados en la técnica OFDM (Multiplexación por División de Frecuencias Ortogonales) son una evolución de los tradicionales sistemas FDM (Multiplexación por División de Frecuencia), gracias a la cual se consigue un mejor aprovechamiento del ancho de banda. En la actualidad los sistemas OFDM y sus variantes ocupan un lugar muy importante en las comunicaciones, estando implementados en diversos estándares como pueden ser: DVB-T (estándar de la TDT), ADSL, LTE, WIMAX, DAB (radio digital), etc. Debido a ello, en este proyecto se implementa un sistema OFDM en el que poder realizar diversas simulaciones para entender mejor su funcionamiento. Para ello nos vamos a valer de la herramienta Matlab. Los objetivos fundamentales dentro de la simulación del sistema es poner a prueba el empleo de turbo códigos (comparándolo con los códigos convolucionales tradicionales) y de un ecualizador. Todo ello con la intención de mejorar la calidad de nuestro sistema (recibir menos bits erróneos) en condiciones cada vez más adversas: relaciones señal a ruido bajas y multitrayectos. Para ello se han implementado las funciones necesarias en Matlab, así como una interfaz gráfica para que sea más sencillo de utilizar el programa y más didáctico. En los capítulos segundo y tercero de este proyecto se efectúa un estudio de las bases de los sistemas OFDM. En el segundo nos centramos más en un estudio teórico puro para después pasar en el tercero a centrarnos únicamente en la teoría de los bloques implementados en el sistema OFDM que se desarrolla en este proyecto. En el capítulo cuarto se explican las distintas opciones que se pueden llevar a cabo mediante la interfaz implementada, a la vez que se elabora un manual para el correcto uso de la misma. El quinto capítulo se divide en dos partes, en la primera se muestran las representaciones que puede realizar el programa, y en la segunda únicamente se realizan simulaciones para comprobar que tal responde nuestra sistema a distintas configuraciones de canal, y las a distintas configuraciones que hagamos nosotros de nuestro sistema (utilicemos una codificación u otra, utilicemos el ecualizador o el prefijo cíclico, etc…). Para finalizar, en el último capítulo se exponen las conclusiones obtenidas en este proyecto, así como posibles líneas de trabajo que seguir en próximas versiones del mismo. ABSTRACT. Systems based on OFDM (Orthogonal Frequency Division Multiplexing) technique are an evolution of traditional FDM (Frequency Division Multiplexing). Due to the use of OFDM systems are achieved by more efficient use of bandwidth. Nowadays, OFDM systems and variants of OFDM systems occupy a very important place in the world of communications, being implemented in standards such as DVB-T, ADSL, LTE, WiMAX, DAB (digital radio) and another more. For all these reasons, this project implements a OFDM system for performing various simulations for better understanding of OFDM system operation. The system has been simulated using Matlab. With system simulation we search to get two key objectives: to test the use of turbo codes (compared to traditional convolutional codes) and an equalizer. We do so with the intention of improving the quality of our system (receive fewer rates of bit error) in increasingly adverse conditions: lower signal-to-noise and multipath. For these reasons necessaries Matlab´s functions have been developed, and a GUI (User Graphical Interface) has been integrated so the program can be used in a easier and more didactic way. This project is divided into five chapters. In the second and third chapter of this project are developed the basis of OFDM systems. Being developed in the second one a pure theoretical study, while focusing only on block theory implemented in the OFDM system in the third one. The fourth chapter describes the options that can be carried out by the interface implemented. Furthermore the chapter is developed for the correct use of the interface. The fifth chapter is divided into two parts, the first part shows to us the representations that the program can perform, and the second one just makes simulations to check that our system responds to differents channel configurations (use of convolutional codes or turbo codes, the use of equalizer or cyclic prefix…). Finally, the last chapter presents the conclusions of this project and possible lines of work to follow in future versions.

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This paper describes the application of language translation technologies for generating bus information in Spanish Sign Language (LSE: Lengua de Signos Española). In this work, two main systems have been developed: the first for translating text messages from information panels and the second for translating spoken Spanish into natural conversations at the information point of the bus company. Both systems are made up of a natural language translator (for converting a word sentence into a sequence of LSE signs), and a 3D avatar animation module (for playing back the signs). For the natural language translator, two technological approaches have been analyzed and integrated: an example-based strategy and a statistical translator. When translating spoken utterances, it is also necessary to incorporate a speech recognizer for decoding the spoken utterance into a word sequence, prior to the language translation module. This paper includes a detailed description of the field evaluation carried out in this domain. This evaluation has been carried out at the customer information office in Madrid involving both real bus company employees and deaf people. The evaluation includes objective measurements from the system and information from questionnaires. In the field evaluation, the whole translation presents an SER (Sign Error Rate) of less than 10% and a BLEU greater than 90%.

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Automated Teller Machines (ATMs) are sensitive self-service systems that require important investments in security and testing. ATM certifications are testing processes for machines that integrate software components from different vendors and are performed before their deployment for public use. This project was originated from the need of optimization of the certification process in an ATM manufacturing company. The process identifies compatibility problems between software components through testing. It is composed by a huge number of manual user tasks that makes the process very expensive and error-prone. Moreover, it is not possible to fully automate the process as it requires human intervention for manipulating ATM peripherals. This project presented important challenges for the development team. First, this is a critical process, as all the ATM operations rely on the software under test. Second, the context of use of ATMs applications is vastly different from ordinary software. Third, ATMs’ useful lifetime is beyond 15 years and both new and old models need to be supported. Fourth, the know-how for efficient testing depends on each specialist and it is not explicitly documented. Fifth, the huge number of tests and their importance implies the need for user efficiency and accuracy. All these factors led us conclude that besides the technical challenges, the usability of the intended software solution was critical for the project success. This business context is the motivation of this Master Thesis project. Our proposal focused in the development process applied. By combining user-centered design (UCD) with agile development we ensured both the high priority of usability and the early mitigation of software development risks caused by all the technology constraints. We performed 23 development iterations and finally we were able to provide a working solution on time according to users’ expectations. The evaluation of the project was carried out through usability tests, where 4 real users participated in different tests in the real context of use. The results were positive, according to different metrics: error rate, efficiency, effectiveness, and user satisfaction. We discuss the problems found, the benefits and the lessons learned in the process. Finally, we measured the expected project benefits by comparing the effort required by the current and the new process (once the new software tool is adopted). The savings corresponded to 40% less effort (man-hours) per certification. Future work includes additional evaluation of product usability in a real scenario (with customers) and the measuring of benefits in terms of quality improvement.

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A methodology for developing an advanced communications system for the Deaf in a new domain is presented in this paper. This methodology is a user-centred design approach consisting of four main steps: requirement analysis, parallel corpus generation, technology adaptation to the new domain, and finally, system evaluation. During the requirement analysis, both the user and technical requirements are evaluated and defined. For generating the parallel corpus, it is necessary to collect Spanish sentences in the new domain and translate them into LSE (Lengua de Signos Española: Spanish Sign Language). LSE is represented by glosses and using video recordings. This corpus is used for training the two main modules of the advanced communications system to the new domain: the spoken Spanish into the LSE translation module and the Spanish generation from the LSE module. The main aspects to be generated are the vocabularies for both languages (Spanish words and signs), and the knowledge for translating in both directions. Finally, the field evaluation is carried out with deaf people using the advanced communications system to interact with hearing people in several scenarios. In this evaluation, the paper proposes several objective and subjective measurements for evaluating the performance. In this paper, the new considered domain is about dialogues in a hotel reception. Using this methodology, the system was developed in several months, obtaining very good performance: good translation rates (10% Sign Error Rate) with small processing times, allowing face-to-face dialogues.

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Desde que las Tecnologías de la Información y la Comunicación comenzaron a adquirir una gran importancia en la sociedad, uno de los principales objetivos ha sido conseguir que la información transmitida llegue en perfectas condiciones al receptor. Por este motivo, se hace necesario el desarrollo de nuevos sistemas de comunicación digital capaces de ofrecer una transmisión segura y fiable. Con el paso de los años, se han ido mejorando las características de los mismos, lo que significa importantes avances en la vida cotidiana. En este contexto, uno de los sistemas que más éxito ha tenido es la Modulación Reticulada con Codificación TCM, que aporta grandes ventajas en la comunicación digital, especialmente en los sistemas de banda estrecha. Este tipo de código de protección contra errores, basado en la codificación convolucional, se caracteriza por realizar la modulación y codificación en una sola función. Como consecuencia, se obtiene una mayor velocidad de transmisión de datos sin necesidad de incrementar el ancho de banda, a costa de pasar a una constelación superior. Con este Proyecto Fin de Grado se quiere analizar el comportamiento de la modulación TCM y cuáles son las ventajas que ofrece frente a otros sistemas similares. Se propone realizar cuatro simulaciones, que permitan visualizar diversas gráficas en las que se relacione la probabilidad de bit erróneo BER y la relación señal a ruido SNR. Además, con estas gráficas se puede determinar la ganancia que se obtiene con respecto a la probabilidad de bit erróneo teórica. Estos sistemas pasan de una modulación QPSK a una 8PSK o de una 8PSK a una 16QAM. Finalmente, se desarrolla un entorno gráfico de Matlab con el fin de proporcionar un sencillo manejo al usuario y una mayor interactividad. ABSTRACT. Since Information and Communication Technologies began to gain importance on society, one of the main objectives has been to achieve the transmitted information reaches the receiver perfectly. For this reason, it is necessary to develop new digital communication systems with the ability to offer a secure and reliable transmission. The systems characteristics have improved over the past years, what it means important progress in everyday life. In this context, one of the most successful systems is Trellis Coded Modulation TCM, that brings great advantages in terms of digital communications, especially narrowband systems. This kind of error correcting code, based on convolutional coding, is characterized by codifying and modulating at the same time. As a result, a higher data transmission speed is achieved without increasing bandwidth at the expense of using a superior modulation. The aim of this project is to analyze the TCM performance and the advantages it offers in comparison with other similar systems. Four simulations are proposed, that allows to display several graphics that show how the Bit Error Ratio BER and Signal Noise Ratio SNR are related. Furthermore, it is possible to calculate the coding gain. Finally, a Matlab graphic environment is designed in order to guarantee the interactivity with the final user.

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Pairwise sequence comparison methods have been assessed using proteins whose relationships are known reliably from their structures and functions, as described in the scop database [Murzin, A. G., Brenner, S. E., Hubbard, T. & Chothia C. (1995) J. Mol. Biol. 247, 536–540]. The evaluation tested the programs blast [Altschul, S. F., Gish, W., Miller, W., Myers, E. W. & Lipman, D. J. (1990). J. Mol. Biol. 215, 403–410], wu-blast2 [Altschul, S. F. & Gish, W. (1996) Methods Enzymol. 266, 460–480], fasta [Pearson, W. R. & Lipman, D. J. (1988) Proc. Natl. Acad. Sci. USA 85, 2444–2448], and ssearch [Smith, T. F. & Waterman, M. S. (1981) J. Mol. Biol. 147, 195–197] and their scoring schemes. The error rate of all algorithms is greatly reduced by using statistical scores to evaluate matches rather than percentage identity or raw scores. The E-value statistical scores of ssearch and fasta are reliable: the number of false positives found in our tests agrees well with the scores reported. However, the P-values reported by blast and wu-blast2 exaggerate significance by orders of magnitude. ssearch, fasta ktup = 1, and wu-blast2 perform best, and they are capable of detecting almost all relationships between proteins whose sequence identities are >30%. For more distantly related proteins, they do much less well; only one-half of the relationships between proteins with 20–30% identity are found. Because many homologs have low sequence similarity, most distant relationships cannot be detected by any pairwise comparison method; however, those which are identified may be used with confidence.

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This report documents the error rate in a commercially distributed subset of the IMAGE Consortium mouse cDNA clone collection. After isolation of plasmid DNA from 1189 bacterial stock cultures, only 62.2% were uncontaminated and contained cDNA inserts that had significant sequence identity to published data for the ordered clones. An agarose gel electrophoresis pre-screening strategy identified 361 stock cultures that appeared to contain two or more plasmid species. Isolation of individual colonies from these stocks demonstrated that 7.1% of the original 1189 stocks contained both a correct and an incorrect plasmid. 5.9% of the original 1189 stocks contained multiple, distinct, incorrect plasmids, indicating the likelihood of multiple contaminating events. While only 739 of the stocks purchased contained the desired cDNA clone, agarose gel pre-screening, colony isolation and similarity searching of dbEST allowed for the identification of an additional 420 clones that would have otherwise been discarded. Considering the high error rate in this subset of the IMAGE cDNA clone set, the use of sequence verified clones for cDNA microarray construction is warranted. When this is not possible, pre-screening non-sequence verified clones with agarose gel electrophoresis provides an inexpensive and efficient method to eliminate contaminated clones from the probe set.

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We present an approach for assessing the significance of sequence and structure comparisons by using nearly identical statistical formalisms for both sequence and structure. Doing so involves an all-vs.-all comparison of protein domains [taken here from the Structural Classification of Proteins (scop) database] and then fitting a simple distribution function to the observed scores. By using this distribution, we can attach a statistical significance to each comparison score in the form of a P value, the probability that a better score would occur by chance. As expected, we find that the scores for sequence matching follow an extreme-value distribution. The agreement, moreover, between the P values that we derive from this distribution and those reported by standard programs (e.g., blast and fasta validates our approach. Structure comparison scores also follow an extreme-value distribution when the statistics are expressed in terms of a structural alignment score (essentially the sum of reciprocated distances between aligned atoms minus gap penalties). We find that the traditional metric of structural similarity, the rms deviation in atom positions after fitting aligned atoms, follows a different distribution of scores and does not perform as well as the structural alignment score. Comparison of the sequence and structure statistics for pairs of proteins known to be related distantly shows that structural comparison is able to detect approximately twice as many distant relationships as sequence comparison at the same error rate. The comparison also indicates that there are very few pairs with significant similarity in terms of sequence but not structure whereas many pairs have significant similarity in terms of structure but not sequence.

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The replication of double-stranded plasmids containing a single adduct was analyzed in vivo by means of a sequence heterology that marks the two DNA strands. The single adduct was located within the sequence heterology, making it possible to distinguish trans-lesion synthesis (TLS) events from damage avoidance events in which replication did not proceed through the lesion. When the SOS system of the host bacteria is not induced, the C8-guanine adduct formed by the carcinogen N-2-acetylaminofluorene (AAF) yields less than 1% of TLS events, showing that replication does not readily proceed through the lesion. In contrast, the deacetylated adduct N-(deoxyguanosin-8-yl)-2-aminofluorene yields approximately 70% of TLS events under both SOS-induced and uninduced conditions. These results for TLS in vivo are in good agreement with the observation that AAF blocks DNA replication in vitro, whereas aminofluorene does so only weakly. Induction of the SOS response causes an increase in TLS events through the AAF adduct (approximately 13%). The increase in TLS is accompanied by a proportional increase in the frequency of AAF-induced frameshift mutations. However, the polymerase frameshift error rate per TLS event was essentially constant throughout the SOS response. In an SOS-induced delta umuD/C strain, both US events and mutagenesis are totally abolished even though there is no decrease in plasmid survival. Error-free replication evidently proceeds efficiently by means of the damage avoidance pathway. We conclude that SOS mutagenesis results from increased TLS rather than from an increased frameshift error rate of the polymerase.

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This paper introduces the session on advanced speech recognition technology. The two papers comprising this session argue that current technology yields a performance that is only an order of magnitude in error rate away from human performance and that incremental improvements will bring us to that desired level. I argue that, to the contrary, present performance is far removed from human performance and a revolution in our thinking is required to achieve the goal. It is further asserted that to bring about the revolution more effort should be expended on basic research and less on trying to prematurely commercialize a deficient technology.

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In the past decade, tremendous advances in the state of the art of automatic speech recognition by machine have taken place. A reduction in the word error rate by more than a factor of 5 and an increase in recognition speeds by several orders of magnitude (brought about by a combination of faster recognition search algorithms and more powerful computers), have combined to make high-accuracy, speaker-independent, continuous speech recognition for large vocabularies possible in real time, on off-the-shelf workstations, without the aid of special hardware. These advances promise to make speech recognition technology readily available to the general public. This paper focuses on the speech recognition advances made through better speech modeling techniques, chiefly through more accurate mathematical modeling of speech sounds.

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Although the n-back task has been widely applied to neuroimagery investigations of working memory (WM), the role of practice effects on behavioural performance of this task has not yet been investigated. The current study aimed to investigate the effects of task complexity and familiarity on the n-back task. Seventy-seven participants (39 male, 38 female) completed a visuospatial n-back task four times, twice in two testing sessions separated by a week. Participants were required to remember either the first, second or third (n-back) most recent letter positions in a continuous sequence and to indicate whether the current item matched or did not match the remembered position. A control task, with no working memory requirements required participants to match to a predetermined stimulus position. In both testing sessions, reaction time (RT) and error rate increased with increasing WM load. An exponential slope for RTs in the first session indicated dual-task interference at the 3-back level. However, a linear slope in the second session indicated a reduction of dual-task interference. Attenuation of interference in the second session suggested a reduction in executive demands of the task with practice. This suggested that practice effects occur within the n-back ask and need to be controlled for in future neuroimagery research using the task.

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Univariate linkage analysis is used routinely to localise genes for human complex traits. Often, many traits are analysed but the significance of linkage for each trait is not corrected for multiple trait testing, which increases the experiment-wise type-I error rate. In addition, univariate analyses do not realise the full power provided by multivariate data sets. Multivariate linkage is the ideal solution but it is computationally intensive, so genome-wide analysis and evaluation of empirical significance are often prohibitive. We describe two simple methods that efficiently alleviate these caveats by combining P-values from multiple univariate linkage analyses. The first method estimates empirical pointwise and genome-wide significance between one trait and one marker when multiple traits have been tested. It is as robust as an appropriate Bonferroni adjustment, with the advantage that no assumptions are required about the number of independent tests performed. The second method estimates the significance of linkage between multiple traits and one marker and, therefore, it can be used to localise regions that harbour pleiotropic quantitative trait loci (QTL). We show that this method has greater power than individual univariate analyses to detect a pleiotropic QTL across different situations. In addition, when traits are moderately correlated and the QTL influences all traits, it can outperform formal multivariate VC analysis. This approach is computationally feasible for any number of traits and was not affected by the residual correlation between traits. We illustrate the utility of our approach with a genome scan of three asthma traits measured in families with a twin proband.