Contribution to quality of user experience provision over wireless networks

The widespread expansion of wireless networks has brought new attractive possibilities to end users. In addition to the mobility capabilities provided by unwired devices, it is worth remarking the easy configuration process that a user has to follow to gain connectivity through a wireless network. Furthermore, the increasing bandwidth provided by the IEEE 802.11 family has made possible accessing to high-demanding services such as multimedia communications. Multimedia traffic has unique characteristics that make it greatly vulnerable against network impairments, such as packet losses, delay, or jitter. Voice over IP (VoIP) communications, video-conference, video-streaming, etc., are examples of these high-demanding services that need to meet very strict requirements in order to be served with acceptable levels of quality. Accomplishing these tough requirements will become extremely important during the next years, taking into account that consumer video traffic will be the predominant traffic in the Internet during the next years. In wired systems, these requirements are achieved by using Quality of Service (QoS) techniques, such as Differentiated Services (DiffServ), traffic engineering, etc. However, employing these methodologies in wireless networks is not that simple as many other factors impact on the quality of the provided service, e.g., fading, interferences, etc. Focusing on the IEEE 802.11g standard, which is the most extended technology for Wireless Local Area Networks (WLANs), it defines two different architecture schemes. On one hand, the infrastructure mode consists of a central point, which manages the network, assuming network controlling tasks such as IP assignment, routing, accessing security, etc. The rest of the nodes composing the network act as hosts, i.e., they send and receive traffic through the central point. On the other hand, the IEEE 802.11 ad-hoc configuration mode is less extended than the infrastructure one. Under this scheme, there is not a central point in the network, but all the nodes composing the network assume both host and router roles, which permits the quick deployment of a network without a pre-existent infrastructure. This type of networks, so called Mobile Ad-hoc NETworks (MANETs), presents interesting characteristics for situations when the fast deployment of a communication system is needed, e.g., tactics networks, disaster events, or temporary networks. The benefits provided by MANETs are varied, including high mobility possibilities provided to the nodes, network coverage extension, or network reliability avoiding single points of failure. The dynamic nature of these networks makes the nodes to react to topology changes as fast as possible. Moreover, as aforementioned, the transmission of multimedia traffic entails real-time constraints, necessary to provide these services with acceptable levels of quality. For those reasons, efficient routing protocols are needed, capable of providing enough reliability to the network and with the minimum impact to the quality of the service flowing through the nodes. Regarding quality measurements, the current trend is estimating what the end user actually perceives when consuming the service. This paradigm is called Quality of user Experience (QoE) and differs from the traditional Quality of Service (QoS) approach in the human perspective given to quality estimations. In order to measure the subjective opinion that a user has about a given service, different approaches can be taken. The most accurate methodology is performing subjective tests in which a panel of human testers rates the quality of the service under evaluation. This approach returns a quality score, so-called Mean Opinion Score (MOS), for the considered service in a scale 1 - 5. This methodology presents several drawbacks such as its high expenses and the impossibility of performing tests at real time. For those reasons, several mathematical models have been presented in order to provide an estimation of the QoE (MOS) reached by different multimedia services In this thesis, the focus is on evaluating and understanding the multimedia-content transmission-process in wireless networks from a QoE perspective. To this end, firstly, the QoE paradigm is explored aiming at understanding how to evaluate the quality of a given multimedia service. Then, the influence of the impairments introduced by the wireless transmission channel on the multimedia communications is analyzed. Besides, the functioning of different WLAN schemes in order to test their suitability to support highly demanding traffic such as the multimedia transmission is evaluated. Finally, as the main contribution of this thesis, new mechanisms or strategies to improve the quality of multimedia services distributed over IEEE 802.11 networks are presented. Concretely, the distribution of multimedia services over ad-hoc networks is deeply studied. Thus, a novel opportunistic routing protocol, so-called JOKER (auto-adJustable Opportunistic acK/timEr-based Routing) is presented. This proposal permits better support to multimedia services while reducing the energy consumption in comparison with the standard ad-hoc routing protocols.Universidad Politécnica de CartagenaPrograma Oficial de Doctorado en Tecnologías de la Información y Comunicacione

Proceedings of the Fifth International Mobile Satellite Conference 1997

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial communications services. While previous International Mobile Satellite Conferences have concentrated on technical advances and the increasing worldwide commercial activities, this conference focuses on the next generation of mobile satellite services. The approximately 80 papers included here cover sessions in the following areas: networking and protocols; code division multiple access technologies; demand, economics and technology issues; current and planned systems; propagation; terminal technology; modulation and coding advances; spacecraft technology; advanced systems; and applications and experiments

LTE Optimization and Resource Management in Wireless Heterogeneous Networks

Mobile communication technology is evolving with a great pace. The development of the Long Term Evolution (LTE) mobile system by 3GPP is one of the milestones in this direction. This work highlights a few areas in the LTE radio access network where the proposed innovative mechanisms can substantially improve overall LTE system performance. In order to further extend the capacity of LTE networks, an integration with the non-3GPP networks (e.g., WLAN, WiMAX etc.) is also proposed in this work. Moreover, it is discussed how bandwidth resources should be managed in such heterogeneous networks. The work has purposed a comprehensive system architecture as an overlay of the 3GPP defined SAE architecture, effective resource management mechanisms as well as a Linear Programming based analytical solution for the optimal network resource allocation problem. In addition, alternative computationally efficient heuristic based algorithms have also been designed to achieve near-optimal performance

Recent Advances in Signal Processing

The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity

Application of advanced on-board processing concepts to future satellite communications systems

An initial definition of on-board processing requirements for an advanced satellite communications system to service domestic markets in the 1990's is presented. An exemplar system architecture with both RF on-board switching and demodulation/remodulation baseband processing was used to identify important issues related to system implementation, cost, and technology development

Learning efficient temporal information in deep networks: From the viewpoints of applications and modeling

With the introduction of deep learning, machine learning has dominated several technology areas, giving birth to high-performance applications that can even challenge human-level accuracy. However, the complexity of deep models is also exploding as a by-product of the revolution of machine learning. Such enormous model complexity has raised the new challenge of improving the efficiency in deep models to reduce deployment expense, especially for systems with high throughput demands or devices with limited power. The dissertation aims to improve the efficiency of temporal-sensitive deep models in four different directions. First, we develop a bandwidth extension mapping to avoid deploying multiple speech recognition systems corresponding to wideband and narrowband signals. Second, we apply a multi-modality approach to compensate for the performance of an excitement scoring system, where the input video sequences are aggressively down-sampled to reduce throughput. Third, we formulate the motion feature in the feature space by directly inducing the temporal information from intermediate layers of deep networks instead of relying on an additional optical flow stream. Finally, we model a spatiotemporal sampling network inspired by the human visual perception mechanism to reduce input frames and regions adaptively

Economically sustainable public security and emergency network exploiting a broadband communications satellite

The research contributes to work in Rapid Deployment of a National Public Security and Emergency Communications Network using Communication Satellite Broadband. Although studies in Public Security Communication networks have examined the use of communications satellite as an integral part of the Communication Infrastructure, there has not been an in-depth design analysis of an optimized regional broadband-based communication satellite in relation to the envisaged service coverage area, with little or no terrestrial last-mile telecommunications infrastructure for delivery of satellite solutions, applications and services. As such, the research provides a case study of a Nigerian Public Safety Security Communications Pilot project deployed in regions of the African continent with inadequate terrestrial last mile infrastructure and thus requiring a robust regional Communications Satellite complemented with variants of terrestrial wireless technologies to bridge the digital hiatus as a short and medium term measure apart from other strategic needs. The research not only addresses the pivotal role of a secured integrated communications Public safety network for security agencies and emergency service organizations with its potential to foster efficient information symmetry amongst their operations including during emergency and crisis management in a timely manner but demonstrates a working model of how analogue spectrum meant for Push-to-Talk (PTT) services can be re-farmed and digitalized as a “dedicated” broadband-based public communications system. The network’s sustainability can be secured by using excess capacity for the strategic commercial telecommunication needs of the state and its citizens. Utilization of scarce spectrum has been deployed for Nigeria’s Cashless policy pilot project for financial and digital inclusion. This effectively drives the universal access goals, without exclusivity, in a continent, which still remains the least wired in the world

Fast-tracking the digital revolution : broadband for Latin America and the Caribbean

This study is based on the premise that broadband is central to a technological, organizational and social innovation system, one which calls for new policies that are flexible, long-term approaches encompassing the objectives of greater productivity, innovation, competitiveness, social inclusion and sustainability. The study focuses on improving and developing broadband accessibility in Latin America and Caribbean (LAC) countries, alongside strategies for the development of public policies, including the role of the State, the role of regulation in market growth and service access, demand incentives, and the proper management of scarce resources such as spectrum availability

Assessing the quality of audio and video components in desktop multimedia conferencing

This thesis seeks to address the HCI (Human-Computer Interaction) research problem of how to establish the level of audio and video quality that end users require to successfully perform tasks via networked desktop videoconferencing. There are currently no established HCI methods of assessing the perceived quality of audio and video delivered in desktop videoconferencing. The transport of real-time speech and video information across new digital networks causes novel and different degradations, problems and issues to those common in the traditional telecommunications areas (telephone and television). Traditional assessment methods involve the use of very short test samples, are traditionally conducted outside a task-based environment, and focus on whether a degradation is noticed or not. But these methods cannot help establish what audio-visual quality is required by users to perform tasks successfully with the minimum of user cost, in interactive conferencing environments. This thesis addresses this research gap by investigating and developing a battery of assessment methods for networked videoconferencing, suitable for use in both field trials and laboratory-based studies. The development and use of these new methods helps identify the most critical variables (and levels of these variables) that affect perceived quality, and means by which network designers and HCI practitioners can address these problems are suggested. The output of the thesis therefore contributes both methodological (i.e. new rating scales and data-gathering methods) and substantive (i.e. explicit knowledge about quality requirements for certain tasks) knowledge to the HCI and networking research communities on the subjective quality requirements of real-time interaction in networked videoconferencing environments. Exploratory research is carried out through an interleaved series of field trials and controlled studies, advancing substantive and methodological knowledge in an incremental fashion. Initial studies use the ITU-recommended assessment methods, but these are found to be unsuitable for assessing networked speech and video quality for a number of reasons. Therefore later studies investigate and establish a novel polar rating scale, which can be used both as a static rating scale and as a dynamic continuous slider. These and further developments of the methods in future lab- based and real conferencing environments will enable subjective quality requirements and guidelines for different videoconferencing tasks to be established

Speech enhancement algorithms for audiological applications

Texto en inglés y resumen en inglés y españolPremio Extraordinario de Doctorado de la UAH en el año académico 2013-2014La mejora de la calidad de la voz es un problema que, aunque ha sido abordado durante muchos años, aún sigue abierto. El creciente auge de aplicaciones tales como los sistemas manos libres o de reconocimiento de voz automático y las cada vez mayores exigencias de las personas con pérdidas auditivas han dado un impulso definitivo a este área de investigación. Esta tesis doctoral se centra en la mejora de la calidad de la voz en aplicaciones audiológicas. La mayoría del trabajo de investigación desarrollado en esta tesis está dirigido a la mejora de la inteligibilidad de la voz en audífonos digitales, teniendo en cuenta las limitaciones de este tipo de dispositivos. La combinación de técnicas de separación de fuentes y filtrado espacial con técnicas de aprendizaje automático y computación evolutiva ha originado novedosos e interesantes algoritmos que son incluidos en esta tesis. La tesis esta dividida en dos grandes bloques. El primer bloque contiene un estudio preliminar del problema y una exhaustiva revisión del estudio del arte sobre algoritmos de mejora de la calidad de la voz, que sirve para definir los objetivos de esta tesis. El segundo bloque contiene la descripción del trabajo de investigación realizado para cumplir los objetivos de la tesis, así como los experimentos y resultados obtenidos. En primer lugar, el problema de mejora de la calidad de la voz es descrito formalmente en el dominio tiempo-frecuencia. Los principales requerimientos y restricciones de los audífonos digitales son definidas. Tras describir el problema, una amplia revisión del estudio del arte ha sido elaborada. La revisión incluye algoritmos de mejora de la calidad de la voz mono-canal y multi-canal, considerando técnicas de reducción de ruido y técnicas de separación de fuentes. Además, la aplicación de estos algoritmos en audífonos digitales es evaluada. El primer problema abordado en la tesis es la separación de fuentes sonoras en mezclas infra-determinadas en el dominio tiempo-frecuencia, sin considerar ningún tipo de restricción computacional. El rendimiento del famoso algoritmo DUET, que consigue separar fuentes de voz con solo dos mezclas, ha sido evaluado en diversos escenarios, incluyendo mezclas lineales y binaurales no reverberantes, mezclas reverberantes, y mezclas de voz con otro tipo de fuentes tales como ruido y música. El estudio revela la falta de robustez del algoritmo DUET, cuyo rendimiento se ve seriamente disminuido en mezclas reverberantes, mezclas binaurales, y mezclas de voz con música y ruido. Con el objetivo de mejorar el rendimiento en estos casos, se presenta un novedoso algoritmo de separación de fuentes que combina la técnica de clustering mean shift con la base del algoritmo DUET. La etapa de clustering del algoritmo DUET, que esta basada en un histograma ponderado, es reemplazada por una modificación del algoritmo mean shift, introduciendo el uso de un kernel Gaussiano ponderado. El análisis de los resultados obtenidos muestran una clara mejora obtenida por el algoritmo propuesto en relación con el algoritmo DUET original y una modificación que usa k-means. Además, el algoritmo propuesto ha sido extendido para usar un array de micrófonos de cualquier tamaño y geometría. A continuación se ha abordado el problema de la enumeración de fuentes de voz, que esta relacionado con el problema de separación de fuentes. Se ha propuesto un novedoso algoritmo basado en un criterio de teoría de la información y en la estimación de los retardos relativos causados por las fuentes entre un par de micrófonos. El algoritmo ha obtenido excelente resultados y muestra robustez en la enumeración de mezclas no reverberantes de hasta 5 fuentes de voz. Además se demuestra la potencia del algoritmo para la enumeración de fuentes en mezclas reverberantes. El resto de la tesis esta centrada en audífonos digitales. El primer problema tratado es el de la mejora de la inteligibilidad de la voz en audífonos monoaurales. En primer lugar, se realiza un estudio de los recursos computacionales disponibles en audífonos digitales de ultima generación. Los resultados de este estudio se han utilizado para limitar el coste computacional de los algoritmos de mejora de la calidad de la voz para audífonos propuestos en esta tesis. Para resolver este primer problema se propone un algoritmo mono-canal de mejora de la calidad de la voz de bajo coste computacional. El objetivo es la estimación de una mascara tiempo-frecuencia continua para obtener el mayor parámetro PESQ de salida. El algoritmo combina una versión generalizada del estimador de mínimos cuadrados con un algoritmo de selección de características a medida, utilizando un novedoso conjunto de características. El algoritmo ha obtenido resultados excelentes incluso con baja relación señal a ruido. El siguiente problema abordado es el diseño de algoritmos de mejora de la calidad de la voz para audífonos binaurales comunicados de forma inalámbrica. Estos sistemas tienen un problema adicional, y es que la conexión inalámbrica aumenta el consumo de potencia. El objetivo en esta tesis es diseñar algoritmos de mejora de la calidad de la voz de bajo coste computacional que incrementen la eficiencia energética en audífonos binaurales comunicados de forma inalámbrica. Se han propuesto dos soluciones. La primera es un algoritmo de extremado bajo coste computacional que maximiza el parámetro WDO y esta basado en la estimación de una mascara binaria mediante un discriminante cuadrático que utiliza los valores ILD e ITD de cada punto tiempo-frecuencia para clasificarlo entre voz o ruido. El segundo algoritmo propuesto, también de bajo coste, utiliza además la información de puntos tiempo-frecuencia vecinos para estimar la IBM mediante una versión generalizada del LS-LDA. Además, se propone utilizar un MSE ponderado para estimar la IBM y maximizar el parámetro WDO al mismo tiempo. En ambos algoritmos se propone un esquema de transmisión eficiente energéticamente, que se basa en cuantificar los valores de amplitud y fase de cada banda de frecuencia con un numero distinto de bits. La distribución de bits entre frecuencias se optimiza mediante técnicas de computación evolutivas. El ultimo trabajo incluido en esta tesis trata del diseño de filtros espaciales para audífonos personalizados a una persona determinada. Los coeficientes del filtro pueden adaptarse a una persona siempre que se conozca su HRTF. Desafortunadamente, esta información no esta disponible cuando un paciente visita el audiólogo, lo que causa perdidas de ganancia y distorsiones. Con este problema en mente, se han propuesto tres métodos para diseñar filtros espaciales que maximicen la ganancia y minimicen las distorsiones medias para un conjunto de HRTFs de diseño