Realisering av streamingtjenester i P2P-nettverk er en utfordring fordi slike nettverk er preget av dynamikk og stadige endringer i forhold til nodenes tilgjengelighet. Hvis en node som leverer en mediestrøm blir utilgjengelig, vil det oppstå avbrudd i avspillingen hos mottakernoden. Utgangspunktet for denne hovedoppgaven er et ønske om å kunne gjøre streaming med tilfredsstillende tjenestekvalitet til en realitet i P2P-nettverk. Målet er å spesifisere en metode som bidrar til å forbedre den opplevde ressurstilgjengeligheten i slike nettverk. P2P-systemer kjennetegnes generelt ved at de består av noder på randen av Internett, at samtlige noder bidrar med ressurser og kommuniserer direkte med hverandre og at hver node er en selvstyrende enhet. Det finnes to hovedkategorier P2P-systemer, “ekte” og “hybride”, samt en rekke applikasjonsområder. Streaming innebærer at innholdsleveranse gjøres i sanntid. Sanntidsleveranse krever at data leveres til mottaker i tide og at pakker leveres i rett rekkefølge. I P2P-nettverk, hvor omgivelsene preges av dynamikk og heterogenitet, er dette en stor utfordring. Flere teknikker kan benyttes for å håndtere slike omgivelser, men utfordringene knyttet til tilgjengelighet framstår som uløste og samtidig kritiske med tanke på oppnåelig tjenestekvalitet. Vårt bidrag i retning av å forbedre den opplevde ressurstilgjengeligheten er en metode som blant annet benytter multinodestreaming som leveranseteknikk. Multinodestreaming innebærer å dele en strøm i flere delstrømmer og hente disse fra forskjellige noder. Delstrømmene kan da flettes ved mottak og spilles av som én strøm. Konsekvensen av at en node blir utilgjengelig vil kun bli kvalitetsdegradering av avspillingen, og ikke totalt avspillingsbrudd. Tilgjengeligheten kan forbedres ytterligere ved å utføre effektiv overtakelse av tapte delstrømmer. Ved å implementere metoden i en prototyp, er det mulig å teste dens funksjon og effekt. Testene viser at kvalitetsdegraderingen på avspillingen, som følge av at en node blir utilgjengelig, kun er midlertidig hvis en annen node kan overta leveransen. Reetablering av en delstrøm kan gjøres på gjennomsnittlig 4,7 sekunder. Enkeltnodestreaming ville til sammenligning medført fullt avspillingsbrudd. Resultatene sannsynliggjør derfor at multinodestreaming kan bidra til å forbedre den opplevde ressurstilgjengeligheten for streamingtjenester i P2P-nettverk.

Med den här uppsatsen vill vi undersöka vilka förutsättningar, samt vilka hinder och möjligheter, som finns för multimediala presentationsformer av journalistik på svenska tidningars webbplatser. Vi har undersökt vilka attityder som finns kring multimedia, på vilka sätt de undersökta tidningarna arbetar med multimedia i dag och hur de vill arbeta med det i framtiden. Vi har också undersökt hur de tänker kring själva utformningen av de multimediala presentationsformerna. Vi har använt oss av explorativa intervjuer med fem personer som har nära anknytning till den multimediaproduktion som bedrivs i dag, och analyserat dessa utifrån Gunnar Nygrens studier av konvergens och divergens, Mark Deuzes teorier om onlinejournalistik, samt Richard E. Mayer och Julianne H. Newtons teorier om den mänskliga hjärnans betydelse i inlärning och journalistik. De medieföretag vi besökt är Dagens Nyheter, Svenska Dagbladet och Svenska Grafikbyrån. Resultatet av vår undersökning visar att attityderna kring multimediala presentationsformer är positiva och att respondenterna ser ljust på framtiden, men alla är överens om att Sverige ligger långt efter exempelvis USA och Spanien när det gäller hur mycket multimedia som används. Faktorer som har bromsat utvecklingen i Sverige är, enligt våra resultat, att man inte riktigt vet hur produktionen ska organiseras, vem som ska producera multimedia. Bristen på inspiration och goda svenska exempel är det kanske största hindret. Goda exempel och inspiration finns utomlands och i nöjes- och reklamindustrin, men traditioner och journalistiska ideal hindrar journalister från att hitta dem. Klart är också att journalisterna inte har reflekterat djupare kring utformning och inlärning ur ett teoretiskt perspektiv, utan går mest på magkänsla – en magkänsla som kanske ännu inte finns på redaktionerna.

Video streaming applications over wireless networks have turned out to be immensely popular recently. In this thesis, we first study the buffering schemes for the VBR video streaming in heterogeneous wireless networks. An analytical framework is presented to derive the expected number of jitters and average buffering delay. Through experimenting with a wide range of buffering schemes, we quantify the bene¯t of incorporating user location information in streaming over heterogeneous wireless networks. Second, we consider the delivery of scalable VBR video streams over wireless channels. We propose adaptive rate control algorithms to improve the combined system performance of video frame quality and playout smoothness based on the feedback information of wireless network estimation, buffer content and playback situation. The proposed adaptive rate control algorithms provide significantly improved streaming quality compared with the non-control policy.

Video streaming applications over wireless networks have turned out to be immensely popular recently. In this thesis, we first study the buffering schemes for the VBR video streaming in heterogeneous wireless networks. An analytical framework is presented to derive the expected number of jitters and average buffering delay. Through experimenting with a wide range of buffering schemes, we quantify the bene¯t of incorporating user location information in streaming over heterogeneous wireless networks. Second, we consider the delivery of scalable VBR video streams over wireless channels. We propose adaptive rate control algorithms to improve the combined system performance of video frame quality and playout smoothness based on the feedback information of wireless network estimation, buffer content and playback situation. The proposed adaptive rate control algorithms provide significantly improved streaming quality compared with the non-control policy.

Un sistema de Vídeo bajo Demanda (Video on Demand - VoD) proporciona que un conjunto de clientes acceda a contenidos multimedia de manera independiente; los usuarios se conectan al sistema, eligen el contenido a visualizar y empiezan a disfrutar del servicio en cualquier instante de tiempo. El vídeo es enviado al cliente, que recibe, descodifica y visualiza el contenido siempre esperando garantía de Calidad de Servicio (Quality of Service - QoS) por parte del sistema. Uno de los objetivos principales en el diseño de servicios de VoD es soportar un gran número de peticiones concurrentes generadas por clientes geográficamente distribuidos; los sistemas de VoD deben conseguir un servicio factible a gran escala y de alta calidad, imponiendo bajos costes de operación y pocas restricciones de despliegue. Recientemente, la distribución de contenidos multimedia en forma de flujo de datos en Internet viene presentando un crecimiento espectacular. La Internet es el entorno más popular de usuarios interconectados y está presente en todo el mundo. Debido a las características de escala global y entorno publico de Internet, esta se ha hecho el ambiente más importante para desplegar el servicio de Vídeo bajo Demanda a gran escala (Large-scale Video on Demand - LVoD). Debido a las limitaciones del modelo cliente-servidor centralizado, los paradigmas peer-to-peer (P2P) y multicast son extensamente aplicados en la distribución multimedia para mejorar la escalabilidad y prestaciones del sistema a través de la compartición de recursos. El P2P está basado en la libre cooperación de iguales con vistas al desarrollo de una tarea común; aprovecha recursos disponibles en el lado del usuario final (almacenamiento, contenido, ancho de banda, poder de procesamiento etc.). El multicast a su vez es una estrategia de comunicación donde un origen tiene la capacidad de transmitir información que puede ser recibida simultáneamente por un grupo de destinos interesados en el mismo contenido. Sin embargo, los paradigmas P2P y multicast añaden nuevas cuestiones en el diseño servicios de VoD para Internet. Los peers son heterogéneos en sus recursos y actúan por su propio libre albedrío, llegando y dejando el sistema en cualquier momento; la carencia o el cambio de la fuente de datos provocada por fallos de peers afectan fuertemente la QoS en sistemas de VoD basados en técnicas de multicast y P2P. Así, la tolerancia a fallos se ha hecho una cuestión crucial en servicios de VoD basados en P2P a fin de garantizar QoS. El mecanismo de tolerancia a fallos se consigue a través del intercambio de mensajes de control; además, el tratamiento de fallos es limitado en el tiempo para proporcionar ausencia de errores y por consiguiente mantener la QoS. Un buen esquema de control se hace imprescindible y su diseño debe ser cuidadoso debido a la restricción de tiempo real del servicio multimedia y el overhead impuesto al sistema por los mensajes de control. Esta tesis presenta un Esquema de Tolerancia a Fallos (Fault Tolerance Scheme - FTS) que trabaja construyendo un sistema de backup distribuido, basado en las capacidades de los propios peers. El FTS está diseñado para organizar un pequeño conjunto de peers que almacenan estáticamente porciones de los archivos multimedia en un buffer llamado 'buffer altruista'. Los clientes que componen el backup distribuido colaboran en el mecanismo de tolerancia a fallos del sistema reservando espacio de almacenamiento (buffer) y capacidad de ancho de banda de subida; los peers seleccionados forman un Grupo de Tolerancia a Fallos (Fault Tolerance Group - FTG). Los resultados obtenidos muestran que el mecanismo de control tiene gran impacto sobre el sistema y exige un diseño cuidadoso; el Esquema de Tolerancia a Fallos propuesto colabora para reducir el overhead impuesto al sistema y es capaz de conseguir tiempos de respuesta bajos en el manejo de fallos; esto mejora la experiencia del usuario reduciendo el retraso en el inicio de la visualización y garantiza un mejor uso de recursos de almacenamiento (buffer). El FTS también distribuye las tareas de control proporcionando fiabilidad y robustez al sistema de VoD. / A Video on Demand (VoD) system provides multimedia content to a set of clients in independent manner; users connect to the system, choose the content to view and start enjoying the service at any given moment. The video is down-streamed to the client, who receives, decodes and displays the content always expecting guaranteed Quality of Service (QoS) from the system. One of the main goals in designing VoD services is to support a great number of concurrent requests generated by geographically distributed clients; VoD systems must achieve a feasible large-scale and high-quality service with the lower costs and fewer deployment restrictions. Recently, multimedia streaming distribution in the Internet presented a spectacular growing. The Internet is the most popular environment of connected users and is deployed throughout the world. Owing to the public and global scale features of Internet, it has become the most important environment to deploy large-scale Video on Demand service (LVoD). Owing to the limitations of centralised server-client model, Peer-to-Peer (P2P) and multicast approaches are widely applied in the multimedia distribution to improve system scalability and performance by sharing resources. P2P is based in the free cooperation of equals in view of the performance of a common task; it takes advantage of available resources at the end host side (storage, content, bandwidth, power processing etc.). The multicast is a communication strategy where a sender has the capability to transmit information that can be received concurrently by a group of interested destinations. Nevertheless, P2P and multicast paradigms add new issues in the design of Internet VoD services. Peers are heterogeneous in their resources and act by their own free will, coming and leaving the system at any time; the lack or the change of data source provoked by peer faults strongly affects the QoS in VoD systems based in P2P and multicast techniques. This way, fault tolerance has become a major issue in P2P-based VoD services in order to guarantee QoS. The fault tolerance mechanism is achieved through the exchange of control messages; moreover, the failure treatment is time limited for providing error absence and consequently maintaining the QoS. A good control scheme is needed and its design must be careful owing to the soft real-time restriction of multimedia service and the overhead imposed on the system. This thesis presents a Fault Tolerance Scheme (FTS) that works by constructing a backup system in a distributed manner, based in own peers' capabilities. The FTS is designed to organise a small set of peers to store portions of the multimedia files statically in a buffer called the 'altruist buffer'. The clients that make up the distributed backup collaborate in system fault tolerance mechanism by reserving buffer space and upload bandwidth capacity; the selected peers form a Fault Tolerance Group (FTG). Results show that the control mechanism has great impact over the system and demands a caution design; the proposed Fault Tolerance Scheme collaborates to reduce the overhead imposed on the system and is able to achieve low response times in dealing with failures; this improves user experience by reducing start-up delays and guarantees a better usage of buffer resources. The FTS also distributes the control tasks providing reliability and robustness to the VoD system.

No description available.

In this thesis, the JPEG 2000 quality scalability feature was investigated in thecontext of transporting video over IP networks. The goals of the investigation wastwo-fold. First, it was desired to nd a way of choosing the number of quality layersto embed in a JPEG 2000 codestream. In previous work, this choice has been moreor less arbitrary. Second, it was desired to nd how low the video bitrate could bedropped before it became perceptible to a viewer. This information can be usedin an IP networking scenario to e.g. adapt the video bitrate blindly according tothe measured channel capacity as long as the drop in bitrate is expected to beimperceptible. When the drop in bitrate is expected to be perceptible, a switchcould be made to a smoother bitrate adaptation.A way of choosing the total number of quality layers to embed in a codestreamwas found by minimizing the dierence in predicted quality between direct andscaled compression. Scaled compression is the compression which is achieved byextracting quality layers. The minimization procedure was bound by the speed ofthe encoder, as it takes longer for an encoder to embed more quality layers. It wasfound that the procedure was highly dependent on the desired bitrate range.A subjective test was run in order to measure how large a drop in video bitrate hadto be for it to become perceptible. A newly developed JPEG 2000 quality layerscaler was used to produce the dierent bitrates in the test. The number of qualitylayers to embed in codestream was found by using the minimization procedurementioned above. It was found that, for the bitrate range used in the test, 2 - 30Mbits/s for a resolution of 1280x720 at 25 frames per second, the magnitude ofthe drop in bitrate had to be at least 10 Mbits/s before the participants in the testnoticed it. A comparison with objective quality metrics, SSIM and PSNR, revealedthat it was very dicult to predict the visibility of the drops in bitrate by usingthese metrics. Designing the type of rate control mentioned in the rst paragraphwill therefore have to wait until a parameter with good predictive properties canbe found.

Pipes with corrugations or cavities are used in a wide variety of applications. In recent years the natural gas industry has experienced "singing" risers, where pipes transporting natural gas excite loud whistling sounds limiting the flow rate of which the gas can be transported. There has been a number of publications regarding this phenomenon, investigating corrugated pipes and pipes with one or more cavities. In this thesis the most basic situation will be studied; A smooth pipe with a single small cavity. This is studied by simulations and experiments. The effects of changing the length of the cavity, and the pipe section between the inlet and the cavity is investigated. The simulations were conducted in Palabos, a Lattice Boltzmann solver which proves to be a promising piece of software for acoustic simulations. The experiments were conduced using a metal pipe with variable inlet and cavity length. Initial vortices created at the inlet are amplified in the cavity by a cavity flow. The results strongly suggest that these inlet vortices are essential for the excitation of whistling sounds. The number of vortices traveling across a cavity at the time is called a hydrodynamic mode. When the frequency of vortices crossing a cavity coincides with an acoustic pipe mode, a whistling sound close to this frequency is excited. Cancellation of the whistling sound with an added cancellation frequency is possible for certain cavity and inlet lengths.

The NTNU Test Satellite (NUTS) is planned to have a payload for observation of atmospheric gravity waves. The gravity waves will be observed by means of an infrared camera imaging the perturbations in the OH airglow layer. So far, no suitable camera has been found that complies with the restrictions that follows when building a small satellite. Uncooled InGaAs has however been concluded to be the most suitable detector type in terms of wavelength response and weight.InGaAs sensors are known to have a high dark current when not cooled, and processing must therefore be applied to remove the background offset and noise.The combination of the high speed of the satellite and the long exposure time that is required for the camera will create motion blur. Simulations with synthetic test images in MATLAB showed that the integration time should at least be kept under 1 second in order not to destroy the wave patterns. Longer integration times may however be required in order to get a sufficient SNR.Two signal processing solutions to this problem was investigated: motion blur removal by deconvolution and image averaging with motion compensation. The former strategy is to apply a long exposure time to get a strong signal, and then remove the blur with deconvolution techniques using knowledge of the blur filter.Simulations applying the Lucy-Richardson (LR) algorithm showed that it was not able to remove strong blur, and was very sensitive to errors in the blur filter and noise in the image. The other approach is to obtain a sequence of images with short exposure time in order to avoid motion blur, and provide the necessary SNR by shifting the images according to the known motion and combine them into one image. This concept is simpler and more reliable than the deconvolution approach, and simulations showed that it is less sensitive to errors in the speed estimate than the deconvolution algorithm. It was concluded that this is the most suitable approach for the NUTS application, and it should be implemented on-board the satellite in order to provide a good SNR for the compression to function optimally. The downlink datarate of NUTS is of only 9600 bit/s, and it has been estimated that 2.45 Mb of payload data can be downloaded on average per day. This corresponds to less than 5 uncompressed images of 256 × 256 pixels with 8 bit per pixel.A sequence of overlapping combined images should be obtained to provide a scan of a desired area, and it was suggested that it should be encoded as video to enable efficient compression and transmission of as many images as possible to the ground station. A three-dimensional DPCM algorithm combined with a deadzone quantizer and stack-run coding was implemented in MATLAB. Simulations demonstrated that this simple compression scheme can provide a bit rate of less than 1 bit/px for a sequence of ravity wave images. One of the quantizers that was tried gave 0.83 bits per pixel with reasonable quality. If this number can be achieved in practice, the image transfer ate would be increased to 45 images per day, which is a significant improvement.

This thesis has taken a closer look at the implementation of the back-end of a language recognition system. The front-end of the system is a Universal Attribute Recognizer (UAR), which is used to detect phonetic characteristics in an utterance. When a speech signal is sent through the UAR, it is decoded into a sequence of attributes which is used to generate a vector of term-count. Vector Space Modeling (VSM) have been used for training the language classifiers in the back-end. The main principle of VSM is that term-count vectors from the same language will position themselves close to eachother when they are mapped into a vector space, and this property can be exploited for recognizing languages. The implemented back-end has trained vectors space classifiers for 12 different languages, and a NIST recognition task has been performed for evaluating the recognition rate of the system. The NIST task was a verification task and the system achived a equal error rate (EER) of $6.73 %$. Tools like Support Vector Machines (SVM) and Gaussian Mixture Models (GMM) have been used in the implementation of the back-end. Thus, are quite a few parameters which can be varied and tweaked, and different experiments were conducted to investigate how these parameters would affect EER of the language recognizer. As a part test the robustness of the system, the language recognizer were exposed to a so-called out-of-set language, which is a language that the system has not been trained to handle. The system showed a poor performance at rejecting these speech segments correctly.