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

Infra-estrutura de transcodificação para distribuição de áudio em uma Intranet adaptável ao estado da rede

BOTELHO, Fábio Pereira January 2004 (has links)
Made available in DSpace on 2014-06-12T15:58:56Z (GMT). No. of bitstreams: 2 arquivo4774_1.pdf: 2496019 bytes, checksum: feac7bc4e882b9034f0e91a01ab90df8 (MD5) license.txt: 1748 bytes, checksum: 8a4605be74aa9ea9d79846c1fba20a33 (MD5) Previous issue date: 2004 / Com o advento das Redes de Computadores e a sua popularização, principalmente nas décadas de 80 e 90, a tradicional, segura e robusta arquitetura centralizada personificada nos mainframes, deu lugar rapidamente a uma arquitetura descentralizada que além correr atrás do prejuízo quando se analisa os benefícios já consolidados da arquitetura centralizada, ainda se volta a partir da década de 90 para a disponibilização de serviços ditos de nova geração, caracterizados por tratar além da mídia textual, ainda áudio e vídeo. Comparativamente ao ambiente complexo, amplo e heterogêneo observado na Internet, o ambiente de uma Intranet é bem comportado para se disponibilizar os serviços de nova geração uma vez que o universo de equipamentos e tecnologias envolvidos, para se garantir os requisitos necessários aos novos serviços, é previsível. A presente dissertação possui como principal contribuição o desenvolvimento de uma infra-estrutura de transcodificação e a implementação de uma aplicação para distribuição de áudio em uma rede corporativa (Intranet) adaptável ao estado da rede que se utiliza da infra-estrura elaborada. Embora focada na distribuição apenas da mídia de áudio, a infraestrutura proposta pode evoluir para permitir a criação de aplicações de distribuição de áudio e vídeo e ainda áudio e vídeo conferências. Usa-se o protocolo de Aplicação/Transporte RTP - Real Time Transport Protocol, disponível em vários formatos de áudio existentes na API JMF - Java Media Framework, além de se abordar as necessidades a nível de rede, tais como Multicast e mecanismos de priorização de quadros e pacotes nas camadas 2 e 3 do modelo de referência OSI (i.e. IEEE 802.1P e DiffServ). São identificados quatro Casos de Uso básicos na infra-estrutura proposta: Transmitir ao Transcodificador; Iniciar Sessões de Áudio no Transcodificador; Receber Stream de Áudio RTP no Receptor; e, Trocar a Sessão Multicast de Recepção. Os três primeiros são especificados, implementados e validados; o quarto, é apenas especificado, sendo a sua implementação e validação em cenário com filas diferenciadas de priorização de pacotes na camada de rede, um trabalho futuro. O trabalho explora o conceito de transcodificação como ideal para lidar com a heterogeneidade de recursos de rede existentes em uma Intranet
2

Komunikační klient v JavaMe / JavaME communication client

Svoboda, Pavel January 2009 (has links)
This diploma thesis deals with developing multimedia applications on Java Micro Edition platform. The aim of this work is to design and implement the application which could establish a call between two users. The first part of the work describes J2ME platform, its two configurations and profiles. Next part is focused on Session Initiation Protocol and Real-time Transport Protocol. The application design consists of choosing the suitable virtual machine JVM, SIP and RTP libraries. The main part of this work describes application structure, graphic user interface and installation packages creating. It also shows a way of customizing the media stack - Java Media Framework, version Cross Platform.
3

Multimediální služby založené na IMS / IMS based multimedia services

Novotný, Roman January 2010 (has links)
Submitted work describes IMS (IP Multimedia Subsystem) and focuses on services offered by this technology. In the theoretical part of the thesis is a detailed description of the IMS architecture, in the terms of its components and interfaces. Also a SIP protocol is described as the main signalling protocol, which represents the basis of the IMS architecture and RTP protocol that allows the transmission of multimedia data in real-time. Further, there are provided services that the IMS system brings. The practical part of the work describes the design and implementation of services within the IMS. It is a client-server service. Server part of the system is implemented as a SIP Servlet application. It has access to the MySQL database, which stores records of multimedia files. Client registered in the IMS network can communicate with the SIP servlet sending a request for information on such files. Following this communication, the client can play the file in his online ICP application, or download it into his PC and then play. To play and view multimedia fines is used JMF. The service is implemented in a development environment SDS Ericsson 4.1, which allows simulation of the IMS network.
4

Σύστημα VoIP με χρήση δορυφορικών επικοινωνιών

Τσουκαλής, Αχιλλέας 03 March 2008 (has links)
Για περιοχές με μικρή ή καθόλου επίγεια τηλεπικοινωνιακή υποδομή η επικοινωνία μέσω δορυφορικής σύνδεσης είναι μία αποτελεσματική και λογικού κόστους λύση. Το DVB-RCS (Digital Video Broadcast - Return Channel System) standard επιτρέπει την αμφίδρομη μετάδοση IP κίνησης πάνω από δορυφορικό κανάλι κάνοντας έτσι δυνατή την VoIP επικοινωνία μέσω δορυφόρου. Η μετάδοση VoIP μέσω μιας δορυφορικής σύνδεσης όμως, μπορεί προκαλέσει ορισμένα προβλήματα τόσο στην ίδια την ποιότητα της VoIP συνδιάλεξης, όσο και σε άλλες υπηρεσίες που ενδεχομένως μοιράζονται με τις VoIP ροές το διαθέσιμο εύρος ζώνης του δορυφορικού καναλιού. Η διπλωματική αυτή εργασία επικεντρώνεται στη μελέτη και ανάπτυξη end to end μηχανισμών ελέγχου του ρυθμού κωδικοποίησης και μετάδοσης στα VoIP τερματικά τηλέφωνα ανάλογα με τον βαθμό συμφόρησης του καναλιού, ώστε να αξιοποιείται πιο αποτελεσματικά το διαθέσιμο εύρος ζώνης του καναλιού μετάδοσης (δορυφορικού ή μη), να αντιμετωπίζεται ενδεχόμενη συμφόρησή του, να βελτιώνεται η ποιότητα της συνδιάλεξης, να αυξάνεται ο μέγιστος αριθμός των ταυτόχρονων συνδιαλέξεων για ένα δεδομένο εύρος ζώνης και να εξασφαλίζεται ο δίκαιος διαμοιρασμός του διαθέσιμου εύρους ζώνης ανάμεσα στις TCP και VoIP ροές. Σε αυτά τα πλαίσια αρχικά εξετάζονται οι παράγοντες που επιδρούν στην ποιότητα της VoIP υπηρεσίας, παρουσιάζονται ορισμένοι τρόποι για την αξιολόγηση της και γίνεται μελέτη της χρήσης του TFRC (TCP- Friendly Rate Control) μηχανισμού σε VoIP εφαρμογές. Προτείνεται ένας νέος, πολύ απλός στην υλοποίηση, μηχανισμός ελέγχου μετάδοσης για VoIP ροές, που αντιθέτως με τους υπάρχοντες μηχανισμούς, στοχεύει ταυτόχρονα στην βελτίωση της ποιότητας της συνδιάλεξης και στην φιλικότητα προς τις TCP ροές. Αναλύονται επίσης η δομή, η λειτουργία και ορισμένα θέματα υλοποίησης του VoIP τερματικού συστήματος που αναπτύχθηκε (σε μορφή λογισμικού) στα πλαίσια αυτής της διπλωματικής εργασίας και που υλοποιεί τον προτεινόμενο μηχανισμό ελέγχου μετάδοσης. / For areas with limited or no terrestrial telecommunication infrastructure, communication via satellite is a cost effective alternative. The DVB-RCS (Digital Video Broadcast - Return Channel System) standard supports the bidirectional transmission of IP data, making VoIP communication via satellite possible. However, the transmission of VoIP through a satellite link raises some serious issues concerning the VoIP quality of service and the plain functionality of other applications that might share the same link with the VoIP flows. This thesis focuses in the study and the development of end to end rate control mechanisms in VoIP terminal phones, which mechanisms can enhance the utilization of the available channel (satellite or not) bandwidth, tackle potential congestion, enhance the conversational quality, increase the maximum number of simultaneous VoIP conversations for a given bandwidth, and ensure that bandwidth is being fairly shared between TCP and VoIP flows. The factors that affect the VoIP quality of service, the ways this quality can be evaluated and the use of TFRC (TCP- Friendly Rate Control) mechanism in VoIP are discussed. A new, easy-to-implement rate control mechanism which, in contrast to the existing mechanisms, targets on both conversational quality enhancement and TCP friendliness is proposed. Finally, some implementation issues, regarding the VoIP terminal software system that it has been developed as part of this thesis and implements the proposed rate control mechanism, are discussed.
5

Návrh nové metody pro stereovidění / Design of a New Method for Stereovision

Kopečný, Josef January 2008 (has links)
This thesis covers with the problems of photogrammetry. It describes the instruments, theoretical background and procedures of acquiring, preprocessing, segmentation of input images and of the depth map calculating. The main content of this thesis is the description of the new method of stereovision. Its algorithm, implementation and evaluation of experiments. The covered method belongs to correlation based methods. The main emphasis lies in the segmentation, which supports the depth map calculation.

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