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Dissertations / Theses on the topic 'Streaming audio'
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Čeněk, Radek. "Audio/Video streaming." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220405.
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This thesis introduces the reader in detail with the problem of audio / video streaming overtheInternet.IntroducestechnologiesMySQL,PHP,JavaScriptandffmpegfurthermore. There is little research which examines presented similar solutions. Creating video library program shows the complexity of the problem and its possible solution. The work also finding how is compression difficult for the server.
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Afzaal, Qasim, and Usman Ahmad. "Audio Video Streaming Solution for Bambuser." Thesis, Umeå universitet, Institutionen för datavetenskap, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-58494.
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Audio/Video streaming has widely been used in different applications but the social communication applications have especially raised its usage. The aim of this thesis is to design and develop an improved Audio/Video streaming solution for a Swedish company Bambuser and can easily be extended with new features where necessary. Currently Bambuser is using the Flash Media Server (FMS) for streaming the media, but it is license based and adds the extra cost to the company's budget. It does not support a wide range of platforms (e.g. OpenBSD and various Linux distributions) and also has limited options for the streaming. There is no real time monitoring and controlling functionality, which can show the status of essential services to the user, needed for the streaming (for example if the camera is working, microphone is turned on, battery power status. etc.). In order to solve these issues the GStreamer is used, which is an Open source multimedia streaming framework. The GStreamer environment was tested on different Linux distributions. The research and implementation includes the creation of the streaming pipeline and analyzing which options (i.e. GStreamer elements and plugins) are required to stream the media. It also includes the testing of different pipeline parameters (for example video rate, audio rate etc.) and noting their effects in a real working environment. Python binding with GStreamer is used to have better control over the pipeline. Another requirement of this project was to add the functionality of monitoring and control that shows the status of essential services to the user. Implementation of this part is done by using server and client side coding. Further improvements and suggestions are also proposed in this report.
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Bansal, Deepak 1978. "Congestion control for streaming video and audio applications." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/86577.
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Marks, Stuart Keith. "Joint Source/Channel Coding for Mobile Audio Streaming." Thesis, Griffith University, 2006. http://hdl.handle.net/10072/367550.
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This thesis investigates mechanisms for providing audio streaming on portable devices over the Wireless Internet. This is no trivial task, as Wireless Internet channels are highly erratic and experience highly variable bandwidth and large bursty error rates; and portable devices have limited storage and computational power. To provide an audio stream of suitable performance highly adaptive mechanisms must be utilised within session layer protocols. The core of this thesis investigates the use of a joint/source channel coder which streams the audio as a set of autonomous audio objects. This is a considerable shift away from the traditional frame-based streaming paradigm, but is warranted as such a coder has the flexibility to accommodate large error bursts and bandwidth variations while producing decoded audio with respectable perceived quality. The goal is to provide an audio coder that is able to stream audio over a channel with low bandwidth and high-error rates. Such an audio coder does not currently exist.
The majority of the material presented in this thesis is devoted to the encoding of audio into a set of autonomous objects. The approach taken is to build high-level objects from mid-level items generated from sinusoidal, transient and noise modelling. The high-level objects are autonomous, flexible, have a low bitrate and a high perceptual-relevance. The penultimate chapter of this thesis demonstrates that the streaming of these autonomous audio objects is superior to the streaming of audio in the traditional frame-based manner. As object-based audio streaming can both automatically mask the effects of packet-loss without the need for an expensive error-concealment scheme at the decoder, and scale bitrate, quality, complexity and memory in a graceful and natural manner.Thesis (PhD Doctorate)
Doctor of Philosophy (PhD)
School of Information and Communication Technology
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Claesén, Daniel. "MCapture; An Application Suite for Streaming Audio over Networks." Thesis, Linköping University, Department of Computer and Information Science, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-4387.
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The purpose of this thesis is to develop software to stream input and output audio from a large number of computers in a network to one specific computer in the same network. This computer will save the audio to disk. The audio that is to be saved will consist mostly of spoken communication. The saved audio is to be used in a framework for modeling and visualization.
There are three major problems involved in designing a software to fill this purpose: recording both input and output audio at the same time, efficiently receiving multiple audio-streams at once and designing an interface where finding and organizing the computers to record audio from is easy.
The software developed to solve these problems consists of two parts; a server and a client. The server captures the input (microphone) and output (speaker) audio from a computer. To capture the output and input audio simultaneously an external application named Virtual Audio Cable (VAC) is used. The client connects to multiple servers and receives the captured audio. Each one of the client’s server-connections is handled by its own thread. To make it easy to find available servers an Automatic Server Discovery System has been developed. To simplify the organization of the servers they are displayed in a tree-view specifically designed for this purpose.
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Sahd, Curtis Lee. "Bluetooth audio and video streaming on the J2ME platform." Thesis, Rhodes University, 2010. http://hdl.handle.net/10962/d1006521.
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With the increase in bandwidth, more widespread distribution of media, and increased capability of mobile devices, multimedia streaming has not only become feasible, but more economical in terms of space occupied by the media file and the costs involved in attaining it. Although much attention has been paid to peer to peer media streaming over the Internet using HTTP and RTSP, little research has focussed on the use of the Bluetooth protocol for streaming audio and video between mobile devices. This project investigates the feasibility of Bluetooth as a protocol for audio and video streaming between mobile phones using the J2ME platform, through the analysis of Bluetooth protocols, media formats, optimum packet sizes, and the effects of distance on transfer speed. A comparison was made between RFCOMM and L2CAP to determine which protocol could support the fastest transfer speed between two mobile devices. The L2CAP protocol proved to be the most suitable, providing average transfer rates of 136.17 KBps. Using this protocol a second experiment was undertaken to determine the most suitable media format for streaming in terms of: file size, bandwidth usage, quality, and ease of implementation. Out of the eight media formats investigated, the MP3 format provided the smallest file size, smallest bandwidth usage, best quality and highest ease of implementation. Another experiment was conducted to determine the optimum packet size for transfer between devices. A tradeoff was found between packet size and the quality of the sound file, with highest transfer rates being recorded with the MTU size of 668 bytes (136.58 KBps). The class of Bluetooth transmitter typically used in mobile devices (class 2) is considered a weak signal and is adversely affected by distance. As such, the final investigation that was undertaken was aimed at determining the effects of distance on audio streaming and playback. As can be expected, when devices were situated close to each other, the transfer speeds obtained were higher than when devices were far apart. Readings were taken at varying distances (1-15 metres), with erratic transfer speeds observed from 7 metres onwards. This research showed that audio streaming on the J2ME platform is feasible, however using the currently available class of Bluetooth transmitter, video streaming is not feasible. Video files were only playable once the entire media file had been transferred.
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Moscote, Freire Ariana. "Tuning into you: personalized audio streaming services and their remediation of radio." Thesis, McGill University, 2008. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=18799.
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This thesis sets out to study and contextualize the discourses and structures surrounding an emerging form of music promotion, distribution, and consumption, namely personalized audio streaming driven by music recommender systems. It finds that services including Last.fm, Yahoo! LAUNCHcast, Pandora.com, and Radiolibre.ca rely on a discursive construct of 'radioness' in order to frame and legitimize their activities to their listeners, to the industry, and to the State. Simultaneously, these emergent media forms claim to surpass conventional radio by offering users agency over their listening experiences and promising artists more equal and relevant access to the 'airwaves,' with potentially revolutionary consequences. The argument of this thesis is that it is a particular conception of 'radio' that is at play in these articulations, and that furthermore, we should recognize the structuring impact of the regulatory context, industrial practices, and technological design of these personalized music streaming systems on their development and implementation, rather than take their promotional rhetoric at face value.Ce mémoire a pour but d'étudier et de mettre en contexte les discours et les structures liés à une forme naissante de promotion, de distribution et de consommation de musique, soit des services de programmation musicale personnalisés par des systèmes de recommandation. Il y est démontré que ces services, dont Yahoo! LAUNCHcast, Last.fm, Pandora.com et Radiolibre.ca s'appuient sur une notion discursive de ce qu'est la radio afin d'encadrer et de rendre légitimes leurs activités aux yeux de leurs membres, de l'industrie musicale et de l'État. Ils rejettent simultanément cette même notion lorsqu'ils clament dépasser la radio conventionnelle en offrant à leurs usagers de contrôler leurs expériences auditives, et aux artistes émergents un accès plus égal et pertinent aux « ondes », aux conséquences potentiellement révolutionnaires. La thèse présentée est qu'un concept particulier de la « radio » est exploité dans ces situations, et que la valeur apparente des promesses des services personnalisés ne correspond pas nécessairement à la réalité, définie par l'impact structurel de la réglementation, des pratiques de l'industrie et de la technologie sur le développement de ces nouveaux médias.
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Buffet, Julien. "Techniques de protection contre les erreurs pour le streaming audio sur IP." Châtenay-Malabry, Ecole centrale de Paris, 2002. http://www.theses.fr/2002ECAP0857.
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Lorsque l'on transfère en temps réel des données audio au-dessus du service "best effort" donné sur l'Internet, les pertes de données non contrôlées peuvent dégrader significativement la qualité d'écoute. Pour améliorer cette qualité, une politique de protection contre les erreurs est nécessaire. Les techniques de protection contre les erreurs se répartissent en deux types : celles qui dépendent du codage et celles qui en sont indépendantes. Les techniques dépendant du codage s'appuient sur les propriétés du codage sous jacent pour la protection contre les erreurs. Une technique de protection contre les erreurs adaptée au codage MPEG-4 Audio combinant une adaptation au débit, un entrelacement de paquets et une récupération d'erreurs s'appuyant sur la FEC a été développée. Pour l'adaptation de débit et la récupération d'erreur, les propriétés de granularité et de "scalabilité" du flux MPEG-4 Audio sont utilisées. Un mécanisme d'entrelacement s'adaptant au processus de perte est mis en oeuvre. La combinaison de ces mécanismes donne lieu à un protocole "TCP fiendly" pour transferer en temps réel des données MPEG-4 au-dessus d'IP. Il a été impléménté pour l" streaming unicast. La plupart des technique FEC indépendantes du codage sont des adaptations de la théorie générale du codage au cas particulier des erreurs de streaming. Les codes correcteurs tel que les codes Hamming ou les codes Reed-Solomon peuvent détecter et corriger une ou plusieurs erreurs apparaissant sporadiquement dans un canal. Mais dans le streaming Internet qui nous concerne, les erreurs ont déjà été détectées par les protocoles bas niveau, le seul problème est la correctio. C'est un problème bien plus facile que celui de la détection-correction et nous n'avons pas besoin d'utiliser la théorie des codes correcteurs pour y répondre. Une nouvelle méthode dédiée aux problèmes de streaming a été développée en utilisant la théorie des systèmes linéaires sur des corps ou des anneaux finis.
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Chiappetta, Marco. "Composizione musicale e streaming peer-to-peer con web audio e webrtc." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2013. http://amslaurea.unibo.it/6200/.
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La tesi descrive lo sviluppo di un'applicazione web per comporre musica tramite la tecnica del "live looping" che fornisce anche la possibilità di effettuare lo streaming di ciò che si crea in tempo reale e in maniera peer-to-peer. L'applicazione in oggetto (chiamata WebLooper) fa uso di due tecnologie web emergenti in ambito multimediale: Web Audio e WebRTC, attualmente in attesa di diventare standard W3C.
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Mathews, Abraham. "Smart Home Based Li-Fi System : Stereo Audio & Image Streaming by Visible light." Thesis, Mittuniversitetet, Avdelningen för elektronikkonstruktion, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:miun:diva-32835.
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To light up the world of technology, where wireless communication has bloomed to a great extend which requires a lot of data to be transmitted and received every fraction of the second a new era is coming. Electro-magnetic waves i.e., radio waves are the main way to transmit wireless data but certain limitations are there because radio waves can only support less bandwidth because of compact spectrum availability and intrusions. Visible Light Communication (VLC) has come to take way those issues. The new technology Li-Fi which stands for Light-Fidelity is a new kind of wireless communication system which uses light waves as a medium instead of radio frequency electromagnetic waves. This pro-ject presents an eco-friendly data communication system through visible light which consists of LEDs that transmit audio signals and sensor data to the receiver. A connection protection mechanism that co-operates with wireless network and visible light communication to achieve relia-bility and performance overcoming the drawbacks from the pre-existing system is proposed here.
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Timoncini, Riccardo. "Streaming audio e video nei sistemi Peer-To-Peer TV: il caso Sopcast P2PTV." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2012. http://amslaurea.unibo.it/3670/.
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La tesi si propone di affrontare il tema del Live Streaming in sistemi P2P con particolare riferimento a Sopcast, un applicativo di P2PTV. Viene fatto un ricorso storico riguardo alla nascita dello streaming e al suo sviluppo, vengono descritte le caratteristiche, il protocollo di comunicazione e i modelli più diffusi per il live streaming P2P. Inoltre si tratterà come viene garantita la qualità del servizio e valutate le performance di un servizio P2PTV.
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Foulkes, Philip James. "An investigation into the control of audio streaming across networks having diverse quality of service mechanisms." Thesis, Rhodes University, 2012. http://hdl.handle.net/10962/d1004865.
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The transmission of realtime audio data across digital networks is subject to strict quality of service requirements. These networks need to be able to guarantee network resources (e.g., bandwidth), ensure timely and deterministic data delivery, and provide time synchronisation mechanisms to ensure successful transmission of this data. Two open standards-based networking technologies, namely IEEE 1394 and the recently standardised Ethernet AVB, provide distinct methods for achieving these goals. Audio devices that are compatible with IEEE 1394 networks exist, and audio devices that are compatible with Ethernet AVB networks are starting to come onto the market. There is a need for mechanisms to provide compatibility between the audio devices that reside on these disparate networks such that existing IEEE 1394 audio devices are able to communicate with Ethernet AVB audio devices, and vice versa. The audio devices that reside on these networks may be remotely controlled by a diverse set of incompatible command and control protocols. It is desirable to have a common network-neutral method of control over the various parameters of the devices that reside on these networks. As part of this study, two Ethernet AVB systems were developed. One system acts as an Ethernet AVB audio endpoint device and another system acts as an audio gateway between IEEE 1394 and Ethernet AVB networks. These systems, along with existing IEEE 1394 audio devices, were used to demonstrate the ability to transfer audio data between the networking technologies. Each of the devices is remotely controllable via a network neutral command and control protocol, XFN. The IEEE 1394 and Ethernet AVB devices are used to demonstrate the use of the XFN protocol to allow for network neutral connection management to take place between IEEE 1394 and Ethernet AVB networks. User control over these diverse devices is achieved via the use of a graphical patchbay application, which aims to provide a consistent user interface to a diverse range of devices.
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Olaleye, Olufunke I. "Symbiotic Audio Communication on Interactive Transport." Kent State University / OhioLINK, 2007. http://rave.ohiolink.edu/etdc/view?acc_num=kent1176438067.
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Janovič, Jakub. "Webový prohlížeč audio/video záznamů přednášek: převod prohlížeče na MySQL databázi." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2010. http://www.nusl.cz/ntk/nusl-237118.
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This project deals with a web-based lecture browser, whose goal is to simplify the gaining of knowledge with the use of multimedia. It presents an existing lecture browser that was created for a diploma thesis at FIT VUT Brno. Demonstrated are the technologies that are used and which will be used to migrate the browser to a MySQL database and to develop a transcription module for speeches. The reader will be acquainted with an analysis and model of the new application. Furthermore, implementation methods for development and subsequent testing are discussed. At the end of the project is a conclusion about the future development of web-based lecture browsers.
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Haandel, Johan Cavalcanti Van. "Formatos emergentes de criação e transmissão de áudio online: a construção do webcasting sonoro." Pontifícia Universidade Católica de São Paulo, 2009. https://tede2.pucsp.br/handle/handle/5224.
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In the last years the transmission of audio is being transformed for the new technologies of the digital support, which had generated new processes and products, among them is the sonorous webcasting. This work investigates the creation of content of the four existing formats of sonorous webcasting: web radio, playlist, on demand audio and the audio portal, which comprises the corpus of the research. As methodological strategy, firstly, examples of the sonorous webcasting formats and of others transmission processes for the digital support had been observed, to distinguish them; the second step was to observe the assembly of audio of the sonorous webcasting formats and the third step was to investigate its graphical interfaces. With basis in the methodological strategy, the research presents itself in three stages: in the first one, it was outlined a mapping of the existing digital transmissions of audio and a classification of sonorous webcasting formats, in which the main authors of reference are Trigo-De-Souza and Kischinhevsky, among others; in the second stage, the construction of the transmission of sonorous webcasting through audio assembly is investigated with data based on Ferraretto, Cyro César, Barbosa Filho and McLeish, among others; and in third stage, it is investigated the construction of the transmission of sonorous webcasting through the graphical interface, where the hypermedia interfaces study referring data are based on Beiguelman, Manovich and Johnson. It is concluded that sonorous webcasting consists of a transmission process of audio that does not have to be confounded with radio and it sets up a new type of content reading, mediated by texts and images, implying, therefore, in studies of audio assembly and graphical interface
Nos últimos anos a transmissão de áudio está sendo transformada pelas novas tecnologias do suporte digital, as quais geraram novos processos e produtos, entre eles o webcasting sonoro. A presente pesquisa investiga a criação de conteúdo dos quatro formatos existentes do webcasting sonoro: a web rádio, a playlist, o áudio on demand e o portal de áudio, os quais compõem o corpus da pesquisa. Como estratégia metodológica, primeiro foram observados exemplos dos formatos do webcasting sonoro e de outros processos de transmissão pelo suporte digital, para particularizá-los; o segundo passo foi observar a montagem de áudio dos formatos do webcasting sonoro e o terceiro passo foi investigar as suas interfaces gráficas. Baseada na estratégia metodológica, a pesquisa se apresenta em três etapas: na primeira, há um mapeamento das transmissões digitais de áudio existentes e uma classificação dos formatos do webcasting sonoro, em que os principais autores de referência são Trigo-De-Souza e Kischinhevsky, entre outros; na segunda, é investigada a construção da transmissão do webcasting sonoro através da montagem de áudio, com dados baseados em Ferraretto, Cyro César, Barbosa Filho e McLeish, entre outros; e na terceira, é investigada a construção da transmissão do webcasting sonoro através da interface gráfica, em que os dados referentes ao estudo de interfaces da hipermídia são baseados em Beiguelman, Manovich e Johnson. Conclui-se que o webcasting sonoro consiste em um processo de transmissão de áudio que não deve ser confundido com rádio e que institui um novo tipo de leitura de conteúdo, mediado por textos e imagens, implicando, por isso, em estudos de montagem sonora e de interface gráfica
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Igumbor, Osedum Peter. "An investigation of protocol command translation as a means to enable interoperability between networked audio devices." Thesis, Rhodes University, 2014. http://hdl.handle.net/10962/d1011128.
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Digital audio networks allow multiple channels of audio to be streamed between devices. This eliminates the need for many different cables to route audio between devices. An added advantage of digital audio networks is the ability to configure and control the networked devices from a common control point. Common control of networked devices enables a sound engineer to establish and destroy audio stream connections between networked devices that are distances apart. On a digital audio network, an audio transport technology enables the exchange of data streams. Typically, an audio transport technology is capable of transporting both control messages and audio data streams. There exist a number of audio transport technologies. Some of these technologies implement data transport by exchanging OSI/ISO layer 2 data frames, while others transport data within OSI/ISO layer 3 packets. There are some approaches to achieving interoperability between devices that utilize different audio transport technologies. A digital audio device typically implements an audio control protocol, which enables it process configuration and control messages from a remote controller. An audio control protocol also defines the structure of the messages that are exchanged between compliant devices. There are currently a wide range of audio control protocols. Some audio control protocols utilize layer 3 audio transport technology, while others utilize layer 2 audio transport technology. An audio device can only communicate with other devices that implement the same control protocol, irrespective of a common transport technology that connects the devices. The existence of different audio control protocols among devices on a network results in a situation where the devices are unable to communicate with each other. Furthermore, a single control application is unable to establish or destroy audio stream connections between the networked devices, since they implement different control protocols. When an audio engineer is designing an audio network installation, this interoperability challenge restricts the choice of devices that can be included. Even when audio transport interoperability has been achieved, common control of the devices remains a challenge. This research investigates protocol command translation as a means to enable interoperability between networked audio devices that implement different audio control protocols. It proposes the use of a command translator that is capable of receiving messages conforming to one protocol from any of the networked devices, translating the received message to conform to a different control protocol, then transmitting the translated message to the intended target which understands the translated protocol message. In so doing, the command translator enables common control of the networked devices, since a control application is able to configure and control devices that conform to different protocols by utilizing the command translator to perform appropriate protocol translation.
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Eriksson, Mattias. "Speech recognition availability." Thesis, Linköping University, Department of Computer and Information Science, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2651.
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This project investigates the importance of availability in the scope of dictation programs. Using speech recognition technology for dictating has not reached the public, and that may very well be a result of poor availability in today’s technical solutions.
I have constructed a persona character, Johanna, who personalizes the target user. I have also developed a solution that streams audio into a speech recognition server and sends back interpreted text. Johanna affirmed that the solution was successful in theory.
I then incorporated test users that tried out the solution in practice. Half of them do indeed claim that their usage has been and will continue to be increased thanks to the new level of availability.
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Uttermalm, Johan. "Audio streaming on top of 802.11n in an IoT context : An implementation along with a literature study of wireless IoT standards." Thesis, Karlstads universitet, Institutionen för matematik och datavetenskap, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-42961.
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The Internet of Things (IoT) is a concept that revolves around ordinary devices that are connected to the internet for extended control and ease of use. Altran, a company dealing in high technology and innovation consultancy, predicts a large growth in business opportunities in the IoT area in the coming years, and therefore wants to invest in knowledge about the Internet of Things. Altran wanted a report that described popular wireless IoT communication technologies along with a proposal for a general IoT communication platform or base that could be used to implement many of these technologies. Additionally, an audio streaming application were to be implemented on the proposed platform to validate its credibility. The project resulted in a report on 6 different wireless IoT technologies: Z-wave, ZigBee, Thread, Bluetooth, 802.11n, and 802.11ah. A hardware and software base was proposed that could implement 4 of 6 of these technologies. This base was the Raspberry Pi 2 along with the Raspbian Jessie operating system. Finally an audio streaming system that could stream data to a set of smart Speaker nodes over wireless links based on IEEE 802.11n was implemented on the proposed base.
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Dibley, James. "An investigation of the XMOS XSl architecture as a platform for development of audio control standards." Thesis, Rhodes University, 2014. http://hdl.handle.net/10962/d1011789.
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This thesis investigates the feasiblity of using a new microcontroller architecture, the XMOS XS1, in the research and development of control standards for audio distribution networks. This investigation is conducted in the context of an emerging audio distribution network standard, Ethernet Audio/Video Bridging (`Ethernet AVB'), and an emerging audio control standard, AES-64. The thesis describes these emerging standards, the XMOS XS1 architecture (including its associated programming language, XC), and the open-source implementation of an Ethernet AVB streaming audio device based on the XMOS XS1 architecture. It is shown how the XMOS XS1 architecture and its associated features, focusing on the XC language's mechanisms for concurrency, event-driven programming, and integration of C software modules, enable a powerful implementation of the AES-64 control standard. Feasibility is demonstrated by the implementation of an AES-64 protocol stack and its integration into an XMOS XS1-based Ethernet AVB streaming audio device, providing control of Ethernet AVB features and audio hardware, as well as implementations of advanced AES-64 control mechanisms. It is demonstrated that the XMOS XS1 architecture is a compelling platform for the development of audio control standards, and has enabled the implementation of AES-64 connection management and control over standards-compliant Ethernet AVB streaming audio devices where no such implementation previously existed. The research additionally describes a linear design method for applications based on the XMOS XS1 architecture, and provides a baseline implementation reference for the AES-64 control standard where none previously existed.
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Soldi, Giovanni. "Diarisation du locuteur en temps réel pour les objets intelligents." Electronic Thesis or Diss., Paris, ENST, 2016. http://www.theses.fr/2016ENST0061.
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La diarisation du locuteur en temps réel vise à détecter "qui parle maintenant" dans un flux audio donné. La majorité des systèmes de diarisation en ligne proposés a mis l'accent sur des domaines moins difficiles, tels que l’émission des nouvelles et discours en plénière, caractérisé par une faible spontanéité. La première contribution de cette thèse est le développement d'un système de diarisation du locuteur complètement un-supervisé et adaptatif en ligne pour les données de réunions qui sont plus difficiles et spontanées. En raison des hauts taux d’erreur de diarisation, une approche semi-supervisé pour la diarisation en ligne, ou les modèles des interlocuteurs sont initialisés avec une quantité modeste de données étiquetées manuellement et adaptées par une incrémentale maximum a-posteriori adaptation (MAP) procédure, est proposée. Les erreurs obtenues peuvent être suffisamment bas pour supporter des applications pratiques. La deuxième partie de la thèse aborde le problème de la normalisation phonétique pendant la modélisation des interlocuteurs avec petites quantités des données. Tout d'abord, Phone Adaptive Training (PAT), une technique récemment proposé, est évalué et optimisé au niveau de la modélisation des interlocuteurs et dans le cadre de la vérification automatique du locuteur (ASV) et est ensuite développée vers un système entièrement un-supervise en utilisant des transcriptions de classe acoustiques générées automatiquement, dont le nombre est contrôlé par analyse de l'arbre de régression. PAT offre des améliorations significatives dans la performance d'un système ASV iVector, même lorsque des transcriptions phonétiques précises ne sont pas disponiblesOn-line speaker diarization aims to detect “who is speaking now" in a given audio stream. The majority of proposed on-line speaker diarization systems has focused on less challenging domains, such as broadcast news and plenary speeches, characterised by long speaker turns and low spontaneity. The first contribution of this thesis is the development of a completely unsupervised adaptive on-line diarization system for challenging and highly spontaneous meeting data. Due to the obtained high diarization error rates, a semi-supervised approach to on-line diarization, whereby speaker models are seeded with a modest amount of manually labelled data and adapted by an efficient incremental maximum a-posteriori adaptation (MAP) procedure, is proposed. Obtained error rates may be low enough to support practical applications. The second part of the thesis addresses instead the problem of phone normalisation when dealing with short-duration speaker modelling. First, Phone Adaptive Training (PAT), a recently proposed technique, is assessed and optimised at the speaker modelling level and in the context of automatic speaker verification (ASV) and then is further developed towards a completely unsupervised system using automatically generated acoustic class transcriptions, whose number is controlled by regression tree analysis. PAT delivers significant improvements in the performance of a state-of-the-art iVector ASV system even when accurate phonetic transcriptions are not available
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Žižka, Josef. "Webový prohlížeč přednášek." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2009. http://www.nusl.cz/ntk/nusl-236668.
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This thesis deals with a web-based lecture browser. Its goal is to facilitate the access to information with the use of modern speech and multimedia technologies. Technologies used for this browser are discussed. Video recordings play a very important role in the browser, and therefore the big portion of this work is aimed at the digital video and methods of its delivery using streaming servers. Solutions of similar multimedia browsers are mentioned. The reader is acquainted with the browser design. This includes describing the various components of the browser and how their mutual synchronization is done. The final version of the browser is introduced and the problems that occurred during the development process and deployment into service are mentioned. In the conclusion of this work the future development of the web-based lecture browser is discussed.
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Bonneau-Crépin, Charlotte. "Les pratiques de réception comme marques de capital symbolique : le cas du mass-streaming chez les fans de musique populaire coréenne." Master's thesis, Université Laval, 2020. http://hdl.handle.net/20.500.11794/66325.
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Les transformations engendrées par l’avènement des réseaux sociaux et des plateformes d’écoute musicale numériques ont vu naître de nouvelles pratiques de réception musicale. Ce mémoire de maîtrise porte sur l’une d’entre elles, exclusive aux admirateurs de musique populaire coréenne. Plus précisément, il traite d’une activité inédite du nom de mass-streaming, populaire au sein de la communauté d’admirateurs de K-pop (Korean Pop). Cette pratique consiste en l’utilisation de méthodes visant à faire augmenter le plus rapidement possible le nombre de lectures associées à une chanson sur les plateformes d’écoute musicale numériques. L’objectif de ce mémoire est de déterminer la mesure dans laquelle le mass-streaming influe sur l’industrie musicale et les admirateurs qui s’y adonnent. Plusieurs modèles théoriques sont employés afin de formuler une réponse à ce questionnement, notamment ceux des mondes de l’art (Becker 1984), de capital symbolique (Bourdieu 1994) et de capital sous-culturel (Thornton 2013). En guise d’étude de cas, c’est le boyband coréen Beyond The Scene (BTS) et sa communauté de fans qui sont observés. La collecte d’informations a été effectuée en deux phases. La première, qui s’est déroulée entièrement sur Twitter, a consisté en la création d’une collection de tweets sur le sujet du mass-streaming rédigés par des admirateurs. La seconde a été la tenue d’entrevues individuelles semi-dirigées avec des professionnels de l’industrie musicale. Afin de présenter un portrait complet du mass-streaming, une description du contexte historique menant à son avènement est effectuée, de même que la présentation des méthodes employées par les admirateurs ainsi que les causes de l’existence du mass-streaming. Finalement, les différents impacts de la pratique sont envisagés en tant que différents types de capital symbolique (économique, social, culturel, sous-culturel) afin de représenter le plus fidèlement possible les répercussions de cette activité sur les différents partis qui y sont impliqués.The transformations brought upon by the rise of social networks and music streaming platforms have caused the birth of new ways to consume music. This thesis is about one of them, exclusive to fans of Korean popular music. More precisely, it is about a novel activity called mass-streaming, popular amongst the K-pop (Korean Pop) fan community. This practice consists in the use of specific methods that are geared towards the quickest rise possible in streams associated to a particular song on streaming platforms. The aim of this thesis is to determine the measure in which mass-streaming has an influence on the music industry and the fan community. Different conceptual models are applied in order to formulate an answer to that question, namely those of Monde des Arts (Becker 1984), symbolic capital (Bourdieu 1994) and subcultrual capital (Thornthon 2013). As a case study, it is Korean boyband Beyond The Scene (BTS) and its fan community that are studied. Data collection took place in two distinct phases. The first, which took place entirely on microblogging site Twitter, consisted in the creation of a collection of tweets about mass-streaming. The second was made up of a series of semi-directed interviews with music industry professionals. In order to present a complete picture of what mass-streaming is, a description of the historic context and causes that lead to its emergence is made, as well as the presentation of the methods used by fans. Finally, the different impacts of the studied practice are considered as different types of symbolic capital (economic, social, cultural, subcultural) in order to represent as precisely as possible the potential repercussions of this new practice on all parties involved in its functioning.
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LINDSTRÖM, CARL, and TORA BYGREN. "Att Lyssna På Boken Som Lyssnar På Dig : Hur användaren och dess beteende integreras i produktutvecklingen av ljudbokstjänster." Thesis, KTH, Skolan för industriell teknik och management (ITM), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-279780.
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De senaste 10 åren har bokbranschen genomgått en omfattande digitalisering. Ett av resultaten från denna digitalisering är framväxten en helt ny marknad för ljudböcker och ur denna marknad har ett flertal ljudbokstjänster vuxit fram. Dessa ljudbokstjänster erbjuder sina användare en produkt i form av en applikation, i produkten får de tillgång till en obegränsad mängd streamat innehåll, huvudsakligen ljudböcker. Produktutvecklingen av dessa ljudbokstjänster har skett under en tid då agila arbetsprocesser har blivit allt populärare, vilka främjar en delad kunskap mellan utvecklare och kunder. Samtidigt har tillgången till kvantitativt loggad användardata vuxit enormt, vilket har möjliggjort en större och mer kontinuerlig närhet till produktens användare än någonsin tidigare. Denna studie ämnar att undersöka hur svenska ljudbokstjänster arbetar för att integrera sina användare i utvecklingen av deras produkt, med vilka metoder och i vilka syften det utförs. Med den obegränsade tillgången till användare i form av kvantitativt loggad användardata, är det av intresse att undersöka hur det påverkar användandet av andra traditionella metoder för användarinvolvering. En litteraturstudie genomfördes för att ta reda på det rådande kunskapsläget gällande metoder för användarinvolvering inom produktutvecklingen av streamingtjänster. Utifrån denna litteraturstudie utformades en intervjuguide. Därefter genomfördes tre kvalitativa intervjuer med en representant vardera från tre av de största och mest etablerade svenska ljudbokstjänsterna. Intervjuerna transkriberades och analyserades mot den undersökta teorin. Ur den analyserade empirin kunde två huvudsakliga syften till ljudbokstjänsternas användarinvolvering fastställas: • Att involvera användare som en medskapare av produkten • Att involvera användaren som testobjekt för prototyper Samtliga av de undersökta ljudboksföretagen använde sig av loggad kvantitativ användardata i båda dessa fall av användarinvolvering. Med användaren som ett testobjekt visar sig A/B-testning spela en stor roll vid användandet av loggad kvantitativ användardata. Två tredjedelar av ljudboksföretagen använde sig av kvalitativa metoder som fokusgrupper, vilket visade sig vara av stor vikt både i syftet att låta användaren testa och medskapa. Fokusgruppens unika fördelar exempelvis empatisk förståelse av användaren, är något som den kvantitativt loggade användardatan inte kan erbjuda.The audiobook industry has undergone extensive digitisation. One of the results of this digitalisation is the emergence of a whole new market for audio books and from this a number of audiobook services have emerged. These audiobook services offer their users a product as an application where the users have access to an unlimited amount of streamed content in form of audiobooks. The product development of these audiobook services has taken place during a time when agile work processes have become increasingly popular, which promotes shared knowledge between developers and customers. At the same time, access to quantitatively logged user data has grown tremendously, which has enabled a greater and more continuous proximity to the product's users than ever before. This study aims to investigate how Swedish audiobook services work to integrate their users in the development of their product, with what methods and purposes it is performed. With the unlimited access to users in the form of quantitatively logged user data, it is of interest to investigate how it affects the use of other traditional methods of user involvement. A literature study was conducted to find out the current state of knowledge regarding methods of user involvement in the product development of streaming services. Based on this literature study, an interview guide was designed. Subsequently, three qualitative interviews were conducted with one representative each from three of the largest and most established Swedish audiobook services. The interviews were transcribed and analysed against the theory investigated. From the analysed empiric study, two main purposes for the user involvement of audiobook services could be established: • Involving users as a co-creator of the product • Involving the user as a test object for prototypes All of the audiobook companies surveyed used logged quantitative user data in both of these cases of user involvement. With the user as a test object, A / B testing proves to play a major role in the use of logged quantitative user data. Two-thirds of audiobook companies used qualitative methods such as focus groups, which proved to be of great importance to both user testing and co-creation. The focus group's unique advantages, for example empathic understanding of the user, is something that the quantitatively logged user data cannot offer.
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Kotouček, Filip. "Vícekanálový přenos zvukových signálů po lokální počítačové síti." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2017. http://www.nusl.cz/ntk/nusl-316879.
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This thesis deals with design and implementation of multi-channel audio signal streaming via local network. The aim was to choose suitable processor for the transmission of up to 32 channels of audio signal, which will be used for implementation. Also the low-latency ASIO driver was studied and was used to provide timestamps for synchronization. The transmission protocol was designed for signaling and for the real time stream. The actual transmission is provided by TCP protocol. In conclusion, I created the application for development board with choosen MCU. Finally whole solution was tested with real data.
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Lanfranchi, Laetitia I. "MPEG-4 AVC traffic analysis and bandwidth prediction for broadband cable networks." Thesis, Atlanta, Ga. : Georgia Institute of Technology, 2008. http://hdl.handle.net/1853/29776.
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Thesis (M. S.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2008.Committee Chair: Bing Benny; Committee Co-Chair: Fred B-H. Juang; Committee Member: Gee-Kung Chang. Part of the SMARTech Electronic Thesis and Dissertation Collection.
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Liese, Christin. "Kaufst du noch oder streamst du schon?" Bachelor's thesis, Saechsische Landesbibliothek- Staats- und Universitaetsbibliothek Dresden, 2017. http://nbn-resolving.de/urn:nbn:de:bsz:14-qucosa-223500.
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Die Zeiten der Plattensammlung sind vorbei, Kassetten und CDs sind der MP3-Datei gewichen und nun wird Musik ausschließlich gestreamt. Dieses Zukunftsszenario ist bis dato noch nicht eingetreten, aber wird dies überhaupt passieren? Wird der Kauf von physischen Musikdatenträgern und digitalen Musikdateien dank der immer stärker ansteigenden Streaming Aktivitäten komplett eingestellt? Oder können beide Formen nebeneinander existieren? Um diesen Fragen auf den Grund zu gehen, wurde im Rahmen dieser Arbeit eine Umfrage mit 1.661 Studenten der Technischen Universität Dresden durchgeführt. Die Ergebnisse geben Aufschluss über die Nutzungshäufigkeiten von kostenfreien und kostenpflichtigen Streaming Anbietern sowie von CDs / Schallplatten und MP3 Musikdateien. Zudem wird aufgezeigt, dass eine geringe Zahlungsbereitschaft bei den Studenten besteht. Es werden bereits selten mehr als 5 € in Musik investiert, doch seitdem die Studenten Streaming Dienste nutzen, geben sie nach eigenen Angaben noch weniger Geld für Musik aus als zuvor. Diesem Negativtrend steht die Erkenntnis gegenüber, dass die Probanden seit der Nutzung von Streaming Angeboten weniger Musik illegal herunterladen. Auch wenn der Großteil weniger Musik kauft, so ist es etwa der Hälfte aller Befragten sehr wichtig, Musik zu besitzen, vor allem in physischer Form. Zudem wurden Nutzungsmotive der Möglichkeiten des Musikhörens erfasst, um deren Stärken und Schwächen aufzuzeigen. Die Ergebnisse verdeutlichen, dass die kostenfreie Variante des Streamens zwar häufig genutzt wird, sich die traditionellen Musikdatenträger und Musikdateien jedoch immer noch großer Beliebtheit erfreuen. Von einer kompletten Verdrängung des Kaufens von Musik kann demnach nicht ausgegangen werden.
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KNOBEL, KARIN, and LOVISA LÆSTADIUS. "Big Data in Performance Measurement: : Towards a Framework for Performance Measurement in a Digital and Dynamic Business Climate." Thesis, KTH, Skolan för industriell teknik och management (ITM), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-238689.
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In today’s business climate permeated by Big Data, an opportunity to drive performance lies in analysing consumer behaviour from user data. In particular for online content providers, user data is available in abundance and logged continuously. This leads to new possibilities for design and usage of metrics, as businesses can benefit from smart and timely decision-making. However, in order to profit from user data in performance measurement (PM), it is critical to identify metrics that truly guide decisions. Thus, an effective and efficient PM process is imperative. Despite its promise, Big Data’s role in PM has been scarcely researched. Research has studied user behaviour from data, for instance in the context of video or audio streaming and web search, but primarily with a focus on technical performance. In addition, the research on online content providers’ PM is fragmented, and has mainly been conducted by practitioners. Thus, the PM field needs to be updated to reflect today’s dynamic and digital business climate. Therefore, the purpose of this research was to explore how online content providers, generating a large amount of user data, work with PM, and also practically illustrate how metrics can be designed from user data. The research was carried out as a case study at an audio streaming company, but empirics was also gathered from other online content providers with the aim to increase the generalisability. The illustration of metric design was based on quantitative analysis of commuters’ in-car audio streaming. For commuters’ audio streaming it was found that suitable metrics should capture the habitual nature. Therefore engagement metrics were found to be applicable, for instance the fraction having sessions both in the morning and afternoon, and the fraction having more than one day commuting with the streaming service per week. In regard to online content providers’ PM process, this research contributes with a proposed framework, which was developed from three existing frameworks; HEART reflected as important measurement dimensions and translation of goals to metrics, OKR which sets the focus in terms of high-level goals, and design-implement-use reflected as the process’ phases. It was found that insights from user data and explicit user feedback are complementary and can arise throughout the whole process, and that mutual communication between data scientists and product managers is crucial. Further, four types of iterations were identified in the process; modifying a metric, designing new metrics, completely changing a metric, and starting new initiatives. Moreover, metrics were found to be highly context dependent. Additionally, four important aspects were identified in metric design; data availability and proxy assessment, characteristics and form of metric, metric trade-offs, and metric movement interpretation.I dagens affärsklimat genomsyrat av Big Data finns en möjlighet att driva resultat framåt genom analys av kundbeteenden från användardata. I synnerhet för online-tjänsteföretag samlas användardata kontinuerligt och finns tillgänglig i en oerhörd mängd. Detta skapar nya möjligheter för design och användande av mätetal då företag kan utveckla smartare och snabbare beslutsfattande. För att verkligen dra fördel av användardata i prestationsmätning (PM) är det dock kritiskt att identifiera mätetal som faktiskt bistår beslutsfattande, vilket följaktligen kräver en effektiv PM-process. Trots potentialen är forskning på Big Data inom PM begränsad. Studier har analyserat kundbeteenden från användardata, exempelvis i kontexten av strömmad video eller audio och webbsökningar, men primärt med fokus på tjänstens tekniska prestanda. Vidare är forskning på PM hos online-tjänsteföretag fragmenterad, och huvudsakligen genomförd av företag inom industrin. Följaktligen bör fältet aktualiseras för att reflektera dagens digitala och dynamiska affärsklimat. Därför var syftet med denna studie att utforska hur online-tjänsteföretag, som besitter stora mängder användardata, arbetar med PM, men även praktiskt illustrera hur mätetal kan designas från denna data. Studien genomfördes som en fallstudie på ett ljud-strömningsföretag, men empiri insamlades även från andra online-tjänsteföretag med avsikt att öka generaliserbarheten. Den praktiska illustrationen av mätetals-design baserades på en kvantitativ analys av pendlares audio-strömning i bil. För pendlares audio-strömning i bil fann denna studie att lämpliga mätetal bör fånga den vanemässiga aspekten associerad med pendling. Därmed anses mätetal som reflekterar engagemang lämpliga, exempelvis andelen som har sessioner både på förmiddagen och eftermiddagen och andelen som har mer än en dag med pendlar-sessioner i veckan. Gällande PM-processen hos online-tjänsteföretag bidrar denna studie med ett föreslaget ramverk som utvecklades från tre existerande ramverk; HEART som reflekteras i form av viktiga mätetalsdimensioner samt översättning av mål till mätetal, OKR vilket sätter fokus för processen i termer av mål på högre nivå, och designa-implementera-använda som reflekterar processens faser. I studien kom det fram att insikter från användardata och explicit användaråterkoppling kompletterar varandra, och att dessa kan uppkomma under hela processen. Vidare konstaterar denna studie att ömsesidig kommunikation mellan dataforskare och produktchefer är essentiellt. Dessutom identifierades fyra typer av iterationer som kan förekomma vid användning av mätetal; modifiera mätetal, designa nya mätetal, fullständigt förändra mätetal samt påbörja nya initiativ. Därutöver kan studien konstatera att mätetal är högst kontextberoende, och att det finns fyra viktiga aspekter att ta hänsyn till i mätetals-design; data-tillgänglighet och proxy-utvärdering, karaktäristik och form på mätetal, trade-off mellan mätetal, samt tolkning av mätetals-förändringar.
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Gonçalves, Neto Jahyr 1980. "Desenvolvimento de uma plataforma multimidia utilizando a linguagem Python." [s.n.], 2007. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259651.
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Orientador: Max Henrique Machado CostaDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
Made available in DSpace on 2018-08-10T00:12:12Z (GMT). No. of bitstreams: 1 GoncalvesNeto_Jahyr_M.pdf: 950657 bytes, checksum: f62691d16e5db013d1b8a9c4e4a32c88 (MD5) Previous issue date: 2007
Resumo: Nesta dissertação apresentamos o desenvolvimento de uma plataforma multimídia baseada no modelo cliente-servidor voltada para aplicações de streaming de áudio e vídeo. Essa plataforma deverá evoluir para um sistema de videoconferência em um projeto futuro. A plataforma permite a comunicação de áudio, vídeo e texto a partir de um ponto (o servidor) para vários outros pontos (os clientes). Uma das inovações do projeto está no desenvolvimento em Python, que é uma linguagem interpretada, orientada a objetos e dinamicamente tipada
Abstract: This dissertation presents the development of a client-server platform designed initially for audio and video streaming applications. This platform will evolve into a videoconference system as part of a future project. The platform allows audio, video and text communication from a point (the server) to several others points (the clients). One of the project innovations is the implementation Python Language, which is an interpreted, objectoriented and dynamically typed language
Mestrado
Telecomunicações e Telemática
Mestre em Engenharia Elétrica
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Wen, Jing Yao, and 溫景堯. "Streaming audio classification for smart home environments." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/24322561547704851822.
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碩士國立政治大學
資訊科學學系
98
Human receive sounds such as language and music through audition. Therefore, audition and vision are viewed as the two most important aspects of human perception. Computational auditory scene analysis (CASA) defined a possible direction to close the gap between computerized audition and human perception using the correlation between features of ears and mental perception in psychology of hearing. In this research, we develop and integrate methods for real-time streaming audio classification based on the principles of psychology of hearing as well as techniques in pattern recognition. There are three major parts in this research. The first is audio processing, translating sounds into information that can be enhanced by computers; the second part uses the principles of CASA to design a framework for audio signal description and event detection by means of computer vision and image processing techniques; the third part defines the distance of image feature vectors and uses K-Nearest Neighbor (KNN) classifier to accomplish audio recognition and classification in real-time. Experimental results show that the proposed approach is quite effective, achieving an overall recognition rate of 80-90% for 8 types of audio input. The performance degrades only slightly in the presence of noise and other interferences.
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Chieu, An-bang, and 邱安邦. "Audio streaming system design Based on Embedded system." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/w2xkm4.
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碩士國立臺灣科技大學
電子工程系
95
The object of this thesis is to design and develop a set to apply the system which applys to the multimedia audio streaming. We use the most popular embedded application at present, uses the existing platform to develop the audio streaming correlation the audio player software and the system core, then porting on the development platform which we select, finally penetrates the wireless network, may the selective reception or the transmission audio streaming, realizes the multimedia audio streaming to flow the development the goal. The key point discusses which in the paper divides into two major parts, respectively the hardware specification and system actualization. For the hardware specification , mainly introduces the development board each item of overhead construction, including CPU, Flash Memory, LCD, I/O equipment and so on main installment. For the system realization contains the operation system, the application program, the decoding software and GUI(Graphic User Interface).
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鄭淙毅. "Development of an Embedded Multimedia Audio Streaming System." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/93228156850978632766.
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Sun, Shih-Jung, and 孫世榮. "An Implementation of Wireless MP3 Audio Streaming with Bluetooth Technique." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/39681848502391687017.
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碩士國立中正大學
電機工程研究所
91
The Bluetooth specification version 1.1 has only defined a SCO link for audio transmission of voice quality. With SCO link, none of the high quality audio can be transmitted. The other ACL link for data transmission which can offer up to 723.2kbps data rate in asymmetric mode is able to transmit some compressed audio file such as MP3,WMA,ACC etc. To do audio streaming in data transmission channel needs extra scheme for pseudo-isochronous operation to ensure real time audio playback. The capability of high quality audio streaming is an added value in Bluetooth products. The possible applications are MP3 walkman with wireless headphone or MP3 player with wireless speaker. The hardware contains two sets of personal computers and Bluetooth USB dongles. One of the computers transmits MP3 file while the other receives, decodes and plays it back.. The control functions such as “PLAY”,”PAUSE”,”STOP” are added to form a complete system. The software stack follows the basic Bluetooth protocols such as L2CAP,SDP,HCI,AVDTP,A2DP etc. and the MP3 decoder fallows the ISO 11172-3 and ISO 13818-3 standard. During transmission the received data is saved to be compared with the original one for testing system performance. Finally, an idea of non-standard profile called “Simple Audio on Demand Profile” is issued from the process of implementation. This non-standard profile is based on a different point of view as A2DP and trying to fit better for some kind of applications.
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Wu, Jheng-Jhong, and 吳正中. "The Investigation and Implementation of Audio and Video Streaming Techniques." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/21815164028894351420.
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碩士義守大學
資訊工程學系碩士班
95
Streaming media is a new type of internet multimedia transport mechanism. Traditionally, user must download the interested file first, and then play the file by some multimedia player software. This may cause a long waiting time and large memory consumption. Sometimes the intellectual property is an issue in this kind of transport mechanism, because after the whole file is downloaded, the user may distribute it to unauthorized user. Streaming media is a technique which can solve the problems mentioned above. When the server receives the client’s request, the server splits the file into packets and delivers them to the client. After filling an initial buffer, the player can start playing the video and audio in real-time without downloading the whole file into the hard disk first. The video and audio signals then transfer from the server to the client continuously. This is also why this kind of transport mechanism is called streaming. Imaging technology is a key technique in multimedia technology development. Image compression technology continuously improves from MPEG-2 (DVD) and MPEG-4 to H.264 (MPEG-4 Part 10). H.264 enhances the compression efficiency by 50% compared with MPEG-2, and will play an important role in the future. The development and application of multimedia communication is an evolving technology. It also changes the way we communicate to each other. How to transfer high quality multimedia streaming data is an ongoing topic and has caught many researchers’ interests. We investigated video/audio compression and multimedia streaming techniques, and finally developed a multimedia streaming server in practice.
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Chen, Fang-Jie, and 陳芳傑. "SIP-based Video/Audio Streaming System in Home Security Application." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/41883725889655047461.
Full textAbstract:
碩士雲林科技大學
資訊工程研究所
97
Due to the popularity of wireless networking and home security, this paper propose architecture combining the doorbell and real-time monitor in the wireless network based on the Session Initiation Protocol, and combine embedded system with PDA device, to achieve a real-time video and audio streaming of the ubiquitous home security monitoring environment. At the same time, due to the often packet loss in wireless network environment, we use MPEG-4 encoding features with proposed XOR-based FEC algorithm, a mechanism to recover the lost packets and thus to improve streaming video quality.
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Huang, Xin-Zhang, and 黃信璋. "Real-time Video/Audio Streaming Mechanisms and Applications on Embedded System." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/5b6553.
Full textAbstract:
碩士國立成功大學
工程科學系碩博士班
92
With advances in communication, multimedia, and the growth of embedded system,the applications of multimedia become diverse. Base on cost-effective design, we use a low cost hardware with free software – LWIP to develop a stable embedded A/V streaming system.We combined embedded system with LWIP to develop a system which is inexpensive and easy to develop software. In this thesis, we studied the free software – LWIP, and designed a mechanism for A/V streaming sytem to allow our products can provide better A/V streaming services. In addition, the network protocol and A/V streaming system that we developed is independent of the system, so it is easy to portable to other platforms.
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Lin, Yi-Wei, and 林宜瑋. "Real-time Video/Audio Streaming Mechanisms and Applications overWireless/Mobile Networks." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/08900081270263144964.
Full textAbstract:
碩士國立成功大學
資訊工程學系碩博士班
90
Due to the characteristics of (1) smaller bandwidth and (2) unreliable transmission media, real-time media streaming over wireless networks is not trivial. To have smooth media streaming over wireless networks, we propose schemes for video and audio streaming respectively. For audio, two sending modes, in which redundant information is embedded in each packet, that the proposed scheme contains are (1) the "redundant" mode and (2) the "duplicated" mode. Let a packet i can contain three audio frames i, i-1, and i-2. In the redundant mode, frame i uses a codec of better quality than that for frames i-1 and i-2. In the duplicated mode, frames i, i-1, and i-2 use the same codec, which has lower quality than that for frame i used in the redundant mode. The "redundant" mode may give better quality of sound but consumes more bandwidth, while the "duplicated" mode gives lower quality of sound but consumes less bandwidth. For video, an adaptive real-time video streaming scheme that uses the layered video technique is proposed. Two attributes that are used to determine the network situation and then adjust the sending rate accordingly are loss-rate and round-trip time (RTT). Two thresholds named "upper-ratio" and "lower-ratio" are set. When the average RTT exceeds the upper-ratio multiplies the maximum RTT, the network situation is set to congested; when the average RTT is under the lower-ratio multiplies the maximum RTT, the network situation is set to unloaded. Since the unreliable media cause of packet loss in the wireless environment is rate-independent, a method that can separate these rate-independent loss from the congestion loss is needed. We use inter-arrival time between two received packets to identify if an out of order packet was received in time. If the out of order packet was received in time, then the packet loss between the two received packets is caused by unreliable media; otherwise, the packet loss is caused by congestion and the network situation is congested. Upon a congestion situation is determined, the sending rate of the sender is dropped down; upon unloaded situation is determined, the sending rate of the sender is raised up. In this way, the adaptive real-time media streaming can be achieved for wireless networks.
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Carvalhido, Eugénio Bettencourt. "Real-Time Audio Fingerprinting for Advertising Detection in Streaming Broadcast Content." Master's thesis, 2018. https://hdl.handle.net/10216/114095.
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Carvalhido, Eugénio Bettencourt. "Real-Time Audio Fingerprinting for Advertising Detection in Streaming Broadcast Content." Dissertação, 2018. https://hdl.handle.net/10216/114095.
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Chen, Ying-Long, and 陳穎隆. "Mining Behavior Patterns of Learner from Streaming Technique Video/Audio Forum." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/37988111204129393291.
Full textAbstract:
碩士大葉大學
資訊管理研究所
90
Nowadays, the multimedia learning via PC has become a mega trend. Thanks to many scholars indicate the digitized presentation of multimedia is more active than the single media way. To improve the interaction degree of learning and the knowledge sharing between trainees, this thesis adopts the video/audio streaming technology into the Video/Audio Forum. With the modified TAM(Technology Acceptance Model), which issued by this thesis, the forecast capability of learning performance has been improved. TAM can evaluate the acceptance performance after an IT system was introduced into an organization. However, without the learning history Portfolio, TAM still suffers the uncertainty of “actual usage aspect”. This thesis tries to improve TAM via the clustering of Media Log Mining method. With this method to analyze the cluster-pattern of learning-behavior of trainees from the WEB log and streaming log in the Video/Audio Forum. The contribution of this thesis is issuing a new TAM model by replacing the “actual usage aspect” of the original TAM with the cluster-pattern of learning-behavior by Media Log Mining Method and forecast the learning performance via “technique acceptance of trainee ”. Consequently, the correlation between the total aspect and learning performance of the modified TAM is experimental better than the original TAM.
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Käs, Christian Haardt Martin. "Analyse und Simulation von Streaming Audio- und Videoanwendungen unter Berücksichtigung von Dienstgüteanforderungen /." 2006. http://www.gbv.de/dms/ilmenau/abs/515081124kaes.txt.
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Liu, Hsiang-Chun, and 劉祥俊. "Design and Implement a Mobile Audio / Video Streaming System for Digital Home." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/02401019907343637805.
Full textAbstract:
碩士元智大學
資訊工程學系
93
Now, people get into a digital life style by the prevalence and development of Internet and digitalize content. Now the digital home living is more and more popular. Every hardware factory invests positively in creating an environment that people could enjoy the digital living and the convenient of information technology more easily and convenient. In this research uses the guideline that propose by DLNA, to establish a multimedia digital home living entertainment platform, which makes it easier to connect the information appliance, digital content, and information service. Users can simply choose their favorite multimedia content by the controller. This platform all application build above Linux operation system, uses TCP/IP stacks and device discovery technology (Universal Plug and Play, UPnP) . Using real tine transport streaming protocol (RTP) suites to control the network flow. In this research, we embedded the network flow adaptive of transfer the digital content processing on Digital Media Server, DMS. In another side, we uses cache and adaptive of hardware ability on Digital Media Player, DMP to play digital content more smooth, to make the platform to attain quality of service(QoS) for playing multimedia content.
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Aires, Nuno Miguel. "Audio-guiding imersivo." Master's thesis, 2018. http://hdl.handle.net/10773/24947.
Full textAbstract:
Desenvolveu-se um protótipo de um sistema de audio-guiding que permite associar fontes sonoras virtuais aos pontos focais de cada rota de visita e aplicar, na reprodução estereofónica através de auscultadores, um efeito de espacialização controlado em tempo real pela pose (posição e orientação da cabeça) do utilizador relativamente a esses pontos. É assim criado um efeito de realidade aumentada, com os conteúdos áudio (virtuais) a parecerem provir de pontos (reais) especificados.
Foi desenvolvido um programa de gestão de rotas para especificar previamente os pontos focais (através de coordenadas GPS), o conteúdo áudio (ficheiro monofónico) pretendido para cada e o ponto do trajeto em que deve ser iniciada a reprodução.
Para detetar posição, usou-se um recetor GPS ligado à plataforma de computação transportada pelo utilizador; para monitorizar continuamente os movimentos da sua cabeça (head-tracking), recorreu-se a um sensor inercial (InertiaCube) acoplado aos auscultadores.
A aplicação principal, desenvolvida em linguagem C++, recorre a buffers circulares implementados em software para realizar, com a mínima latência possível, o streaming áudio a partir dos ficheiros de entrada. O processamento baseia-se numa biblioteca (‘motor’) de auralização em tempo real que utiliza bases de dados de HRTF de acesso público. Para escolher o par de HRTF a utilizar em cada ciclo de processamento, o azimute e a elevação da fonte virtual são continuamente recalculados em função dos dados recolhidos sobre a pose do utilizador.
Para avaliar o funcionamento da aplicação, identificar possíveis problemas e caracterizar as gamas mais adequadas para determinados parâmetros (e.g. tamanho de buffers, número de amostras por bloco de áudio, janela de filtragem dos dados GPS), efetuaram-se testes subjetivos preliminares, com dez sujeitos a percorrer uma rota criada para o efeito. Embora confirmando a capacidade de obter o efeito sonoro pretendido, os testes evidenciaram a necessidade de melhorar a precisão dos dados de posição, principal fator a afetar negativamente a qualidade da experiência.An audio-guiding system prototype was developed which makes it possible to associate virtual sound sources to the focal points of each tourist route and apply, in the stereophonic reproduction over headphones, a spatialisation effect controlled in real time by user pose (position and head orientation) relative to those points. An augmented reality effect is thus achieved, with the (virtual) audio content seemingly originating from specified (real) points. A route management program was developed to allow specification of the focal points (through GPS coordinates), audio content (monophonic file) intended for each and route point where its playback should be triggered. Position was detected by a GPS receptor plugged into the computing platform carried by the user; head-tracking was based on an inertial sensor (InertiaCube) attached to the headphone strap. The main application, developed in C++, implemented software buffers to stream audio through from the input files with the lowest possible latency. The processing resorts to a real-time auralisation engine using a public access HRTF database. In order to choose the appropriate HRTF pair for each processing cycle, the azimuth and elevation of the virtual source are continuously recalculated according to the acquired user pose data. In order to assess performance, identify possible problems and characterise the most appropriate application parameter ranges (e.g. buffer size, number of samples per audio block, filtering window for GPS data), preliminary subjective tests were carried out, with ten subjects following a route created for this purpose. Whilst confirming the ability to provide the desired audio spatialisation effects, the tests also evidenced the need to improve the precision of position data, as this was the main factor negatively affecting the experience.
Mestrado em Engenharia Eletrónica e Telecomunicações
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Hsu, Ching-Yun, and 許菁云. "A Study of Video-Audio Streaming Technology and anImplementation of Integrated Management System." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/nmcnsp.
Full textAbstract:
碩士國立高雄第一科技大學
電腦與通訊工程所
96
An audiovisual streaming technology is becoming even more important recently with the widespread use of Internet. It conveys video and audio streams in real time through Internet. A client player can play an audiovisual stream that transmitted from server. In addition, it doesn’t have to wait until the downloaded streams are completed. The advantages of this technology are to save the waiting time and reduce the disk storage. In order to make it convenient, the administrator can operate and manage the Video-Audio Streaming service in remote. So, the thesis is an implementation about an integrated management system of Video-Audio streaming service. This system integrates three major components, the process of the video and audio, server management, and Video-Audio test player. Using the Integrated Management System can control WebCam to catch video and audio. And, compressing it into MPEG-4 format, then upload and store the content to the streaming server. The content can be outputted by streaming technology via the streaming server. Not only can it operate and manage the Video-Audio streaming server by means of the remote Management system, but the authentication and authorization access control functions provide users the better video quality. Finally, using the test player can check all the settings of streaming service correctly. Besides implementing an integrated management system of Video-Audio streaming service, this thesis studys the streaming network packet, the processes of Video-Audio streaming service that contain the protocol of RTSP, RTP, RTCP , and decoding the MPEG-4 video by MPlayer.
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Wong, Hon-Long, and 黃漢龍. "A Ubiquitous IAs and Audio Streaming Access Platform in the UPnP Home Network." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/94482795023513578585.
Full textAbstract:
碩士國立成功大學
資訊工程學系碩博士班
95
The radical development of information technology makes the new generation of home appliance become Information Appliances (IA). IAs can interact and share information inside the UPnP home network. In this thesis, we proposes the Ubiquitous IAs and Audio Streaming Access Platform (UIAP) that includes two main services: (i) IA Access Service and (ii) Audio Access Service. User can use a PC, a handheld device and a light-weight Bluetooth headset to access the above two services via the home gateway. IA Access Service enables users access IAs anytime and anywhere. Inter-IA interactions which involve more than one IAs to cooperate together are achieved and speci ed in Service Interaction Script (SIS) language. Audio Access Service enables users to access audio files which are distributed in different media servers by using the light-weight Bluetooth headset in the digital home network. When users with the light weight Bluetooth headset roam from one room to another, it may cause the degradation of signal quality, UIAP will perform the Computing Powerless Handoff (CP-Handoff) procedure to keep continuous audio streaming. In UIAP, the home gateway provides Web-based interface for users to access IAs, audio files and to define inter-IA interaction no matter they are at home or out of home.
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Lai, Chih-Yun, and 賴志昀. "Design and Implementation of Intelligent Audio/Video Streaming System Based on OSGi Service Platform." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/33849046019288290116.
Full textAbstract:
碩士國立成功大學
工程科學系碩博士班
96
The study of this thesis focuses on the design and implementation of intelligent audio/video streaming system based on OSGi service platform. This system is split into three sub-system: Indoor location sub-system、Multimedia streaming service sub-system and Intelligent A/V management sub-system. These sub-systems are implement as bundles on OSGi Service platform. We will write multiple bundles to combine the ZigBee wireless sensor network and the UPnP A/V Architecture. In this system，we will collect information from the sensor network，analysis these data and then control the UPnP A/V Devices in the home network.
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Lu, Pao-Lin, and 呂寶麟. "Broadband audio/video streaming-media industry and their applications and services-Realma.com case study-." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/36305273420524450021.
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碩士國立臺灣大學
國際企業學研究所
93
In recent years as overall Internet infrastructure begins to mature, numerous Internet applications and services such as online games, shopping, auction, search and advertisements have been gradually assuming more important roles in our daily lives, demonstrating that so long as the overall environment is ready, in addition to having a successful business model, Internet businesses are not simply “bubbles”. As a result, this research will provide an analysis on broadband audio/video streaming industry and their applications and services. Such analysis will be based on the example of the leading broadband content leader, Realma DigiMedia, and its realma.com services, to further discuss various broadband content applications’ commercial feasibility as well as to analyze the future and potential of broadband content/service industry. The broadband industry for the overall Chinese market is now capable of significant growth, based on the fact that the number of broadband subscribers has increased, the Internet is becoming a more important part of daily lives, and the lifting of Greater China economy’s significance. As a result, Taiwan broadband industry should focus on the global Chinese market. Take Realma DigiMedia’s realma.com for instance, such service has a number of advantages, such as partnership with an international leading company, ability to provide service across platforms and borders, larger subscriber base and strong management team and relationships. However, Realma also faces problems of slow change of consumer habits, small size of Taiwan market and competition from similar service providers. In addition, smaller resources compared to large media players is also a concern. Although confronted with certain threats and disadvantages, Realma can still skillfully maneuver strategic partnerships to accomplish objectives. Realma will also leverage its future businesses in providing richer content, entering into China market, self-producing original content to own usage rights, expanding e-commerce application realm, to prepare itself for soon coming mobile content distribution service and integrated marketing business.
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Gomes, João Francisco Vaz Brandão. "Evaluating the Performance of the AES70/AES 67-based Network Architectures for Audio Streaming." Master's thesis, 2021. https://hdl.handle.net/10216/137345.
Full textAbstract:
O consumo de mutimédia via streaming tornou-se um aspecto central da vida moderna. Tal papel acarreta enormes desafios para garantir geral satisfação com a qualidade dos serviços, o que naturalmente tende a acelerar o seu desenvolvimento, isto é, tende a traduzir-se em sucessivos aperfeiçoamentos. O utilizador tem tremenda facilidade em avaliar a qualidade de um destes serviços: não pode apresentar latência e deve apresentar a maior fidelidade possível.
São precisamente estes parâmetros que o presente projeto pretende avaliar na performance do protocolo AES70 e do standard AES67, constituindo assim o seu principal objetivo. Esta avaliação está contextualizada dentro daquilo que são as opções de transmissão de áudio sobre uma rede local Ethernet, principalmente quando aplicada a micro-controladores embutidos.
Para tal, o trabalho realizado para esta dissertação prendeu-se inicialmente por dominar o controlador disponibilizado para testagem, no caso, uma placa STM com um micro-controlador ARM Cortex-M7. Durante este processo foi possível analisar sobretudo a idiossincrasia do controlador, descobrindo-se lacunas que põem em causa não só o processo de avaliação dos referidos protocols, mas antes ainda, a sua implementação no sistema embutido.
Contudo, foi possível implementar uma aplicação de áudio streaming sobre Ethernet, a qual se espera ser útil para trabalho futuro. Este é, na verdade, o factor que esta dissertação mais acaba por desempenhar, uma sucinta explicação para entender todas as vertentes do controlador disponibilizado, nomeadamente as mais relevantes a aplicações de áudio e Ethernet, deixando-o perparado para cenários de teste e posterior comparação com abordagens semelhantes.The consumption of multimedia services via streaming has become a core aspect of modern life. Such a role brings enormous challenges to ensure general satisfaction with the quality of those services, which naturally tends to accelerate their development, i.e., it reflects in a tendency for successive improvements. Their quality, for an user, is measured quite elementally: the service must not present any discernible latency and must deliver the highest possible fidelity. These parameters are precisely the ones this project intends to evaluate in the performance of the AES70 protocol and the AES67 standard, thus constituting the project's main objective. This evaluation is contextualized within the available options for audio transmission over an Ethernet local area network, especially when applied to embedded microcontrollers. As such, the work that comprises this dissertation was initially concerned with mastering the testing environment, i.e., the controller provider for that role, in this case, an STM board with an ARM Cortex-M7 microcontroller. During this process, the analysis of the controller's idiosyncrasies was the main object of focus, contributing to the uncovering of important shortcomings which further challenged not only the evaluation process of these protocols, but particularly, their implementation in the embedded system. However, an implementation of an audio streaming over Ethernet application was accomplished and it is expected to be useful for future work. This is, in fact, the most valuable factor of this dissertation: it incorporates a precise explanation for understanding all key aspects of the STM controller, namely those most relevant to audio and Ethernet applications, rendering it ready for test scenarios and, subsequently, for the comparison of results with similar audio streaming approaches.
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Lin, Yu-Ching, and 林毓慶. "A Study on Developing a Streaming Client-Server Architecture for MPEG-4 Video and Audio." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/74699973759935510775.
Full textAbstract:
碩士義守大學
資訊管理學系碩士班
91
An audiovisual streaming technology is becoming even more important now with the widespread use of Internet. It conveys video and audio streams in real time through Internet. A client player can play an audiovisual stream that transmitted from server. In addition, it doesn’t have to wait until the downloaded streams are completed. The advantages of this technology are saving the waiting time and reducing the disk storage. The standard MPEG-4 provides the high compressions rate and streaming delivery service for audiovisual objects. Therefore, it can be applied to the applications of communication network and streaming delivery. This thesis proposes a streaming client-server architecture for MPEG-4 audiovisual objects. Also, a media player which can receive and play the MPEG-4 streaming over Internet. In order to achieve playing audiovisual objects simultaneously, an audiovisual synchronized method with AVI format is also proposed in this thesis. Finally, a streaming client-server architecture is mentioned and implemented.
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Liese, Christin. "Kaufst du noch oder streamst du schon?: Der Einfluss von Musik Streaming Diensten auf den Kauf von Musikdateien und Musikdatenträgern." Bachelor's thesis, 2015. https://tud.qucosa.de/id/qucosa%3A30277.
Full textAbstract:
Die Zeiten der Plattensammlung sind vorbei, Kassetten und CDs sind der MP3-Datei gewichen und nun wird Musik ausschließlich gestreamt. Dieses Zukunftsszenario ist bis dato noch nicht eingetreten, aber wird dies überhaupt passieren? Wird der Kauf von physischen Musikdatenträgern und digitalen Musikdateien dank der immer stärker ansteigenden Streaming Aktivitäten komplett eingestellt? Oder können beide Formen nebeneinander existieren? Um diesen Fragen auf den Grund zu gehen, wurde im Rahmen dieser Arbeit eine Umfrage mit 1.661 Studenten der Technischen Universität Dresden durchgeführt. Die Ergebnisse geben Aufschluss über die Nutzungshäufigkeiten von kostenfreien und kostenpflichtigen Streaming Anbietern sowie von CDs / Schallplatten und MP3 Musikdateien. Zudem wird aufgezeigt, dass eine geringe Zahlungsbereitschaft bei den Studenten besteht. Es werden bereits selten mehr als 5 € in Musik investiert, doch seitdem die Studenten Streaming Dienste nutzen, geben sie nach eigenen Angaben noch weniger Geld für Musik aus als zuvor. Diesem Negativtrend steht die Erkenntnis gegenüber, dass die Probanden seit der Nutzung von Streaming Angeboten weniger Musik illegal herunterladen. Auch wenn der Großteil weniger Musik kauft, so ist es etwa der Hälfte aller Befragten sehr wichtig, Musik zu besitzen, vor allem in physischer Form. Zudem wurden Nutzungsmotive der Möglichkeiten des Musikhörens erfasst, um deren Stärken und Schwächen aufzuzeigen. Die Ergebnisse verdeutlichen, dass die kostenfreie Variante des Streamens zwar häufig genutzt wird, sich die traditionellen Musikdatenträger und Musikdateien jedoch immer noch großer Beliebtheit erfreuen. Von einer kompletten Verdrängung des Kaufens von Musik kann demnach nicht ausgegangen werden.:1. Einführung und Relevanzbegründung
2. Musik Streaming Dienste
2.1. Begriffsdefinition
2.2 Technologische Aspekte
2.3 Rechtliche Aspekte
2.4 Wirtschaftliche Aspekte
3. Der Musikkonsum im Umbruch
3.1 Der Musikkonsum im Wandel
3.1.1 Die fortschreitende Digitalisierung
3.1.2 Die aktuelle Musiknutzung
3.2 Die deutsche Musikindustrie – Nutzung, Absatz und Umsatz
3.2.1 Aktuelle Absatz- und Umsatzzahlen
3.2.2 Zwei Zukunftsszenarien
4. Musik Streaming Dienste im Fokus der Forschung
4.1 Aktuelle Studien zum Musik Streaming
4.2 Die Digital Natives als Zielgruppe
5. Das Forschungsvorhaben
5.1 Herleitung der Forschungsfragen und Hypothesen
5.2 Erhebungsmethode
5.3 Zielgruppenbestimmung und Grundgesamtheit
5.4 Die Online-Befragung
5.4.1 Aufbau und Durchführung
5.4.2 Beschreibung der Stichprobe
6. Darstellung und Auswertung
6.1 Die Nutzung von Musik als Stream, physisches und digitales Medium
6.2 Einflüsse der Musik Streaming Dienste auf das Kaufverhalten
6.3 Zahlungsbereitschaft für Musik
6.4 Nutzungsmotive für die vier Optionen des Musikhörens
7. Diskussion
7.1 Kritik und Interpretation der Ergebnisse
7.2 Ein Ausblick auf die Zukunft
8. Literatur
9. Anhang
A. Fragebogen
B. Email-Anschreiben an alle TU Dresden Studenten
C. Weitere Tabellen
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Hung, Hsi-Hsuan, and 洪晳瑄. "Peer to Peer(P2P) Synchronous Transmission of Audio and Video Signals ─A case study on the Wei-Wu-Ying Center for the Performing Arts and Pier 2 SAR performances with a Synchronous Transmission of Video Streaming." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/94519952562656002082.
Full textAbstract:
碩士國立高雄應用科技大學
電機工程系博碩士班
101
In recent years, due to many scholars in the world involved in the research and marketing experiments on P2P (Peer to Peer) based audio and video transmission, enabling a rapid growth of audio and video streaming technology. A significant example is the recent progress of related Internet and networking technologies in Mainland China. In this work we focus on developing a method for P2P audio and video synchronous transmission, and propose an experimental prototype through a case study on data transmission between two landmarks in Kaohsiung, say, “Wei-Wu-Ying Center for the Performing Arts” and “Pier-2 SAR”. Such a solution is not only more convenient but also able to save money for the promotion of cultural business. For instance, for the promotion of cultural and artistic activities or performances in previous years it was often considered only about the limitation of specific performance area; if all activities or performances are being performed simultaneously in many regions, SNG satellite connection provided by several funding television companies for synchronous transmission should be utilized for promoting such activity or performance purposes. Under such a circumstance, program scheduling and planning should be constrainted to the arrangement of the funding companies, the expected performance of the activity organizer will not be fully achieved. As a result, it is expected that by means of P2P audio and video signal synchronous transmission, the issues will be solved. Furthermore, the application domain can be expanded from the concept of “point” to that of “line”, then eventually reach to that of “surface”, for offering an alternative approach for an effective audio and video synchronous transmission, in order to make a practical contribution to the development of cultural business.