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Miller, Robin J. "COFDM for HF digital broadcasting." Thesis, University of Brighton, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.287067.
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Paradise, Richard A. "Modeling and simulation of the physical layer of the single channel ground and airborne radio system (SINCGARS)." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Jun%5FParadise.pdf.
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Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, June 2005.Thesis Advisor(s): Frank Kragh, Herschel H. Loomis. Includes bibliographical references (p. 55-56). Also available online.
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Patel, Dipankumar Dalubhai. "Subjective effects of cell loss and bit error on compressed audio-visual applications over ATM." Thesis, Imperial College London, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.314077.
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Ligeti, Agnes. "Coverage optimization in digital audio broadcasting network." Licentiate thesis, KTH, Signaler, sensorer och system, 1997. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-712.
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Lin, Xiao. "High quality audio coding." Thesis, University of Southampton, 1993. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.357991.
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Taylor, Gregory. "Canadian broadcasting regulation and the digital television transition." Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86874.
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This thesis examines the evolution of Canadian broadcasting regulation since the adoption of the 1991 Broadcasting Act with a specific focus on the digital television transition. For Canada, this technological shift exemplifies many of the greater changes in public policy in the last two decades: a faith in market mechanisms, light-touch regulation, co-regulatory approaches, and the powerful influence of new technologies. This dissertation will address the following central question: does the policy surrounding the transition to digital television broadcasting pose a challenge to traditional concerns of the role of broadcasting in Canadian democracy? Utilizing the approach of institutional political economy, this study is informed by primary government and industry documents and interviews conducted with key actors within Canadian broadcasting. This dissertation offers a unique contribution to knowledge in three areas: establishing clear parameters for the differences between policy and regulations in Canadian broadcasting; offering the first comprehensive study of the Canadian digital television transition; and analyzing the growth and impact of self and co-regulation in Canadian broadcasting policy. The results of this study speak to the power dynamics amid the range of actors involved in the Canadian policy process, the influence of new technologies, and the greater prevailing policy directions in broadcasting since the 1991 Broadcasting Act was adopted.Cette thèse examine l'évolution de la réglementation canadienne de la radiodiffusion depuis l'adoption de la Loi sur la radiodiffusion (fédérale) en 1991 en se concentrant plus particulièrement sur la transition à la télévision numérique. Pour le Canada, ce changement de cap est révélateur des grandes modifications que les politiques publiques ont connues dans les deux dernières décennies : une foi dans les mécanismes de marché, une réglementation en pointillé, des approches axées sur la coréglementation et la puissante influence des nouvelles technologies. La question centrale étudiée dans ce mémoire est la suivante : la politique sous-jacente à la transition à la télédiffusion numérique constitue-t-elle un défi en regard des préoccupations traditionnelles afférentes au rôle de la radiodiffusion dans la démocratie canadienne ? Adoptant une approche de économie politique institutionnelle, cette étude se fonde sur l'étude de documents gouvernementaux ou provenant des entreprises privées oeuvrant dans le secteur ainsi que sur des entretiens effectués avec certains acteurs clés de la radiodiffusion canadienne. Cette thèse offre une contribution unique au savoir quant à trois aspects : établir des paramètres clairs permettant de différencier les politiques de la réglementation dans le domaine de la radiodiffusion ; offrir la première étude détaillée sur la transition canadienne à la télévision numérique, et présenter une analyse du développement et de l'impact de l'autoréglementation et de la coréglementation dans les politiques canadiennes relatives à la radiodiffusion. Les conclusions de cette étude mettent en lumière les dynamiques de pouvoir existant entre les divers acteurs engagés dans le processus d'élaboration et d'adoption des politiques canadiennes, l'influence des nouvelles technologies et les grandes orientations politiques ayant prévalu dans le domaine de la radiodiffusion depuis l'adoption d
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Moreno, Martinez Eduardo. "Implementation of Digital Audio Broadcasting System based in SystemC Library." Thesis, Linköping University, Department of Electrical Engineering, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2411.
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This thesis describes the design and implementation of a Digital Audio Broadcasting (DAB) System developed using C++ Language and SystemC libraries. The main aspects covered within this report are the data structure of DAB system, and some interesting points of SystemC Library very useful for the implementation of the final system.
It starts with a introduction of DAB system and his principals advantages. Next it goes further into the definition of data structures of DAB, they are FIC, MSC, and DAB audio frame, explained with MPEG and PAD packets. Later on this chapter there is an explanation of the SystemC library with special attention on the features that I used to implement the system. This features are the events used in the communication between processes and the interfaces needed for sending and receiving the data.
With all these points covered is quite easy for a reader to understand the implementation of the system, despite this point is covered in the last chapter of the thesis. The implementation is here explained in two different steps. The first one explain how is formed the DAB audio frame by means of MPEG frames that are wrote in channel by producer interface, this frames are readed by consumer interface. For this purpose I have created some classes and structures that are explained in this part. The second part explain how I obtain the DAB transmission frame which is obtained creating MSC frames, that are big data structures formed by groups of DAB audio frames, therefore there are some functions that act like a buffer and add audio frames to the MSC data structure. Of independent way there is the FIC frame that is generated of random way and its added to the transmission frame.
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Kim, Dukhyun. "Orthogonal frequency division multiplexing for digital broadcasting." Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/13704.
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Hewavithana, Thushara Chinthaka. "Algorithms for long delay equalization and soft decision decoding in digital audio broadcasting receivers." Thesis, Imperial College London, 2005. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.415088.
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Igumbor, Osedum Peter. "A proxy approach to protocol interoperability within digital audio networks." Thesis, Rhodes University, 2010. http://hdl.handle.net/10962/d1004852.
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Digital audio networks are becoming the preferred solution for the interconnection of professional audio devices. Prominent amongst their advantages are: reduced noise interference, signal multiplexing, and a reduction in the number of cables connecting networked devices. In the context of professional audio, digital networks have been used to connect devices including: mixers, effects units, preamplifiers, breakout boxes, computers, monitoring controllers, and synthesizers. Such networks are governed by protocols that define the connection management rocedures, and device synchronization processes of devices that conform to the protocols. A wide range of digital audio network control protocols exist, each defining specific hardware requirements of devices that conform to them. Device parameter control is achieved by sending a protocol message that indicates the target parameter, and the action that should be performed on the parameter. Typically, a device will conform to only one protocol. By implication, only devices that conform to a specific protocol can communicate with each other, and only a controller that conforms to the protocol can control such devices. This results in the isolation of devices that conform to disparate protocols, since devices of different protocols cannot communicate with each other. This is currently a challenge in the professional music industry, particularly where digital networks are used for audio device control. This investigation seeks to resolve the issue of interoperability between professional audio devices that conform to different digital audio network protocols. This thesis proposes the use of a proxy that allows for the translation of protocol messages, as a solution to the interoperability problem. The proxy abstracts devices of one protocol in terms of another, hence allowing all the networked devices to appear as conforming to the same protocol. The proxy receives messages on behalf of the abstracted device, and then fulfills them in accordance with the protocol that the abstracted device conforms to. Any number of protocol devices can be abstracted within such a proxy. This has the added advantage of allowing a common controller to control devices that conform to the different protocols.
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Farrugia, Maria. "Combined speech and audio coding with bit rate and bandwidth scalability." Thesis, University of Surrey, 2001. http://epubs.surrey.ac.uk/843624/.
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The past two decades have witnessed a rapid expansion within the telecommunications industry. This growth has been primarily motivated by the proliferation of digital communication systems and services which have become easily available through wired and wireless systems. Current research trends involve the integration of speech, audio, video and data channels into true multimedia communications over fixed and mobile networks. However, while the available bandwidth in wired terrestrial networks is relatively cheap and expandable, it becomes a limited resource in satellite and cellular-radio systems. In order to accommodate an ever growing number of users while maintaining high quality and low operational costs, it is necessary to maximise spectral efficiency. This has given rise to the development of high rate compression techniques with the ability to adapt to a broad class of input signals and to varying network resources. The research carried out in this thesis has mainly focused on the design of a single algorithm for compressing speech and audio signals sampled at different rates. The algorithms are based on the analysis-by-synthesis linear prediction coding (AbS-LPC) scheme, which has been widely employed in various speech coding standards. However, this bit rate reduction technique is based on the speech production mechanism and as such provides a rigid structure which presents a major limitation for audio coding. In order to improve the audio quality at low rates and to compensate for the errors incurred by the linear prediction during segments of high transitions, the algorithms employ an efficient pulse excitation structure which represents the short innovation sequences with sparse unit magnitude pulses. The scheme proposed for the compression of telephone bandwidth speech and audio signals at 12kb/s achieves similar quality to the G.728 coder at 16kb/s and higher audio quality than the GSM-EFR standard at 12.2kb/s. Wideband speech and audio coding schemes have been designed using both the fullband approach at bit rates of 17 and 19kb/s and also the split band technique at a bit rate of 20kb/s. The perceptual quality is comparable to the G.722 coder operating at 48kb/s. The subband decomposition technique is also adapted to code speech and audio signals sampled at 32kHz. The quality of the coder at 28kb/s is similar to the quality achieved by the MP3 coder at 32kb/s. The algorithm also provides bandwidth and bit rate scalability ranging from 12 to 64kb/s, making it ideal for deployment in rate-adaptive communication systems.
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Hu, Anqiao. "Multi-modulus divider in fractional-N frequency synthesizer for direct conversion DVB-H receiver." Columbus, Ohio : Ohio State University, 2008. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1196105249.
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Lee, Hyangsun. "How to manage the issue of unauthorized copying and distribution of digital audio broadcasts constructive alternatives to technological mandates /." [Bloomington, Ind.] : Indiana University, 2008. http://gateway.proquest.com/openurl?url_ver=Z39.88-2004&rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&res_dat=xri:pqdiss&rft_dat=xri:pqdiss:3331294.
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Thesis (Ph.D.)--Indiana University, Dept. of Telecommunications, 2008.Title from PDF t.p. (viewed on Jul 24, 2009). Source: Dissertation Abstracts International, Volume: 69-11, Section: A, page: 4174. Adviser: Michael McGregor.
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Rennie, Elinor Mary. "The Future of Community Broadcasting: Civil Society and Communications Policy." Queensland University of Technology, 2003. http://eprints.qut.edu.au/15829/.
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Will community television one day be lamented in the same way as the Glenn Valley Bridge Club in Pennsylvania, where no one remains 'who can tell us precisely when or why the group broke up' (Putnam, 2000: 15)? Robert Putnam's bestseller Bowling Alone proposed that people 'need to reconnect with one another' and rebuild their communities for the good of society. Although he may not have succeeded in instigating a revival of lawn bowls and bridge, Putnam did spark a debate about the meaning of "community" today and its role in bringing about positive social change. At a time when the communications landscape is set to transform with the introduction of digital broadcasting technology, this thesis looks at the status of community broadcasting and its role within civil society. Taking Australia's community television sector as its starting point, it aims to define the pressures, public philosophies and policy decisions that make community broadcasting what it is. This thesis is structured thematically and geographically. The introductory chapters establish the research question in relation to Australia's community broadcasting sector. As well as tracing the intellectual path of community media studies, it sets out to locate community broadcasting within broader intellectual debates around notions of community, governance and the media. These are brought back to the "on-the-ground" reality throughout the thesis by means of policy analysis, interviews and anecdotal evidence. Chapters Three to Five map out the themes of access, the public interest and development by reference to community broadcasting in different regions. In North America I explore notions of free speech and first-come-first served models of access. In Europe, notions of "quality", public service broadcasting and the difficult relationship that community broadcasting has with public interest values. Through the Third World and the Third Way I examine how community broadcasting is implicated within development discourse and ideas of social change. The final chapter of the thesis moves into the virtual region of the Internet, looking at changing notions of access and the relevance of new communications rationales to the community broadcasting project. At the intersection of the various themes and models discussed throughout the thesis exists a strong rationale for the future of community broadcasting. Although new technologies may be interpreted as the beginning of the end of community broadcasting, I have argued that in fact it is an idea whose time has come.
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Devlin, John P. "The relationship between the BBC and the commercial radio sector in promoting DAB (Digital Audio Broadcasting) in the United Kingdom." Thesis, Bournemouth University, 2016. http://eprints.bournemouth.ac.uk/25359/.
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The place of Digital Audio Broadcasting (DAB) in the UK provides an interesting area of study. As of 2015, it is a platform which has been in existence for twenty years. During that time its position has been nebulous. It represents only one method of listening to radio output in a digital format but one in which the BBC, and latterly, the commercial radio sector, placed a huge degree of trust. It has however not become the standard mode of radio listening. One can argue that after much proactive effort from within the UK radio industry, its position may be beginning to falter somewhat in the commercial sector while remaining a paramount policy objective for the BBC. The aim of this thesis is to account for the emergence of DAB and its impact on the relationship between the BBC and the commercial radio sector. It examines the separate and joint roles of each in promoting DAB and highlights how this led to a unique period of cooperation between the two. A historical backdrop is necessary in order to establish the previous state of relations, before embarking upon the core of the thesis which argues that both parties adopted significant changes of posture in order to promote DAB. I argue that the implementation of DAB as a new radio technology represents a marked period of institutional change within the UK radio industry. In playing distinct, separate roles, I show how this reveals a maturity within the industry whereby the BBC and the commercial sector reached a point of equality. In promoting a joint role, based on a policy of overt cooperation, I demonstrate how this represented a major shift in the historical relationship between both parties. Relying on a small canon of secondary sources and an extensive survey of primary source documentation, as well as interviews with some of the key players in DAB and the UK radio industry as a whole, this thesis provides a study of a significant period in British radio history which witnessed the development of a nascent radio platform, coupled with a significant change in the dynamic of existing relationships within the industry. In doing so, it delivers a wider historical interpretation of a particular point in radio history and marks a critical phase when the very nature of the industry changed.
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Reese, Stephanie. "Der Funktionsauftrag des öffentlich-rechtlichen Rundfunks vor dem Hintergrund der Digitalisierung : zur Konkretisierung des Funktionsauftrages in 11 Rundfunkstaatsvertrag /." Frankfurt am Main [u.a.] : Lang, 2006. http://www.gbv.de/dms/ilmenau/toc/511997515.PDF.
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Stensgaard, Pontus, Anders Alléus, and Jesper Palm. "Adaptive Mood Audio : Rethinking Audio for Games." Thesis, Blekinge Tekniska Högskola, Sektionen för planering och mediedesign, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-2869.
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The focus of this thesis is to study the way that adaptive audio can be used in digital games and how it can be used to portray different moods to the listener, how games can reflect different feelings and how quickly those feelings can change. Games audio environment is significant to be able to adapt to the ever changing narrative of the game. The purpose is to gain insight in how immersion in digital games can be improved with the use of adaptive audio to study if there’s an easy way to implement a system where audio can be mixed and adjusted in realtime to mirror the events in the game and project accurate feelings. To study this we will create a parameter based system in the sound engine of the game we will make during the production phase, with different parameters based on a number of different factors . Keywords: Adaptive Audio, Parameter System, Mood Music, Digital Games, Game Production Syftet med detta kandidatarbete är att studera hur adaptiva ljud kan användas i digitala spel, hur de kan användas för att beskriva olika stämningar till lyssnaren, och hur spel kan spegla olika känslor och hur snabbt dessa känslor kan förändras. Det är viktigt att ett spels ljudmiljö kan anpassa sig till den ständigt föränderliga berättelsen i spelet. Syftet är att få en inblick i hur immersion i digitala spel kan förbättras med användning av adaptiva ljud att studera om det finns ett enkelt sätt att implementera ett system där ljudet kan mixas och anpassas i realtid för att spegla händelserna i spelet och återskapa äkta känslor. För att undersöka detta kommer vi skapa ett parameterbaserat system i ljudmotorn till spelet som vi kommer göra under produktionsfasen, med olika parameterar som är baserade på ett antal olika faktorer. Nyckelord: Adaptivt Ljud, Parametersystem, Stämningsmusik, Digitala Spel, Spelproduktion.Ett arbete som tar upp hur man kan få immersionen i digitala spel att förbättras med användning av adaptivt ljud och musik för att spegla narrativet.
Pontus Stensgaard: 0769123182 pontus.stensgaard@gmail.com
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Okai-Tettey, Harold A. "High speed end-to-end connection management in a bridged IEEE 1394 network of professional audio devices." Thesis, Rhodes University, 2006. http://hdl.handle.net/10962/d1006638.
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A number of companies have developed a variety of network approaches to the transfer of audio and MIDI data. By doing this, they have addressed the configuration complications that were present when using direct patching for analogue audio, digital audio, word clock, and control connections. Along with their approaches, controlling software, usually running on a PC, is used to set up and manage audio routings from the outputs to the inputs of devices. However one of the advantages of direct patching is the conceptual simplicity it provides for a user in connecting plugs of devices, the ability to connect from the host plug of one device to the host plug of another. The connection management or routing applications of the current audio networks do not allow for such a capability, and instead employ what is referred to as a two-step approach to connection management. This two-step approach requires that devices be first configured at the transport layer of the network for input and output routings, after which the transmit and receive plugs of devices are manually configured to transmit or receive data. From a user’s point of view, it is desirable for the connection management or audio routing applications of the current audio networks to be able to establish routings directly between the host plugs of devices, and not the audio channels exposed by a network’s transport, as is currently the case. The main goal of this work has been to retain the conceptual simplicity of point-to-point connection management within digital audio networks, while gaining all the benefits that digital audio networking can offer.
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Gorthy, Anantha Surya Raghu. "A Study on the Effects of Decoder Quantization of Digital Video Broadcasting - Return Channel over Satellite (DVB-RCS) Turbo Codes." Ohio University / OhioLINK, 2008. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1226965326.
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Chigwamba, Nyasha. "An investigation into the hardware abstraction layer of the plural node architecture for IEEE 1394 audio devices." Thesis, Rhodes University, 2009. http://hdl.handle.net/10962/d1004841.
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Digital audio network technologies are becoming more prevalent in audio related environments. Yamaha Corporation has created a digital audio network solution, named mLAN (music Local Area Network), that uses IEEE 1394 as its underlying network technology. IEEE 1394 is a digital network technology that is specifically designed for real-time multimedia data transmission. The second generation of mLAN is based on the Plural Node Architecture, where the control of audio and MIDI routings between IEEE 1394 devices is split between two node types, namely an Enabler and a Transporter. The Transporter typically resides in an IEEE 1394 device and is solely responsible for transmission and reception of audio or MIDI data. The Enabler typically resides in a workstation and exposes an abstract representation of audio or MIDI plugs on each Transporter to routing control applications. The Enabler is responsible for configuring audio and MIDI routings between plugs on different Transporters. A Hardware Abstraction Layer (HAL) within the Enabler allows it to uniformly communicate with Transporters that are created by various vendors. A plug-in mechanism is used to provide this capability. When vendors create Transporters, they also create device-specific plug-ins for the Enabler. These plug-ins are created against a Transporter HAL Application Programming Interface (API) that defines methods to access the capabilities of Transporters. An Open Generic Transporter (OGT) guideline document which models all the capabilities of Transporters has been produced. These guidelines make it possible for manufacturers to create Transporters that make use of a common plug-in, although based on different hardware architectures. The introduction of the OGT concept has revealed additional Transporter capabilities that are not incorporated in the existing Transporter HAL API. This has led to the underutilisation of OGT capabilities. The main goals of this investigation have been to improve the Enabler’s plug-in mechanism, and to incorporate the additional capabilities that have been revealed by the OGT into the Transporter HAL API. We propose a new plug-in mechanism, and a new Transporter HAL API that fully utilises both the additional capabilities revealed by the OGT and the capabilities of existing Transporters.
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Scoma, David. "THE DEVELOPMENT OF LOOP-BASED CINEMATIC TECHNIQUES IN TWENTIETH CENTURY MOTION PICTURES AND THEIR APPLICATION IN EARLY DIGITAL C." Doctoral diss., University of Central Florida, 2008. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/2227.
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For centuries, repetition in one form or another has been seen as a significant element in the artistic palette. In numerous formats of expression, duplication and looping became a significant tool utilized by artisans in a multitude of creative formats. Yet within the realm of film, the Griffith and Eisenstein models of cinematic editing techniques (as the most popular-- and near-monolithic--narrative aesthetic criteria) effectively disregarded most other approaches, including looping. Despite the evidence for the consistent use of repetition and looping in multiple ways throughout the course of cinematic history, some theorists and practitioners maintain that the influx of the technique within digital cinema in recent years represents a sudden breakthrough, one that has arrived simply because technology has currently advanced to a point where their utilization within digital formats now makes sense both technologically and aesthetically. This situation points to a cyclical problem. Students of film and video frequently are not taught aesthetical or editorial options other than standard industry procedures. Those who are interested in varying techniques are therefore put in the position of having to learn alternative practices on their own. When they do look beyond visual norms to try applying different approaches in their projects, they risk going against the views of their instructors who are only interested in implementations of the standard methods which have been in the forefront for so long. Yet the loop s importance and prevalence as a digital language tool will only likely grow with the evolution of digital cinema. With this is mind, the dissertation addresses the following questions: To what extent can various forms of repetitive visuals be found throughout film history, and are not simply technical manifestations that have merely emerged within digital cinema? How might current educational practices in the realm of film and video work to inform students of techniques outside of the common narrative means? Finally, what other sources or strategies might be available to enlighten students and practitioners exploring both the history surrounding--and possible applications of--techniques based upon early cinema practices such as the loop?Ph.D.
Department of English
Arts and Humanities
Texts and Technology PhD
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Sibanda, Phathisile. "Connection management applications for high-speed audio networking." Thesis, Rhodes University, 2008. http://hdl.handle.net/10962/d1006532.
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Traditionally, connection management applications (referred to as patchbays) for high-speed audio networking, are predominantly developed using third-generation languages such as C, C# and C++. Due to the rapid increase in distributed audio/video network usage in the world today, connection management applications that control signal routing over these networks have also evolved in complexity to accommodate more functionality. As the result, high-speed audio networking application developers require a tool that will enable them to develop complex connection management applications easily and within the shortest possible time. In addition, this tool should provide them with the reliability and flexibility required to develop applications controlling signal routing in networks carrying real-time data. High-speed audio networks are used for various purposes that include audio/video production and broadcasting. This investigation evaluates the possibility of using Adobe Flash Professional 8, using ActionScript 2.0, for developing connection management applications. Three patchbays, namely the Broadcast patchbay, the Project studio patchbay, and the Hospitality/Convention Centre patchbay were developed and tested for connection management in three sound installation networks, namely the Broadcast network, the Project studio network, and the Hospitality/Convention Centre network. Findings indicate that complex connection management applications can effectively be implemented using the Adobe Flash IDE and ActionScript 2.0.
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Foulkes, Philip James. "A grid based approach for the control and recall of the properties of IEEE 1394 audio devices." Thesis, Rhodes University, 2009. http://hdl.handle.net/10962/d1004836.
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The control of modern audio studios is complex. Audio mixing desks have grown to the point where they contain thousands of parameters. The control surfaces of these devices do not reflect the routing and signal processing capabilities that the devices are capable of. Software audio mixing desk editors have been developed that allow for the remote control of these devices, but their graphical user interfaces retain the complexities of the audio mixing desk that they represent. In this thesis, we propose a grid approach to audio mixing. The developed grid audio mixing desk editor represents an audio mixing desk as a series of graphical routing matrices. These routing matrices expose the various signal processing points and signal flows that exist within an audio mixing desk. The routing matrices allow for audio signals to be routed within the device, and allow for the device’s parameters to be adjusted by selecting the appropriate signal processing points. With the use of the programming interfaces that are defined as part of the Studio Connections – Total Recall SDK, the audio mixing desk editor was integrated with compatible DAW applications to provide persistence of audio mixing desk parameter states. Many audio studios currently use digital networks to connect audio devices together. Audio and control signals are patched between devices through the use of software patchbays that run on computers. We propose a double grid-based FireWire patchbay aimed to simplify the patching of signals between audio devices on a FireWire network. The FireWire patchbay was implemented in such a way such that it can host software device editors that are Studio Connections compatible. This has allowed software device editors to be associated with the devices that are represented on the FireWire patchbay, thus allowing for studio wide control from a single application. The double grid-based patchbay was implemented such that it can be hosted by compatible DAW applications. Through this, the double grid-based patchbay application is able to provide the DAW application with the state of the parameters of the devices in a studio, as well as the connections between them. The DAW application may save this state data to its native song files. This state data may be passed back to the double grid-based patchbay when the song file is reloaded at a later stage. This state data may then be used by the patchbay to restore the parameters of the patchbay and its device editors to a previous state. This restored state may then be transferred to the hardware devices being represented by the patchbay.
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Ernest, Petro Pesha. "DAB implementation in SDR." Thesis, Stellenbosch : University of Stellenbosch, 2005. http://hdl.handle.net/10019.1/2943.
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Thesis (MScEng (Electrical and Electronic Engineering))--University of Stellenbosch, 2005.The aim of this thesis is to implement a Digital Audio Broadcasting (DAB) system in a Software Defined Radio (SDR). The physical modulation part of the DAB transmitter for one of the transmission modes as well as its receiver is to be implemented and tested in the SDR. DAB transmission mode II is implemented.
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Anantharaman, B. "Compressed Domain Processing of MPEG Audio." Thesis, Indian Institute of Science, 2001. http://hdl.handle.net/2005/68.
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MPEG audio compression techniques significantly reduces the storage and transmission requirements for high quality digital audio. However, compression complicates the processing of audio in many applications. If a compressed audio signal is to be processed, a direct method would be to decode the compressed signal, process the decoded signal and re-encode it. This is computationally expensive due to the complexity of the MPEG filter bank. This thesis deals with processing of MPEG compressed audio. The main contributions of this thesis are
a) Extracting wavelet coefficients in the MPEG compressed domain.
b) Wavelet based pitch extraction in MPEG compressed domain.
c) Time Scale Modifications of MPEG audio.
d) Watermarking of MPEG audio.
The research contributions starts with a technique for calculating several levels of wavelet coefficients from the output of the MPEG analysis filter bank. The technique exploits the toeplitz structure which arises when the MPEG and wavelet filter banks are represented in a matrix form, The computational complexity for extracting several levels of wavelet coefficients after decoding the compressed signal and directly from the output of the MPEG analysis filter bank are compared. The proposed technique is found to be computationally efficient for extracting higher levels of wavelet coefficients.
Extracting pitch in the compressed domain becomes essential when large multimedia databases need to be indexed. For example one may be interested in listening to a particular speaker or to listen to male female audio segments in a multimedia document. For this application, pitch information is one of the very basic and important features required. Pitch is basically the time interval between two successive glottal closures. Glottal closures are accompanied by sharp transients in the speech signal which in turn gives rise to a local maxima in the wavelet coefficients. Pitch can be calculated by finding the time interval between two successive maxima in the wavelet coefficients. It is shown that the computational complexity for extracting pitch in the compressed domain is less than 7% of the uncompressed domain processing. An algorithm for extracting pitch in the compressed domain is proposed. The result of this algorithm for synthetic signals, and utterances of words by male/female is reported.
In a number of important applications, one needs to modify an audio signal to render it more useful than its original. Typical applications include changing the time evolution of an audio signal (increase or decrease the rate of articulation of a speaker),or to adapt a given audio sequence to a given video sequence. In this thesis, time scale modifications are obtained in the subband domain such that when the modified subband signals are given to the MPEG synthesis filter bank, the desired time scale modification of the decoded signal is achieved. This is done by making use of sinusoidal modeling [I]. Here, each of the subband signal is modeled in terms of parameters such as amplitude phase and frequencies and are subsequently synthesised by using these parameters with Ls = k La where Ls is the length of the synthesis window , k is the time scale factor and La is the length of the analysis window. As the PCM version of the time scaled signal is not available, psychoacoustic model based bit allocation cannot be used. Hence a new bit allocation is done by using a subband coding algorithm. This method has been satisfactorily tested for time scale expansion and compression of speech and music signals.
The recent growth of multimedia systems has increased the need for protecting digital media. Digital watermarking has been proposed as a method for protecting digital documents. The watermark needs to be added to the signal in such a way that it does not cause audible distortions. However the idea behind the lossy MPEC encoders is to remove or make insignificant those portions of the signal which does not affect human hearing. This renders the watermark insignificant and hence proving ownership of the signal becomes difficult when an audio signal is compressed. The existing compressed domain methods merely change the bits or the scale factors according to a key. Though simple, these methods are not robust to attacks. Further these methods require original signal to be available in the verification process. In this thesis we propose a watermarking method based on spread spectrum technique which does not require original signal during the verification process. It is also shown to be more robust than the existing methods. In our method the watermark is spread across many subband samples. Here two factors need to be considered, a) the watermark is to be embedded only in those subbands which will make the addition of the noise inaudible. b) The watermark should be added to those subbands which has sufficient bit allocation so that the watermark does not become insignificant due to lack of bit allocation. Embedding the watermark in the lower subbands would cause distortion and in the higher subbands would prove futile as the bit allocation in these subbands are practically zero. Considering a11 these factors, one can introduce noise to samples across many frames corresponding to subbands 4 to 8. In the verification process, it is sufficient to have the key/code and the possibly attacked signal. This method has been satisfactorily tested for robustness to scalefactor, LSB change and MPEG decoding and re-encoding.
26
Zhao, Yue. "Independent Component Analysis Enhancements for Source Separation in Immersive Audio Environments." UKnowledge, 2013. http://uknowledge.uky.edu/ece_etds/34.
Full textAbstract:
In immersive audio environments with distributed microphones, Independent Component Analysis (ICA) can be applied to uncover signals from a mixture of other signals and noise, such as in a cocktail party recording. ICA algorithms have been developed for instantaneous source mixtures and convolutional source mixtures. While ICA for instantaneous mixtures works when no delays exist between the signals in each mixture, distributed microphone recordings typically result various delays of the signals over the recorded channels. The convolutive ICA algorithm should account for delays; however, it requires many parameters to be set and often has stability issues. This thesis introduces the Channel Aligned FastICA (CAICA), which requires knowledge of the source distance to each microphone, but does not require knowledge of noise sources. Furthermore, the CAICA is combined with Time Frequency Masking (TFM), yielding even better SOI extraction even in low SNR environments. Simulations were conducted for ranking experiments tested the performance of three algorithms: Weighted Beamforming (WB), CAICA, CAICA with TFM. The Closest Microphone (CM) recording is used as a reference for all three. Statistical analyses on the results demonstrated superior performance for the CAICA with TFM. The algorithms were applied to experimental recordings to support the conclusions of the simulations. These techniques can be deployed in mobile platforms, used in surveillance for capturing human speech and potentially adapted to biomedical fields.
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Jaff, Esua Kinyuy. "IP mobile multicast over next generation satellite networks : design and evaluation of a seamless mobility framework for IP multicast communications over a multi-beam geostationary satellite network." Thesis, University of Bradford, 2016. http://hdl.handle.net/10454/14581.
Full textAbstract:
The inherent broadcast nature of satellites, their global coverage and direct access to a large number of subscribers give satellites unrivalled advantages in supporting IP multicast applications. A new generation of satellite systems that support regenerative on-board processors and multiple spot beam technology have opened new possibilities of implementing IP multicast communication over satellites. These new features enable satellites to make efficient use of their allocated bandwidth resources and provide cost effective network services but equally, create new challenges for mobile satellite terminals. IP mobility support in general and IP mobile multicast support in particular on mobile satellite terminals like the ones mounted on continental flights, maritime vessels, etc., still remain big challenges that have received very little attention from the research community. Up till now, there are no proposed mechanisms to support IP multicast for mobile receivers/sources in multi-beam satellite networks in open literature. This study explores the suitability of IP multicast mobility support schemes defined for terrestrial networks in a satellite environment and proposes novel schemes based on the concepts of Home and Remote subscription-based approaches, multiple interface and PMIPv6 protocol. Detailed analysis and comparison of results obtained from the proposed schemes, Mobile IP (MIP) Home and Remote subscription-based approaches (for terrestrial networks) when implemented on a reference multi-beam satellite network are presented. From these results, the proposed schemes outperform the MIP Home and Remote subscription-based approaches in terms of gateway handover latency, number of multicast packets lost and signalling cost over the satellite air interface.
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Jaff, Esua K. "IP Mobile Multicast over Next Generation Satellite Networks. Design and Evaluation of a Seamless Mobility Framework for IP Multicast Communications over a Multi-beam Geostationary Satellite Network." Thesis, University of Bradford, 2016. http://hdl.handle.net/10454/14581.
Full textAbstract:
The inherent broadcast nature of satellites, their global coverage and direct
access to a large number of subscribers give satellites unrivalled advantages in
supporting IP multicast applications. A new generation of satellite systems that
support regenerative on-board processors and multiple spot beam technology
have opened new possibilities of implementing IP multicast communication over
satellites. These new features enable satellites to make efficient use of their
allocated bandwidth resources and provide cost effective network services but
equally, create new challenges for mobile satellite terminals. IP mobility support
in general and IP mobile multicast support in particular on mobile satellite
terminals like the ones mounted on continental flights, maritime vessels, etc.,
still remain big challenges that have received very little attention from the
research community.
Up till now, there are no proposed mechanisms to support IP multicast for
mobile receivers/sources in multi-beam satellite networks in open literature.
This study explores the suitability of IP multicast mobility support schemes
defined for terrestrial networks in a satellite environment and proposes novel
schemes based on the concepts of Home and Remote subscription-based approaches, multiple interface and PMIPv6 protocol. Detailed analysis and
comparison of results obtained from the proposed schemes, Mobile IP (MIP)
Home and Remote subscription-based approaches (for terrestrial networks)
when implemented on a reference multi-beam satellite network are presented.
From these results, the proposed schemes outperform the MIP Home and
Remote subscription-based approaches in terms of gateway handover latency,
number of multicast packets lost and signalling cost over the satellite air
interface.
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Watier, Maxime. "Le VJing ou la visualisation du son dans les performances audio-visuelles contemporaines." Thesis, Sorbonne Paris Cité, 2016. http://www.theses.fr/2016USPCA023.
Full textAbstract:
Cette thèse a pour objet le VJing, terme désignant l’activité du VJ (pour Video ou Visual Jockey) consistant à créer et manipuler en direct plusieurs sources visuelles par médiation technologique en direction d’un public et en synchronisation avec la musique, et se propose de mettre cette pratique en perspective en l’envisageant comme un travail spécifique opéré sur l’image, interrogeant son paradigme poïétique fondé sur la synesthésie aussi bien que le contexte performatif dans lequel elle se déploie. Empruntant le concept de re-médiation à Jay David Bolter et Richard Grusin, nous étudions les divers processus par lesquels le VJing se réapproprie, tant par la forme que par le contenu, les propriétés constitutives du DJing et des musiques électroniques, tout comme il prolonge directement ou indirectement certains procédés d’union du visuel et du sonore expérimentés sur ces deux médiums audiovisuels que sont le cinéma et la vidéo. S’appuyant sur un corpus de près d’une centaine de créations VJ (extraits de performances, mix réalisés pour support vidéo), notre approche méthodologique consiste à mener plusieurs analyses comparatives avec des œuvres issues du cinéma d’avant-garde, du cinéma expérimental, de l’Art Vidéo, du vidéo-clip, afin de définir progressivement l’identité spécifique du VJing. En combinant perspective théorique, historique et esthétique, nous examinons par quels biais cette pratique a constitué avec le temps un dispositif technologique et projectif visant à la réalisation en temps réel d’un objet audiovisuel basé sur le dialogue constant entre les images en mouvement et la musique (ou la bande-son) et dans quelle mesure ce type de performance se déployant dans le contexte des nouveaux médias, du remix généralisé et du « cinéma élargi » entre en résonnance avec la tradition issue des avant-gardes et celle, aussi bien picturale, musicale que cinématographique, de la Visual MusicThe subject of this thesis is VJing, a term referring to the work performed by the VJ (Video or Visual Jockey) consisting of creating and manipulating several visual sources in real-time through technological mediation and for an audience, in synchronization to music. This study intends to put this practice in perspective by considering it as a specific work made over the image, questioning its poietic paradigm based on synaesthesia as well as the performative context in which it takes place. Borrowing the concept of Remediation from Jay David Bolter and Richard Grusin, we study the various processes by which VJing re-appropriates, both in form and content, some constitutive properties of DJing and electronic music, as it also extends directly or indirectly some processes of union of the visual and the aural as it has been experimented in cinema and video. Based on a corpus of about a hundred of VJ works (performance extracts, mixes made for video distribution), our methodological approach consists of conducting several comparative analysis with different works taken from avant-garde cinema, experimental cinema, video art and music videos, in order to progressively defining the specific identity of VJing. By combining theoretical, historical and aesthetic perspectives, we analyze the means by which this practice has established a technological and projective device over time to achieve producing audio-visual objects in real-time based on the constant dialogue between moving pictures and music (or soundtrack) and we try to see how this kind of performance, taking place in the context of new media, mainstreamed remix and "expanded cinema", echoes with the avant-garde tradition and the pictorial, musical as well as cinematic legacy of Visual Music
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Cullell, March Cristina. "La Política del Espectro Radioeléctrico en la Unión Europea: la Armonización del Dividendo Digital en el Reino Unido y España." Doctoral thesis, Universitat Internacional de Catalunya, 2010. http://hdl.handle.net/10803/9352.
Full textAbstract:
L'objecte d'estudi d'aquesta tesi es centra en la política europea de l'espectre radioelèctric i presta especial atenció a l'harmonització del dividend digital al Regne Unit i Espanya. La metodologia es basa en una revisió bibliogràfica i hemerogràfica completada amb un conjunt d'entrevistes amb representants de diferents organismes de regulació del Regne Unit i Espanya, i de la mateixa Comissió Europea. En la primera part de la tesi es realitza una aproximació conceptual, des d'una vessant tècnica i jurídica, a la xarxa radioelèctrica i al dividend digital, espai alliberat una vegada completada la transició a la TDT. Seguidament, s'aborda la naturalesa, àmbits de regulació de la xarxa radioelèctrica, actors, models i evolució de les formes de gestió. El tercer capítol es centra en la Unió Europea com actor de la política de l'espectre radioelèctric des de mitjans de la dècada dels 80 fins a l'actualitat. En la darrera part de la tesi es descriuen i analitzen les actuacions de la UE destinades a harmonitzar el dividend digital fent especial referència a les conseqüències d'aquesta harmonització sobre les estructures nacionals i la planificació de la TDT en els dos països estudiats, el Regne Unit i Espanya.El objeto de estudio de esta tesis se centra en la política europea del espectro radioeléctrico y presta una especial atención a la armonización del dividendo digital en el Reino Unido y España. La metodología está basada en una revisión bibliográfica y hemerográfica completada con un conjunto de entrevistas con representantes de diferentes organismos de regulación del Reino Unido y España, y de la misma Comisión Europea. En la primera parte de la tesis se realiza una aproximación conceptual, desde una vertiente técnica y jurídica, a la red radioeléctrica y al dividendo digital, espacio liberado una vez se ha completado la transición a la TDT. Seguidamente, se aborda la naturaleza y ámbitos de la regulación de la red radioeléctrica, sus actores, modelos y evolución de sus formas de gestión. El tercer capítulo se centra en la Unión Europea como actor de la política del espectro radioeléctrico desde mediados de la década de los 80 hasta la actualidad. En la última parte de la tesis se describen y analizan las actuaciones de la UE destinadas a armonizar el dividendo digital y se atiende especialmente a las consecuencias que esta armonización tiene sobre las estructuras nacionales y la planificación de la TDT en los dos países estudiados, el Reino Unido y España.
The object of study of this thesis focuses on the European radio spectrum policy in particular the harmonization of the digital dividend in the United Kingdom and Spain. The methodology is based on a multidisciplinary bibliographic review completed with an in-depth interviews with policy-makers from the national regulatory authorities and the European Commission. The first chapter of the dissertation consists of a conceptual approach of radio spectrum network and the digital dividend, the amount space released once the switch-over is completed. Afterwards, it deals with the nature of spectrum regulation, its actors, models and evolution of spectrum management. Its third chapter focuses on the European Union as a spectrum policy actor from the 80's onwards. Finally, this thesis analyses the EU actions oriented to the harmonisation of the digital dividend, and pays special attention to its consequences on national structures and DTT plans of the United Kingdom and Spain.
31
Blake, Greyory. "Good Game." VCU Scholars Compass, 2018. https://scholarscompass.vcu.edu/etd/5377.
Full textAbstract:
This thesis and its corresponding art installation, Lessons from Ziggy, attempts to deconstruct the variables prevalent within several complex systems, analyze their transformations, and propose a methodology for reasserting the soap box within the display pedestal. In this text, there are several key and specific examples of the transformation of various signifiers (i.e. media-bred fear’s transformation into a political tactic of surveillance, contemporary freneticism’s transformation into complacency, and community’s transformation into nationalism as a state weapon). In this essay, all of these concepts are contextualized within the exponential growth of new technologies. That is to say, all of these semiotic developments must be framed within the post-Internet sphere.
32
Huang, Lee-Tsair, and 黃李財. "Digital Audio Broadcasting Principles and Applications." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/15787488217863120845.
Full textAbstract:
碩士淡江大學
資訊工程學系
92
While steps into the 21st century, the whole world is carrying on digital revolutions, the information, the communication, as well as the entertainment presented by the rich lively content forms; Therefore, the broadcast service need to preserves the audience even continues to develop, at present, it then must compete against Internet, mobile phone, as well as digital televisions and so on, in the technology and the application.Today, while the traditional broadcast (the AM amplitude modulation and the FM frequency modulation) have been regarded as the obsolete technology, and will be replace by the other consumer technologies, hence, the broadcast technology must innovate and the modernization, achieves the same as CD and DVD qualities . The broadcast must become multimedia service provider, simultaneously maintains the mobility and may gain the information everywhere. In other words, the broadcast technology and the service needs to carry on the big transformation, then can survives form more compete day by day in the time.In the paper, discusses Digital Audio Broadcasting principles and applications, aims at European EUREKA-147 system, analysis its construction, function and the receiver function request, EUREKA-147 system is at present the system which widely uses, in broadcasts digitize have its contribution to the multimedia dissemination and hope will solve the problem when advance broadcasting digitize.
33
Yung, Kai-Shun, and 楊凱翔. "Design of Antennas for Digital Audio Broadcasting." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/01480812615521224761.
Full textAbstract:
碩士國立臺灣科技大學
電機工程系
97
The research is used by low cost and easy making PCB. To design and compare of utilizing two different kinds of Antenna which are spiral-line and meander-line in digital broadcast reception. The frequency specified by digital audio broadcast(DAB) is Band III (174-240 MHz),and L band (1452-1492 MHz).This research is focus on the design of a Band III antenna. At the first beginning, this research is simulated the behavior of Antenna by using high frequency simulate software HFSS. Making Antenna is based on using PCB which is according to the acceptable result of simulate Antenna route. The advantage of designing the route of spiral-line Antenna is smaller dimension and the disadvantage is extreme narrow frequency .But this structure is not suitable. At the second step, the research is designed the Antenna route of meander-line. The advantage is wide range of frequency and the disadvantage is bigger dimension comparing to the pervious spiral-line experiment. In conclusion ,we have found out the meander-line is the best design in this research.
34
HUANG, KUN-YU, and 黃坤裕. "Practice in Audio Decoding Program of Digital Audio Broadcasting Receiver." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/34980061035859106714.
Full textAbstract:
碩士中華大學
電機工程學系碩士班
98
Digital Audio Broadcasting(DAB) is a new broadcast transmission technology and different from the well-known amplitude modulation(AM) and frequency modulation(FM) broadcast technology. The audio coding-decoding format of Digital Audio Broadcasting takes MPEG-1/2 layer II audio coding method as the foundation, and then adds Program Associated Data (PAD). In this thesis, I design the audio decoding program of Digital Audio Broadcasting Receiver by using a personal computer as the simulation platform. The input of this decoding program is the audio source signal which is decoded from Main Service Channel (MSC), and the output is a decoded PCM audio file or directly audible program voice. Since the main function of PAD is transmitting text or video, that doesn’t have much to do with audio signal, so I don’t discuss PAD in this thesis.
35
Hsieh, Yihung, and 謝宜宏. "FPGA IMPLEMENTATION OF OFDM FOR DIGITAL AUDIO BROADCASTING." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/63435029427937263495.
Full textAbstract:
碩士國立雲林科技大學
電機工程系碩士班
89
The advanced techniques of digital signal processing adopted in Digital Audio Broadcasting system not only provide high quality audio but also conquer the impact of the fading channel. This paper is to evaluate the architecture of Digital Audio Broadcasting and hardware implement, especially focusing on implementing Orthogonal Frequency Division Multiplex and Interleaver by Verilog HDL language. We also investigate the optimum hardware implementation of the Fast Fourier Transform to further reduce the hardware complexity without degrading its efficiency. In this paper, Radix-2 Butterfly, Controller, and Random Access Memory are proposed to realize the OFDM circuit. In practice, real-time processing is as important as the size and cost of hardware. For consideration of hardware in this study, high Density Logic Integration and higher performance complex Programmable Logic Device are chosen to accomplish the overall architecture of Digital Audio Broadcasting.
36
Jen, Cheng-min, and 任正民. "DESIGN OF ANTENNA FOR DIGITAL AUDIO BROADCASTING RECEPTION." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/k492fa.
Full textAbstract:
碩士大同大學
通訊工程研究所
95
In this thesis, three printed antennas were studied for digital broadcast reception. The first one is a meandered loop antenna. Being in a meandered form, it can be made very compact. However, the gain and efficiency are quite low. In our study, the achieved gain is only -30dBi. Therefore, this configuration is not practical in use. We then studied a bow-tie antenna. Bow-tie antenna is typically considered to be a wide-band antenna. It takes more space than the loop antenna. Although the gain is increased up to 1.85dBi, the measured bandwidth of around 8% is less than what is required in DAB reception. The last one understudy is a dipole-patch antenna. This antenna is constructed by roughly half-sizing the previous bow-tie antenna. Measured -10db return loss bandwidth of the third antenna ranges from 192 to 236 MHz, or 21% bandwidth centered at 214MHz. The gain is 1.78dBi within the whole interested band. The third antenna is generally accepted for DAB reception.
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Lee, Ton-Shing, and 李東興. "Simulations and Measurements of Wave Propagation for Digital Audio Broadcasting." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/43467400340266876935.
Full textAbstract:
碩士樹德科技大學
電腦與通訊研究所
91
Simulations and Measurements of Wave Propagation for Digital Audio Broadcasting Student: Ton-Shing Lee Advisors: Dr. Yih-Chuan Lin Dr. Wen-Tzu Chen Department of Computer and Communication Shu-Te University ABSTRACT The purpose of this thesis is to measure and simulate the wave propagation loss for Digital Audio Broadcasting in Taiwan. We measured the electrical field in Chiayi county, Tainan county, and Kaohsiung city. We constructed the wave propagation models by means of analyzing the measured data and comparing with the simulation results. The results showed that Hata model gave better agreement in urban and suburban regions. Lee''s model is also in good agreement in urban and suburban regions. Egli and Walfisch-Ikegami models are both are in good agreement in suburban and open regions.
38
Huang, Chorng-Ming, and 黃崇銘. "A 1.4GHz CMOS Frequency Synthesizer for Digital Audio Broadcasting Receiver." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/11513758514189251274.
Full textAbstract:
碩士國立成功大學
電機工程學系
88
CMOS RF has been a hot topic of circuit design for wireless receivers. One of the research area is the GHz CMOS frequency synthesizer design. In this thesis, several approaches to realize frequency synthesis are compared. The one using PLL technique has the advantages of higher operating frequency, low cost, and less circuit complexity. The frequency synthesizer in this thesis is designed based on PLL architecture. Voltage controlled oscillator ( VCO ) and prescaler are key components in the frequency synthesizer. Ring oscillator is adopted as VCO in the frequency synthesizer loop to generate GHz frequency in the thesis. Low phase noise ring oscillator is obtained due to the use of saturated delay cells which perform fully switching of device current. The designed prescaler using differential D-flipflops has well supply noise rejection. It induces less jitter to the synthesizer loop when there are current switching and signal coupling in the power line. A 1.4GHz frequency synthesizer is designed and implemented in 0.35um, one poly, four metal, CMOS process operating at 3.3V power supply.
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Wang, Nai-Rung, and 王乃宗. "The Competitive Strategy Analysis of Taiwan Digital Audio Broadcasting Industry." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/61505826371774873720.
Full textAbstract:
碩士國立臺灣大學
國際企業學研究所
92
THESIS ABSTRACT GRADUATE INSTITUTE OF BUSINESS ADMINISTRATION NATIONAL TAIWAN UNIVERSITY NAME: NAI-RUNG WANG MONTH/YEAR : JUN/2004 ADVISER: DR.WU, CHING-SUNG THE COMPETITIVE STRATEGY ANALYSIS OF TAIWAN DIGITAL AUDIO BROADCASTING INDUSTRY The purpose of this thesis is to explore the driving forces of the DAB system diffusion and the competition structure of Taiwan audio broadcasting industry. What the key successful factors and its opportunities and threats are. This thesis attempts to formulate a dynamic competitive fit strategy for the industry to cope with this rapidly changing digital world economy. This is accomplished by applying the M. Porter’s five forces model to survey the outside-in relative force of the competition and Prahala & Hamel’s core competence model inside-out approach to empirically study of the difference between UK and Taiwan’s DAB industry. The main achievement of this study is to develop an integrative conceptual framework. This enables the formation of the digital oriented business model that can transform the first mover advantage into a sustainable competitive advantage. By developing the awareness of the disruptive innovation technology, and emerging the DAB network into 4C (consumer, computer, content, communication) multimedia platform will realize the increased profits. This analysis and findings concerning the Taiwan DAB OEM/ODM industry strategic positioning and value adding activities can be of significant help to government policy maker, content provider and its Broadcasters.
40
Hsu, Chih-Chien, and 許志謙. "Conversion of Digital Audio Broadcasting Signal to Analog Radio Frequency Signal." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/62189259048093408104.
Full textAbstract:
碩士國立臺灣科技大學
電機工程系
96
As the diffusion of digitization, people pay much more attention to Digital Audio Broadcasting (DAB), which is adopted in some countries. It offers the advantages that analogue audio broadcasting, such as FM and AM broadcasting, don’t have. These advantages include improved reception, improved clarity of sound, greater radio station choice, better channel utility rate and so on. In this thesis, there is a design about a adapter that has the conversion of a digital audio broadcasting signal to an analog radio frequency signal. The device offers users a cheap way to enjoy listening Digital Audio Broadcasting. People can listen Digital Audio Broadcasting and analogue audio broadcasting programs without a DAB radio receivers. The system of the device include Digital Audio Broadcasting Tuner, Radio Data System Encoder, Baseband Mixer, Subcarrier Modulator, Frequency Modulation Modulator and Microprocessor.
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"Jitter reduction techniques for digital audio." 1997. http://library.cuhk.edu.hk/record=b5889216.
Full textAbstract:
by Tsang Yick Man, Steven.Thesis (M.Phil.)--Chinese University of Hong Kong, 1997.
Includes bibliographical references (leaves 94-99).
ABSTRACT --- p.i
ACKNOWLEDGMENT --- p.ii
LIST OF GLOSSARY --- p.iii
Chapter 1 --- INTRODUCTION --- p.1
Chapter 1.1 --- What is the jitter ？ --- p.3
Chapter 2 --- WHY DOES JITTER OCCUR IN DIGITAL AUDIO ？ --- p.4
Chapter 2.1 --- Poorly-designed Phase Locked Loop ( PLL ) --- p.4
Chapter 2.1.1 --- Digital data problem --- p.7
Chapter 2.2 --- Sampling jitter or clock jitter ( Δti) --- p.9
Chapter 2.3 --- Waveform distortion --- p.12
Chapter 2.4 --- Logic induced jitter --- p.17
Chapter 2.4.1 --- Digital noise mechanisms --- p.20
Chapter 2.4.2 --- Different types of D-type flop-flip chips are linked below for ease of comparison --- p.21
Chapter 2.4.3 --- Ground bounce --- p.22
Chapter 2.5 --- Power supply high frequency noise --- p.23
Chapter 2.6 --- Interface Jitter --- p.25
Chapter 2.7 --- Cross-talk --- p.28
Chapter 2.8 --- Inter-Symbol-Interference (ISI) --- p.28
Chapter 2.9 --- Baseline wander --- p.29
Chapter 2.10 --- Noise jitter --- p.30
Chapter 2.11 --- FIFO jitter reduction chips --- p.31
Chapter 3 --- JITTER REDUCTION TECHNIQUES --- p.33
Chapter 3.1 --- Why using two-stage phase-locked loop (PLL ) ？
Chapter 3.1.1 --- The PLL circuit components --- p.35
Chapter 3.1.2 --- The PLL timing specifications --- p.36
Chapter 3.2 --- Analog phase-locked loop (APLL ) circuit usedin second stage --- p.38
Chapter 3.3 --- All digital phase-locked loop (ADPLL ) circuit used in second stage --- p.40
Chapter 3.4 --- ADPLL design --- p.42
Chapter 3.4.1 --- "Different of K counter value of ADPLL are listed for comparison with M=512, N=256, Kd=2" --- p.46
Chapter 3.4.2 --- Computer simulated results and experimental results of the ADPLL --- p.47
Chapter 3.4.3 --- PLL design notes --- p.58
Chapter 3.5 --- Different of the all digital Phase-Locked Loop (ADPLL ) and the analogue Phase-Locked Loop (APLL ) are listed for comparison --- p.65
Chapter 3.6 --- Discrete transistor oscillator --- p.68
Chapter 3.7 --- Discrete transistor oscillator circuit operation --- p.69
Chapter 3.8 --- The advantage and disadvantage of using external discrete oscillator --- p.71
Chapter 3.9 --- Background of using high-precision oscillators --- p.72
Chapter 3.9.1 --- The temperature compensated crystal circuit operation --- p.73
Chapter 3.9.2 --- The temperature compensated circuit design notes --- p.75
Chapter 3.10 --- The discrete voltage reference circuit operation --- p.76
Chapter 3.10.1 --- Comparing the different types of Op-amps that can be used as a voltage comparator --- p.79
Chapter 3.10.2 --- Precaution of separate CMOS chips Vdd and Vcc --- p.80
Chapter 3.11 --- Board level jitter reduction method --- p.81
Chapter 3.12 --- Digital audio interface chips --- p.82
Chapter 3.12.1 --- Different brand of the digital interface receiver (DIR) chips and clock modular are listed for comparison --- p.84
Chapter 4. --- APPLICATION CIRCUIT BLOCK DIAGRAMS OF JITTER REDUCTION AND CLOCK RECOVERY --- p.85
Chapter 5 --- CONCLUSIONS --- p.90
Chapter 5.1 --- Summary of the research --- p.90
Chapter 5.2 --- Suggestions for further development --- p.92
Chapter 5.3 --- Instrument listing that used in this thesis --- p.93
Chapter 6 --- REFERENCES --- p.94
Chapter 7 --- APPENDICES --- p.100
Chapter 7.1.1 --- Phase instability in frequency dividers
Chapter 7.1.2 --- The effect of clock tree on Tskew on ASIC chip
Chapter 7.1.3 --- Digital audio transmission----Why jitter is important?
Chapter 7.1.4 --- Overview of digital audio interface data structures
Chapter 7.1.5 --- Typical frequency Vs temperature variations curve of Quartz crystals
Chapter 7.2 --- IC specification used in these research project
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李承中. "A Research of the Possibility that a Public Service Broadcasting Station Runs the Digital Audio Broadcasting." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/51691224352628387911.
Full textAbstract:
碩士世新大學
傳播研究所(含博士班)
91
On February 29, 2000, The Directorate General of Telecommunications, Ministry of Transportation and Communications (DGT) announced the result for selecting the candidates to run the Digital Audio Broadcasting (DAB). The selected candidates which were Central Broadcasting System (CBS), The Voice of Han Broadcasting Network（VOH） , Public Radio System（PRS）,National Education Radio（NER）, set up a joint station named "the Public Broadcasting Trial Station" (PBTS) and got trial channel which cover the whole area. Now the trial period is coming to an end. The channel is going to be released to the public. Will the above four Public Service Broadcasting Stations (PSBS) continue to cooperate to run a DAB Station? And what is the possibility? This research is trying to find out how the PBTS should integrate the resources from PSBS and the characters of DAB, and find their privilege to win the channel. This thesis is a quality research. Based on the theory of "Digital Environment and Public Service Broadcasting", "The Strategy of Running Broadcasting Station", "Diffusion of Innovations", and "Media Uses and Gratification Theory", I focused on the analysis of the above mentioned four PSBSs' internal causes and external environment individually. Through interviews, observing and case studies, I try to find out "is it practical if PSBS runs DAB?" The conclusion is that PSBS running DAB is practical. In this research, I also discover that the organization is not suitable for running DAB, a joint venture may reduce costs, the program production team should involve in producing DAB programs as soon as possible, the integration of techniques of software and hardware should be improved, and, public service and disaster prevention should be constantly stressed. The conclusions and discovery may be a reference for further researches.
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Lin, On Sun, and 林昂生. "Design of Digital Audio Broadcasting (DAB) Receiver L-Band Down-Converter and RFICs." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/47879842208354114692.
Full textAbstract:
碩士國立成功大學
電機工程學系
89
This thesis presents the design and implementation of L-band DAB downconverter RFICs, including LNA/mixer/AGC and phase-locked oscillator (PLO), in a standard 0.35m CMOS technology. The RF is from 1.452 to 1.492 GHz, the LO is at 1.28 GHz and the IF is from 172 MHz to 212 MHz. The single-stage LNA has 3.2 dB noise figure and only 3dB gain. The single-balanced mixer has 20dB noise figure , 16dB conversion loss. Simulation results of the 1.28 GHz PLO show that the output power is —2 dBm and the locking time is 15s. An external hybrid-circuit LNA with 26 dB gain and 2.2 dB noise figure is added, due to the low gain result of the implemented LNA RFIC. The LNA and a 250 MHz IF lowpass filter with the LNA/mixer/AGC RFIC construct a hybrid/RFIC downconverter (not including the PLO RFIC) for receiving measurement. The hybrid/RFIC downconverter has 13dB gain, 3.5 dB noise figure, -17 dBm P1dB and —7.5 dBm OIP3. For 1.2Msps 32-QAM digital modulation, the measured error vector magnitude (EVM) is 1.2%. With a DAB test encorder and a DAB test receiver for OFDM signal receiving test, the adjacent channel power ratio (ACPR) is —30.61 dBc and the sensitivity is -90dBm (BER=10-4 ) or -97dBm (BER=10-2 ). Further effort will be devoted to design and implement a complete CMOS L-band DAB downconverter RFIC.
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Chen, Chin-Lung, and 陳志龍. "CMOS Circuit Design of the RF Front-End for Digital Audio Broadcasting (DAB) Receiver." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/51643934538778822779.
Full textAbstract:
碩士國立成功大學
電機工程學系
88
This thesis describes the design of a 2.5V 1.5GHz CMOS RF receiver front-end for Digital Audio Broadcasting(DAB) system. The receiver front-end combines a balanced low noise amplifer (LNA), quadrature downconversion mixers, an LO quadrature generator , LO input buffers, IF gain stages , a polyphase filter, second gain stages, and output buffers all in one single CMOS IC. Using polyphase image-reject architecture, CMOS RF circuits, and on-chip filtering, the receiver is fully integrated and eliminates off-chip SAW filters often seen in heterodyne receivers. Down to the signal path of this front-end design, the low noise amplifier is a differential, cascode common-source with inductive source degeneration type. The planar spiral inductor used in LNA boost the gain performance and enable low voltage RF circuits to be monolithically integrated. Combined with the current-reused architecture and power-constrained noise optimization techniques, the LNA can achieve 29.4dB voltage gain, -29.2dB S11, 25mW power consumption, and 2.38 dB NF, obtained from simulation. The down-conversion mixer employs a double balanced Gilbert Cell type mixer with common mode feedback circuit. Simulation shows it can achieve 7.5dB conversion gain and -5dBV IIP3. This RF receiver front-end is designed in TSMC 0.25$\mu$m single-poly, five-metal mixed-signal CMOS process, operated with 2.5V power supply. The spiral inductor and MOS RF model, from the TSMC process, are adopted in the receiver design.
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Wang, Wen-Chyi, and 王文祺. "Design, Analysis, and Implementation of High-Order ΣΔ Analog- to-Digital Converters for Digital Audio and Wireless Communications." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/63015082732879851224.
Full textAbstract:
碩士國立成功大學
電機工程研究所
84
In this thesis, we verify the various methods for modelingΣΔ modulators, examine their validation, and use these methods to explain some of the nonidealities of ΣΔ modulators. The concept of stability for ΣΔ modulators is also explored. Based on these analyses, we present a novel method named "Internal Linear Feedback(ILF)" to stabilize high-order ΣΔ modulators. With the ILF method, a modulator can be guaranteed to be absolutely stable and its dynamic range can also be improved. Next, a 4th-order audio-bandΣΔ modulator is implemented using switched-capacitor circuitry. With an oversampling ratio of 64 and a clock frequency of 2.56 MHz, the modulator achieves a signal-to- noise ratio (SNR) of 96 dB. The power consumption is about 30 mW for a single 5 V power supply. The circuit die area is only 850 um × 625 um using a 0.6 um double-poly CMOS process. sign methodology for lowpass ΣΔ modulators is extended toodulators. Based on this methodology, a bandpass ΣΔ modulatoreasily
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Tseng, Chu-Min, and 曾竹敏. "The Influence of Music on The Product Image of Audio Devices－A Case Study of Digital Audio Broadcasting Devices." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/27884643130026011677.
Full textAbstract:
碩士大同大學
工業設計學系(所)
99
ABSTRACT Digital Audio Broadcasting DAB provides excellent multimedia information service. Not only does DAB deliver audio broadcasting comparable to that of CD sound quality but it also offers background information and message related to music programs; DAB produces a new form for the function technology of life and a unique lifestyle. The psychological perception for vision is comparably pleasing as that for hearing though the forming designs of digital audio-broadcasting receiver. The study aims to understand the association between the product forms, interface distributions and musical perception, in attempt to discover the product form suitable to the music type. The experiment design is divided into two stages. We discover from the Pre-test experiment that a good form design for digital audio broadcasting receiver will attract more expectation towards the form before even listening to the music broadcasted. In the experiment of body form and state of interface distribution, the understanding of the different consumer perceptions for the form and interface distribution of digital audio broadcasting receivers will acquire different expected audio perceptions. We discover from the experiment that among the 6 form characteristics, three of them were not affected by the music form, regardless of music, graceful, fluent, and fashion. Nonetheless, calm, simple and mellow will have different results under the different music form
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Fu, Fred, and 傅志明. "The Evolution and Application of Digital Audio Broadcasting Industry in Taiwan- The Case Study of company A." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/29323869590135575780.
Full textAbstract:
碩士輔仁大學
科技管理學程碩士在職專班
97
Digital Audio Broadcasting is a new broadcasting technique after the traditional analog broadcasting. Nowadays, various kinds of media transmission systems have been developed rapidly. Thus, broadcasting has become increasingly important and highly focuses on variety services. Since AM and FM radio are only able to convey audio services and unable to provide multimedia services, utilizing Digital Audio Broadcasting is the way to overcome this weakness and making the radio broadcasting technique transfer into a brand-new all-digitalized platform for AM and FM. It is a milestone in broadcasting history. Moreover, by looking at its capability, DAB is able to transfer considerable quantity of digital images, texts, graphs and so on. The development and spread of DAB enhances the capability of us into the abundant and variety information era. This thesis is according to the revolution of domestic and European digital broadcasting and brings the idea that the market trend is meanly affected by abilities of convenience and function. In conclusion, Besides, via reviewing bibliographies from both domestic and aboard and interviewing industrial manager, enabling us to clarify that the problems and bottleneck s of digital broadcast in our country. In order to bring up a perspective to the direction of digital broadcast development, this thesis worked by way of collecting and researching the data which content with the development and application of domestic and European digital broadcast industrial market and further analyzing its strength and weakness. The thesis provides the recommendations to the government and company A based on the development of DAB in Taiwan and future. In addition, it also raises some suggestions for further studies.
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Wang, Tzu-Wen, and 王姿雯. "Pilot Study of Using Motivation and Gratification for The Initial Users on The Digital Audio Broadcasting Constructive Period in Taiwan." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/71577729205063591057.
Full textAbstract:
碩士玄奘大學
資訊傳播研究所
97
Following the rapid advancement and in general use of digital technology, Digital Audio Broadcasting (DAB) is developing accordingly by breaking through the restriction of analogy signal. This study is to explore the using motivation and gratification for the initial users on the DAB constructive period in Taiwan by using the approach of literature data analysis and depth interview to figure out what is going on about the stagnation of the DAB and the initial users’ satisfaction and appraisal. The findings of this study show that the initial users are not satisfied with the DAB and give it a low mark on evaluation. Some recommendations are given as follows: firstly, rules and regulations should be remedied to meet the DAB’s features, co-ownership management and value-added service; secondly, setting up common iron tower and transmitting station by government and promoting the policy of separation of production and broadcasting respectively to integrate the conglomerate between the broadcasting and TV, communication and electronic industries; thirdly, cultivating the talents about the digital content innovation to lower the digital divide; fourthly, taking Japan and Korea digital media success as the role model to market the digital broadcasting industry actively. In short, those findings and recommendations can be a follow-up reference for further study in this field.
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Albertyn, Eben. "Channel characterisation and coding for the FM SST channel." Thesis, 2011. http://hdl.handle.net/10210/3944.
Full textAbstract:
M.Ing.The transmission of digital data at higher data rates and greater reliabilities is becoming increasingly more important in the society of today. The usage of, especially wireless digital data, is increasing at an alarming rate, and the need arises for more bandwidth to be made available for this purpose. Hence, the need arises to optimise and expand the usage of bandwidth currently under use. Existing wireless channels, such as the commercial FM service, need to be re-evaluated and unused bandwidth needs to be developed and used as close to capacity as possible. In this dissertation, a study was launched to develop the FM SST (Supplementary Signal Transmission) commercial analog service as a simplex digital channel that can commercially be used to transmit a myriad of digital data. With the help of a commercial FM radio channel, digital data was transmitted using a number of different modulation schemes to obtain channel measurements for the FM SST channel. The modulation scheme that performed the best, given certain criteria, was GMSK. This raw information was then used to extract first and higher order statistical information from the channel, in order to parameterise discrete channel models, based on a Markov process. The channel models that were used were the Gilbert, GilbertElliott and Fritchman channel models. These channel models were then tested according to their iirst and higher order statistical data to evaluate their effectiveness in modelling the error process on the FM SST channel. It was found that the channel model best approximating the real channels was the Gilbert channel. Once satisfactory results were obtained, these discrete channel models were then used to evaluate various error correction schemes for their ability to correct burst errors on the FM SST channel. From the various schemes evaluated, a (63,39) BCH with an interleaver of index 4 was found to perform the best. The purpose of which is to transmit digital data at the highest data rate possible and at the same time having a bit error rate less than lxl0-6.
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Su, Wen-yenn, and 蘇文彥. "A Research on the Correlations between the Media Conglomerates and the Popularization of the Digital Audio Broadcasting Industry–A case study in DAB development in UK." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/95830853350247907386.
Full textAbstract:
碩士世新大學
傳播管理學研究所(含碩專班)
99
In the backdrop of deregulation policy and media liberalization prompted by the deployment of new technology, and in the trend of media digitization and convergence, the destiny of Digital Audio Broadcasting, or DAB, has been different between UK & Taiwan after more than 10 years of its application. The former has been a model to the world of the employment of this innovative technology, but in Taiwan it has been toppled in to a state of silence. Why is this so? In the light of the Innovation theory by Joseph A. Schumpeter and New Institutionalism by Lance Davis& Douglass North,” Invention” is always prior to “Innovation”. The promotion of the Innovation is caused by the recession of economy. And the Innovation will be meaningless unless there is a strong determination of a leader who take advantage of an invention and the economic benefits the Innovation produced. The determined-leading & mutual benefits of the corporation of British government and industries has played a significant role in the popularity of DAB in UK, while the conglomerate-oriented transformation in the radio broadcasting industry also provide the support to the development of new system in UK. And this is in the line with the principle mentioned above. As to the case in Taiwan, the toppled development of DAB represents a combination results of inadequate media policy, unsuitable consultation to the government and unwillingness of the investment to innovation from the industry which has been constrained in size and finance by the Broadcasting Act in the face of an era of digitization. In the efforts of the study to DAB development & the radio industry policy of UK, the research believe that under the international tide of cross-national & cross industry operation of media, and its pursuit of scale, the first priority for Taiwan in the development of radio digitization is the take up of innovation under a close corporation between government and operators. The deregulation of the restrictions to the industry of its diversified operation and cross-industry merging should be commencing at this very moment to safeguard existence and digital upgrading of the industry. The notions of this research also includes suggestions to the operators and government, expecting it will be a useful contributions to the carry- out of the governments ” Development programs in digital convergence” unveiled in July, 2010.