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Tesi sul tema "Recording digital audio"
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1
Hans, Mathieu Claude. "Optimization of digital audio for Internet transmission". Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/14894.
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LeRoi, Jack. "A forensic investigation of the electrical properties of digital audio recording". Thesis, University of Colorado at Denver, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=1556863.
Testo completoAbstract (sommario):
In media forensics, the devices; e.g. computers, smart phones, still/video cameras, audio recorders, and software; e.g. video, audio, and graphics editors, file and disk utilities, mathematical computation applications, are, for the most part, black boxes. The design specifications are usually proprietary and the operating specifications may be incomplete, inaccurate, or unavailable. This makes it difficult to validate the technology, but using it without validation could discredit a practitioner's findings or testimony. The alternative is to test the device or program to determine relevant characteristics of its performance.
An important and common device in media forensics is the portable digital audio recorder used to record surveillance and interviews. This type can also be used to record the alternating current (AC) waveform from the mains power. While small variations in the AC frequency (ENF) can be forensically important, distortion in the recording can affect its value in adjudication or investigation. A method is presented to evaluate aspects of a recorder's operation that can cause distortion. Specifically, the method measures the noise generated by the recorder's electronics in its input and amplifier circuits. The method includes a procedure to isolate the recorder from environmental sources of noise. The method analyzes the broadband noise floor produced by the range of recording conditions and recorder settings. It also analyzes the noise amplitude for the harmonics for the mains frequency.
3
Hatch, Wesley. "High-level audio morphing strategies". Thesis, McGill University, 2005. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=83108.
Testo completoAbstract (sommario):
This thesis presents high-level strategies for controlling existing audio morphing algorithms. Prominent methods for spectrally representing audio data in the context of audio morphing are surveyed, and techniques involving manipulating and interpolating additive models are explored. Additionally, a combination of past audio-morphing approaches (sinusoidal morphing techniques as well as techniques taken from an MFCC approach) are employed in order to facilitate any type of sonic input. The system presented herein is meant to accommodate a wide range of inputs, and will adjust the impact of a high-level feature depending on the type and quality of the input.**This dissertation is a compound document (contains both a paper copy and a CD as part of the dissertation).
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Linton, Ken N. "Digital mixing consoles : parallel architectures and taskforce scheduling strategies". Thesis, Durham University, 1995. http://etheses.dur.ac.uk/5371/.
Testo completoAbstract (sommario):
This thesis is concerned specifically with the implementation of large-scale professional DMCs. The design of such multi-DSP audio products is extremely challenging: one cannot simply lash together n DSPs and obtain /7-times the performance of a sole device. M-P models developed here show that topology and IPC mechanisms have critical design implications. Alternative processor technologies are investigated with respect to the requirements of DMC architectures. An extensive analysis of M-P topologies is undertaken using the metrics provided by the TPG tool. Novel methods supporting DSP message-passing connectivity lead to the development of a hybrid audio M-P (HYMIPS) employing these techniques. A DMC model demonstrates the impact of task allocation on ASP M-P architectures. Five application-specific heuristics and four static-labelling schemes are developed for scheduling console taskforces on M-Ps. An integrated research framework and DCS engine enable scheduling strategies to be analysed with regard to the DMC problem domain. Three scheduling algorithms — CPM, DYN and AST — and three IPC mechanisms — FWE, NSL and NML — are investigated. Dynamic-labelling strategies and mix-bus granularity issues are further studied in detail. To summarise, this thesis elucidates those topologies, construction techniques and scheduling algorithms appropriate to professional DMC systems.
5
Hlaváček, Ondřej. "Business models of digital audio distribution". Master's thesis, Vysoká škola ekonomická v Praze, 2008. http://www.nusl.cz/ntk/nusl-7907.
Testo completoAbstract (sommario):
This thesis considers current status of recording industry divided into two parts -- music and audiobooks, either in the whole world and Czech Republic. Used technologies, payment systems and business models are examined alongside with fresh information from recording industry. The models part describes an ideal business model, which incorporates current trends and creates a vision of the business in the future. Applied part adjusts this model according to real world factors into a model usable in the Czech Republic. A final SWOT analysis describes all strengths, weaknesses, opportunities and threats of the given model.
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Markle, Blake L. "A comparative study of time-stretching algorithms for audio signals /". Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=31119.
Testo completoAbstract (sommario):
Algorithms exist which will perform independent transformations on frequency or duration of a digital audio signal. These processes have different results different types of audio signals. A comparative study of granular and phase vocoder algorithms, implementation, and their respective effects on audio signals was made to determine which algorithm is best suited to a particular type of audio signal.
7
Fan, Yun-Hui. "A stereo audio coder with a nearly constant signal-to-noise ratio". Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/14788.
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Zarate, Orozco Ismael. "Software and Hardware-In-The-Loop Modeling of an Audio Watermarking Algorithm". Thesis, University of North Texas, 2010. https://digital.library.unt.edu/ark:/67531/metadc33221/.
Testo completoAbstract (sommario):
Due to the accelerated growth in digital music distribution, it becomes easy to modify, intercept, and distribute material illegally. To overcome the urgent need for copyright protection against piracy, several audio watermarking schemes have been proposed and implemented. These digital audio watermarking schemes have the purpose of embedding inaudible information within the host file to cover copyright and authentication issues. This thesis proposes an audio watermarking model using MATLAB® and Simulink® software for 1K and 2K fast Fourier transform (FFT) lengths. The watermark insertion process is performed in the frequency domain to guarantee the imperceptibility of the watermark to the human auditory system. Additionally, the proposed audio watermarking model was implemented in a Cyclone® II FPGA device from Altera® using the Altera® DSP Builder tool and MATLAB/Simulink® software. To evaluate the performance of the proposed audio watermarking scheme, effectiveness and fidelity performance tests were conducted for the proposed software and hardware-in-the-loop based audio watermarking model.
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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.
Testo completoAbstract (sommario):
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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Laubscher, Robert Alan. "An investigation into the use of IEEE 1394 for audio and control data distribution in music studio environments". Thesis, Rhodes University, 1999. http://hdl.handle.net/10962/d1006483.
Testo completoAbstract (sommario):
This thesis investigates the feasibility of using a new digital interconnection technology, the IEEE-1394 High Performance Serial Bus, for audio and control data distribution in local and remote music recording studio environments. Current methods for connecting studio devices are described, and the need for a new digital interconnection technology explained. It is shown how this new interconnection technology and developing protocol standards make provision for multi-channel audio and control data distribution, routing, copyright protection, and device synchronisation. Feasibility is demonstrated by the implementation of a custom hardware and software solution. Remote music studio connectivity is considered, and the emerging standards and technologies for connecting future music studio utilising this new technology are discussed.Microsoft Word
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Chen, Howard. "AZIP, audio compression system: Research on audio compression, comparison of psychoacoustic principles and genetic algorithms". CSUSB ScholarWorks, 2005. https://scholarworks.lib.csusb.edu/etd-project/2617.
Testo completoAbstract (sommario):
The purpose of this project is to investigate the differences between psychoacoustic principles and genetic algorithms (GA0). These will be discussed separately. The review will also compare the compression ratio and the quality of the decompressed files decoded by these two methods.
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Svoboda, Tomáš. "Vestavěné zařízení pro ovládání digitální audio stanice". Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2019. http://www.nusl.cz/ntk/nusl-399180.
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The aim of this work is to design an architecture of the embedded device that will be used for controlling DAW software in recording studio. First of all, attention is given to a brief summary of the necessary knowledge which is needed to design such kind of device. Af- ter that follows short survey of the existing solutions and description of protocols which can be used for communication with the recording software. Then, subsequent part of the thesis builds upon these foundations and further elaborates the device architecture by me- ans of decomposing it into several modules. In fact, two hardware modules are designed and manufactured, when each of them is conceived on a separate PCB with its own microcon- troller. Then the control firmware has been implemented for each of the modules. At the end of the work an aluminium enclosure, which holds both modules, is designed. The result of this work is a functional prototype of the assembled controller which can be used for the purpose of controlling DAW software.
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Rodríguez, Salinas Roberto Javier. "El uso e impacto de las técnicas de grabación y mezcla analógica en la producción musical estadounidense de pop y rock (2011-2017): Alternativas para su aplicación en el estudio digital". Bachelor's thesis, Universidad Peruana de Ciencias Aplicadas (UPC), 2020. http://hdl.handle.net/10757/654919.
Testo completoAbstract (sommario):
El presente trabajo de investigación trata de identificar y analizar la influencia y la aplicación del audio analógico en la producción musical contemporánea, abarcando diversos criterios: sonoro/técnico, de ejecución, social, de preservación o archivo, entre otros. A pesar de que el audio digital, y el uso derivado de las herramientas de edición y mezcla, se consideran el estándar en la industria musical actual, se conoce que un número importante de productores, ingenieros y artistas siguen empleando hoy en día las técnicas de grabación y mezcla analógica desarrolladas décadas atrás. Por ello, el audio analógico surge como una alternativa a los procesos de creación y producción musical modernos, tanto a nivel sonoro como de ejecución. Este trabajo parte del desarrollo técnico y estético en la grabación analógica, realizando comparaciones con su contraparte digital. Para ello, se toman en cuenta las opiniones y los procesos de producción de productores musicales y profesionales a cargo de las producciones discográficas más relevantes en los últimos años en el circuito norteamericano. Es en estas producciones donde se evidencia el uso creativo del audio analógico, así como su adaptación al entorno digital. Finalmente, se tiene en cuenta al productor independiente como principal beneficiario de esta publicación, quien conoce directa o indirectamente los beneficios del audio analógico, pero no necesariamente cuenta con los recursos para integrarlo a su estación de trabajo. Por ello, se busca brindar herramientas y soluciones para poder emplear e integrar las principales funciones del formato analógico y digital en una producción moderna.The hereby thesis attempts to identify and analyze the influence and application of analog audio in contemporary music production by covering diverse criteria: sonic/technical, performative, social, of preservation or archive, among others. Although digital audio, and its derived use of editing and mixing tools, are currently considered standard in the music industry, it is also known that a significant number of producers, engineers and artists still employ analog recording and mixing techniques developed decades ago. Hence, analog audio presents itself as an alternative to the modern creative and production processes, at both sonic and performance levels. This work begins with the technical and aesthetical development within an analog recording, comparing it afterwards with its digital counterpart. To serve this purpose, the opinion and detailed production processes of renowned American producers and professionals, responsible for the most acclaimed records in the last ten years, are taken into consideration. Those records evidence the creative use of analog audio, as well as its adaptation to a digital setup. Finally, independent producers could benefit the most from the information and arguments in this paper, because they directly or indirectly know the benefits of working analog, even when they often lack the resources to acquire these tools. Therefore, the author attempts to bring tools and solutions for the application of analog and digital audio in a single modern production.
Tesis
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Sibanda, Phathisile. "Connection management applications for high-speed audio networking". Thesis, Rhodes University, 2008. http://hdl.handle.net/10962/d1006532.
Testo completoAbstract (sommario):
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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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.
Testo completoAbstract (sommario):
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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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.
Testo completoAbstract (sommario):
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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Camerani, Roberto. "The role of users and suppliers in the adoption and diffusion of consumer electronics : the case of portable digital audio players". Thesis, University of Sussex, 2012. http://sro.sussex.ac.uk/id/eprint/40051/.
Testo completoAbstract (sommario):
The diffusion of innovations is a fundamental aspect of the innovative process, to which the literature on innovation dedicated a lot of attention. This voluminous literature covers a variety of themes, such as different kinds of innovations, potential adopters, and mechanisms by which the innovation spreads among its potential users. However, some aspects of this vast literature still deserve some further investigation. The objective of the thesis is to study the adoption and diffusion of a consumer technology, the portable digital audio player (DAP) market in Europe and Japan. The methodology is quantitative and consists on the collection and analysis of two original datasets. The first dataset regards the demand-side consisting in a survey of 1562 young potential adopters from 9 countries (France, Germany, Italy, Portugal, Netherlands, Spain, Switzerland, UK, and Japan). The other source of data is a dataset of 585 DAPs marketed between 2001 and 2009, including information on product characteristics (storage space, size, etc.) and price. The analysis of the data is carried out at three levels. The first one regards the demand-side, with the aim of assessing how users' characteristics shape the adoption decision, and providing a classification of potential adopters that goes beyond the usual classification based on timing of adoption or on the distribution of a single variable such as income. The second level concentrates on the supply-side, testing if there is a systematic relationship between product price and its objectively measurable characteristics and evaluating how technical change in the sector influences the diffusion path by matching products' quality change with users' preferences and patterns of adoption over time. Finally, the third level aims at providing evidence on whether conventional models of diffusion are able to provide an adequate explanation of the diffusion of DAPs, and moreover, on how the assumptions underlying these models might be combined or synthesised into a coherent framework.
18
Boland, Simon Daniel. "High quality audio coding using the wavelet transform". Thesis, Queensland University of Technology, 1998.
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Hella, Vegard. "Digital Audio Restoration : Denoising phonograph recordings". Thesis, Norges teknisk-naturvitenskapelige universitet, Institutt for elektronikk og telekommunikasjon, 2013. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-23521.
Testo completoAbstract (sommario):
This master's thesis realize an audio noise reduction tool by use of digital signal processing. The tool is used to restore phonograph recordings. The recordings are restored on behalf of Ringve Museum, Norway's national museum of music and musical instruments. Sometimes the noise can be louder than the actual audio. In the view of a museum or library institution, this makes them less valuable as they are not presentable to the general public. A common restoration environment will include multiple tools. We will only specialize in one of them reducing broadband, stationary and additive noise. This is often perceived as static hiss or buzz. To realize the tool we use the numerical computation environment MATLAB. In MATLAB the calculations are written using a high-level programming language with many embedded functions. There are several established algorithms specializing in noise reduction of audio and speech. We will look at some basic and some complex algorithms that are based on the Short Time Fourier Transform (STFT). This technique slices the audio in short time frames to be able to analyze the local complex frequency spectrum. The noise reduction procedure compare the audio spectrum with its estimated noise spectrum to calculate an attenuation at each frequency. The attenuated signal is transformed back into time domain. Some of the algorithms are based on the Wiener filter or AR-modeling. The program will include a user interface with selectable algorithms and parameters. In old recordings a certain level of noise may be wanted to preserve authenticity. Thus a noise floor generator will be implemented.Some necessary theory of digital signal processing will be given, but some general knowledge will be required. The noise reduction theory is presented before the realization and program flow is explained. A listening test will be conducted. Audio examples are used to illustrate the general results, and the development process, results and further work is discussed. The program gave better results than one of the commercial available softwares. Another important result is that the stationary property is a poor approximation. The phonograph noise exhibits periodical properties with longer time periods than used in the short time transform. A model incorporating this feature should be implemented.
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Cooper, Alan John. "Detection of copies of digital audio recordings for forensic purposes". Thesis, Open University, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.430588.
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Tansony, R. W. (Robert W. ). "A variable rate adaptive transform coder for the digital storage of audio signals /". Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63841.
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Pando, Rosimeire Aparecida [UNESP]. "As contribuições do audiovisual para a formação de jovens produtores: a construção de conteúdos por meio de dispositivos móveis". Universidade Estadual Paulista (UNESP), 2012. http://hdl.handle.net/11449/89549.
Testo completoAbstract (sommario):
Made available in DSpace on 2014-06-11T19:24:06Z (GMT). No. of bitstreams: 0
Previous issue date: 2012-03-23Bitstream added on 2014-06-13T20:28:07Z : No. of bitstreams: 1
pando_ra_me_bauru.pdf: 1232045 bytes, checksum: 47bf57c7307388b317f73f748ac6702b (MD5)Apresenta-se a possibilidade de criação de núcleos de produção audiovisual com conteúdos informativos que correlacionem, entre outros, arte, meio ambiente e tecnologia. Nesse contexto, a televisão digital e a internet surgem como suporte na cadeia produtiva de conteúdos realizados a partir de dispositivos móveis. Tal realidade conta com o incentivo, dentre outros, do Centro Nacional de Excelência em Produção de Conteúdos Digitais Interativos e Interoperáveis, coordenado pelo Ministério de Ciência e Tecnologia (MCT). As tecnologias de informação e comunicação (TICs) podem reestabelecer às produções midiáticas a vocação de fonte cultural formadora na vida dos indivíduos, principalmente jovens. Neste sentido, os meios de comunicação interativos prometem reduzir as fronteiras da desigualdade de acesso, fortalecendo a democratização de informações, o compartilhamento de saberes e práticas cidadãs
This paper presents the possibility of creating audiovisual production cores with informative and educative contents which correlate art, environment and technology, among others. In this context, digital television and internet appear as support to the production chain of contents generated from mobile internet devices. Such reality is encouraged, among others, by the Centro Nacional de Excelência em Produção de Conteúdos Digitais interativos e interoperáveis, coordinated by the Ministério de Ciência e Tecnologia (MCT). The information and communication technologies (ICT) can re-establish to the media productions the vocation of cultural source in one's life, especially the young. So, interactive communication means promise to reduce the bordes of access inequality, building up information democratization, knowledge sharing and civic practices
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Pando, Rosimeire Aparecida. "As contribuições do audiovisual para a formação de jovens produtores : a construção de conteúdos por meio de dispositivos móveis /". Bauru : [s.n.], 2012. http://hdl.handle.net/11449/89549.
Testo completoAbstract (sommario):
Orientador: João Baptista de Mattos Winck FilhoBanca: Sânderson Reginaldo de Mello
Banca: Regina Célia Baptista Belluzzo
O Programa de Pós Graduação em Televisão Digital: informação e conhecimento capacita profissionais para atuarem em televisão digital nas três áreas: comunicação, educação e tecnologia
Resumo: Apresenta-se a possibilidade de criação de núcleos de produção audiovisual com conteúdos informativos que correlacionem, entre outros, arte, meio ambiente e tecnologia. Nesse contexto, a televisão digital e a internet surgem como suporte na cadeia produtiva de conteúdos realizados a partir de dispositivos móveis. Tal realidade conta com o incentivo, dentre outros, do Centro Nacional de Excelência em Produção de Conteúdos Digitais Interativos e Interoperáveis, coordenado pelo Ministério de Ciência e Tecnologia (MCT). As tecnologias de informação e comunicação (TICs) podem reestabelecer às produções midiáticas a vocação de fonte cultural formadora na vida dos indivíduos, principalmente jovens. Neste sentido, os meios de comunicação interativos prometem reduzir as fronteiras da desigualdade de acesso, fortalecendo a democratização de informações, o compartilhamento de saberes e práticas cidadãs
Abstract: This paper presents the possibility of creating audiovisual production cores with informative and educative contents which correlate art, environment and technology, among others. In this context, digital television and internet appear as support to the production chain of contents generated from mobile internet devices. Such reality is encouraged, among others, by the Centro Nacional de Excelência em Produção de Conteúdos Digitais interativos e interoperáveis, coordinated by the Ministério de Ciência e Tecnologia (MCT). The information and communication technologies (ICT) can re-establish to the media productions the vocation of cultural source in one's life, especially the young. So, interactive communication means promise to reduce the bordes of access inequality, building up information democratization, knowledge sharing and civic practices
Mestre
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Fassler, Andreas. "Merging Task-Centered Social Work and Motivational Interviewing in Outpatient Medication Assisted Substance Abuse Treatment: Model Development for Social Work Practice". VCU Scholars Compass, 2007. http://hdl.handle.net/10156/1977.
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"Jitter reduction techniques for digital audio". 1997. http://library.cuhk.edu.hk/record=b5889216.
Testo completoAbstract (sommario):
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
26
Dowdall, Peter Robert. "The impact of technology on the role and function of the bass in jazz". Thesis, 2012. http://hdl.handle.net/2440/109819.
Testo completoAbstract (sommario):
The thesis explores the critical impact of technology on the bass, and the extent to which that has fundamentally influenced the course of jazz. It does so according to a three-tiered framework involving technologies that are intrinsic to the instrument, including its eventual electrification; technologies extrinsic to the bass, most notably evolving recording technologies; and an evaluation of the ways in which the implementation of these technologies has transformed jazz as an art form. Topics include: the impact of organological changes brought about by the transition from tuba to string bass and the adoption of the electric bass; audio technology’s role in the presentation of the jazz bassist’s performance; and the influence of technical developments in the recording studio on the advancement of jazz bass style. The study also considers the challenges facing the bassist as a result of the increasing power of the recording engineer. Finally, it assesses the ways in which emerging technologies might threaten the security of the bassist within the jazz ensemble. Among the conclusions drawn is that the technological empowerment of the bassist is largely responsible for the innovations of jazz’s more celebrated soloists. In this regard the study describes the stylistic evolution of jazz from the bass player’s perspective – that is, from the musical foundation up. A clear and valid link emerges between the emancipation of the bass in jazz and the role of technology in delivering its full potential.Thesis (Ph.D.) -- University of Adelaide, Elder Conservatorium of Music, 2012.
27
Naidoo, Throshni. "The development of a behavioural video analysis system and a toolbox prototype". Thesis, 1996. http://hdl.handle.net/10413/5910.
Testo completoAbstract (sommario):
Behavioural research studies often include a data acquisition process, during which
subjects are observed, and the displayed behaviours recorded, and a subsequent analysis
process, during which the observed behaviours are analysed. In addition the data
acquisition process could occur in real-time, or at a later stage, by referring to a recording
of the original session.
This dissertation addresses the latter approach through the use of computer based digital
video technology. A windows based video analysis system, called VAS, that was
developed to assist with the acquisition of observed behaviours from the video, and the
analysis thereof is discussed. This is followed by a discussion of the implementation of
VAS. Finally the directions for further research are identified, and discussed.Thesis (M.Sc.)-University of Natal, Pietermaritzburg, 1996.