In reviewing the events of the advances of science and technology of the past two decades one is struck by the magnificence of man's accomplishments. The first moon walk televised live in 1969, the widespread utilization of color television receivers, the educational uses of sophisticated audio-visual equipment such as overhead projectors, slide projection, sound tape recorders, close-circuit television and videotape and the almost omniscient usage of television, is ample evidence to demonstrate that the tools necessary to accomplish a task are readily accessible items in this age. We are introduced to new gadgets, innovations, and events (like the moon walk) at such a rapid pace that we have become more flexible and more receptive to change; so much so that at times many of us are even blase about nearly miraculous accomplishments.

Thesis (M. Phil. (Music Technology)) -- University of Stellenbosch, 2006. / ENGLISH ABSTRACT: Modern digital signal processing, allowing a much greater degree of flexibility in audio processing and therefore greater potential for noise removal, pitch correction, filtering and editing, has allowed transfer and audio restoration engineers a diversity of ways in which to “improve” or “reinterpret” (in some cases even drastically altering) the original sound of recordings. This has lead to contrasting views regarding the role of the remastering engineer, the nature and purpose of audio restoration and the ethical implications of the restoration process. The influence of audio restoration on the recorded legacy of a performing artist is clearly illustrated in the case of Maria Callas (1923 - 1977), widely regarded not only as one of the most influential and prolific of opera singers, but also one of the greatest classical musicians of all time. EMI, for whom Callas recorded almost exclusively from 1953 - 1969, has reissued her recordings repeatedly, continually adapting their sound “to the perceived preferences of the record-buying public” (Seletsky 2000: 240). Their attempts at improving the sound of Callas’s recordings to meet with the sonic quality expected of modern recordings, as reissued in the latest releases that form part of EMI’s Callas Edition, Great Recordings of the Century (GROTC) and Historical Series, have resulted in often staggeringly different reinterpretations of the same audio material that bear no resemblance to previous CD or LP incarnations or “evince no consolidated conviction about exactly how Callas’s voice should sound.” In essence, some commentators argue that the “Callas sound” we hear on recent CD releases is not necessarily exactly as the great diva might have sounded. The purpose of this study is to consider the influence of audio restoration and remastering techniques on the recorded legacy of Callas, by illustrating the sometimes startlingly different ways in which her voice has been made to sound, examining and comparing the way in which different remasterings of the same audio material can vary in quality, as well as demonstrating how vastly different sonic reinterpretations of a single recording can affect our perception of an artist’s “true” sound. To this end, various reissues of six different complete opera recordings, including four studio recordings: Tosca (1953), Lucia di Lammermoor (1953), Norma (1954), Madama Butterfly (1955), as well as two “live” performances of Macbeth (1953) and La Traviata (1958), have been evaluated and compared, using the “true” sound of Callas’s voice as reference in comparing the different remasterings. Pitch and frequency spectrum analysis was used to confirm or support any subjective claims and observations and further analysis performed with the aid of a specialised Matlab algorithm. / AFRIKAANSE OPSOMMING: Moderne digitale seinprossesering bied kragtige en veelsydige moontlikhede vir die verwerking van klankseine. Die groter potensiaal vir ruisverwydering, toonhoogte verstelling, filtrering en redigering van opnames bied klankingenieurs ‘n wye verskeidenheid van maniere om die oorspronklike klank van opnames te verbeter, te interpreteer en soms ingrypend te verander. Dit het aanleiding gegee tot teenstrydige en uiteenlopende menings oor die funksie van die klankrestourasie-ingenieur, die aard en doel van klankrestourasie en die etiese gevolge van die restourasieproses. Die invloed van klankrestourasie op die klanknalatenskap van ‘n uitvoerende kunstenaar kan duidelik bestudeer word in die geval van Maria Callas (1923 – 1977), algemeen aanvaar as een van die mees invloedryke en grootse klassieke musici van alle tye. Die platemaatskappy EMI, vir wie Callas feitlik uitsluitlik vanaf 1953 tot 1969 opgeneem het, het haar klankopnames reeds verskeie kere heruitgereik en die klank daarvan deurlopend aangepas om aanklank te vind by die “veronderstelde voorkeure van die publiek” (Seletsky 2000: 240). EMI se pogings om die klank van Callas se opnames te verbeter om aan die klankvereistes van moderne opnames te voldoen, het ontaard in dikwels aangrypend verskillende interpretasies van dieselfde audio materiaal wat geen ooreenkomste toon met vorige laserskyf of langspeelplaat uitgawes nie, asook “geen vasgestelde oortuigings openbaar oor hoe Callas se stem presies moet klink nie.” Sommige critici argumenteer dat die “Callas klank” wat ons op hedendaagse CD uitgawes hoor, nie noodwendig klink soos wat Callas werklik geklink het nie. Die doel van hierdie studie is om die invloed van klankrestourasie op die klanknalatenskap van Callas te bestudeer deur die verskillende wyses waarop die klank van haar stem aangepas is te illustreer, die verskille in klankkwaliteit tussen verskillende uitgawes van dieselfde materiaal te ondersoek en te vergelyk, asook te demonstreer hoe uiteenlopend verskillende interpretasies van ‘n enkele opname die persepsie van ‘n kunstenaar se “ware” klank kan affekteer. Vir hierdie doel is verkeie uitgawes van ses verskillende volledige opera opnames, insluitend vier studio opnames van onderskeidelik Tosca (1953), Lucia di Lammermoor (1953), Norma (1954) en Madama Butterfly (1955), asook twee “lewendige” opnames van Macbeth (1952) en La Traviata (1958) bestudeer deur Callas se “ware” klank as maatstaf te gebruik om die onderskeie opnames te vergelyk. Toonhoogte- en frekwensie spektrum analise, asook analise deur middel van ‘n gespesialiseerde Matlab algoritme, is deurlopend gebruik om enige subjektiewe gevolgtrekkings en waarnemings te staaf.

Submitted by Filipe dos Santos (fsantos@pucsp.br) on 2018-07-04T12:24:09Z No. of bitstreams: 1 Fernanda Manzo Ceretta.pdf: 64197918 bytes, checksum: 89e46d8729046a5162627527f8dcd7bc (MD5) / Made available in DSpace on 2018-07-04T12:24:09Z (GMT). No. of bitstreams: 1 Fernanda Manzo Ceretta.pdf: 64197918 bytes, checksum: 89e46d8729046a5162627527f8dcd7bc (MD5) Previous issue date: 2018-06-26 / Conselho Nacional de Pesquisa e Desenvolvimento Científico e Tecnológico - CNPq / This research analyzes the sound design of movie monsters, especially their voices. Chewbacca (Star Wars, 1977), Godzilla (1954) and Predator (1987) constitute our corpus. These monsters have voices composed by designers who have experimented different creation processes, using sounds generated by nature, body and manipulated objects, in order to create the sound identity of these characters. We investigate the contexts of these creation processes and the resulting sounds in their particularities to make a proposition of a method for creating the sound of monsters. Our method covers the potential sources of base sounds and other sonic characteristics such as frequencies, timbre and intensity. The research was based in the observation of the selected audio-visual materials and in the documentation available regarding the making ofs (which is vast, given the popularity of the selected monsters). The thesis is based mainly on articulations with the works of Rick Altman, Michel Chion and William Whittington, on the cinematographic sound, and of Theo Van Leeuwen in his proposition of sonorous analysis / Este trabalho analisa o sound design de monstros do cinema, sobretudo suas vozes. Chewbacca (Star Wars, 1977), Godzilla (1954) e Predador (1987) constituem o corpus da presente pesquisa. Estes monstros possuem vozes compostas por designers que experimentaram diferentes processos de criação, utilizando sons na natureza, do corpo e de objetos manipulados para criar a identidade sonora destas personagens. investigamos os contextos destes processos de criação e os sons criados, em suas particularidades, para compor uma proposta de método de composição de som de monstros, o qual abarca as potenciais fontes dos sons de base e demais características sonoras, como frequências, timbres e intensidade. A pesquisa foi feita a partir da observação dos materiais audiovisuais selecionados e do resgate da documentação disponível sobre os bastidores da criação dos mesmos (bastante vasta dada a popularidade dos monstros selecionados). A tese se baseia sobretudo em articulações com obras de Rick Altman, Michel Chion, William Whittington acerca do som cinematográfico e em Theo Van Leeuwen em sua proposição de análise sonora

Virtual three-dimensional (3D) auditory displays utilize signal-processing techniques to alter sounds presented through headphones so that they seem to originate from specific spatial locations around the listener. In some circumstances bone-conduction headsets (bonephones) can provide an alternative sound presentation mechanism. However, existing 3D audio rendering algorithms need to be adjusted to use bonephones rather than headphones. This study provided anchor points for a function of shift values that could be used to adapt virtual 3D auditory displays for use with bonephones. The shift values were established by having participants adjust phase and amplitude of two waves in order to cancel out the signal and thus produce silence. These adjustments occurred in a listening environment consisting of air-conducted and bone-conducted tones, as well as air- conducted masking. Performance in the calibration condition suggested that participants understood the task, and could do this task with reasonable accuracy. In the bone-to-air listening conditions, the data produced a clear set of anchor points for an amplitude shift function. The data did not reveal, however, anchor points for a phase shift function the data for phase were highly variable and inconsistent. Application of shifts, as well as future research to establish full functions and better understand phase are discussed, in addition to validation and follow-up studies.

Thesis (MPhil) -- University of Stellenbosch, 2000. / ENGLISH ABSTRACT: The most neglected element in television reporting is the sound track. The problem is illustrated by the fact that there are currently no specialised textbooks, or training courses, on sound recording for television reporting, anywhere in the world. Textbooks that deal with television reporting dedicate very little space to sound recordings. With the growing competition in television news, news teams are increasingly becoming smaller. It is common practice these days to have a news team consisting of only a cameraperson and a reporter. The cameraperson is also responsible for the sound. Two television stations, namely NYl in New York and Channel One Television in England, have already dispensed with the cameraperson and send out only a reporter. This dissertation addresses the above-mentioned problem by doing research on the sound equipment, recording techniques and production techniques that are useful for effective sound recordings during single camera television reporting. Chapter two explains the functions of the different departments in a television station, as well as the duties of the staff. Chapter three explains basic television principles, terminology and equipment that the sound person uses daily and needs to understand to perform his work optimally. Chapterfour gives the basic terms that are needed to follow a conversation on sound recording. Terms like decibel, stereo and digital sound are explained. The chapter also covers basic electricity and sound equipment. Chapter five covers microphones under three headings, namely: electrical characteristics, acoustic characteristics and microphone design. The chapter also covers associated equipment, explains the sound facilities on video cameras and gives a list of possible sound equipment that can be used during a production. Chapter six covers sound recordings, principles and techniques under the following headings: • Perspective and boom swinging, which deals with sound perspective and boom swinging. • Rigging of cables, which gives practical tips for laying cables inside and outside buildings. • Recording principles, which gives practical tips on sound recordings. • Interviews, which includes recording tips for television interviews and reporting. • Reporting, which covers reporting, media conferences and public events. • Commentary recordings, which deals with the preparation and recording of voice over. • Music recordings, which deals with instruments and bands, and suggests microphone positions. • Telephone lines, which covers the use of telephone lines for reporting. • Location reconnaissance, which gives practical tips on pre-production planning. • Guidelines for sound persons during productions, which concludes the chapter and the dissertation with practical tips on behaviour during local, foreign and/or dangerous productions. / AFRIKAANSE OPSOMMING: Die veranderlike wat die meeste afgeskeep word in televisieverslaggewing, is die klankbaan. Die probleem word onderstreep deur die feit dat daar wêreldwyd tans geen gespesialiseerde handboeke of opleidingskursusse bestaan wat oor klankopnames vir televisieverslaggewing handel nie. Handboeke wat handeloor televisieverslaggewing wy ook baie min ruimte aan klankopnames. Met die groeiende kompetisie in televisienuus raak nuusspanne toenemend kleiner. Dit is reeds algemene praktyk dat die nuusspan slegs uit 'n kamerapersoon en 'n verslaggewer bestaan. Die kamerapersoon moet dus ook die klankopnames doen. Twee televisiestasies, naamlik NYl in New York en Channel One Television in Engeland, het reeds die kamerapersoon uitgeskakel en stuur slegs 'n verslaggewer uit. Hierdie verhandeling spreek bogenoemde probleem aan deur navorsing te doen oor die klanktoerusting, opnametegnieke en produksietegnieke wat nuttig is vir effektiewe klankopnames tydens enkelkameratelevisieverslaggewing. Hoofstuk twee verduidelik die funksies van die verskillende departemente in 'n televisiestasie sowel as die pligte van die personeel. Hoofstuk drie verduidelik basiese televisiebeginsels, -terminologie en -toerusting wat die klankpersoon daagliks mee werk en dus moet verstaan om sy werk optimaal te verrig. Hoofstuk vier gee die basiese terme wat nodig is om 'n gesprek oor klankopnames te volg. Begrippe soos desibel, stereo en digitale klank word verduidelik. Die hoofstuk behandelook basiese elektriese beginsels en klanktoerusting. Hoofstuk vyf bespreek mikrofone onder drie indelings naamlik: elektriese eienskappe, akoestiese eienskappe en mikrofoonontwerp. Die hoofstuk dek ook aanverwante toerusting, verduidelik die klankfasiliteite op videokameras en gee 'n lys van klanktoerusting wat tydens produksies gebruik kan word. Hoofstuk ses is die belangrikste en bespreek klankopnames, beginsels en tegnieke onder die volgende opskrifte: • Perspektief en boomhantering, waaronder klankperspektief en boomhantering behandel word. • Lê van kabels, wat praktiese wenke gee vir die lê van kabels binne en buite geboue. • Opnamebeginsels, wat praktiese wenke gee in verband met klankopnames. • Onderhoude, wat opnamewenke gee in verband met televisieonderhoude en verslaggewing. • Verslaggewing, wat verslaggewing, nuuskonferensies en openbare geleenthede dek. • Kommentaaropnames, wat handeloor die voorbereiding vir, en opneem van kommentaar. • Musiekopnames, wat musiekinstrumente en orkeste bespreek en mikrofoonposisies voorstel. • Telefoonlyne, waaronder die gebruik van telefoonlyne vir verslaggewing bespreek word. • Terreinverkenning, wat praktiese riglyne gee vir voorproduksie-ondersoeke. • Riglyne vir klankpersone tydens produksies, wat die hoofstuk en die studie afsluit met praktiese wenke vir gedrag tydens plaaslike, buitelandse en/of gevaarlike produksies.

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.

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.

Acompanha 1 CD-ROM / Orientador: Jose Armando Valente / Tese (doutorado) - Universidade Estadual de Campinas, Instituto de Artes / Made available in DSpace on 2018-08-12T09:43:21Z (GMT). No. of bitstreams: 1 Tonus_Mirna_D.pdf: 2485563 bytes, checksum: 4314568417f46f70cc65fd3e9ce62375 (MD5) Previous issue date: 2007 / Resumo: O radiojornalismo atual tem requisitado uma nova formação, voltada à edição digital, frente às exigências da atividade profissional. Para concretizá-la, é necessário recorrer às tecnologias da informação e da comunicação (TIC), empregando-as como dispositivo de interação com os aprendizes e de interatividade entre eles e a máquina, o que inclui o desenvolvimento de habilidades para lidar com computadores, softwares e, especificamente, com o áudio digital. Este é o foco desta tese, que traça um dos caminhos possíveis para essa formação, tratando dos cenários científico e documentarista aplicados no radiojornalismo, a partir de teorias da aprendizagem envolvendo o aprender fazendo de Piaget, a zona proximal de desenvolvimento (ZPD) de Vygotsky, a construção social do conhecimento de Freire, e o ciclo de ações e a espiral de aprendizagem de Valente. Com base em pesquisa-ação junto a estudantes de Jornalismo da Universidade de Sorocaba (Uniso), a tese apresenta uma análise de similaridade e implicativa, evidenciando que o processo de aprendizagem é auxiliado com o uso das TIC em uma proposta de formação plurimodal, utilizando elementos da educação presencial e da educação a distância (EAD). / Abstract: The contemporary radiojournalism requests a new formation, directed to the digital edition, because of the requirements of the professional activity. To materialize it, it's necessary to appeal to the information and communication technologies (ICT), using them as device of interaction with the apprenticees and interactivity between them and the machine, what includes the development of abilities to deal with computers, softwares and, specifically, with digital audio one. This is the focus of this thesis, that traces one of the possible ways for this formation, treating to the scientific and documentarist scenes applied in the radiojournalism, from learning theories involving learning by doing of Piaget, the zone of proximal development (ZPD) of Vygostky, social construction of the knowledge of Freire, and the actions cycle and learning spiral of Valente. On the basis of action-research with Journalism students at University of Sorocaba (Uniso), the thesis presents a similarity and implicative analysis, evidencing that the learning process is assisted with the use of ICT in a proposal of plural mode education, using traditional and distance education elements. / Doutorado / Doutor em Multimeios

With the rapid adoption of multimedia network technologies, a number of companies and standards bodies are introducing technologies that enhance user experience in networked multimedia environments. These technologies focus on device discovery, connection management, control, and monitoring. This study focused on control and monitoring. Multimedia networks make it possible for devices that are part of the same network to reside in different physical locations. These devices contain parameters that are used to control particular features, such as speaker volume, bass, amplifier gain, and video resolution. It is often necessary for changes in one parameter to affect other parameters, such as a synchronised change between volume and bass parameters, or collective control of multiple parameters. Thus, relationships are required between the parameters. In addition, some devices contain parameters, such as voltage, temperature, and audio level, that require constant monitoring to enable corrective action when thresholds are exceeded. Therefore, a mechanism for monitoring networked devices is required. This thesis proposes relationships that are essential for the proper functioning of a multimedia network and that should, therefore, be incorporated in standard form into a protocol, such that all devices can depend on them. Implementation mechanisms for these relationships were created. Parameter grouping and monitoring capabilities within mixing console implementations and existing control protocols were reviewed. A number of requirements for parameter grouping and monitoring were derived from this review. These requirements include a formal classification of relationship types, the ability to create relationships between parameters with different underlying value units, the ability to create relationships between parameters residing on different devices on a network, and the use of an event-driven mechanism for parameter monitoring. These requirements were the criteria used to govern the implementation mechanisms that were created as part of this study. Parameter grouping and monitoring mechanisms were implemented for the XFN protocol. The mechanisms implemented fulfil the requirements derived from the review of capabilities of mixing consoles and existing control protocols. The formal classification of relationship types was implemented within XFN parameters using lists that keep track of the relationships between each XFN parameter and other XFN parameters that reside on the same device or on other devices on the network. A common value unit, known as the global unit, was defined for use as the value format within value update messages between XFN parameters that have relationships. Mapping tables were used to translate the global unit values to application-specific (universal) units, such as decibels (dB). A mechanism for bulk parameter retrieval within the XFN protocol was augmented to produce an event-driven mechanism for parameter monitoring. These implementation mechanisms were applied to an XFN-protocol-compliant graphical control application to demonstrate their usage within an end user context. At the time of this study, the XFN protocol was undergoing standardisation within the Audio Engineering Society. The AES-64 standard has now been approved. Most of the implementation mechanisms resulting from this study have been incorporated into this standard.

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.