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

We live in a dynamic world full of noises and interferences. The conventional acoustic echo cancellation (AEC) framework based on the least mean square (LMS) algorithm by itself lacks the ability to handle many secondary signals that interfere with the adaptive filtering process, e.g., local speech and background noise. In this dissertation, we build a foundation for what we refer to as the system approach to signal enhancement as we focus on the AEC problem. We first propose the residual echo enhancement (REE) technique that utilizes the error recovery nonlinearity (ERN) to "enhances" the filter estimation error prior to the filter adaptation. The single-channel AEC problem can be viewed as a special case of semi-blind source separation (SBSS) where one of the source signals is partially known, i.e., the far-end microphone signal that generates the near-end acoustic echo. SBSS optimized via independent component analysis (ICA) leads to the system combination of the LMS algorithm with the ERN that allows for continuous and stable adaptation even during double talk. Second, we extend the system perspective to the decorrelation problem for AEC, where we show that the REE procedure can be applied effectively in a multi-channel AEC (MCAEC) setting to indirectly assist the recovery of lost AEC performance due to inter-channel correlation, known generally as the "non-uniqueness" problem. We develop a novel, computationally efficient technique of frequency-domain resampling (FDR) that effectively alleviates the non-uniqueness problem directly while introducing minimal distortion to signal quality and statistics. We also apply the system approach to the multi-delay filter (MDF) that suffers from the inter-block correlation problem. Finally, we generalize the MCAEC problem in the SBSS framework and discuss many issues related to the implementation of an SBSS system. We propose a constrained batch-online implementation of SBSS that stabilizes the convergence behavior even in the worst case scenario of a single far-end talker along with the non-uniqueness condition on the far-end mixing system. The proposed techniques are developed from a pragmatic standpoint, motivated by real-world problems in acoustic and audio signal processing. Generalization of the orthogonality principle to the system level of an AEC problem allows us to relate AEC to source separation that seeks to maximize the independence, hence implicitly the orthogonality, not only between the error signal and the far-end signal, but rather, among all signals involved. The system approach, for which the REE paradigm is just one realization, enables the encompassing of many traditional signal enhancement techniques in analytically consistent yet practically effective manner for solving the enhancement problem in a very noisy and disruptive acoustic mixing environment.

In a large number of electronic devices, users interact with the system by navigating through various menus. Auditory menus can complement or even replace visual menus, so research on auditory menus has recently increased with mobile devices as well as desktop computers. Despite the potential importance of auditory displays on touch screen devices, little research has been attempted to enhance the effectiveness of auditory menus for those devices. In the present study, I investigated how advanced auditory cues enhance auditory menu navigation on a touch screen smartphone, especially for new input gestures such as tapping, wheeling, and flicking methods for navigating a one-dimensional menu. Moreover, I examined if advanced auditory cues improve user experience, not only for visuals-off situations, but also for visuals-on contexts. To this end, I used a novel auditory menu enhancement called a "spindex" (i.e., speech index), in which brief audio cues inform the users of where they are in a long menu. In this study, each item in a menu was preceded by a sound based on the item's initial letter. One hundred and twenty two undergraduates navigated through an alphabetized list of 150 song titles. The study was a split-plot design with manipulated auditory cue type (text-to-speech (TTS) alone vs. TTS plus spindex), visual mode (on vs. off), and input gesture style (tapping, wheeling, and flicking). Target search time and subjective workload for the TTS + spindex were lower than those of the TTS alone in all input gesture types regardless of visual type. Also, on subjective ratings scales, participants rated the TTS + spindex condition higher than the plain TTS on being 'effective' and 'functionally helpful'. The interaction between input methods and output modes (i.e., auditory cue types) and its effects on navigation behaviors was also analyzed based on the two-stage navigation strategy model used in auditory menus. Results were discussed in analogy with visual search theory and in terms of practical applications of spindex cues.

Thesis (M.Sc. (Computer Science)) - Rhodes University, 2008.

Thesis (Ph.D.)--University of Wollongong, 2007. / Typescript. Includes 2 sound discs and 1 DVD-ROM in back pocket. CD 1: The animation of lists; CD 2: And the archytan transpositions. DVD-ROM contains Part Three - Appendix. Includes bibliographical references: leaf 291-301.

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.

Orientador: Max Henrique Machado Costa / Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação / Made available in DSpace on 2018-08-10T00:12:12Z (GMT). No. of bitstreams: 1 GoncalvesNeto_Jahyr_M.pdf: 950657 bytes, checksum: f62691d16e5db013d1b8a9c4e4a32c88 (MD5) Previous issue date: 2007 / Resumo: Nesta dissertação apresentamos o desenvolvimento de uma plataforma multimídia baseada no modelo cliente-servidor voltada para aplicações de streaming de áudio e vídeo. Essa plataforma deverá evoluir para um sistema de videoconferência em um projeto futuro. A plataforma permite a comunicação de áudio, vídeo e texto a partir de um ponto (o servidor) para vários outros pontos (os clientes). Uma das inovações do projeto está no desenvolvimento em Python, que é uma linguagem interpretada, orientada a objetos e dinamicamente tipada / Abstract: This dissertation presents the development of a client-server platform designed initially for audio and video streaming applications. This platform will evolve into a videoconference system as part of a future project. The platform allows audio, video and text communication from a point (the server) to several others points (the clients). One of the project innovations is the implementation Python Language, which is an interpreted, objectoriented and dynamically typed language / Mestrado / Telecomunicações e Telemática / Mestre em Engenharia Elétrica

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.