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.

This Bachelor Thesis in composition explores an alternative compositional process for creating improvised music through the approach of sound design. The questions addressed in the thesis are concerned with how the soundscape can be controlled by the composer while still maintaining the expressivity of an improvised performance. Thus, the composer’s role as both sound designer, producer and performer is investigated using live electronics and score design. The theoretical framework for the thesis is to be found in the intersection between Anglo-American experimental music, electronic music, sound design and electroacoustic improvisation. Key influences include indeterminacy as expressed in the music of John Cage, Deep Listening practices pioneered by Pauline Oliveros as well as the aesthetical paradigms of electronic and electroacoustic music. The artistic methods revolved around the following steps: studio workshops with musicians (Fredrik Ekenvi and José Louis Relova Gallego) exploring timbres, playing techniques and music technology in real-time followed by associative discussions about the soundscapes created; composing audio and graphical drafts influenced by the workshops; designing diagrams for the electronics and compiling these to a design document; workshopping and performing the final piece (Towards the Event Horizon). Towards the Event Horizon (for bass clarinet, electric guitar and live electronics) is the final result of the work, comprising a score with accompanying design document as well as a recording of the original performance at Studio Acusticum in Piteå 20th of March 2021. The artefacts and associated documentation are presented in their entirety on the dedicated Research Catalogue exposition. Conclusions include how composition and sound design are tied together and cannot be considered as isolated processes, especially in the context of live electronics; the interrelationship between the improviser, the acoustic soundscape and the artistic associations that inform the improvisation; the importance of form as the defining element of a composition and lastly; how a score for improvised music can be devised in the shape of a blueprint of form and texture, setting certain technical, sonic and gestural conditions while leaving the performer to play freely within this frame. / <p>Konstnärliga artefakter presenteras via Research Catalogue: https://www.researchcatalogue.net/view/1250652/1250651</p>

Esta dissertação apresenta um estudo comparativo de diferentes métodos computacionais de segmentação estrutural musical, onde o principal objetivo é delimitar fronteiras de seções musicais em um sinal de áudio, e rotulá-las, i.e. agrupar as seções encontradas que correspondem a uma mesma parte musical. São apresentadas novas propostas para segmentação estrutural nãosupervisionada, incluindo métodos para processamento em tempo real, alcançando resultados com taxas de erro inferiores a 12%. O método utilizado compreende um estudo dos descritores sonoros e meios de modelá-los temporalmente, uma exposição das técnicas computacionais de segmentação estrutural e novos métodos de avaliação dos resultados que penalizam tanto a incorreta detecção das fronteiras quanto o número incorreto de rótulos encontrados. O desempenho de cada técnica computacional é calculado utilizando diferentes conjuntos de descritores sonoros e os resultados são apresentados e analisados tanto quantitativa quanto qualitativamente. / A comparative study of different music structural segmentation methods is presented, where the goal is to delimit the borders of musical sections and label them, i.e. group the sections that correspond to the same musical part. Novel proposals for unsupervised segmentation are presented, including methods for real-time segmentation, achieving expressive results, with error ratio less then 12%. Our method consists of a study of sound descriptors, an exposition of the computational techniques for structural segmentation and the description of the evaluation methods utilized, which penalize both incorrect boundary detection and incorrect number of labels. The performance of each technique is calculated using different sound descriptor sets and the results are presented and analysed both from quantitative and qualitative points-of-view.

Cette thèse concerne l’anatomie macroscopique des gyri de Heschl (GH) en relation avec la Préférence Manuelle (PM) et la Spécialisation Hémisphérique (SH) pour le langage étudiée dans une base de données multimodale dédiée à l’étude de la SH (la BIL&GIN), équilibrée pour le sexe et la PM. Le GH, situé à la surface du lobe temporal, abrite l’aire auditive primaire. Des études ont montré que le volume du GH est asymétrique gauche, et que le GH gauche (GHG) covarie avec les performances phonologiques et avec la quantité de cortex dévolu au traitement temporel des sons, suggérant une relation entre GHG et SH pour le langage. Pourtant l’anatomie des GH, très variable en terme de gyrification, est mal connue. Nous avons : 1- Décrit la distribution inter-hémisphérique de la gyrification des GH sur les images IRM anatomiques de 430 sujets. 2- Etudié les variations de surface et d’asymétrie du premier gyrus ou GH antérieur (GHa), montré sa diminution en présence de duplication et l’existence d’une asymétrie gauche pour les configurations les plus fréquentes avec GHG unique. Les gauchers présentaient moins de duplications droites et une perte de l’asymétrie gauche de GHa. 3- Testé si la variance de l’anatomie du GH expliquait la variabilité interindividuelle des asymétries mesurées en IRM fonctionnel pendant une tâche d’écoute de mots chez 281 sujets, et si les différences anatomiques liées à la PM étaient en relation avec une diminution de la SH pour le langage des gauchers. La distribution du nombre de GH expliquait 11% de la variance de l’asymétrie fonctionnelle du GH, les configurations à GHG unique étant les plus asymétriques gauches, sans effet de la PM sur la latéralisation fonctionnelle du GH. / This thesis concerns the macroscopical anatomy of Heschl’s gyri (HG) in relation with Manual Preference (MP) and the Hemispheric Specialization (HS) for language studied in a multimodal database dedicated to the investigation of HS and balanced for sex and MP (BIL&GIN). HG, located on the surface of the temporal lobe, hosts the primary auditory cortex. Previous studies have shown that HG volume is leftward asymmetrical and that the left HG (LHG) covaries with phonological performance and with the amount of cortex dedicated to the processing of the temporal aspects of sounds, suggesting a relationship between LHG and HSL. However HG anatomy is highly variable and little known. In this thesis we have: 1- Described HG inter-hemispheric gyrification pattern on the anatomical MRI images of 430 healthy participants. 2- Studied the variation of the first or anterior HG (aHG) surface area and its asymmetry and shown its reduction in the presence of duplication and that its leftward asymmetry was present only in the case of a single LHG. Left-handers exhibited a lower incidence of right duplication and a loss of aHG leftward asymmetry. 3- Tested whether the variance of HG anatomy explained the interindividual variability of asymmetries measured with fMRI during the listening of a list of words in 281 participants, and whether differences in HG anatomy with MP were related to decreased HS for language in left-handers. HG inter-hemispheric gyrification pattern explained 11% of the variance of HG functional asymmetry, the patterns including a unique LHG being those with the strongest leftward asymmetry. There was no incidence of MP on HG functional lateralization.