This thesis focuses on a time in the mid-1960s where practice in the studio changed from a formal arena where previously rehearsed songs were recorded, to a playground where sonic possibilities were explored and sound manipulation became normal practice. This abuse of technology and manipulation of reality became part of the creative process in the studio, providing soundscapes that resonated with the counter-cultural ethos of upsetting the established order, and were adopted by the mainstream during the 1967 ‘Summer of Love”. Following a discussion of current literature, practice as research is applied to demonstrate how interaction with historical technology reveals the performative nature of the tacit knowledge that created many of the aural effects under consideration. The research then focuses through the prism of two case studies, “Eight Miles High” recorded by The Byrds in Los Angeles in January 1966, and “Rain”, recorded by The Beatles in London in April 1966. Through re-enactment of these historical recording sessions, I recreate the closed envirnment of the 1960’s recording studio. By interacting with historical technology and following a similar structure to the original sessions, I investigate how the methodology was influenced by collaborative actions, situational awareness and the demarcation of roles. Post session video analysis reveals the flow of decision making as the sessions unfold, and how interaction with the technological constraints recreates ‘forgotten’ techniques that were deemed everyday practice at the time and were vital to the outcome of the soundscapes. The thesis combines theory and practice to develop an understanding of how the engineers interacted with technology (Polanyi, 1966), often abusing the equipment to create manipulated soundscapes (Akrich and Latour, 1992), and how the sessions responded to musicians demanding innovation and experimentation, circumventing the constraints of established networks of practice (Law and Callon, 1986) during the flow of the recording session (Ingold, 2013).

This thesis explores the historical development of the electronic organ via the survey, analysis and comparison of stylistic practices heard in historic recordings. This project establishes that the instrument went through several significant stages of development since its introduction in 1935, which have hitherto been undocumented in scholarly work. As this thesis will show, the changing design of the instrument can be aligned with an evident expansion in the stylistic lexicon of musical arrangement and performance. This aural-based micro-genre of electronic music is rediscovered via a multi-faceted survey model that triangulates the results of transcribed recordings, reconstructive performance on period instruments and practitioner survey. This addresses the typical challenge of historical instrument study: that of defining the degree to which technology shapes musical performance. Chapter One places the instrument within a cultural context via a review of literature. The reason for the instrument’s lack of appeal to musicologists is explained as the result of an image problem: the instrument is often regarded as a dated appliance of home entertainment and exists within a method of practice which aligns more closely to that of jazz than Western art music. By removing stereotypes and establishing the displaced cultural values that the instrument embodies, it is possible to see the true value of the research process. Chapter Two begins to present the findings of the survey by examining some of the earliest recordings made on the Hammond organ. The chapter illustrates how certain design flaws in an instrument that was originally Christopher Stanbury Introduction 6 intended as a low-cost replacement for a pipe organ led to an entirely different trajectory than the inventor’s initial ecclesiastical application. Chapter Three details further updates to the original Hammond design whilst correcting and expanding upon previous definitions of features that are defined in literature. The Lowrey organ is also introduced, along with an illustration of why the unique features and tonal qualities of the instrument resulted in a different approach to musical arrangement and performance. Chapter Four documents the introduction of emulative voicing, whereby instruments of the nineteen seventies and early eighties were designed to imitate the sound of other acoustic instruments. The resultant change in arrangement and performance style is illustrated and compared to the results of previous chapters. Chapter Five details instruments made by the Yamaha Corporation that feature digital synthesis technologies. The vast distance between these instruments and previous models, both in terms of technological profile and resultant performance practice, is illustrated and discussed. Chapter Six provides a summary of the survey findings and reexamines the evident changes in the instrument and performance practice. The nature of the relationship between organist and instrument is discussed, along with a return to some of the literature reviewed in Chapter One. Discrepancies between the conclusions of some authors and those of this thesis are outlined and discussed.

Tese de mestrado. Multimédia - Música Interactiva e Sound Design. Faculdade de Engenharia. Universidade do Porto. 2012

Research in the field of music information retrieval (MIR) is concerned with methods to effectively retrieve a piece of music based on a user's query. An important goal in MIR research is the ability to successfully retrieve music stored as recorded audio using note-based queries. In this work, we consider the searching of musical audio using symbolic queries. We first examined the effectiveness of using a relative pitch approach to represent queries and pieces. Our experimental results revealed that this technique, while effective, is optimal when the whole tune is used as a query. We then suggested an algorithm involving the use of pitch classes in conjunction with the longest common subsequence algorithm between a query and target, also using the whole tune as a query. We also proposed an algorithm that works effectively when only a small part of a tune is used as a query. The algorithm makes use of a sliding window in addition to pitch classes and the longest common subsequence algorithm between a query and target. We examined the algorithm using queries based on the beginning, middle, and ending parts of pieces. We performed experiments on an audio collection and manually-constructed symbolic queries. Our experimental evaluation revealed that our techniques are highly effective, with most queries used in our experiments being able to retrieve a correct answer in the first rank position. In addition, we examined the effectiveness of duration-based features for improving retrieval effectiveness over the use of pitch only. We investigated note durations and inter-onset intervals. For this purpose, we used solely symbolic music so that we could focus on the core of the problem. A relative pitch approach alongside a relative duration representation were used in our experiments. Our experimental results showed that durations fail to significantly improve retrieval effectiveness, whereas inter-onset intervals significantly improve retrieval effectiveness.

Les systèmes de réalité virtuelle interactifs combinent des représentations visuelle, sonore et haptique, afin de simuler de manière immersive l'exploration d'un monde tridimensionnel représenté depuis le point de vue d'un observateur contrôlé en temps réel par l'utilisateur. La plupart des travaux effectués dans ce domaine ont historiquement port'e sur les aspects visuels (par exemple des méthodes d'affichage interactif de modèles 3D complexes ou de simulation réaliste et efficace de l'éclairage) et relativement peu de travaux ont été consacrés 'a la simulation de sources sonores virtuelles 'également dénommée auralisation. Il est pourtant certain que la simulation sonore est un facteur clé dans la production d'environnements de synthèse, la perception sonore s'ajoutant à la perception visuelle pour produire une interaction plus naturelle. En particulier, les effets sonores spatialisés, dont la direction de provenance est fidèlement reproduite aux oreilles de l'auditeur, sont particulièrement importants pour localiser les objets, séparer de multiples signaux sonores simultanés et donner des indices sur les caractéristiques spatiales de l'environnement (taille, matériaux, etc.). La plupart des systèmes de réalité virtuelle immersifs, des simulateurs les plus complexes aux jeux vidéo destin'es au grand public mettent aujourd'hui en œuvre des algorithmes de synthèse et spatialisation des sons qui permettent d'améliorer la navigation et d'accroître le réalisme et la sensation de présence de l'utilisateur dans l'environnement de synthèse. Comme la synthèse d'image dont elle est l'équivalent auditif, l'auralisation, appel'ee aussi rendu sonore, est un vaste sujet 'a la croisée de multiples disciplines : informatique, acoustique et 'électroacoustique, traitement du signal, musique, calcul géométrique mais également psycho-acoustique et perception audio-visuelle. Elle regroupe trois problématiques principales: synthèse et contrôle interactif de sons, simulation des effets de propagation du son dans l'environnement et enfin, perception et restitution spatiale aux oreilles de l'auditeur. Historiquement, ces trois problématiques émergent de travaux en acoustique architecturale, acoustique musicale et psycho-acoustique. Toutefois une différence fondamentale entre rendu sonore pour la réalité virtuelle et acoustique réside dans l'interaction multimodale et dans l'efficacité des algorithmes devant être mis en œuvre pour des applications interactives. Ces aspects importants contribuent 'a en faire un domaine 'a part qui prend une importance croissante, tant dans le milieu de l'acoustique que dans celui de la synthèse d'image/réalité virtuelle.

L'objectif de cette thèse est la recherche d'outils de détection et de reconnaissance de la structure rythmique d'un morceau de musique. Il existe de nombreuses applications nécessitant l'information rythmique, parmi lesquelles la transcription musicale par ordinateur, l'alignement rythmique de plusieurs instruments, tâches de "music information retrieval". L'axe principal du travail de recherche a consisté en le développement de techniques pour l'estimation des paramètres rythmiques (cadences et localisation de beats à plusieurs niveaux). Parmi les originalités de ce travail de recherche, nous avons proposé de effectuer l'analyse rythmique de façon séparée sur la partie déterministe du signal musical (contenant les sons harmoniques) et sur la partie stochastique (contenant le signal résiduel après extraire la partie harmonique du son original). Dans le domaine du traitement des signaux audio, la détection robuste et efficace des attaques (onsets) des signaux musicaux est considérée comme un sujet difficile. A ce propos, nous utilisons le concept du flux énergétique spectral comme le taux auquel l'énergie du contenu fréquentiel du signal audio change à un instant donné. Et nous nous sommes servis de ce principe pour développer un détecteur d'attaques très performant qui améliore de façon considérable la méthode de calcule traditionnelle. Une autre originalité de ce travail consiste en l'utilisation d'un algorithme de programmation dynamique modifiée pour suivre simultanément plusieurs trajectoires rythmiques à l'intérieur d'un signal musical. L'efficacité de cette approche a été validé sur une base de données comportant 1435 morceaux musicaux appartenant à une dizaine de genres. Les algorithmes développés ont été soumis à évaluation dans le cadre du concours international "Music Information Retrieval Evaluation eXchange" (MIREX) dans la catégorie de "tempo extraction" et dans la l'édition 2005 un des algorithmes obtenu la première place parmi plus d'une dizaine de participants.

Detta arbete inspirerades av Chions (1994) term det audiovisuella kontraktet samt Nørretranders (1993) forskning om medvetandes bandbredd. Problemställningen för detta arbete var: kan man utan uppmaning höra ett ljud som betraktas som ett svåruppmärksammat ljud i en ljudbild som består av ljud som är lättare att höra, med hjälp av en bild. Testpersonerna fick lyssna på ett ljudklipp som bestod av fem ljud. Ett av dessa ljud var förhållandevis mycket svår att höra i jämförelse till de andra ljuden. Medan testpersonerna lyssnade på ljudklippet fick de se en av två bilder. Totalt skapades två ljudklipp och två bilder till genomförandet av undersökningen. Det som undersöktes var hur många testpersoner hörde klockan och hur resultaten varierade beroende på den visuella rekvisitan. Utav 28 personer hörde tre människor klockan som påvisar att svaret till problemställningen skulle kunna vara ”ja”. I framtida arbeten skulle detta kunna utvecklas för att användas som ett verktyg inom dataspelsutveckling.

In wireless transmissions the communication is often degraded by random fades, noise and other performance reducing phenomena. One way of improving the stability and reducing the error rates is to use relaying techniques where several nodes cooperate in a transmission between two of them. This thesis analyzes some of the available Decode-and-Forward relaying schemes for wireless transmission. The investigated schemes are conventional repetition coding, partial repetition coding and non-collaborative direct transmission. I have developed a three-node communication system using an audio channel to test the performance of repetition coding and direct transmission. This audio communication system can also be used to demonstrate some basic phenomena in wireless transmissions and how different scenarios change the performance of the communication. A theoretical performance analysis and computer simulations of the schemes performance over a Rayleigh fading channel are done as a basis for comparison. As a result we see that in the audio communication system repetition coding actually degrades the performance, compared to direct transmission, when using a relatively slow data rate in comparison to the speed of the fading in the audio channel.

This report describes the design and implementation of a fixed audio equalizer based on a scheme where parts of the signal spectrum are downsampled and treated differently for the purpose of reducing the computational complexity and memory requirements. The primary focus has been on finding a way of taking an equalizer based on a simple minimum-phase FIR filter and transform it to the new type of equalizer. To achieve this, a number of undesireable effects such as aliasing distortion and upsampling imaging had to be considered and dealt with. In order to achieve a good amplitude response of the system, optimization procedures were used. As part of the thesis, a cost-effective implementation of the filter has been made for an FPGA, in order to verify that the scheme is indeed usable for equalizing an audio signal.

Originally presented as author's thesis: der Technischen Universität Berlin, 2008.