Wave Field Synthesis in a listening room

This thesis investigates the influence of the listening room on sound fields synthesised by Wave Field Synthesis. Methods are developed that allow for investigation of the spatial and timbral perception of Wave Field Synthesis in a reverberant environment using listening experiments based on simulation by binaural synthesis and room acoustical simulation. The results can serve as guidelines for the design of listening rooms for Wave Field Synthesis.Diese Dissertation untersucht den Einfluss des Wiedergaberaums auf Schallfelder, die mit Wellenfeldsynthese synthetisiert werden. Es werden Methoden zur Untersuchung von räumlicher und klangfarblicher Wahrnehmung von Wellenfeldsynthese in einer reflektierenden Umgebung mittels Hörversuchen entwickelt, die auf Simulation mit Binauralsynthese und raumakustischer Simulation beruhen. Die Ergebnisse können als Richtlinien zur Gestaltung von Wiedergaberäumen für Wellenfeldsynthese dienen

The Oramics Machine: From Vision to Reality

The pioneering contributions of Daphne Oram to visual music, notably the construction of her unique synthesiser known as the Oramics Machine during the 1960s, have yet to be fully recognised. The development of this synthesiser, in terms of both the creative objectives that inspired its design and also the functional characteristics of the resulting technology, is all the more remarkable for being the product of highly individual endeavour, working entirely without the support and resources normally provided by an institution or a commercial manufacturer. Oram's background in both music and electronics was to prove invaluable in this regard, and her appointment as the founding director of the BBC Radiophonic Workshop in 1958, having previously lobbied within the organisation for such a facility for several years, provides testament to her standing in both regards. Her decision within a year of appointment to resign from this post and establish her own private studio specifically to develop Oramics is indicative of her determination and commitment to explore new horizons in the medium of electronic music, and this paper provides a perspective of her achievements, drawing on materials in the Oram archive that have hitherto not been studied

Allpass Feedback Delay Networks

In the 1960s, Schroeder and Logan introduced delay line-based allpass filters, which are still popular due to their computational efficiency and versatile applicability in artificial reverberation, decorrelation, and dispersive system design. In this work, we extend the theory of allpass systems to any arbitrary connection of delay lines, namely feedback delay networks (FDNs). We present a characterization of uniallpass FDNs, i.e., FDNs, which are allpass for an arbitrary choice of delays. Further, we develop a solution to the completion problem, i.e., given an FDN feedback matrix to determine the remaining gain parameters such that the FDN is allpass. Particularly useful for the completion problem are feedback matrices, which yield a homogeneous decay of all system modes. Finally, we apply the uniallpass characterization to previous FDN designs, namely, Schroeder's series allpass and Gardner's nested allpass for single-input, single-output systems, and, Poletti's unitary reverberator for multi-input, multi-output systems and demonstrate the significant extension of the design space

Making things louder : amplified music and multimodality

University of Technology, Sydney. Faculty of Arts and Social Sciences.This thesis looks at the use of electronic amplification at concerts of music. A broad introduction, constituting both a technological and a musical history, precedes a literature review that identifies the topic as under-researched in musical, technological and critical discourse. Proceeding from that broad approach which covers the first three chapters the analytical focus is narrowed by applying key concepts from social semiotic multimodal discourse analysis, as developed by amongst others Gunther Kress and Theo van Leeuwen and rooted in the work of linguist Michael Halliday. In addition, elements of the work of sociologist Erving Goffman are explored; notably his use of ‘decorum’, the ‘participation framework’ and ‘production formats’. Amplification is treated as a semiotic mode; the different meaning potentials of the use of technology are outlined and contrasted with the notion of reproduction technology as a neutral channel. Questions of the relation between original (or acoustic, ie the sound that is amplified) and the amplified sound are analysed using the concept of linguistic modality, so as to investigate how notions of musical truth such as authenticity or fidelity, are encoded in expressions. Music is considered as social action, and this encompasses both the music itself and the musical experience in which it is embedded. In social semiotics making meaning is an activity, and technological practices form an integral part of this. The final chapter therefore interrogates matters of agency in relation to the use of amplification and its use in musical performances

Tuning to Trust: System Calibration as Creative Enabler

This paper presents a mixed-music composition methodology emerging from the author’s latest practice-based research in the field over the last five years. The calibration of the interactive performance systems has enabled trust in reproducible sound quality for both the composer and the performers, enhancing the portability and adaptability of the works, and permitting increasingly daring creative experiments without compromising the rehearsal and concert experiences. A set of general, transferable responsibilities and solutions are presented and assessed against clear design criteria in the author’s latest pieces

Modelling the cocktail party : a binaural model for speech intelligibility in noise

We often listen to speech in an imperfect environment, with noise and reverberation; there will be voices around us, in complex acoustics. In this “cocktail-party” situation (Cherry 1953) listeners are helped by two binaural processes to segregate the desired voice from the competing noise: Better-Ear listening (BE) and Binaural Unmasking (BU). The aim of this thesis was to develop a model capable of efficiently predicting the benefits of BE and BU from Binaural Room Impulse Responses (BRIR). The developed model is a computationally efficient version of that created by Lavandier & Culling (2010) that predicts speech reception thresholds which include the benefit of binaural-unmasking, as explained by Equalization-Cancellation theory (Durlach 1963, 1972), and the benefit of better-ear listening, through Target-to-Interferer ratio analysis. The model accurately predicted a number of appropriate data sets from the literature that measure speech reception thresholds as a function of target and interferer source locations. Application of the model to a number of novel situations allowed environmental factors affecting intelligibility to be predicted and explored. In most situations, the effect of reverberation is to reduce the level of BE and BU, except when the listener is close to the interfering source, but this is when the benefits are needed the most. Depending on the spatial separation and source distances, the inclusion of multiple interferers again reduces the benefits in the majority of situations. To examine the benefits of head orientation a number of configurations were tested, whilst rotating the listener relative to the sound field. Benefits exceeding 12 dB can be achieved through modest rotations, particularly showing the benefits of BE. According to the model, the current literature on the benefits of bilateral cochlear implantation has underestimated that benefit by employing sub-optimal spatial configurations; using optimum orientations the model predicts an extra 6 dB of benefit being available to the listener. In a simulated restaurant situation, the model predicts that orientation of a table can affect the ability of a listener by up to 5 dB.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

The Biometric Evolution of Sound and Space

Auditoria in the late 20th and 21st centuries have evolved into a series of spatial conventions that are an established and accepted norm. The relationship between space and music now exists in a decoupled condition, and music is no longer reliant on volumetric and material conditions to define its form (Glantz 2000). This thesis looks at a series of novel approaches to investigate how the links between music and space can be reconnected though evolutionary computation, parametric modelling, virtual acoustics and biometric sensing. The thesis describes in detail the experiments undertaken in developing methodologies in linking music, space and the body. The thesis will show how it is possible to develop new form finding and musical generation tools that allow new room shapes and acoustic measures to inform how new acoustic and musical forms can be developed unconsciously and objectively by a listener, in response to sound and site

Application of sound source separation methods to advanced spatial audio systems

This thesis is related to the field of Sound Source Separation (SSS). It addresses the development and evaluation of these techniques for their application in the resynthesis of high-realism sound scenes by means of Wave Field Synthesis (WFS). Because the vast majority of audio recordings are preserved in twochannel stereo format, special up-converters are required to use advanced spatial audio reproduction formats, such as WFS. This is due to the fact that WFS needs the original source signals to be available, in order to accurately synthesize the acoustic field inside an extended listening area. Thus, an object-based mixing is required. Source separation problems in digital signal processing are those in which several signals have been mixed together and the objective is to find out what the original signals were. Therefore, SSS algorithms can be applied to existing two-channel mixtures to extract the different objects that compose the stereo scene. Unfortunately, most stereo mixtures are underdetermined, i.e., there are more sound sources than audio channels. This condition makes the SSS problem especially difficult and stronger assumptions have to be taken, often related to the sparsity of the sources under some signal transformation. This thesis is focused on the application of SSS techniques to the spatial sound reproduction field. As a result, its contributions can be categorized within these two areas. First, two underdetermined SSS methods are proposed to deal efficiently with the separation of stereo sound mixtures. These techniques are based on a multi-level thresholding segmentation approach, which enables to perform a fast and unsupervised separation of sound sources in the time-frequency domain. Although both techniques rely on the same clustering type, the features considered by each of them are related to different localization cues that enable to perform separation of either instantaneous or real mixtures.Additionally, two post-processing techniques aimed at improving the isolation of the separated sources are proposed. The performance achieved by several SSS methods in the resynthesis of WFS sound scenes is afterwards evaluated by means of listening tests, paying special attention to the change observed in the perceived spatial attributes. Although the estimated sources are distorted versions of the original ones, the masking effects involved in their spatial remixing make artifacts less perceptible, which improves the overall assessed quality. Finally, some novel developments related to the application of time-frequency processing to source localization and enhanced sound reproduction are presented.Cobos Serrano, M. (2009). Application of sound source separation methods to advanced spatial audio systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/8969Palanci

Modelling the cocktail party: a binaural model for speech intelligibility in noise

We often listen to speech in an imperfect environment, with noise and reverberation; there will be voices around us, in complex acoustics. In this “cocktail-party” situation (Cherry 1953) listeners are helped by two binaural processes to segregate the desired voice from the competing noise: Better-Ear listening (BE) and Binaural Unmasking (BU). The aim of this thesis was to develop a model capable of efficiently predicting the benefits of BE and BU from Binaural Room Impulse Responses (BRIR). The developed model is a computationally efficient version of that created by Lavandier & Culling (2010) that predicts speech reception thresholds which include the benefit of binaural-unmasking, as explained by Equalization-Cancellation theory (Durlach 1963, 1972), and the benefit of better-ear listening, through Target-to-Interferer ratio analysis. The model accurately predicted a number of appropriate data sets from the literature that measure speech reception thresholds as a function of target and interferer source locations. Application of the model to a number of novel situations allowed environmental factors affecting intelligibility to be predicted and explored. In most situations, the effect of reverberation is to reduce the level of BE and BU, except when the listener is close to the interfering source, but this is when the benefits are needed the most. Depending on the spatial separation and source distances, the inclusion of multiple interferers again reduces the benefits in the majority of situations. To examine the benefits of head orientation a number of configurations were tested, whilst rotating the listener relative to the sound field. Benefits exceeding 12 dB can be achieved through modest rotations, particularly showing the benefits of BE. According to the model, the current literature on the benefits of bilateral cochlear implantation has underestimated that benefit by employing sub-optimal spatial configurations; using optimum orientations the model predicts an extra 6 dB of benefit being available to the listener. In a simulated restaurant situation, the model predicts that orientation of a table can affect the ability of a listener by up to 5 dB