• Extending realities: creativity, artistry and technology

      Wilson, Chris; Brown, Michael; University of Derby (KIE, 2013-09-10)
    • The generation of panning laws for irregular speaker arrays using heuristic methods

      Wiggins, Bruce; University of Derby (Audio Engineering Society, 2007-06-25)
      Currently, the ITU standard surround sound speaker arrangement is based on an irregular 5 speaker array. However, this may change to an irregular 7 speaker array (as is now the standard on computer hardware) or more in the future. The Ambisonic system, pioneered by Micheal Gerzon, among others, in the late 1960’s, is very well suited to situations where the end system speaker configuration is not fixed in terms of number or position while also offering a simple way (via energy and velocity vector analysis) of quantifying the performance of such systems. However, while the derivation of the decoders is well documented for regular speaker arrangements [1], optimising the decoders for irregular layouts is not a simple task, where optimisation requires the solution of a set of non linear simultaneous equations, complicated further by the fact that multiple solutions are possible [2]. Craven [3] extended the system to use higher order circular harmonics and presented a 4th order Ambisonic decoder (9 input channels), although the derivation method used was not presented. In this paper a semi-automated decoder optimisation system using heuristic methods will be presented that will be shown to be robust enough to generate higher order Ambisonic decoders based on the energy and velocity vector parameters. This method is then analytically compared to Craven’s decoder using both energy/velocity vector and head related transfer function based methods.
    • Towards a generalized theory of low-frequency sound source localization

      Hill, Adam J.; Lewis, Simon P.; Hawksford, Malcolm O. J.; University of Derby; University of Essex (Institute of Acoustics, 2012-11)
      Low-frequency sound source localization generates considerable amount of disagreement between audio/acoustics researchers, with some arguing that below a certain frequency humans cannot localize a source with others insisting that in certain cases localization is possible, even down to the lowest audible of frequencies. Nearly all previous work in this area depends on subjective evaluations to formulate theorems for low-frequency localization. This, of course, opens the argument of data reliability, a critical factor that may go some way to explain the reported ambiguities with regard to low-frequency localization. The resulting proposal stipulates that low-frequency source localization is highly dependent on room dimensions, source/listener location and absorptive properties. In some cases, a source can be accurately localized down to the lowest audible of frequencies, while in other situations it cannot. This is relevant as the standard procedure in live sound reinforcement, cinema sound and home-theater surround sound is to have a single mono channel for the low-frequency content, based on the assumption that human’s cannot determine direction in this band. This work takes the first steps towards showing that this may not be a universally valid simplification and that certain sound reproduction systems may actually benefit from directional low-frequency content.
    • Low-frequency temporal accuracy of small-room sound reproduction

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Derby; University of Essex (Audio Engineering Society, 2012-10)
      Small-room sound reproduction is strongly affected by room-modes in the low-frequency band. While the spectral impact of room-modes is well understood, there is less information on how modes degrade the spatiotemporal response of a sound reproduction system. This topic is investigated using a bespoke finite-difference time-domain (FDTD) simulation toolbox to virtually test common subwoofer configurations using tone bursts to judge waveform fidelity over a wide listening area. Temporal accuracy is compared to the steady-state frequency response to determine any link between the two domains. The simulated results are compared to practical measurements for validation.
    • Analysis, modeling and wide-area spatiotemporal control of low-frequency sound reproduction

      Hill, Adam J.; University of Essex (University of Essex, 2012-01)
      This research aims to develop a low-frequency response control methodology capable of delivering a consistent spectral and temporal response over a wide listening area. Low-frequency room acoustics are naturally plagued by room-modes, a result of standing waves at frequencies with wavelengths that are integer multiples of one or more room dimension. The standing wave pattern is different for each modal frequency, causing a complicated sound field exhibiting a highly position-dependent frequency response. Enhanced systems are investigated with multiple degrees of freedom (independently-controllable sound radiating sources) to provide adequate low-frequency response control. The proposed solution, termed a chameleon subwoofer array or CSA, adopts the most advantageous aspects of existing room-mode correction methodologies while emphasizing efficiency and practicality. Multiple degrees of freedom are ideally achieved by employing what is designated a hybrid subwoofer, which provides four orthogonal degrees of freedom configured within a modest-sized enclosure. The CSA software algorithm integrates both objective and subjective measures to address listener preferences including the possibility of individual real-time control. CSAs and existing techniques are evaluated within a novel acoustical modeling system (FDTD simulation toolbox) developed to meet the requirements of this research. Extensive virtual development of CSAs has led to experimentation using a prototype hybrid subwoofer. The resulting performance is in line with the simulations, whereby variance across a wide listening area is reduced by over 50% with only four degrees of freedom. A supplemental novel correction algorithm addresses correction issues at select narrow frequency bands. These frequencies are filtered from the signal and replaced using virtual bass to maintain all aural information, a psychoacoustical effect giving the impression of low-frequency. Virtual bass is synthesized using an original hybrid approach combining two mainstream synthesis procedures while suppressing each method‟s inherent weaknesses. This algorithm is demonstrated to improve CSA output efficiency while maintaining acceptable subjective performance.
    • Chameleon subwoofer arrays in live sound

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (Institute of Acoustics, 2011-06)
      Live-sound subwoofer systems should deliver low-frequency sound evenly distributed throughout the audience area while simultaneously minimizing sound pressure levels on stage. Approximate solutions generally exploit cardioid subwoofers and/or steerable subwoofer clusters, yet require venue-specific manual fine tuning limited mainly by practical positioning issues. Enhanced live-sound systems are explored using a virtual three-dimensional acoustic space to model dominant venue characteristics. Specifically the Chameleon Subwoofer Array (CSA) is incorporated, already proposed as a solution to small-room low-frequency sound reproduction by extending the available degrees of freedom to control sound distribution in the target space. The CSA is adapted and scaled to match the large-scale dimensions typical of live events with 3-D simulation used to optimize and validate performance. Adaptation of existing industry-standard equipment with only minor modification is presented as a core feature.
    • Visualization and analysis tools for low-frequency propagation in a generalized 3D acoustic space

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (Audio Engineering Society, 2011-05)
      A software toolbox is described that enables three-dimensional animated visualization and analysis of low-frequency wave propagation within a generalized acoustic environment. The core computation exploits a finite-difference time-domain (FDTD) algorithm selected because of its known low-frequency accuracy. Multiple sources can be configured and analyses performed at user-selected measurement locations. Arbitrary excitation sequences enable virtual measurements embracing both time-domain and spatio-frequency-domain analyses. Examples are presented for a variety of low-frequency loudspeaker placements and room geometries to illustrate the utility of the toolbox for various acoustical design challenges.
    • A hybrid virtual bass system for optimized steady-state and transient performance

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (IEEE, 2010-09)
      Bandwidth extension of a constrained loudspeaker system is regularly achieved employing nonlinear bass synthesis. The method operates on the doctrine of the missing fundamental whereby humans infer the presence of a fundamental tone when presented with a signal consisting of higher harmonics of said tone. Nonlinear devices and phase vocoders are commonly used for signal generation; both exhibiting deficiencies. A system is proposed where the two approaches are used in tandem via a mixing algorithm to suppress these deficiencies. Mixing is performed by signal transient content analysis in the frequency domain using constant-Q transforms. The hybrid approach is rated subjectively against various nonlinear device and phase vocoder techniques using the MUSHRA test method.
    • Kick-Drum signal acquisition, isolation and reinforcement optimization in live sound

      Hill, Adam J.; Hawksford, Malcolm O. J.; Rosenthal, Adam P.; Gand, Gary; University of Essex; Gand Concert Sound (Audio Engineering Society, 2011-05)
      A critical requirement for popular music in live-sound applications is the achievement of a robust kick-drum sound presented to the audience and the drummer while simultaneously achieving a workable degree of acoustic isolation for other on-stage musicians. Routinely a transparent wall is placed in parallel to the kick-drum heads to attenuate sound from the drummer’s monitor loudspeakers, although this can cause sound quality impairment from comb filter interference. Practical optimization techniques are explored, embracing microphone selection and placement (including multiple microphones in combination), isolation-wall location, drum-monitor electronic delay and echo cancellation. A system analysis is presented augmented by real-world measurements and relevant simulations using a bespoke Finite-Difference Time-Domain (FDTD) algorithm.
    • Individualized low-frequency response manipulation for multiple listeners using chameleon subwoofer arrays

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (IEEE, 2011-07)
      Low-frequency acoustical responses are naturally position dependent across wide listening areas. This is predominantly due to room modes in small, closed spaces. Numerous methodologies have been proposed targeting room mode compensation to give an objectively even response across all listening locations. These techniques cannot guarantee, however, that every listener receives an equally pleasing subjective response. Chameleon subwoofer arrays (CSA) were originally developed to minimize low-frequency spatiotemporal variations by addressing frequency response errors at multiple listening locations using a subwoofer system consisting of multiple degrees of freedom. The CSA system can alternatively be utilized to control listening locations independently, allowing each listener to adjust their localized low-frequency response to their liking. This alternate CSA implementation is evaluated using a bespoke finite-difference time-domain (FDTD) algorithm for small home theater applications.
    • Practical applications of chameleon subwoofer arrays

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (Audio Engineering Society, 2012-04)
      Spatiotemporal variations of the low-frequency response in a closed-space are predominantly caused by room-modes. Chameleon subwoofer arrays (CSA) were developed to minimize this variance over a listening area using multiple independently-controllable source components and calibrated with one-time measurements. Although CSAs are ideally implemented using hybrid (multiple source component) subwoofers, they can alternatively be realized using conventional subwoofers. This capability is exploited in this work where various CSA configurations are tested using commercially-available subwoofers in a small-sized listening room. Spectral and temporal evaluation is performed using tone-burst and maximum length sequence (MLS) measurements. The systems are implemented with practicality in mind, keeping the number of subwoofers and calibration measurements to a minimum while maintaining correction benefits.
    • Wide-area psychoacoustic correction for problematic room-modes using nonlinear bass synthesis

      Hill, Adam J.; Hawksford, Malcolm O. J.; University of Essex (Audio Engineering Society, 2011-11)
      Small room acoustics are characterized by a limited number of dominant low-frequency room-modes that result in wide spatio-pressure variations that traditional room correction systems may find elusive to correct over a broad listening area. A psychoacoustic-based methodology is proposed whereby signal components coincident only with problematic modes are filtered and substituted by virtual bass components to forge an illusion of the suppressed frequencies. Although this approach can constitute a standalone correction system, the impetus for development is for use within well-established correction methodologies. A scalable and hierarchical approach is studied using subjective evaluation to confirm uniform wide-area performance. Bass synthesis exploits parallel nonlinear and phase vocoder generators with outputs blended as a function of transient and steady-state signal content.
    • Sound, space, image and music: hybridity in creative process through technology, interactivity and collaboration

      Wilson, Chris; Brown, Michael; University of Derby (Intellect, 2012-05-22)
      This article explores the dynamic interaction of sound and image creativity through technology. Focusing on the potential significance of the blurring of boundaries between the visual and the auditory in artistic perception and creative procedure, and more fluid approaches to collaboration and artistic interaction through technology and virtual environments, issues are explored through the development and exhibition of original artwork. Developing mixed-media outcomes, reflections of particular aspects of human interaction with physical spaces emerged as a persistent theme in collaborative work, technology providing an adaptable mechanism and medium of craft, as well as an influence on perspective and artistic perception. This article develops a contextual evaluation of the project, whilst focusing on the implications of technology for artistic practice and higher education. With an emphasis on the development and understanding of musical creativity, the virtualization of compositional process and potential for enrichment of pedagogy and artistry are considered.
    • An investigation into the real-time manipulation and control of three-dimensional sound fields

      Wiggins, Bruce; University of Derby (University of Derby, 2004)
      This thesis describes a system that can be used for the decoding of a three dimensional audio recording over headphones or two, or more, speakers. A literature review of psychoacoustics and a review (both historical and current) of surround sound systems is carried out. The need for a system which is platform independent is discussed, and the proposal for a system based on an amalgamation of Ambisonics, binaural and transaural reproduction schemes is given. In order for this system to function optimally, each of the three systems rely on providing the listener with the relevant psychoacoustic cues. The conversion from a five speaker ITU array to binaural decode is well documented but pair-wise panning algorithms will not produce the correct lateralisation parameters at the ears of a centrally seated listener. Although Ambisonics has been well researched, no one has, as yet, produced a psychoacoustically optimised decoder for the standard irregular five speaker array as specified by the ITU as the original theory, as proposed by Gerzon and Barton (1992) was produced (known as a Vienna decoder), and example solutions given, before the standard had been decided on. In this work, the original work by Gerzon and Barton (1992) is analysed, and shown to be suboptimal, showing a high/low frequency decoder mismatch due to the method of solving the set of non-linear simultaneous equations. A method, based on the Tabu search algorithm, is applied to the Vienna decoder problem and is shown to provide superior results to those shown by Gerzon and Barton (1992) and is capable of producing multiple solutions to the Vienna decoder problem. During the write up of this report Craven (2003) has shown how 4th order circular harmonics (as used in Ambisonics) can be used to create a frequency independent panning law for the five speaker ITU array, and this report also shows how the Tabu search algorithm can be used to optimise these decoders further. A new method is then demonstrated using the Tabu search algorithm coupled with lateralisation parameters extracted from a binaural simulation of the Ambisonic system to be optimised (as these are the parameters that the Vienna system is approximating). This method can then be altered to take into account head rotations directly which have been shown as an important psychoacoustic parameter in the localisation of a sound source (Spikofski et al., 2001) and is also shown to be useful in differentiating between decoders optimised using the Tabu search form of the Vienna optimisations as no objective measure had been suggested. Optimisations for both Binaural and Transaural reproductions are then discussed so as to maximise the performance of generic HRTF data (i.e. not individualised) using inverse filtering methods, and a technique is shown that minimises the amount of frequency dependant regularisation needed when calculating cross-talk cancellation filters.
    • Distance coding and performance of the mark 5 and st350 SoundField microphones and their suitability for Ambisonic reproduction.

      Wiggins, Bruce; Spenceley, Thomas; University of Derby (Institute of Acoustics, 2009-11-19)
      Capturing and replaying distance cues for multi-channel audio is currently an under-explored and under-exploited area. Panners that successfully give control of distance do not, currently exist. However, recordings made with 1st order ambisonic, Soundfield microphones replayed over an ambisonic rig can give realistic results with respect to distance perception (particularly when bringing sound sources inside the speaker array). Near-field effect, resulting from the wave front curvature of near-field sources, is one cue recorded by the microphone, but not reproduced by software or hardware panners. Papers by Daniel (2003, 2004) discuss the encoding and decoding of ambisonic material with particular reference to higher-order ambisonics, and describe ‘near-field coding’ filters which encode near-field effect while pre-compensating for finite loudspeaker reproduction distance. While existing research concentrates on its simulation, this report documents an investigation into near-field effect in Soundfield ST350 and MK V tetrahedral microphones. It is found that, as a result of calibration for a flat frequency response at a practical source distance, the Soundfield microphone responses bear strong similarity to various near-field coding filters, suggesting the existence of an optimum loudspeaker array radius for positional localisation. On determination of this distance, recordings may be adapted for proper reproduction at any chosen reference distance using the WigWare ambisonic plug-ins created at the University of Derby.