Sound field synthesis for line source array applications in large-scale sound reinforcement

The thesis deals with optimized large-scale sound reinforcement using line source arrays. This is treated as a sound field synthesis problem. The synthesis of a virtual source via the line source array allows for audience adapted wavefront shaping. For practical array designs and setups this is affected by the deployed loudspeakers and their arrangement, its electronic control and spatial aliasing occurrence. The influence of these parameters is discussed with array signal processing revisiting the Wavefront Sculpture Technology and proposing Wave Field Synthesis as a suitable control method.Die Arbeit beschäftigt sich mit optimaler Beschallung großer Auditorien mit Line Source Arrays. Das Problem wird mit Schallfeldsynthese beschrieben. Die Synthese einer virtuellen Quelle mit einem Line Source Array ermöglicht eine für das Auditorium angepasste Wellenfront. In der Praxis wird dies beeinflusst von den verwendeten Lautsprechern, ihrer Anordnung, ihrer Ansteuerung und räumlichem Aliasing. Der Einfluss der Parameter wird mit Array-Signalverarbeitung diskutiert, wofür Wavefront Sculpture Technology aufgegriffen und Wellenfeldsynthese als Ansteuerungsmethode vorgeschlagen wird

Digital acoustics: processing wave fields in space and time using DSP tools

Systems with hundreds of microphones for acoustic field acquisition, or hundreds of loudspeakers for rendering, have been proposed and built. To analyze, design, and apply such systems requires a framework that allows us to leverage the vast set of tools available in digital signal processing in order to achieve intuitive and efficient algorithms. We thus propose a discrete space-time framework, grounded in classical acoustics, which addresses the discrete nature of the spatial and temporal sampling. In particular, a short-space/time Fourier transform is introduced, which is the natural extension of the localized or short-time Fourier transform. Processing in this intuitive domain allows us to easily devise algorithms for beam-forming, source separation, and multi-channel compression, among other useful tasks. The essential space band-limitedness of the Fourier spectrum is also used to solve the spatial equalization task required for sound field rendering in a region of interest. Examples of applications are show

Sound field planarity characterized by superdirective beamforming

The ability to replicate a plane wave represents an essential element of spatial sound field reproduction. In sound field synthesis, the desired field is often formulated as a plane wave and the error minimized; for other sound field control methods, the energy density or energy ratio is maximized. In all cases and further to the reproduction error, it is informative to characterize how planar the resultant sound field is. This paper presents a method for quantifying a region's acoustic planarity by superdirective beamforming with an array of microphones, which analyzes the azimuthal distribution of impinging waves and hence derives the planarity. Estimates are obtained for a variety of simulated sound field types, tested with respect to array orientation, wavenumber, and number of microphones. A range of microphone configurations is examined. Results are compared with delay-and-sum beamforming, which is equivalent to spatial Fourier decomposition. The superdirective beamformer provides better characterization of sound fields, and is effective with a moderate number of omni-directional microphones over a broad frequency range. Practical investigation of planarity estimation in real sound fields is needed to demonstrate its validity as a physical sound field evaluation measure. © 2013 Acoustical Society of America

Blind identification of Ambisonic reduced room impulse response

Recently proposed Generalized Time-domain Velocity Vector (GTVV) is a generalization of relative room impulse response in spherical harmonic (aka Ambisonic) domain that allows for blind estimation of early-echo parameters: the directions and relative delays of individual reflections. However, the derived closed-form expression of GTVV mandates few assumptions to hold, most important being that the impulse response of the reference signal needs to be a minimum-phase filter. In practice, the reference is obtained by spatial filtering towards the Direction-of-Arrival of the source, and the aforementioned condition is bounded by the performance of the applied beamformer (and thus, by the Ambisonic array order). In the present work, we suggest to circumvent this problem by properly modelling the GTVV time series, which permits not only to relax the initial assumptions, but also to extract the information therein is a more consistent and efficient manner, entering the realm of blind system identification. Experiments using measured room impulse responses confirm the effectiveness of the proposed approach.Comment: Submitte

The Acoustic Hologram and Particle Manipulation with Structured Acoustic Fields

This book shows how arbitrary acoustic wavefronts can be encoded in the thickness profile of a phase plate - the acoustic hologram. The workflow for design and implementation of these elements has been developed and is presented in this work along with examples in microparticle assembly, object propulsion and levitation in air. To complement these results, a fast thermographic measurement technique has been developed to scan and validate 3D ultrasound fields in a matter of seconds

Adaptive Optics Progress

For over four decades there has been continuous progress in adaptive optics technology, theory, and systems development. Recently there also has been an explosion of applications of adaptive optics throughout the fields of communications and medicine in addition to its original uses in astronomy and beam propagation. This volume is a compilation of research and tutorials from a variety of international authors with expertise in theory, engineering, and technology. Eight chapters include discussion of retinal imaging, solar astronomy, wavefront-sensorless adaptive optics systems, liquid crystal wavefront correctors, membrane deformable mirrors, digital adaptive optics, optical vortices, and coupled anisoplanatism

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

Efficient Hybrid Virtual Room Acoustic Modelling

This thesis investigates approaches to virtual room acoustic modelling and auralisation in order to a develop hybrid modelling solution that is capable of efficient and accurate simulation of enclosed sound propagation. Emphasis is placed on the advantages and disadvantages of state of the art numerical and geometric acoustic modelling methods. Numerical methods have been shown to preserve important sound wave characteristics such as diffraction and room modes, and are considered more accurate for low frequency acoustic modelling than geometric techniques which fail to preserve such wave effects. However, the implementation of numerical acoustic models inherently requires large computational effort compared to more efficient geometric techniques such as ray-tracing. This is particularly problematic for simulations of large-scale 3D acoustic environments and for high spatio-temporal sampling rates. A novel acoustic modelling solution is presented, which seeks to circumvent the disadvantageous computational requirements of 3D numerical models while producing a suitable approximation to low frequency sound behaviour. This modelling technique incorporates multiple planar cross-sectional 2D Finite Difference schemes that are utilised in combination to synthesise low frequency wave propagation throughout the target acoustic field. In this way a subset of prominent low frequency sound wave characteristics may be emulated with low computational cost compared to 3D numerical schemes. These low-frequency results can then be combined with the high-frequency output from efficient geometric simulations to create a hybrid model providing accurate broadband results at a relatively low computational cost. Investigation of room impulse response rendering for a series of theoretic and real spaces demonstrates advantages of this new hybrid acoustic modelling technique over purely ray-based methods in terms of low frequency accuracy, and over 3D numerical methods in terms of computational efficiency. Conclusions are drawn from objective measurements obtained from simulation results of the virtual models produced. Results demonstrate the applicability of the novel hybrid approach to the enhancement of purely ray-based room impulse response rendering by which a more realistic representation of low frequency wave phenomena may be simulated in an efficient manner, improving the theoretical accuracy of objective and audible results. This study contributes towards research and design of high-speed, interactive virtual acoustic simulations appropriate for industrial and creative virtual reality applications

非球形のアレイ形状を持つAmbisonicsの定式化

Tohoku University鈴木陽一課

Movements in Binaural Space: Issues in HRTF Interpolation and Reverberation, with applications to Computer Music

This thesis deals broadly with the topic of Binaural Audio. After reviewing the literature, a reappraisal of the minimum-phase plus linear delay model for HRTF representation and interpolation is offered. A rigorous analysis of threshold based phase unwrapping is also performed. The results and conclusions drawn from these analyses motivate the development of two novel methods for HRTF representation and interpolation. Empirical data is used directly in a Phase Truncation method. A Functional Model for phase is used in the second method based on the psychoacoustical nature of Interaural Time Differences. Both methods are validated; most significantly, both perform better than a minimum-phase method in subjective testing. The accurate, artefact-free dynamic source processing afforded by the above methods is harnessed in a binaural reverberation model, based on an early reflection image model and Feedback Delay Network diffuse field, with accurate interaural coherence. In turn, these flexible environmental processing algorithms are used in the development of a multi-channel binaural application, which allows the audition of multi-channel setups in headphones. Both source and listener are dynamic in this paradigm. A GUI is offered for intuitive use of the application. HRTF processing is thus re-evaluated and updated after a review of accepted practice. Novel solutions are presented and validated. Binaural reverberation is recognised as a crucial tool for convincing artificial spatialisation, and is developed on similar principles. Emphasis is placed on transparency of development practices, with the aim of wider dissemination and uptake of binaural technology