Computation of spherical harmonic representations of source directivity based on the finite-distance signature

The measurement of directivity for sound sources that are not electroacoustic transducers is fundamentally limited because the source cannot be driven with arbitrary signals. A consequence is that directivity can only be measured at a sparse set of frequencies—for example, at the stable partial oscillations of a steady tone played by a musical instrument or from the human voice. This limitation prevents the data from being used in certain applications such as time-domain room acoustic simulations where the directivity needs to be available at all frequencies in the frequency range of interest. We demonstrate in this article that imposing the signature of the directivity that is obtained at a given distance on a spherical wave allows for all interpolation that is required for obtaining a complete spherical harmonic representation of the source’s directivity, i.e., a representation that is viable at any frequency, in any direction, and at any distance. Our approach is inspired by the far-field signature of exterior sound fields. It is not capable of incorporating the phase of the directivity directly. We argue based on directivity measurement data of musical instruments that the phase of such measurement data is too unreliable or too ambiguous to be useful. We incorporate numerically-derived directivity into the example application of finite difference time domain simulation of the acoustic field, which has not been possible previously

Incorporating source directivity in wave-based virtual acoustics:Time-domain models and fitting to measured data

The modeling of source directivity is a problem of longstanding interest in virtual acoustics and auralisation. This remains the case for newer time domain volumetric wave-based approaches to simulation such as the finite difference time domain method. In this article, a spatio-temporal model of acoustic wave propagation, including a source term is presented. The source is modeled as a spatial Dirac delta function under the action of a series of differential operators associated with the spherical harmonic functions. Each term in the series gives rise to the directivity pattern of a given spherical harmonic, and is separately driven through a time domain filtering operation of an underlying source signal. Such a model is suitable for calibration against measured frequency-dependent directivity patterns and a procedure for arriving at time domain filters for each spherical harmonic channel is illustrated. It also yields a convenient framework for discretisation, and a simple strategy is presented, yielding a locally-defined operation over the spatial grid. Numerical results, illustrating various features of source directivity, including the comparison of measured and synthetic directivity patterns, are presented

Efficient Techniques for Wave-based Sound Propagation in Interactive Applications

Sound propagation techniques model the effect of the environment on sound waves and predict their behavior from point of emission at the source to the final point of arrival at the listener. Sound is a pressure wave produced by mechanical vibration of a surface that propagates through a medium such as air or water, and the problem of sound propagation can be formulated mathematically as a second-order partial differential equation called the wave equation. Accurate techniques based on solving the wave equation, also called the wave-based techniques, are too expensive computationally and memory-wise. Therefore, these techniques face many challenges in terms of their applicability in interactive applications including sound propagation in large environments, time-varying source and listener directivity, and high simulation cost for mid-frequencies. In this dissertation, we propose a set of efficient wave-based sound propagation techniques that solve these three challenges and enable the use of wave-based sound propagation in interactive applications. Firstly, we propose a novel equivalent source technique for interactive wave-based sound propagation in large scenes spanning hundreds of meters. It is based on the equivalent source theory used for solving radiation and scattering problems in acoustics and electromagnetics. Instead of using a volumetric or surface-based approach, this technique takes an object-centric approach to sound propagation. The proposed equivalent source technique generates realistic acoustic effects and takes orders of magnitude less runtime memory compared to prior wave-based techniques. Secondly, we present an efficient framework for handling time-varying source and listener directivity for interactive wave-based sound propagation. The source directivity is represented as a linear combination of elementary spherical harmonic sources. This spherical harmonic-based representation of source directivity can support analytical, data-driven, rotating or time-varying directivity function at runtime. Unlike previous approaches, the listener directivity approach can be used to compute spatial audio (3D audio) for a moving, rotating listener at interactive rates. Lastly, we propose an efficient GPU-based time-domain solver for the wave equation that enables wave simulation up to the mid-frequency range in tens of minutes on a desktop computer. It is demonstrated that by carefully mapping all the components of the wave simulator to match the parallel processing capabilities of the graphics processors, significant improvement in performance can be achieved compared to the CPU-based simulators, while maintaining numerical accuracy. We validate these techniques with offline numerical simulations and measured data recorded in an outdoor scene. We present results of preliminary user evaluations conducted to study the impact of these techniques on user's immersion in virtual environment. We have integrated these techniques with the Half-Life 2 game engine, Oculus Rift head-mounted display, and Xbox game controller to enable users to experience high-quality acoustics effects and spatial audio in the virtual environment.Doctor of Philosoph

Perceptually Driven Interactive Sound Propagation for Virtual Environments

Sound simulation and rendering can significantly augment a user‘s sense of presence in virtual environments. Many techniques for sound propagation have been proposed that predict the behavior of sound as it interacts with the environment and is received by the user. At a broad level, the propagation algorithms can be classified into reverberation filters, geometric methods, and wave-based methods. In practice, heuristic methods based on reverberation filters are simple to implement and have a low computational overhead, while wave-based algorithms are limited to static scenes and involve extensive precomputation. However, relatively little work has been done on the psychoacoustic characterization of different propagation algorithms, and evaluating the relationship between scientific accuracy and perceptual benefits.In this dissertation, we present perceptual evaluations of sound propagation methods and their ability to model complex acoustic effects for virtual environments. Our results indicate that scientifically accurate methods for reverberation and diffraction do result in increased perceptual differentiation. Based on these evaluations, we present two novel hybrid sound propagation methods that combine the accuracy of wave-based methods with the speed of geometric methods for interactive sound propagation in dynamic scenes.Our first algorithm couples modal sound synthesis with geometric sound propagation using wave-based sound radiation to perform mode-aware sound propagation. We introduce diffraction kernels of rigid objects,which encapsulate the sound diffraction behaviors of individual objects in the free space and are then used to simulate plausible diffraction effects using an interactive path tracing algorithm. Finally, we present a novel perceptual driven metric that can be used to accelerate the computation of late reverberation to enable plausible simulation of reverberation with a low runtime overhead. We highlight the benefits of our novel propagation algorithms in different scenarios.Doctor of Philosoph

Perceptual Evaluation of Spatial Room Impulse Response Extrapolation by Direct and Residual Subspace Decomposition

Six-degrees-of-freedom rendering of an acoustic environment can be achieved by interpolating a set of measured spatial room impulse responses (SRIRs). However, the involved measurement effort and computational expense are high. This work compares novel ways of extrapolating a single measured SRIR to a target position. The novel extrapolation techniques are based on a recently proposed subspace method that decomposes SRIRs into a direct part, comprising direct sound and salient reflections, and a residual. We evaluate extrapolations between different positions in a shoebox-shaped room in a multi-stimulus comparison test. Extrapolation using a residual SRIR and salient reflections that match the reflections at the target position is rated as perceptually most similar to the measured reference

Perceptual Evaluation of Spatial Room Impulse Response Extrapolation by Direct and Residual Subspace Decomposition

Six-degrees-of-freedom rendering of an acoustic environment can be achieved by interpolating a set of measured spatial room impulse responses (SRIRs). However, the involved measurement effort and computational expense are high. This work compares novel ways of extrapolating a single measured SRIR to a target position. The novel extrapolation techniques are based on a recently proposed subspace method that decomposes SRIRs into a direct part, comprising direct sound and salient reflections, and a residual. We evaluate extrapolations between different positions in a shoebox-shaped room in a multi-stimulus comparison test. Extrapolation using a residual SRIR and salient reflections that match the reflections at the target position is rated as perceptually most similar to the measured reference

The acoustics of concentric sources and receivers – human voice and hearing applications

One of the most common ways in which we experience environments acoustically is by listening to the reflections of our own voice in a space. By listening to our own voice we adjust its characteristics to suit the task and audience. This is of particular importance in critical voice tasks such as actors or singers on a stage with no additional electroacoustic or other amplification (e.g. in ear monitors, loudspeakers, etc.). Despite the usualness of this situation, there are very few acoustic measurements aimed to quantify it and even fewer that address the problem of having a source and receiver that are very closely located. The aim of this thesis is to introduce new measurement transducers and methods that quantify correctly this situation. This is achieved by analysing the characteristics of the human as a source, a receiver and their interaction in close proximity when placed in acoustical environments. The characteristics of the human voice and human ear are analysed in this thesis in a similar manner as a loudspeaker or microphone would be analysed. This provides the basis for further analysis by making them analogous to measurement transducers. These results are then used to explore the consequences of having a source and receiver very closely located using acoustic room simulation. Different techniques for processing data using directional transducers in real rooms are introduced. The majority of the data used in this thesis was obtained in rooms used for performance. The final chapters of this thesis include details of the design and construction of a concentric directional transducer, where an array of microphones and loudspeakers occupy the same structure. Finally, sample measurements with this transducer are presented

Ambisonics

This open access book provides a concise explanation of the fundamentals and background of the surround sound recording and playback technology Ambisonics. It equips readers with the psychoacoustical, signal processing, acoustical, and mathematical knowledge needed to understand the inner workings of modern processing utilities, special equipment for recording, manipulation, and reproduction in the higher-order Ambisonic format. The book comes with various practical examples based on free software tools and open scientific data for reproducible research. The book’s introductory section offers a perspective on Ambisonics spanning from the origins of coincident recordings in the 1930s to the Ambisonic concepts of the 1970s, as well as classical ways of applying Ambisonics in first-order coincident sound scene recording and reproduction that have been practiced since the 1980s. As, from time to time, the underlying mathematics become quite involved, but should be comprehensive without sacrificing readability, the book includes an extensive mathematical appendix. The book offers readers a deeper understanding of Ambisonic technologies, and will especially benefit scientists, audio-system and audio-recording engineers. In the advanced sections of the book, fundamentals and modern techniques as higher-order Ambisonic decoding, 3D audio effects, and higher-order recording are explained. Those techniques are shown to be suitable to supply audience areas ranging from studio-sized to hundreds of listeners, or headphone-based playback, regardless whether it is live, interactive, or studio-produced 3D audio material