Technical aspects of a demonstration tape for three-dimensional sound displays

This document was developed to accompany an audio cassette that demonstrates work in three-dimensional auditory displays, developed at the Ames Research Center Aerospace Human Factors Division. It provides a text version of the audio material, and covers the theoretical and technical issues of spatial auditory displays in greater depth than on the cassette. The technical procedures used in the production of the audio demonstration are documented, including the methods for simulating rotorcraft radio communication, synthesizing auditory icons, and using the Convolvotron, a real-time spatialization device

Synthetic Aperture Radar (SAR) data processing

The available and optimal methods for generating SAR imagery for NASA applications were identified. The SAR image quality and data processing requirements associated with these applications were studied. Mathematical operations and algorithms required to process sensor data into SAR imagery were defined. The architecture of SAR image formation processors was discussed, and technology necessary to implement the SAR data processors used in both general purpose and dedicated imaging systems was addressed

Measurement and modelling of head-related transfer function for spatial audio synthesis

There has been a growing interest in spatial sound generation arising from the development of new communications and media technologies. Binaural spatial sound systems are capable of encoding and rendering sound sources accurately in three dimensional space using only two recording/playback channels. This is based on the concept of the Head-Related Transfer Function (HRTF), which is a set of acoustic filters from the sound source to a listener's eardrums and contains all the listening cues used by the hearing mechanism for decoding spatial information encoded in binaural signals. The HRTF is usually obtained from acoustic measurements on different persons. In the case of discrete data and sets of measurements corresponding to different human subjects, it is desirable to have a continuous functional representation of the HRTF for efficiently rendering moving sounds in the virtual spatial audio systems; further this representation should be well-suited for customization to an individual listener. In this thesis, modal analysis is applied to examine the HRTF data structure, that is to employ the wave equation solutions to expand the HRTF with separable basis functions. This leads to a general representation of the HRTF into separated spatial and spectral components, where the spatial basis functions modes account for the HRTF spatial variations and the remaining HRTF spectral components provide a new means to examine the human body scattering behavior. The general model is further developed into the HRTF continuous functional representations. We use the normalized spatial modes to link near-field and far-field HRTFs directly, which provides a way to obtain the HRTFs at different ranges from measurements conducted at only a single range. The spatially invariant HRTF spectral components are represented continuously using an orthogonal series. Both spatial and spectral basis functions are well known functions, thus the developed analytical model can be used to easily examine the HRTF data feature-individualization. An important finding of this thesis is that the HRTF decomposition with the spatial basis functions can be well approximated by a finite number, which is defined as the HRTF spatial dimensionality. The dimensionality determines the least number of the HRTF measurements in space. We perform high resolution HRTF measurements on a KEMAR mannequin in a semi-anechoic acoustic chamber. Both signal processing aspects to extract HRTFs from the raw measurements and a practical high resolution spatial sampling scheme have been given in this thesis

Spatial auditory display for acoustics and music collections

PhDThis thesis explores how audio can be better incorporated into how people access information and does so by developing approaches for creating three-dimensional audio environments with low processing demands. This is done by investigating three research questions. Mobile applications have processor and memory requirements that restrict the number of concurrent static or moving sound sources that can be rendered with binaural audio. Is there a more e cient approach that is as perceptually accurate as the traditional method? This thesis concludes that virtual Ambisonics is an ef cient and accurate means to render a binaural auditory display consisting of noise signals placed on the horizontal plane without head tracking. Virtual Ambisonics is then more e cient than convolution of HRTFs if more than two sound sources are concurrently rendered or if movement of the sources or head tracking is implemented. Complex acoustics models require signi cant amounts of memory and processing. If the memory and processor loads for a model are too large for a particular device, that model cannot be interactive in real-time. What steps can be taken to allow a complex room model to be interactive by using less memory and decreasing the computational load? This thesis presents a new reverberation model based on hybrid reverberation which uses a collection of B-format IRs. A new metric for determining the mixing time of a room is developed and interpolation between early re ections is investigated. Though hybrid reverberation typically uses a recursive lter such as a FDN for the late reverberation, an average late reverberation tail is instead synthesised for convolution reverberation. Commercial interfaces for music search and discovery use little aural information even though the information being sought is audio. How can audio be used in interfaces for music search and discovery? This thesis looks at 20 interfaces and determines that several themes emerge from past interfaces. These include using a two or three-dimensional space to explore a music collection, allowing concurrent playback of multiple sources, and tools such as auras to control how much information is presented. A new interface, the amblr, is developed because virtual two-dimensional spaces populated by music have been a common approach, but not yet a perfected one. The amblr is also interpreted as an art installation which was visited by approximately 1000 people over 5 days. The installation maps the virtual space created by the amblr to a physical space

Nearfield Acoustic Holography using sparsity and compressive sampling principles

Regularization of the inverse problem is a complex issue when using Near-field Acoustic Holography (NAH) techniques to identify the vibrating sources. This paper shows that, for convex homogeneous plates with arbitrary boundary conditions, new regularization schemes can be developed, based on the sparsity of the normal velocity of the plate in a well-designed basis, i.e. the possibility to approximate it as a weighted sum of few elementary basis functions. In particular, these new techniques can handle discontinuities of the velocity field at the boundaries, which can be problematic with standard techniques. This comes at the cost of a higher computational complexity to solve the associated optimization problem, though it remains easily tractable with out-of-the-box software. Furthermore, this sparsity framework allows us to take advantage of the concept of Compressive Sampling: under some conditions on the sampling process (here, the design of a random array, which can be numerically and experimentally validated), it is possible to reconstruct the sparse signals with significantly less measurements (i.e., microphones) than classically required. After introducing the different concepts, this paper presents numerical and experimental results of NAH with two plate geometries, and compares the advantages and limitations of these sparsity-based techniques over standard Tikhonov regularization.Comment: Journal of the Acoustical Society of America (2012

Proceedings of the EAA Spatial Audio Signal Processing symposium: SASP 2019

International audienc

Development of a dynamic underwater acoustic communication channel simulator with configurable sea surface parameters to explore time-varying signal distortion

A wide-band phase-coherent multi-path underwater acoustic channel simulation is developed using an approximate quantitative model of the acoustic wave response to a time-varying three-dimensional rough surface. It has been demonstrated over transmission ranges from 100 m to 8 km by experimental channel probing and comparable synthetic replication of the channel probing through the simulated channel, that the simulation is capable of reproducing fine-time-scale Doppler and delay distortions consistent with those generated in real shallow channels

Acoustic Performance of a Real-Time Three-Dimensional Sound-Reproduction System

The Exterior Effects Room (EER) is a 39-seat auditorium at the NASA Langley Research Center and was built to support psychoacoustic studies of aircraft community noise. The EER has a real-time simulation environment which includes a three-dimensional sound-reproduction system. This system requires real-time application of equalization filters to compensate for spectral coloration of the sound reproduction due to installation and room effects. This paper describes the efforts taken to develop the equalization filters for use in the real-time sound-reproduction system and the subsequent analysis of the system s acoustic performance. The acoustic performance of the compensated and uncompensated sound-reproduction system is assessed for its crossover performance, its performance under stationary and dynamic conditions, the maximum spatialized sound pressure level it can produce from a single virtual source, and for the spatial uniformity of a generated sound field. Additionally, application examples are given to illustrate the compensated sound-reproduction system performance using recorded aircraft flyover