Elena Grassi

The bat head-related transfer function reveals binaural cues for sound localization in azimuth and elevation

Directional properties of the sound transformation at the ear of four intact echolocating bats, Eptesicus fuscus, were investigated via measurements of the head-related transfer function (HRTF). Contributions of external ear structures to directional features of the transfer functions were examined by remeasuring the HRTF in the absence of the pinna and tragus. The investigation mainly focused on the interactions between the spatial and the spectral features in the bat HRTF. The pinna provides gain and shapes these features over a large frequency band (20-90 kHz), and the tragus contributes gain and directionality at the high frequencies (60 to 90 kHz). Analysis of the spatial and spectral characteristics of the bat HRTF reveals that both interaural level differences (ILD) and monaural spectral features are subject to changes in sound source azimuth and elevation. Consequently, localization cues for horizontal and vertical components of the sound source location interact. Availability of multiple cues about sound source azimuth and elevation should enhance information to support reliable sound localization. These findings stress the importance of the acoustic information received at the two ears for sound localization of sonar target position in both azimuth and elevation.

Integrated system identification and PID controller tuning by frequency loop-shaping

A systematic design methodology for proportional-integral-derivative (PID) controllers is presented. Starting from data sets, a model of the system and its uncertainty bounds are obtained. The parameters of the controller are tuned by a convex optimization algorithm, minimizing a weighted difference between the actual loop transfer function and a target in an /spl Lscr//sub 2///spl Lscr//sub /spl infin// sense. The target selection is guided by the identified model and its uncertainty. The problem of disjoint data sets and/or different models for the same system is also addressed. The method has proved successful in numerous practical cases showing both expediency in controller design and implementation and improved performance over existing controllers.

Fast head-related transfer function measurement via reciprocity.

An efficient method for head-related transfer function (HRTF) measurement is presented. By applying the acoustical principle of reciprocity, one can swap the speaker and the microphone positions in the traditional (direct) HRTF measurement setup, that is, insert a microspeaker into the subjects ear and position several microphones around the subject, enabling simultaneous HRTF acquisition at all microphone positions. The setup used for reciprocal HRTF measurement is described, and the obtained HRTFs are compared with the analytical solution for a sound-hard sphere and with KEMAR manikin HRTF obtained by the direct method. The reciprocally measured sphere HRTF agrees well with the analytical solution. The reciprocally measured and the directly measured KEMAR HRTFs are not exactly identical but agree well in spectrum shape and feature positions. To evaluate if the observed differences are significant, an auditory localization model based on work by J. C. Middlebrooks [J. Acoust. Soc. Am. 92, 2607-2624 (1992)] was used to predict where a virtual sound source synthesized with the reciprocally measured HRTF would be localized if the directly measured HRTF were used for the localization. It was found that the predicted localization direction generally lies close to the measurement direction, indicating that the HRTFs obtained via the two methods are in good agreement.

Flexible layout and optimal cancellation of the orthonormality error for spherical microphone arrays

This paper describes an approach to achieving a flexible layout of microphones on the surface of a spherical microphone array for beamforming. Our approach achieves orthonormality of spherical harmonics to higher order for relatively distributed layouts. This gives great flexibility in microphone layout on the spherical surface. One direct advantage is that it makes it much easier to build a real world system, such as those with cable outlets and a mounting base, with minimal effects on the performance. Simulation results are presented.

Temperature-composition determination based on modeling of optical constants of III-V compound semiconductors measured by spectroscopic ellipsometry

A general procedure to fit optical constants, using a transfer function model with temperature-and/or-composition-dependent coefficients, is presented. The model is further inverted by a simple algorithm to retrieve temperature and composition information from optical measurements obtained by spectroscopic ellipsometry. The method was applied to fit: (1) the complex index of refraction of the system AlXGa1−xAs at 600 °C, for values of X between 0 and 1. (2) Two data bases of complex dielectric constants, for near-lattice-matched InGaAs and InAlAs, and around temperatures of 500 °C. The parameters of the model are determined with a least squares algorithm with recursive “whitening” of the error, which shows fast convergence to a near-optimal solution, even when handling a large number of parameters. The level of accuracy achieved makes this method an adequate sensor for temperature, composition, and thickness control during molecular beam epitaxy growth.

A spherical microphone array system for traffic scene analysis

This paper describes a practical spherical microphone array system for traffic auditory scene capture and analysis. Our system uses 60 microphones positioned on the rigid surface of a sphere. We then propose an optimal design of a robust spherical beamformer with minimum white noise gain (WNG) of -6 dB. We test this system in a real-world traffic environment. Some preliminary simulation and experimental results are presented to demonstrate its performance. This system may also find applications in broader areas such as 3D audio, virtual environment, etc.

Auditory models of suprathreshold distortion in persons with sensorineural hearing loss.

Hearing technology has advanced to where it is now reasonable to ask whether signal processing algorithms can be developed to compensate for an individual’s hearing loss, thus allowing them to hear functionally in a manner similar to persons with normal hearing. Clinically, the pure‐tone audiogram is the primary tool used to represent the patient’s hearing status. However, it has been well established, experimentally and theoretically, that the audiogram cannot reflect fully all aspects of the hearing loss, most notably that part which pertains to suprathreshold distortion. Much has been written about the distortion component of sensorineural hearing loss, yet there is little agreement on estimating its importance to speech recognition, nor much consensus on which hearing factors (e.g., spectral and/or temporal resolution) most accurately represent the distortion. Recent attempts to use biologically inspired models of auditory processing to represent a patient’s internal auditory experiences have shown pr...

Understanding the complex modulation spectrum for consonants and consonant features

Speech intelligibility is highly dependent on the magnitude and phase characteristics of the low‐frequency modulation spectrum. However, unlike more traditional representations of speech, such as the spectrogram, associating details of the modulation spectrum to specific phonemes and subphonemic units of speech has not been readily forthcoming. In the present study we used local time reversals of the speech waveform between 20‐160 ms to selectively distort portions of the complex modulation spectrum. Normal‐hearing subjects were tested on a consonant recognition task and a detailed analysis of the perceptual confusions was performed. Consistent with earlier results using sentence‐length materials, average consonant intelligibility declined as the length of the time reversal segment increased. Further analyses were conducted to determine the effect of time‐reversal segment duration on the amount of information transmitted for individual consonants (including specific consonant productions) and acoustic fea...

Spherical and hemispherical microphone arrays for capture and analysis of sound fields

The capture of the spatial structure of a sound field and analysis is important in many fields including creating virtual environments, source localization and detection, noise suppression, and beamforming. Spherical microphone arrays are a promising development to help achieve such capture and analysis, and have been studied by several groups. We develop a practical spherical microphone array and demonstrate its utility in applications for sound capture, room measurement and for beamforming and tracking. To accommodate equipment failure and manufacturing imprecision we extend their theory to handle arbitrary microphone placement. To handle speech capture and surveillance we describe the development of a new sensor, the hemispherical microphone array. For each array the practical performance follows that predicted by theory. Future applications and improvements are also discussed. [Work supported by NSF.]