bluuebunny/crossref_metadata_2025 huggingface dataset:
Distance information transmission using first-order reflections
Authors: Brungart Douglas S. , Durlach N. I.
Published: October 2005

An experiment is underway to determine the viability of synthetic first-order room reflections as a source of audio distance information for virtual environments. Although reflections have been considered possible sources of audio distance cues in real environments for some time, little basic research has been done to determine the ability of human listeners to identify the relative intensity of the primary and reflected signals and the time delay between the primary and reflected signals. A preliminary experiment showed little improvement in distance identification with the addition of a single-floor reflection when a distance cue based on the overall intensity of the signal was present. A second experiment has been designed that examines the amount of information transfer that can be achieved through variations in the delay and the relative intensity of the reflection when the overall intensity of the signal is randomized. The results of these experiments are discussed, along with the implications of these results for the implementation of reflection-based audio cues in virtual environments.
Detecting changes in distance in an anechoic and echoic environment
Authors: Hafter Ervin R. , Jensen Eric R. , Saberi Kourosh , Cohen Ephram
Published: October 2005

Increasing the distance to an acoustic source produces: (a) low-pass filtering of the spectrum through absorption by the medium, (b) reduction in the overall sound-pressure level of the stimulus, and (c) changes in the temporal and directional relations between sounds along the primary acoustic pathway and those on secondary pathways (echoes). A simulated open-field environment [Hafter etal., J. Acoust. Soc. Am. 92, 2334 (1992)] consisting of an anechoic chamber and multiple computer-controlled speakers was used to study the discriminability of changes in distance. Simulated acoustic reflections were from a single, fully reflective wall. Assuming spectral cues would be ineffective with small changes of distance, the primary concern was with the roles of level and echoes. Other considerations included the changing temporal overlap of primary and secondary stimuli for lengthening trains of wideband, midfrequency (4-kHz center) impulses, and distance perception for single words and short phrases. Subjects reported that with echoes they always heard a change in distance, but without echoes, they heard either a change in distance orloudness. Thresholds were smallest when the changes were interpreted as distance, thus performance was generally better with echoes.
Radial basis function neural network for modeling auditory space
Authors: Jenison Rick L. , Fissell Kate
Published: October 2005

A neural network model is presented for approximating free-field-to-eardrum transfer functions (HRTFs) from a set of measured HRTFs. Learning an input–output mapping from examples is something that neural networks have been designed to perform and can be thought of as forming an approximation of a multivariate function. The mapping of input spherical coordinates (azimuth and elevation) to output HRTFs can be accomplished using an approximating function composed of a fixed number of basis functions and parameters that are estimated through a process of optimization. Radially symmetric functions, known as radial basis functions (RBFs), can serve as the set of nonorthogonal bases with the Gaussian often the basis function of choice. An important consideration in setting up the RBF architecture is choosing a sufficient number of basis functions as well as choosing the placement of the basis functions in order to adequately cover the two-dimensional input space. A gradient descent algorithm is presented that automatically learns the optimal placement and size of the RBFs such that the predicted HRTF mean-squared error is minimized. Other basis functions that are more appropriate for working on the sphere, such as the von Mises–Fisher function, will also be discussed. [Work supported by ONR and Wisconsin Alumni Research Foundation.]
The MIT acoustics and vibration laboratory: An overview
Authors: Leehey Patrick
Published: October 2005

Founded in 1964, this laboratory has served as an interdepartmental focal point for research and education in acoustics. Its activities in the fields of random vibrations, flow noise, building and urban acoustics, structural acoustics, machine dynamics, and signal processing are reviewed. Its influence on education in acoustics at MIT and elsewhere is also noted.
Aeroacoustic sources of high speed maglev trains
Authors: Hanson Carl E.
Published: October 2005

Noise has been identified as a potential source of concern associated with the introduction of high speed maglev trains into the existing transportation system in the United States. The first operational maglev systems have been shown to generate high noise levels at high speeds, with noise levels over 100 dBA at 25 m typical for the maximum cruising speeds. At speeds greater than 150 miles per hour, the dominant noise source is of aerodynamic origin, proportional to the 6th power of velocity at high speeds, and 8th power dependency at very high speeds. An investigation of the noise generated by maglev systems was conducted to determine noise sources, develop noise criteria for high speed maglev systems, prepare design guidelines for noise control, and recommend an acoustic test facility for maglev research. This paper describes the findings of the study including recommendations for further research to develop a better understanding of the aerodynamic sound generation associated with high speed maglev operations. [Work performed under contract to U.S. Department of Transportation as part of the National Maglev Initiative.]
A fractional fractorial experiment to reduce disk drive noise
Authors: Wittig Larry E. , Murphy Brian
Published: October 2005

Hard disk drives used in computer equipment make noise while at idle and while seeking because the outer shell vibrates. A number of treatments thought to be useful to reduce this vibration were identified, and the most effective ones were determined by conducting one-at-a-time screening tests. Since no single treatment provided sufficient noise reduction, a fractional fractorial experiment was conducted using a Taguchi L16(215) array to determine the best combination of seven different treatments. This approach reduced the total number of combinations for the experiment from 128 to 16. The experimental results showed that a combination of just two or three of the seven treatments gave the most noise reduction.
Optimization of structural vibration using energy concepts
Authors: DeJong Richard G.
Published: October 2005

Energy concepts are used to develop techniques for optimizing the design and control of structural vibrations. In searching for a design that minimizes the vibrational amplitude of a structure at a number of points and frequencies, the minimization function is very irregular, having many local minima in the design parameter space. This makes it difficult for automated search algorithms to find a global minimum. The use of frequency-averaged energy levels as the response variable provides a smoother minimization function and better results from optimization algorithms. Similarly, in active control systems, the use of measured point vibration amplitudes as the minimization function suffers when the measured points correspond to the nodes in a modal response. The use of structural intensity measurements as the minimization function helps to overcome this problem. Several examples are given for beam and plate structures.
Visualization of statistical energy analysis results
Authors: Moeller Mark J.
Published: October 2005

Statistical energy analysis (SEA) is a tool for estimating the response of complex dynamic systems at high modal density. SEA is seeing ever wider application in a range of industries. It is used in the aerospace industry for launch vehicles, commercial, light aircraft, and helicopters. It has been successfully applied in the marine industry for both submarine and surface vehicles. SEA has seen extensive application to the building trades. The automotive industry is beginning to explore the application of SEA to vehicle acoustic design. The SEA model of vibrational power transmission has a direct analogy to thermal power transmission (diffusion). Just as thermal power flow is proportional to temperature difference, vibrational power flow is proportional to modal energy difference. This thermal analogy can be exploited to visualize the SEA results. This was accomplished by color coding a finite element representation of the structure. Visualization of SEA results is explored in two examples. The first example is a network of flat plates reported in the literature. The second example is a SEA model of a D-class car body-in-white with closures. The use of the thermal analogy provides an effective means of visualizing the diffusion process when a finite element surrogate is available.
Mach- and Reynolds-number dependence of the wall-pressure spectrum in a flat-plate turbulent boundary layer
Authors: Haj-Hariri H. , Herbert K. , Leehey P.
Published: October 2005

Wall-pressure measurements are performed using an array of pinhole microphones mounted into the test wall of a low noise, low turbulence flow facility. The frequency-wave-number spectra for both streamwise and spanwise directions are obtained through the spatial Fourier transformation of cross-spectral measurements in the said directions. The results of the wind-tunnel measurements reveal an interesting behavior of the low wave number portion of the wall-pressure spectrum. Namely, this portion of the spectrum does not exhibit any Mach-number dependence for nearly incompressible flows. The bulk of this talk addresses the analytical investigations undertaken in order to establish the validity of the experimental observations. The existing theories—which are inviscid—predict a low-wave-number wall pressure proportional to the Mach number. Initially, the complete problem is analyzed while allowing for the effect of a finite Reynolds number (viscosity). The low-Mach-number analytical model shows the wall-pressure spectrum to have a leading term independent of the Mach number. There is good frequency trend agreement between the theoretical and the experimental spectra. The physical mechanism for the Mach-number independence of the spectrum is attributed to a hydrodynamic contribution to the wall pressure resulting from a Stokes layer induced by the sound-generating turbulent eddies. Next, a simple model problem is presented which isolates the viscous mechanism responsible for the hydrodynamic contribution to the wall pressure and the subsequent contribution to the sound field brought about by this interaction. It is argued that the physical process involved is akin (albeit inversely) to that operative in the boundary-layer receptivity problem for nearly incompressible flows.
Source waveform recovery using minimum phase cepstrum
Authors: Hirobayashi Sigeki , Tohyama Mikio , Koike Tsunehiko , Lyon R.
Published: October 2005

Source waveform recovery under reverberant conditions is quite important for machine diagnostics. This talk investigates cepstrum dereverberation without using inverse filters of reverberant transfer functions. Source waveforms are typically modeled here by a pulselike signal, pulse trains, and modulated pulse trains. The dereverberation experiments demonstrate that a pulselike waveform, a periodic pulse train, and a modulated periodic train are recovered from the reverberant minimum phase cepstrum without nonminimum phase all-pass components. The all-pass components do not include important signature elements for those source signals. However, aperiodic pulse trains convey their signatures also in their nonminimum phase components. The recovery for those waveforms requires their all-pass components. The authors notice that speech waveforms are typical examples whose signature elements are included also in their nonminimum phase components due to their pitch fluctuations. The authors will also illustrate the zero locations of the source signals to describe not only the power spectra but also the phase property of the source signals. [Work supported by Sound Technology Foundation, Japan.]
Effect of spectral notches on robust waveform recovery methods for machinery diagnostics
Authors: Bowen David L.
Published: October 2005

Recovery of pulselike waveforms from vibration response measurements is often a goal in reciprocating machinery diagnostics. One such method, which makes use of inverse filtering, uses cepstral windowing in an attempt to make the procedure ‘‘robust.’’ It has been observed that, under certain conditions, the timing of the (cepstrally windowed) waveforms recovered from several consecutive machine cycles may occur at one of two distinct values. This behavior stems from a zero lying close to the unit circle in the response spectrum. Small changes in the measured response will cause such a zero to fluctuate between minimum and nonminimum phase conditions, resulting in corresponding phase differences of 2π between cycles. Subsequent cepstral windowing (smoothing) of the phase curves tends to ‘‘smear’’ this difference out over a broader frequency range, causing the recovered waveforms to be grouped into two distinctly different times, depending on whether π was added or subtracted at the frequency of the zero. The timing differences between these two groups is a function of the frequency of the zero, the characteristics of the transfer function, and the cepstral window width. Waveform recovery procedures that try to take this behavior into account have been investigated, and will be discussed. [Work supported by Robert Bosch GmbH.]
Multidimensional visualization of ultrasonic images
Authors: Greenleaf James F. , Belohlavek Marek
Published: October 2005

Modern commercial medical ultrasound instruments are capable of obtaining up to 40 real-time, two-dimensional images per second. One of the most difficult organs to evaluate with ultrasound is the heart. The heart is a complex organ that achieves its pumping function through cyclic motion of its chambers and valves. Computerized visualization of the heart is being developed to aid in training and to provide comprehensible images of this complex organ to the patient, the surgeon, and the researcher. To obtain the required data, the two-dimensional imaging plane of an ultrasound instrument is scanned through the heart from the surface of the chest, or more commonly, from within the esophagus. Sequences of digitized two-dimensional images taken at incremental positions through the heart are gated with the cardiac cycle and assembled by computer programs into volumetric images. Visualization methods of filtering, segmentation, and rendering are used to produce comprehensible, data dense, three-dimensional images of the heart. Sequences of the three-dimensional images obtained at sequential points through the cardiac cycle produce what are called ‘‘four’’-dimensional images. Mathematical projections of elements of these images onto selected two-dimensional planes can be recorded on videotape to produce movies of the structure or function of interest at various angles of view. Metrics such as distance, area, volume, or speed can be used on these data to produce quantitative measurements.
Time domain visualizations of simulated acoustic propagation and scattering processes in complex structures using a cellular automaton algorithm run on a parallel processor
Authors: Cole James B. , Krutar Rudolph A. , Numrich Susan K.
Published: October 2005

A cellular automaton realization of a finite difference time domain algorithm that can exploit the full power of parallel processors to simulate acoustic wave fields as they evolve in time, forming animated visualizations of wave propagation and scattering processes has been developed. Since the computation time is independent of environmental complexities (e.g., boundary shape, medium inhomogeneities), it is possible to simulate and visualize acoustic propagation in complex structures. Moreover the algorithm does not employ the far-field approximation so it is particularly useful to study time-dependent near-field and intermediate-field processes. Two-dimensional views of a four-dimensional space-time lattice of acoustic field values are selected.
Phase space footprints of truncated periodicity: Frequency domain wave-oriented data processing
Authors: Felsen L. B. , Carin L. , Hsu T.-T. , Kralj D.
Published: October 2005

An asymptotic theory for frequency and time domain acoustic scattering due to truncated periodicity modeled by filamentary scatterers has been presented recently [L. B. Felsen and L. Carin, J. Acoust. Soc. Am. 95, 638–649 (1994)]. The solution is parametrized in ray-acoustic forms which contrast individual scattering events with collectively restructured events over the equivalent Bragg-modulated aperture distribution covering the extent of the array; the resulting scattered fields involve truncated Bragg beams and Bragg-modulated edge diffractions. Wave-oriented processing is now applied to the frequency domain scattering data to extract the ray-acoustic phenomenologies. The setting is the (x,z,kx,kz)ω phase space where kx and kz are the spatial spectral wave numbers corresponding to the x and z coordinates, with the frequency ω as a parameter. The phase space footprints are extracted by windowed and global transforms that yield various local and global distributions displayed by alternative visualizations. Of special interest are the footprints pertaining to the structurally dispersive Bragg modes. Discussion includes linkage of phase space footprints with the scattering configuration. [Work supported by AFOSR.]
A new nonlinear approximation for elastic wave scattering
Authors: Kostek S. , Habashy T. M. , Torres-Verdín C.
Published: October 2005

A novel approximation to simulate elastic wave scattering in arbitrarily heterogeneous media is introduced. The approximation, formulated in the frequency domain, is derived from a volume integral equation governing the displacement vector within the scatterer. It is shown that if the displacement vector is assumed a locally smooth function of position within the scatterer then it can be expressed as the projection of the background displacement onto a scattering operator. This scattering operator is nonlinear with respect to the spatial variations of density and Lamé constants and can be computed via simple explicit formulas. The scattering operator adjusts the background displacement by way of amplitude, phase, and polarization corrections which are needed to estimate the internal displacement due to an arbitrary source excitation. It is also shown that the new approximation is substantially more efficient than iterative Born techniques whose convergence is hardly guaranteed for large contrasts in material properties. Validation tests are presented which confirm that the new approximation remains accurate for large contrasts of the elastic parameters and over a wide frequency range. Moreover, the new approximation has nearly the computational efficiency of the first-order Born approximation but is much more accurate. These two features make the scattering formulation extremely attractive to approach multifrequency inversion problems involving a multitude of scatterers.
A modified T-matrix approach for scatterers with mixed boundary conditions or concavities
Authors: Varadan Vasundara V. , Kim Yong Woo , Varadan Vijay K.
Published: October 2005

Many scattering problems involve specification of different types of boundary conditions on different parts of the surface of the same scatterer or the geometry of the scatterer may involve concavities. Both these problems are difficult to model using available methods. The applications of such problems are however many—debonded fibers in fiber reinforced composites, partially coated inclusions in engineered coatings, to name just two. A modification of the usual T-matrix approach is proposed by constructing a mathematical surface which partially includes the actual surface of the scatterer. The integral representations for the fields for the two regions—the actual scatterer volume and the volume enclosed by the extended surface are simultaneously solved. A general formulation is presented with numerical results for P- and S-wave scattering by partially debonded fibers in an elastic matrix. a)Present address: Samsung Electronics, Suwon, Korea.
Comparison of linear and nonlinear elastic moduli for rocks by use of a granular medium model
Authors: Belyaeva Irina Yu. , Ostrovsky Lev A. , Sutin Alexander M. , Zaitsev Vladimir Yu.
Published: October 2005

Comparative analysis of linear and nonlinear elastic parameters variability depending on material structure and initial stress is given based on the model of the medium as of a granular structure with fluid pore filling. Examples of such variations for some geological conditions presumably interesting for seismoprospecting, are presented. It is demonstrated that the nonlinearity parameter may be used in exploration seismology as a characteristic much more sensitive to the changes of the medium structure than conventially measured linear moduli.
Contactless characterization of antisymmetric edge wave dispersion along truncated wedge using electromagnetic acoustic transducer
Authors: Chamuel Jacques R.
Published: October 2005

Accurate laboratory measurements are compared with approximate theoretical calculations on the dispersion of antisymmetric flexural edge waves propagating along the apex of a truncated 6061 aluminum wedge with an apex angle Θ=26.7°. The wedge waves are detected with a specially designed electromagnetic acoustic transducer in the frequency range 30–500 kHz. Phase velocity dispersion plots are obtained from gated sinusoidal signals and wavelength measurements averaged over 10–20 wavelengths. The general characteristics of the experimental phase velocity dispersion data closely matched the approximate theoretical parabolic model by McKenna etal. [IEEE Trans. Sonics Ultrason. SU-21(3), 178–186 (1974)] derived for small apex angle. The measured dispersion curve was duplicated using a truncated wedge tip width 57 μm in the parabolic approximation calculations while the actual measured tip width was 62 μm Antisymmetric flexural wedge waves obtained from a sharp apex (3 μm) are practically nondispersive as expected in the same frequency range. Previously reported laser ultrasonic dispersion measurements obtained by Jia etal. [Appl. Phys. Lett. 61, 2970–2972 (1992) and IEEE Ultrason. Symp. Proc. 1993, paper B4] revealed unexplained measured dispersion behavior of antisymmetric wedge flexural waves based on a phase analysis method. Contactless electromagnetic acoustic direction of antisymmetric wedge waves provided dispersion results in agreement with theoretical calculations. [Work supported by ONR.]
A lumped-parameter model for the acoustic power output from a vibrating body
Authors: Fahnline John B. , Koopmann Gary H.
Published: October 2005

A general lumped-parameter model for the acoustic power radiated by a vibrating body is derived. The derivation starts from an analytical expression for the acoustic power output written in terms of the Green’s function of the second kind and the specified surface velocity. By discretizing this expression in terms of elemental volume velocities and ‘‘averaged’’ Green’s functions, a consistent approximation for the power output from a vibrating body is determined. The approximation assumes that two solutions for the power output from a vibrating body are equivalent if they have the same volume velocity distribution over each of the surface elements. The limitations of the approximation are examined and it is shown through several example problems that the approximate expression for the power output is accurate as long as each of the surface elements is small compared to both the acoustic and structural wavelengths.
Lateralization of comodulated bands of noise
Authors: Saberi Kourosh
Published: October 2005

Lateralization thresholds were measured in a 2IFC task for two conditions. The 400-ms signal consisted of two high-frequency narrow bands of noise with common interaural delays (cf.=2.55 and 3.35 kHz; bandwidths=50,100,200,400 Hz). In the first condition, the temporal envelopes of the two bands were the same (comodulated) and, in the second condition, they were independent. Results of this experiment showed improvements in lateralization when the bands were comodulated compared to independently modulated bands. In a second experiment, the two noise bands were replaced by two sinusoidally amplitude modulated (SAM) tones (cf.=2.55 and 4 kHz; modulation rate=100 Hz) and lateralization thresholds were measured as a function of the relative phase of modulation between the two SAMs. Thresholds were lowest when the envelope of the two SAMs were in phase and monotonically increased as the envelopes were increasingly shifted out of phase up to 180°. In a third experiment, binaural interference was examined. Results are discussed in terms of cross-correlation models of binaural hearing. [Work supported by NIH and AFOSR.]
Quantitative analysis of onset dominance in lateralization
Authors: Saberi Kourosh
Published: October 2005

A statistical method for measuring the weight which an observer gives to each component of a multicomponent signal [COSS Analysis; Berg, J. Acoust. Soc. Am. 86, 1743–1746 (1989)] was applied to precedence. Signals consisted of trains of dichotic clicks whose time envelopes were Gaussian and centered at the cosine phase of a 6-kHz carrier. The interaural delay of each click was perturbed by adding or subtracting a small Gaussian random variable. In a single-interval task, subjects were required to determine whether the lateralized image was to the right or left of the head. Using COSS analysis, weights were determined for each click of the multiclick stimulus. For short interclick intervals (ICI=1.8 ms) as the number of clicks increased (n=2 to 16), the relative weight given to the onset click diminished. The weights estimated for all clicks following the first were nearly the same. Only the relative weight of the first click varied as stimulus parameters were altered. Frequency-channel effects and the idea of restarting in binaural adaptation were also examined. [Work supported by NIH and AFOSR.]
The relative contribution of target IDTs and distractor IDLs in judgments of laterality
Authors: Dye R. H. , Stellmack M. A.
Published: October 2005

A stimulus-classification paradigm was used to examine the extent to which interaural differences of time (IDTs) and interaural differences of level (IDLs) interact across the frequency domain. The target was a 753-Hz tone presented with one of ten IDTs (−90 to +90 μs). The distractor was 353, 553, 953, 1153, 1753, or 2753 Hz (fixed within a block of trials), presented with one of ten IDLs (−9 to +9 dB). Each test interval was preceded by a diotic presentation of the target alone. The duration of the signals was 200 ms. Each block of 100 trials presented each combination of target-distractor interaural differences once. Subjects indicated whether the target appeared to the left or right of the intracranial midline. The relative salience of the target IDT and the distractor IDL was assessed by the slope of the best linear boundary between left and right responses. For conditions in which the distractor was 200 Hz above or below the target, distractor and target information ‘‘traded’’ at about 1 dB/10 μs. At wide frequency separations the slopes were steeper for most listeners, indicating greater ability to lateralize the time-delayed target independently of the distractor. [Work supported by NIH.]
Consistency over frequency in high-frequency binaural lateralization
Authors: Lee Wonseok , Stern Richard M.
Published: October 2005

Recent experimental results [e.g., R. M. Stern etal., J. Acoust. Soc. Am. 84, 156–165 (1988)] imply that the auditory system emphasizes the contributions of internal delays that are consistent over frequency in lateralizing binaural stimuli. This phenomenon, known as ‘‘straightness weighting,’’ has been used effectively to predict lateralization of low-frequency bandpass noise. This paper describes a series of experiments that investigate the extent to which a similar type of straightness weighting is observed at high frequencies for which there is no synchronization of the auditory-nerve response to the fine structure of the stimuli. High-frequency bandpass noise stimuli were generated by summing cosines with interaural time delay (ITD) that were systematically varied as a function of frequency. Stimuli were generated that produce varying amounts of consistency over frequency in the putative response of the units in the binaural system that record interaural coincidences of auditory-nerve activity. The results of pilot experiments in which a rivalry is established between ridges of the response at different internal delays indicate that lateralization for these stimuli is dominated by the component of the high-frequency stimulus that produces the component of the response that is most consistent over frequency. [Work supported by NSF.]
Effects of uncertainty on auditory spatial resolution of tonal sequences
Authors: Chandler David W. , Grantham D. Wesley , Leek Marjorie R.
Published: October 2005

The horizontal-plane minimum audible angle (MAA) was measured for single components of a five-tone sequence, where each component was a 1000-Hz burst 100 ms in duration. The five components were presented in a pseudorandom order from the following five azimuths: −54°, −19°, 0°, +23°, and +56°. The target component whose azimuth change was to be detected was at either −54° or 0°, and could occur in temporal position 2 or 5 in the sequence. Stimulus uncertainty was varied from a minimal level to a high level by randomly varying target temporal position, target azimuth, or both from trial to trial. When the target position and azimuth within the sequence was unchanged from trial to trial (minimum stimulus uncertainty), MAAs for targets presented in the last temporal position at both target azimuths were as small as for tones presented in isolation. However, for targets presented earlier in the sequence, MAAs were elevated at both azimuths. The MAAs at 0° at both temporal positions were unaffected as uncertainty increased, but there were large effects of uncertainty at −54° azimuth. Results suggest that in a task requiring spatial resolution in the horizontal plane, when the time of arrival and location of an auditory signal are unknown, listeners focus their attention near midline, where resolution is typically most precise. [Work supported by U.S. Army and NIH DC00185.]
Adaptation to supernormal auditory localization cues as a function of rearrangement strength
Authors: Shinn-Cunningham Barbara , Delhorne Lorraine , Durlach Nat , Held Richard
Published: October 2005

The ability of humans to adapt to intermodal discrepancies is an important factor in the design of virtual environments and teleoperator systems. Previously, it was demonstrated that subjects can adapt to supernormal auditory localization cues that enhance position resolution using either standard psychophysical training methods (correct-answer feedback) or methods traditionally employed in studies of sensorimotor rearrangement (active sensorimotor loop tasks without special cognitive feedback). In the present study, the strength of the supernormal rearrangement was parametrically varied in order to ascertain the impact on the rate and asymptote of adaptation achieved. Subjects were repeatedly tested in a hybrid real/virtual environment where auditory stimuli were synthesized (using a PC, Convolvotron and electromagnetic head tracker) for thirteen positions in the horizontal plane marked by real lights. Testing consisted of identifying the azimuth of the virtual sound source without significant head motion, but with correct-answer feedback. Preliminary results will be given for resultant changes in bias and resolution as well as for rate and asymptote of adaptation as a percentage of the distortion caused by the rearrangement. [Work supported by MIT Lincoln Laboratory & AFOSR, Grant No. 90-200.]
Auditory localization in noise. II. The effects of masker location
Authors: Good Michael D. , Gilkey Robert H.
Published: October 2005

Good and Gilkey [preceding abstract] report that the ability to determine the direction of a masked, broadband, click-train signal in the free-field is dependent on signal-to-noise ratio. In the present study, localization performance was measured at two signal-to-noise ratios as a function of the spatial location of the masker. Subjects judged the direction of the signal, which could emanate from any of 239 possible spatial directions (completely surrounding the subject in azimuth and ranging from −45° to 90° in elevation), by pointing at a 20-cm-diam spherical model of auditory space. The masker was at one of five possible spatial locations (front, back, left, right, or above, relative to the subject). In general, the judged direction of the signal was biased toward the location of the masker. However, the strength of this effect depended on the signal-to-noise ratio and on the location of the masker. [Work supported by AFOSR-91-0289.]
Directional masking: Detection, localization, and confidence in responses to a target signal presented in directional noise
Authors: Abouchacra Kim S. , Letowski Tomasz R.
Published: October 2005

The purpose of this investigation was to explore auditory spatial hearing perception in environments containing directional noise. For 40 normal hearing subjects, detection thresholds were measured for a target signal that was spatially separated from directional noise (65 dBA) by either 0°, 45°, 135°, or 180° azimuth. Following these measurements, localization accuracy and confidence ratings were determined for the target signal presented in directional noise at 0 dB SL, 6 dB SL, 12 dB SL, and 18 dB SL. As expected, detection results showed that as the spatial separation between the target signal and noise source increased, the target signal was more easily detected, with maximum improvements amounting to as much as 13 dB. As sensation level increased, localization accuracy improved and confidence in localization responses increased. Unlike detection, the extent of improvement in localization performance depended more on the spatial location of the target signal than on the amount of spatial separation between the target and directional noise. A general description of detection data, followed by a detailed description of localization and confidence data will be presented and discussed. [Work supported by the U.S. Army Research Laboratory.]
Ray-tracing analysis of auditorium acoustics for the Macintosh
Authors: Rieffel Marc , Noer Geoffrey , Dilatush Jeremy , Brown Andrew , Everbach E. Carr
Published: October 2005

An enhanced version of a previously presented geometrical ray-tracing program for the Macintosh [M. Rieffel and E. C. Everbach, J. Acoust. Soc. Am. 91, 2373(A) (1992)] will be demonstrated. The program accepts input files of auditoria created in any of several popular CAD programs that export standard DXF formats, displaying the auditorium in 3-D projection. The user specifies the surface absorption coefficients of all internal surfaces and the placement and characteristics of sound sources. The user also specifies an arbitrary ‘‘monitor’’ plane passing through the hall, each point of which will be a receiving location in the subsequent simulation. When the simulation is run, rays are propagated between source and receiver, allowing the computation of an echogram for each point in the monitor plane. In this way users can discover focusing, dead spots, or late reflections at specific locations in a proposed hall design. The program is available for consultant or instructor use, and will be demonstrated during the session.
End-point detection of microphone-array speech signals
Authors: Srivastava Sanjay , Che Chi-Wei , Lin Qiguang
Published: October 2005

This paper describes algorithms for automatic end-point detection of microphone-array speech signals. Microphone arrays provide a hands-free sound pickup. The captured sound typically has a higher signal-to-noise ratio (SNR) than that captured with conventional microphones used at distances, such as in teleconferencing environments. However, due to multipath distortion (room reverberation) and ambient noise, the detection of starting/ending points of array speech is more difficult than that of close-talking speech. In this paper, short-time energy and short-time zero-crossing rate are computed for the original speech waveform and its high-pass filtered and low-pass filtered versions. These six functions are then utilized in different combinations to determine the end points. Speech data used in the experiments are collected in a hard-walled laboratory room, having a reverberation time of approximately 0.5 s with a one-dimensional beamforming line array. From the experiments, it is found that the high-pass filtered signal gives a more reliable estimate of end points than does the low-pass filtered counterpart. This result is consistent with the fact that reverberation and noise in rooms are typically more prominent at low frequencies and are relatively moderate at mid- and high frequencies. The detection algorithms have been integrated into a dynamic-time-warping- (DTW) based speech recognizer. Recognition performance of the system is evaluated for both array speech and for close-talking speech.
Dynamic information in the identification of Mandarin Chinese vowels and tones
Authors: Suiter Traci L. , Gottfried Terry L.
Published: October 2005

Native speakers of Mandarin Chinese were tested on their ability to identify the vowel and tone of syllables produced in a carrier sentence by a female native speaker. Four vowels /i/, /a/, /o/, /u/ were spoken with the four tones of Mandarin: level, rising, falling–rising, and falling. Stimuli in condition A consisted of intact /b/-vowel syllables followed by /dzbarred eye/, which were excised from the original sentence. Condition B stimuli consisted of the first six pitch periods+silent gap+/dzbarred eye/. Condition C stimuli consisted of all but the first six pitch periods+/dzbarred eye/. Listeners were most accurate in condition A, and only slightly less accurate in condition C. Listeners were significantly less accurate identifying initial-only syllables (condition B). Most errors were made on tone identification. These findings suggest that Mandarin Chinese listeners rely little on the formant transitions from the initial portion of the syllable to identify tone. Further studies investigate how the final portion of the syllable affects identification. [Work supported by NSF-ILI Grant No. USE-9251432.]
Optimum source amplitude for reduction of sound radiation by several point sources
Authors: Wiegandt Ted J.
Published: October 2005

A computer simulation of an array of point sources will be the focus of the noise reduction study. The amplitude of a select few of the point sources will be varied with the goal to reduce total sound output. The computer model will accept an array of up to ten points in any two-dimensional geometry. The phase and amplitude of the point sources can all be set independently. The chosen point sources will then have their amplitude varied over a specified range. The resulting total sound power output in terms of the amplitudes of the two varied sources can then be studied. There will be similarities in the behavior of a simple array of point sources and more complex vibrating structures such as a box. Understanding the behavior of the point source model will help lead to a method of optimization for more complex objects. [Work supported by NSF MSS-9103377.]
Interactive instruments for experts and amateurs
Authors: Machover Tod
Published: October 2005

Since 1986, the Hyperinstrument Group at the MIT Media Lab. has been designing intelligent, interactive musical instruments to measure and interpret musical performance and enhance and expand the possibility of a particular instrument. A series of hyperinstruments was designed, some based on existing musical instruments, others based on new models for musical expression, such as hand gesture. From 1986 to 1991, hyperinstruments were designed for virtuosos and emphasis was placed on measuring very subtle differences in interpretation and articulation from performer to performer, and between different performances by the same virtuoso. Since 1991, concentration has been placed on expanding the hyperinstrument model for use by amateurs not proficient on an instrument. An attempt has been made to create systems that stimulate the imagination and allow room for growth and learning, investigating how musical composition will change when instruments are smart and listeners can actively shape music. This paper will explore recent hyperinstrument experiments for experts and amateurs, and will discuss an opera project currently under development in which an audience of one hundred people will work together at every performance, using specially designed interactive instruments, to help create and perform a complex piece of music they have never heard before.
Reconstructing an improviser
Authors: Haase Kenneth B.
Published: October 2005

A ‘‘rational reconstruction’’ of the jazz improvisation system implemented in [David Levitt, ‘‘A Melody Description System for Jazz Improvization,’’ MIT S.M. thesis (1981)] will be discussed. A rational reconstruction is a reimplementation of a computer program in a way which clearly reveals its basic mechanisms and the fundamental issues it addresses. Levitt’s program did melodic improvisation from an initial chord progression, based on a collection of heuristics operating over a simple constraint representation. Its rational reconstruction reveals a number of basic issues in computational aesthetics: The use of constraints to model aesthetic choice; the use of representation to encode stylistic background; and the tension between randomness and structure in creative process.
Acoustic pattern perception in synchronous music systems
Authors: Vercoe Barry
Published: October 2005

A common fallacy in interactive music systems is that a computer can make musically intelligent use of data from classical DSP-based acoustic analysis. In fact, human auditory sensors and processors are so different from standard silicon ones that machines are unlikely to ‘‘hear’’ music anything like the way humans perceive it. However, a computer model of human auditory sensing and preattentive processing can provide a good approximation of the conditioned stimuli that the musical brain actually works with. A real-time model is described in which the first derivative of a constant-Q filter bank is positively rectified, convolved with an energy integration function, then summed and fed into a temporal pattern detector (a pulse and tempo sensor, or foot-tapper) which provides a robust control for synthetic instruments needing to play ‘‘in sync’’ with the acoustic source. The computational stages are based on solid evidence from psychacoustic research, and appear robust in practice. Videotaped and live demonstrations.
A musical chaotic space
Authors: Dabby Diana S.
Published: October 2005

A procedure for generating different versions of a musical piece based on the properties of a chaotic system—most notably, sensitive dependence on the initial condition—will be described and demonstrated. The procedure uses the Lorenz equations to set up a musical chaotic system. The chaotic system is embedded in a computer program that enables the composer to select which musical attributes (such as melodic line or dynamic level) will be affected by the chaotic trajectory. Mutations of the original work can be heard as well as revisions where many of the musical elements intrinsic to the piece, e.g., rhythm or timbre, remain intact. The composer is able to interact with the program to select various versions and change them, if desired. Musical examples will be played so that listeners can hear the source piece and its concomitant transformations, each of which is initiated by the musical chaotic space. Yet the compositional character of the revised versions remains within the artist’s domain of style, expression, and inventiveness. [Work supported in part by a LeBel Fellowship at MIT and in part by funds from the Media Lab.]
Measurement of harmonic ratios of sounds produced by musical instruments
Authors: Brown Judith C.
Published: October 2005

A recently developed high resolution pitch tracker [J. C. Brown and M. S. Puckette, J. Acoust. Soc. Am. 94, 662–667 (1993)] has made it possible to measure the ratios of the frequencies of the upper harmonics of a sound with respect to its fundamental frequency with high accuracy. Calculations were carried out on digitized sounds produced by a cello, viola, alto flute, clarinet, and piano. The sounds produced by the stringed instruments included examples both with and without vibrato. Measured ratios were exactly equal to integers for all instruments except the piano with an upper bound of error of 3 cents. As expected the piano was found to have stretched harmonics with the measured inharmonicity increasing for harmonics 2 through 5. The major limitation on this method of calculation was found to be extraneous noise, and this was greatest for the stringed instruments.
Synthesizing musical tones by summing successive read-outs of a stored impulse response
Authors: Slaymaker Frank H.
Published: October 2005

By storing a digital representation of the impulse response of an equivalent filter in memory, a set of musical tones can be generated in which every tone in the set has the same spectral envelope rather than the same wave shape as occurs when the wave shape itself is stored. Similarly, in the tone of a violin played with vibrato the spectral envelope, rather than the wave shape, would be maintained throughout the vibrato cycle. Generating the tones by summing successive read-outs of the stored impulse response requires considerably less memory than that required to store a different wave shape for every note in the range of an instrument. In the absence of a synthesizer which generates tones from a stored impulse response, several examples of brass, woodwind, and string instrument tones have been computed in advance and the tones stored on disk so they can be demonstrated by playing them through a sampler synthesizer.
Violin timbre and the picket fence—Part III
Authors: Jansson Erik V.
Published: October 2005

During a bicycle ride along a picket fence the common observation was made that the picket fence becomes transparent at a sufficient speed. The author got the idea that a similar phenomenon may be applicable to hearing, and may explain the similarities of different tones from the same violin [STL-QPSR 2-3/1990, pp. 89–95]. Thereby the fundamental question arises whether one listens to the spectral properties or the time history. A series of pilot synthesis experiments have been started. Digitally realized resonators were excited by repetitive pulses. The pulse frequency and the resonator properties (resonant frequency and bandwidth) were varied. Listening to the synthesis gave the following results. (1) Partial tones dominated over the reverberating resonator tone. (2) The shape of the excitation pulse had a major influence. (3) For pulse frequencies up to 50 Hz a reverberating tone was perceived thereafter dominated the pulse sound. (4) For pulse frequencies up to 500 Hz the resonator frequency influenced the timbre thereafter resonator influence was difficult to hear. (5) A resonator bandwidth up to approximately 50 Hz gave a ringing tone. Some observations support frequency domain analysis and some time domain analysis. [Work supported by Swedish Natural Science Research Council.]