bluuebunny/crossref_metadata_2025 huggingface dataset:
Validating a perceptual framework for dynamic auralization
Authors: Warusfel Olivier , Cerveau Laurent , Jot Jean-Marc
Published: October 2005

The auralization of room impulse responses is generally realized by direct convolution with an anechoic signal. When using a simulated impulse response, the realism of the auralization relies mainly on the accuracy of the propagation model used to estimate time and spatial distributions of early reflections and of the reverberation process. An alternative approach, adopted at Ircam, uses a room effect synthesizer driven by acoustical indices, derived from the analysis of the response in order to describe the corresponding spatial, energetic, and statistical properties. Although the auralization performance then depends on the relevancy of these parameters, this latter solution offers the following advantages: (1) reduction of the computational cost; (2) manipulation of measured, estimated, and virtual configurations in a unified perceptual formalism; (3) adaptation to different reproduction format or setup, without recomputing the corresponding impulse responses. In order to validate the performance of the synthesis approach, a series of comparisons between synthesized and measured responses is undertaken, focusing successively on different time sections of the response. The study of the late reverberation is conducted by comparing measured monaural or binaural responses with modified responses where the late section is replaced by a decreasing white noise, or by a reverberation signal provided by a feedback delay network. These modifications are controlled by analysis/synthesis methods based on time-frequency representations.
New approach for obtaining individualized head-related transfer functions
Authors: Katz Brian F.
Published: October 2005

Presentation of a synthesized 3-D audio environment depends heavily on the use of head-related transfer functions (HRTF’s) when headphones or small numbers of loudspeakers are used for the actual sound sources. The detailed differences between individual’s HRTF’s are a serious hindrance to the widespread use of such virtual acoustic environments, as standard HRTF models do not perform adequately for large populations. Currentmethods of obtaining HRTF’s are very time consuming and inconvenient for the subject, consisting of long periods of immobility during measure-ments. Two new methods under consideration for obtaining an individualized HRTF are discussed which use physical models of the pinnae structure and geometric measurements of the entire head and pinnae. Comparisons of the results as compared with established measurement techniques were made. Subjective comparisons using headphone presentations of virtual complex sound sources were also made.
Broadband recording of social acoustic signals of the Hawaiian spinner and spotted dolphins
Authors: Lammers Marc O. , Au Whitlow W. L.
Published: October 2005

The social and communication signals of several species of free-ranging delphinids have been analyzed and characterized by a number of workers over the past 3 decades. Most of the past works have used bandlimited recording equipment and have focused on the sonic components of the signals. Therefore, most discussions of delphinid whistles and nonecholocation pulsed signals have been limited almost exclusively to below approximately 20 kHz. A technique has been developed that allows one to collect high-fidelity recordings of signals up to 55 kHz in frequency. By digitizing signals directly into a laptop computer through a PCMCIA A/D converter, the sonic and ultrasonic components of the whistles and burst pulses of two populations of Hawaiian spinner and spotted dolphins (Stenella longirostris and S. attenuata) resident along the leeward coast of Oahu, Hawaii are being studied. Preliminary results show that the ultrasonic range above 20 kHz is rich with whistle harmonics and components of pulsed signals.
Quantitative analysis of Tursiops truncatus
whistles from the Gulf of Mexico
Authors: Bazua-Duran Carmen , Evans William E.
Published: October 2005

Quantitative analyses of sounds based on spectral and statistical analyses have been used to compare Tursiops truncatus (bottlenosed dolphin) whistles. Groups and populations of T. truncatus have been quantitatively discriminated using a set of ten variables measured from a large sample (more than 500 for each group) of whistle contours [Wang et al., Aquat. Mamm. 21, 65–77 (1995)]. In the present work, T. truncatus whistle recordings from the Gulf of Mexico animals have been spectrally and statistically analyzed to discriminate between individuals and regional groups. A 13 variable set of measurements (beginning frequency, ending frequency, minimum frequency, maximum frequency, peak frequency, duration, peak time, number of inflection points, beginning sweep, ending sweep, number of contour breaks, type of signal, and number of steps) and a smaller sample size were used. [Work supported by UNAM and TAMU.]
Auditory masking of whale communication by ship noise
Authors: Erbe Christine , Farmer David M. , Yedlin Matthew J.
Published: October 2005

A method is presented to determine the acoustic masking effects of man-made noise on marine mammal communication. In particular, the interference of icebreaker related noise with beluga whale vocalizations is studied. Captive beluga whales have been trained for acoustic experiments during which they try to detect beluga vocalizations in various noisy backgrounds. In a stop/go manner, the animals indicate whether or not they can discriminate call from noise. Results are that bubbler system noise, generated when an icebreaker ejects high pressure air into the sea in order to push ice debris away, has the worst masking effects followed by propeller cavitation noise, generated when an icebreaker is stopped by an iceridge. Naturally occurring thermal icecracking noise has the least masking impact. Based on the experimentally collected data, computer software was developed to model the whale’s auditory abilities. Means of adaptive noise cancellation greatly outperform the whale and are hence not considered to take place in the whale’s brain. A backpropagation neural network showed the best similarity to the whale’s performance and classified the noises in the same order of disturbance as the whale.
Flow measurement with a double acoustic time-reversal mirror
Authors: Roux Philippe , Fink Mathias
Published: October 2005

Acoustic time-reversal invariance is broken when the propagating medium contains a flow. This effect is taken advantage of to achieve a new way of characterizing the vorticity field of a flow with a double time-reversal mirror (TRM). The double TRM is made up of two transducer arrays surrounding the flow. The time-reversal process is performed between the two arrays: When one array emits, the other one receives, time-reverses, and re-emits the acoustic wave. In the absence of flow, time-reversal invariance insures that many back and forth propagations between the two arrays do not modify the emitted wavefront. In the presence of flow, the acoustic wavefront is distorted through the flow. This distortion is related to the amplitude and size of the vorticity field and is often too small after one crossing to be observed experimentally. In order to amplify this wavefront distortion, several back and forth propagations are imposed on the acoustic wave through the flow between the two arrays. The greater number of crossings through the vorticity field induces an amplification of the effect of vorticity on the acoustic wavefront. This effect can then be easily measured and provides the vorticity characteristics even for flow with a small vorticity field.
Modeling of scintillation approach to transversal current measurement with respect to the Fram Strait environment
Authors: Naugolnykh Konstantin A. , Esipov Igor B. , Johannessen Ola M. , Ovchinnikov Oleg B. , Tujilkin Yurii I. , Zosimov Viktor V.
Published: October 2005

The results of a computer simulation of amplitude and travel-time variations in the process of a signal crossing the Strait on a number of paths to infer the intervening fine-scale variability and transverse current are discussed in this paper. The method relies on the advection of small-scale inhomogeneities across the acoustic path. These inhomogeneities produce perturbations in the amplitude and travel time of the sound and the current can be sensed by generating a time-lagged cross correlation of the full acoustic field. A linear four-element transmission array and a four-element receiving array both with equally spaced elements were used for calculations. By combining the signals from each transmitter–receiver pair in different ways, a number of different path positions were probed and a profile of the transverse current along the propagation path was retrieved. The calculations were done in the frame of the ray model of signal propagation. Analysis has shown that the stable rays, which touch the ocean surface four or five times, are stable under the influence of inhomogeneities. These rays make it possible to avoid the phase ambiguity which can arise when a signal is propagated in an inhomogeneous waveguide.
Acoustically focused oceanographic sampling in the Haro Strait experiment
Authors: Bellingham James G. , Schmidt Henrik , Deffenbaugh Max , Johnson Mark
Published: October 2005

The tidal fronts of Haro Strait, British Columbia, are associated with currents in excess of 3 kn, with significant horizontal stratification, leading to significant frontal turbulence and upwelling. Volumetric mapping of these fronts using traditional oceanographic sampling techniques is virtually impossible due to the spatial and temporal aliasing produced by the extremely dynamic oceanography. The concept of acoustically focused oceanographic sampling (AFOS) was developed to overcome this problem. AFOS combines the autonomous oceanographic sampling network (AOSN) [Curtin etal ., Oceanography 6(3) (1993)] with acoustic tomography. The network nodes used for communication with and navigation of AUVs is equipped with vertical arrays and tomography sources. In addition an AUV carries a source for adaptive, moving source tomography. The array data are transmitted to shore via a wireless local-area network and processed in real time for estimation of current distribution and frontal geometry. The results are used for adaptively focusing the AUV’s sampling of the front using surface mapping side-scan sonars and acoustic Doppler current profilers, as well as CTDs. [Work supported by ONR.]
Acoustic wave–internal wave interaction experiment in the Yellow Sea
Authors: Zhang Renhe , Jin Guoliang , Lei L. Y. , Gan Zijun , Zhou Ji-Xun , Rogers Peter H. , McCall George S. , Martin James S. , Caille Gary W. , Tate Duane C. , Dahl Peter H. , Spindel Robert C.
Published: October 2005

Simultaneous observations of internal wave activity and acoustic wave propagation in 70 m water in the Yellow Sea were made in the late summer of 1996. The objective of the experiment was to validate the predicated modal coupling and resulting fluctuations and alterations in propagation loss induced by shallow-water internal waves. Propagation over distances up to 50 km (using narrow- and broadband sources over the frequency range of 50 Hz to 6 kHz) was measured with moored and suspended arrays which spanned the water column. Internal wave activity was monitored using several thermistor chains. Details of the experiment and preliminary data will be presented. [Work supported by ONR and the Chinese Academy of Sciences.]
The coupled mode parabolic equation
Authors: Abawi Ahmad T. , Kuperman W. A. , Collins Michael D.
Published: October 2005

An additional term has been incorporated into the adiabatic mode parabolic equation (AMPE) to account for mode coupling. The coupled mode parabolic equation (CMPE) involves the solution of coupled horizontal wave equations for the mode coefficients. The numerical solution may be obtained efficiently using the splitting method. The first step is identical to the numerical solution of the AMPE. The second step involves Crank-Nicolson integration of the coupling term. The accuracy of the CMPE has been confirmed for two-dimensional problems involving coupling due to sloping bottoms and volume heterogeneities. The CMPE has been applied to three-dimensional problems involving both mode coupling and azimuthal coupling. It is practical to apply the CMPE to large-scale problems and possibly even global-scale problems at low frequencies. [Work supported by ONR.] a)A. T. Abawi is presently at AETC, Inc., 10975 Torreyana Rd. 300, San Diego, CA 92121.
Modeling internal gravity waves in a range-dependent medium
Authors: Lingevitch Joseph F. , Collins Michael D. , Siegmann William L.
Published: October 2005

The parabolic equation (PE) method is a useful approach for solving range-dependent ocean acoustics problems. The PE method can also be used to model the propagation of internal gravity waves in the ocean. One of the possible applications of a range-dependent gravity wave model to ocean acoustics would be to generate a model-based noise spectrum that could be used as an alternative or complement to the empirical spectrum of Garrett and Munk [J. Geophys. Res. 80, 291–297 (1975)]. This could be done using the approach of Kuperman and Ingenito for modeling surface-generated acoustic noise [J. Acoust. Soc. Am. 67, 1988–1996 (1980)] for an appropriate distribution of gravity wave sources. The mathematical details of a PE for gravity waves will be discussed. Examples will be presented to illustrate the propagation of gravity waves in range-dependent environments. [Work supported by ONR.]
Characteristics of low-frequency Arctic ambient noise determined using the SPINNAKER array
Authors: Pawlowicz Rich , Sotirin Barbara
Published: October 2005

The ambient noise field at low frequencies in the Arctic is dominated by sounds resulting from pressure ridging caused by convergences in the surface ice field. These convergences are driven by currents in the ocean and by winds on the surface, and are modulated by the ice strength and compactness. Analysis of long time series of ambient noise might therefore be a useful tool in studying the spatial and temporal changes in these factors. The SPINNAKER array recently deployed at the edge of the continental shelf north of Ellesmere Island and cabled to shore provides a unique opportunity for measuring the structure and variability of the ambient noise field. Preliminary analysis shows a number of interesting features, which will be correlated with environmental factors to monitor long-term changes in the Arctic oceanography.
Sonic guidance of clinical tubes and catheters
Authors: Wodicka George R.
Published: October 2005

The use of tubes and catheters in clinical therapeutics and diagnostics is growing rapidly. The distal ends of these devices are placed in a variety of tubelike body cavities, and in many cases reside there for extended periods to provide therapy or monitor physiologic function. Sonic techniques to guide the placement, monitor the position, and determine the patency of such tubes or catheters are under development. With a miniature microphone and speaker attached to the proximal end of the tube, a pulse–echo approach is used to gather patency and position information from reflections that arise from within the tube and body cavity, respectively. Design parameters such as the input pulse bandwidth can be optimized based upon the tube and body cavity geometry, including wall properties and branching. This approach has been used to monitor breathing tubes in neonates, and allows placement to within 4 mm of the desired location and quantification of decreased lumen area to within 10% of the actual obstruction. Thus these sonic systems offer the potential to reliably and inexpensively monitor tubes and catheters in the clinical environment.
A noninvasive ultrasonic technique for measuring the dynamic properties of biological tissues
Authors: Finneran James J. , Hastings Mardi C.
Published: October 2005

A noninvasive, continuous wave ultrasonic technique has been used to measure the amplitude and phase of biological tissues in response to acoustically induced vibrations. Tissues exposed to a low-frequency sound field were simultaneously insonified by a low-power, continuous wave ultrasonic source operating at 15 MHz. The motion of the target (due to the low-frequency acoustic field) caused ultrasound scattered by the target to be phase modulated. A separate ultrasonic transducer acted as the receiver. Target amplitude information was obtained directly from the receiver output using a high-frequency spectrum analyzer; phase information was obtained by first demodulating the received signal using a double-balanced mixer and low-pass filter. This paper examines the operation of the measurement system, particularly the phase-measurement technique. The spatial resolution of the ultrasonic measurement system is also examined. The theoretical model for the ultrasonic receiver field coupled existing models for focused piston radiators and for pulse-echo ultrasonic fields. Experimental measurements of the resulting receiver fields compare favorably with theoretical predictions. [Work supported by ONR Grant No. N00014-94-1-0337.]
Subjective evaluation of synthesized signals
Authors: Namba Seiichiro , Kuwano Sonoko , Mizunami Tazu
Published: October 2005

Many kinds of synthesized signals are used in our daily life, for example, warning signals for: vehicles moving in reverse, signaling the end of a washing machine cycle, electric pots boiling, and telephone bells. Sometimes these signals are quite similar and are mistaken for each other. The present investigation was designed to examine what kind of synthesized signals are appropriate for each purpose. The impressions of the timbre of the signals were judged using semantic differentials and identification of the signals was also examined. The results of these subjective evaluations were examined in relation to the physical properties of the signals. Since the impressions of sounds are sometimes different in different cultures, cross-cultural comparison was also made in order to find appropriate signals that can be used internationally.
Identification and characterization of automotive powertrain sound quality attributes
Authors: Feng B. John , Otto Norman C.
Published: October 2005

As vehicle interior noise levels decrease and customer expectations of comfort increase, measures of sound-pressure level are no longer sufficient to characterize the acoustic environment of the vehicle. It therefore becomes important to study other sound attributes that affect customer satisfaction. A procedure has been developed by which customers identify and evaluate the important powertrain attributes, using the semantic differential technique. The attributes fall into two categories, psychoacoustic (loudness, roughness, pitch) and ‘‘halo’’ (expensive, reliable, powerful). By correlating the semantic result with preference data, it can be determined which attributes contribute most to customer preferences. Attributes important for one market segment may be unimportant for another segment. The semantic differential results are correlated to objective sound quality metrics. Comparison of the results across compact, midsize, and luxury market segments will be discussed.
Fundamental issues for optimization of laser sources for generation of ultrasound
Authors: Wagner James W. , Spicer James B. , Murray Todd , Hurley David , Ehrlich Michael
Published: October 2005

Laser-based methods for generation and detection of ultrasonic signals have proven useful in the laboratory and for niche applications in industrial and field measurements. Their further utility continues to be limited by the cost of producing systems with high-detection sensitivity. Overall, laser ultrasonic detection sensitivity can be enhanced in part by insuring that, for a given material and surface condition, the laser source couples optimally and is converted efficiently to elastic energy. Variables to be considered for optimization of elastic wave amplitudes include the laser wavelength, pulse shape, pulse duration, and spatial distribution of the incident laser light. In a laboratory setting in particular, the laser ultrasonic practitioner is able to select propagation angles and modes which contain the greatest amount of ultrasonic energy. Several of these source parameters have been considered carefully for both isotropic and anisotropic materials and may be optimized for maximum ultrasonic energy generated by a pulsed laser source. [Work supported, in part, by the U.S. Air Force Wright Laboratories.]
Ultrafast photothermal and photoacoustic phenomena and their applications
Authors: Sawada Tsuguo
Published: October 2005

Photothermal techniques have been applied to investigate various physical and chemical phenomena occurring both in homogeneous systems like liquid solutions and in heterogeneous systems like surfaces and interfaces of opaque materials. It is expected that improvement of the temporal resolution in observing photothermal phenomena will provide a new investigative tool for a variety of material systems in condensed phases [A. Harata and T. Sawada, Trends Anal. Chem. 14, 504–515 (1995); A. Harata, T. Edo, and T. Sawada, Chem. Phys. Lett. 249, 112–116 (1996)]. These ultrafast photothermal methods may give one information about nonradiative relaxation, heat conduction, and acoustic propagation on a molecular scale. Recent results on ultrafast photothermal investigations will be presented.
Noncontacting resonant ultrasound spectroscopy using a new optical vibration spectral measurement technique
Authors: Telschow Ken , Hale Thomas Chatters
Published: October 2005

A new noncontacting optical method of vibration detection has been developed that utilizes the photorefractive effect in select materials to form an optical ‘‘lock-in’’ amplifier. The method synchronously detects the optical phase shift of an object beam scattered from a vibrating specimen surface. Four-wave mixing and conventional sychronous detection allow measurement of both the vibration amplitude and phase directly as a function of the excitation frequency. Narrow bandwidth detection can be achieved at frequencies from the photorefractive response limit to the reciprocal of the photoinduced carrier recombination time. The method has been implemented using Bismuth silicon oxide providing a resolution bandwidth of about 130 Hz and flat frequency response up to the MHz region. Both synchronous and random vibration excitation methods can be used providing a minimum detectable displacement amplitude of 0.002 nm, at present, with the possibility of further improvement. Application of the method to resonant ultrasound spectroscopy is described using materials of various microstructures and shapes. [Work supported by the U.S. Department of Energy.]
Modeling comodulation masking release (CMR) effects with a single-channel analysis
Authors: Verhey Jesko , Dau Torsten
Published: October 2005

In one of the earliest experiments concerning comodulation masking release (CMR), Hall et al. [J. Acoust. Soc. Am. 76, 50–56 (1984)] measured the detectability of a sinusoidal signal in the presence of a bandpass-noise masker centered at the signal frequency. Thresholds for noise maskers multiplied by a low-pass noise (‘‘comodulated’’ noise) were generally lower than for random noise maskers (‘‘noncomodulated’’ noise). In the literature it is generally assumed that the release is a consequence of an across-channel process. However, it is also argued in the literature [e.g., Schoonefeldt and Moore, J. Acoust. Soc. Am. 85, 273–281 (1989)], that some of the release might be due to a single-channel process, where threshold is determined by a change of the amplitude statistics within one critical band. Simulations with a modulation-filterbank model suggested by Dau et al. [J. Acoust. Soc. Am. 97, 3273(A) (1995)] were performed and compared to experimental data. The effects of varying masker bandwidth, center frequency, modulator bandwidth, and signal duration were examined. It is shown that a single-channel analysis accounts quantitatively for the CMR, indicating that across-channel processing is not necessary for simulating results from this kind of experiment.
Reliability of listeners judging compressed audio
Authors: Precoda Kristin , Meng Teresa H. Y. , Kreiman Jody
Published: October 2005

Listener judgments of audio quality are important in designing minimally distorting compression algorithms, yet the reliability of these judgments is often considered questionable. This experiment examines reliability in detail. Nine listeners heard all possible pairs of 15 signals, the original and 14 generated by compression algorithms, for each of two music samples. They rated the similarity of members of a pair on a seven-point scale—generalizing the idea that high-quality compression produces output highly similar to the original. Mean intralistener agreement within one scale value on repeated ratings was 85.9% (range 66.7%–100%). Interlistener reliability was examined without one version determined to be a perceptual outlier. The average correlation between listeners’ similarity judgments was 0.57 (range 0.28–0.76). However, multidimensional scaling analyses of judgments of one music sample showed listeners attended to very much the same acoustic characteristics, but weighted them differently. When ‘‘similarity’’ (distance) matrices were recreated from the coordinates of the stimuli in each listener’s perceptual space, with no attempt at adjusting dimension weights, correlations between listeners averaged 0.80, compared with 0.62 for the original judgments. Related data transformations may help uncover the reliable core in standard audio quality judgments. [Work supported by JSEP.]
A novel nonlinear system identification technique for modeling cochlear transduction using otoacoustic emissions data
Authors: Krishnan Geetha , Chertoff Mark E.
Published: October 2005

A novel mathematical technique is presented for modeling the linear and nonlinear properties of the cochlear transduction process. This new nonlinear system identification procedure (Bendat, 1990) has several advantages over classical techniques used for analyzing nonlinear systems. With this technique, nonlinear systems may be modeled as a linear system in parallel with any number of finite-memory nonlinear systems. Higher-order nonlinear terms can be more easily computed with this technique than with others such as volterra kernel analysis. Furthermore, a broadband signal such as a transient signal can be used to characterize the cochlea over a wide frequency band. This makes the application of this procedure ideal for modeling cochlear transduction process with transient evoked otoacoustic emissions data. Otoacoustic emissions were recorded in response to finite impulse response (FIR) pulses from 11 subjects with normal hearing. System identification procedures were implemented using third- and fifth-order polynomial models. Based on the input and output records, system properties were estimated. Coherence measures tested the goodness-of-fit of the chosen model. The identified systems were used to predict the systems’ response to two-tone stimuli. The fifth-order model predictions were in better accordance with previously reported data on acoustic distortion products.
An automated off-line procedure for detecting SOAEs from a power spectrum
Authors: Pasanen Edward G. , McFadden Dennis
Published: October 2005

Elimination of observer bias is an important advantage of automating the detection of spontaneous otoacoustic emissions (SOAEs), particularly when exploring differences in SOAE prevalence across subject populations. An objective procedure was developed for detecting peaks in the power spectrum, and estimating the likelihood that such peaks arose from SOAES. The procedure involved multiple passes. First, a smoothed estimate of the spectral noise floor was obtained by replacing spectral regions containing possible SOAEs with linear estimates extrapolated from adjacent frequency regions containing no peak deviations larger than a designated criterion, and then calculating a running average on that smoothed function. Potential SOAEs were then compared with this smoothed noise floor. Based on the variance of points in the original spectrum, the likelihood that any spectral peak did not occur by chance could be estimated. The efficiency and power of this procedure was explored for several combinations of smoothing procedures and criteria for eliminating suspected peaks in the first pass. The implications of varying the criterion for acceptance of a peak as an SOAE are discussed in terms of possible differences in SOAE prevalence in population samples. [Supported by NIDCD Grant DC 00153.]
Pitch identification in masking release paradigms
Authors: Hall Joseph W. , Grose John H.
Published: October 2005

One of three pure-tone frequencies was presented in masking noise on a given trial, and the listener was asked to identify which pitch (low, middle, or high) was presented. The level of the signal varied adaptively. In a masking-level difference (MLD) paradigm, the noise was always interaurally in phase (No) and the signal was either interaurally in phase (So) or was interaurally inverted (Sπ). In a comodulation masking release (CMR) paradigm, a monaurally presented masking noise was either unmodulated or square-wave modulated at a rate of 30 Hz. In each paradigm, the parameter was the frequency separation between the signals. Detection thresholds were also measured for each frequency. When the frequency separation was relatively wide (e.g., 100 Hz or more), detection thresholds were similar to identification thresholds. As the frequency separation decreased, the identification thresholds increased steeply, particularly in the masking release conditions (NoSπ and modulated noise). Results will be discussed in terms of the different cues available in the baseline and masking release conditions, and in terms of why those cues may be less informative in identification than in detection conditions. [Work supported by NIDCD.]
Mechanical cochlear-partition parameter variation in a box-cochlea model
Authors: Wilson Timothy A.
Published: October 2005

Place and frequency responses were determined for one-, two-, and three-dimensional fluid motion in a box-cochlea model with a stiffness-damping-mass cochlear-partition representation having exponentially decreasing stiffness, exponentially varying damping, and constant mass densities per unit length. The rational-function frequency-domain impedance for that partition model was represented as a function of a single variable, the generalized cochlear place, a linear combination of cochlear place, and the logarithm of stimulus frequency. When cochlear mass density was relatively large, the best generalized cochlear place was a constant, with the result that the slope of the linear best (ordinary) place versus log-frequency graph depended only on the rate of partition stiffness decrease; in that case, three orders of magnitude of presented stimulus frequency mapped best places over the entire cochlear length only when the stiffness changed by six orders of magnitude. When cochlear mass density was relatively small, the slope of best generalized place versus log frequency could be made more negative for the same stiffness rate, and the same range of best-place versus frequency variation was achieved with less stiffness change along the cochlear length.
Evoked hemodynamic response of the auditory cortex measured by magnetoencephalography and near-infrared spectroscopy
Authors: Ohnishi Michihiro , Masaki Shinobu , Honda Kiyoshi , Kusakawa Naoki , Hayashi Noboru , Hirao Konomu
Published: October 2005

Near-infrared (NIR) spectroscopy, a noninvasive optical method for monitoring hemodynamics, was applied to record the evoked auditory response of the human brain. A newly developed system consists of a halogen light source, an NIR spectroscope, and optical fiber bundles for the light and signal probes. The system detects the amount of oxygenated and deoxygenated hemoglobin in the area of the local brain located beneath the probes. In this study, magnetoencephalography (MEG) and NIR spectroscopy were used to observe the blood component change near the auditory cortex. The sound stimulus of a 1-kHz sine wave at 70 dB SPL was used to elicit evoked responses in both experiments. The active region of the auditory cortex was determined by the MEG, and the same area was examined by the NIR system. The results from NIR experiments showed significant increases of the total hemoglobin and deoxygenated hemoglobin, indicating both blood flow and oxygen consumption at the cortical area increased in response to sound stimuli. The NIR recordings at several different depths revealed that the increases of local blood flow and oxygen consumption were observed only at depths ranging between 20 and 30 mm from the scalp.
Effects of narrow-band contralateral noise on absolute thresholds in the Japanese macaque
Authors: Smith David W.
Published: October 2005

Detection thresholds for pure-tone stimuli were measured in quiet and in the presence of octave-band noise presented simultaneously to the contralateral ear in a group of Japanese macaques. Subjects were trained to respond to the presence of 1-s sinusoids presented to the ipsilateral ear in a go/no-go procedure. All stimuli were presented through insert earphones. Pure-tone stimuli were 1-s in duration and ranged from 125 Hz to 16.0 kHz. Continuous noise stimuli were octave bands, centered at the ipsilateral test-tone frequency, presented at levels of 0 to 70 dB above threshold for the center frequency stimulus. While some variability was evident across subjects, both in the frequency distribution and magnitude (as a function of contralateral noise level) of threshold increases, all subjects exhibited an orderly increase in thresholds in the presence of contralateral noise. Maximum increases in threshold were evident between 1.0 and 4.0 kHz. Measurable threshold elevations were evident at noise levels as low as 20 dB above threshold and, in general, increased systematically with contralateral noise levels. Decreases in threshold were evident at low-noise levels. The characteristics of the measured threshold shifts are similar to efferent suppression effects which have been reported in the physiological literature. [Work supported by NIH.]
The effect of electrical cochlear stimulation on the autonomic nervous system
Authors: Kamekura Nobuhito , Kamada Tsutomu , Kobayashi Hiroshi , Matsushima Junichi , Miyoshi Shigeki , Sakajiri Masatsugu , Uemi Norihiro , Ifukube Tohru
Published: October 2005

Extracochlear electrical stimulation seems to induce relaxation with temporal tinnitus relief in patients. Electrical treatment was accompanied by increased cutaneous blood flow in the digits, although there was no clear tendency among tinnitus suppression, relaxation, and blood flow. To elucidate the effect on the autonomic system in the experimental animals, the blood pressure, the heart rate, and sympathetic and parasympathetic nervous activities were examined in rats and rabbits. Various stimuli such as square pulses, biphasic pulses, or sine waves were given to the cochlea with the electrodes on the round windows. The arterial blood pressures decreased in most cases in rabbits, while they decreased, increased, or represented biphasic changes in rats depending upon the type of stimulation. The autonomic activity seemed to follow the drifts in blood pressure, but the results did not show which autonomic system contributed to the drifts clearly. The effect of electrical stimulation on the autonomic nervous system was quantitatively analyzed and the effects of the stimulus waveforms and frequencies on the autonomic reactions were compared. One would look for the stimulating conditions suitable for tinnitus suppression or hearing improvement without the presumed side effects of stimulation.
Formant transition duration and hearing impairment
Authors: Turner Christopher W. , Smith Sarah J. , Aldridge Patricia L. , Stewart Suzanne L.
Published: October 2005

Listeners with sensorineural hearing loss often have difficulty in discriminating stop consonants even when the speech signals are presented at high levels. One hypothesis offered to explain their deficit is that they cannot use the information contained in the rapid formant transitions as well as normal-hearing listeners. Sets of three CV syllables were synthesized corresponding to /ba, da, ga/ with formant transitions for each set ranging from 5 to 160 ms in duration. The listener’s task was to identify the presented consonant in a three-alternative, closed-set response task. The results for normal-hearing listeners showed nearly perfect performance for transitions of 20 ms and longer, whereas the shortest transitions yielded poorer performance. The hearing-impaired listeners showed large deficits compared to the normals for the longer duration transitions only, for shorter durations their performance was slightly less than, and sometimes equal to that of normals. These results would appear to contradict the original hypothesis, and supports recent psychoacoustic results of Summers and Leek [J. Acoust. Soc. Am. 97, 3825–3832 (1995)]. Therefore, increasing the duration of formant transitions for hearing-impaired listeners is not a helpful solution to their difficulties. [Work supported by NIDCD Grant DC 00377.]
Modeling auditory processing of AM detection and discrimination for sensorineural hearing-impaired listeners
Authors: Derleth Peter , Dau Torsten , Kollmeier Birger
Published: October 2005

Experimental data and model predictions on amplitude-modulation detection are presented. The sensorineural hearing-impaired listeners had a flat hearing loss in the stimulated frequency region and showed recruitment. Normal and impaired listeners showed the same TMTF for different noise carriers (fc=2 kHz; bandwidth=16 Hz, 200 Hz). The presentation level was equal in units of categorical loudness scaling. Corresponding simulations were performed with a modified version of the modulation filterbank model [Dau et al., J. Acoust. Soc. Am. 97, 3273(A) (1996)], in which an increased absolute threshold and a loss of compression were introduced. The model can account for the data, using the same parameters for the scaling of the modulation filters, indicating that frequency selectivity in modulation is comparable to that of normal-hearing listeners. In the case of random noise as the carrier, thresholds are determined by the inherent statistics of the carrier. Therefore no effect of reduced compression can be expected, because test modulation and inherent carrier modulation are treated the same way. Using deterministic carriers, thresholds are only determined by internal noise added after the compression stage in the model. Therefore, a further experiment with a sinusoidal carrier was performed to examine the role of compression and internal noise on amplitude-modulation discrimination.
The experiences of untrained subjects listening to virtual sounds
Authors: Simpson Brian D. , Hale Dennis W. , Isabelle Scott K. , Gilkey Robert H.
Published: October 2005

Anecdotal reports from casual listening in auditory virtual environments suggest that the ‘‘compellingness’’ of virtual sounds depends not only on the fidelity of the synthesized environment but also on the characteristics of the real environment in which the sounds are presented (i.e., even though the sounds are presented through headphones, nonauditory characteristics of, or previous auditory experience in, the listening environment may influence the perception of the virtual environment). In particular, when the synthesized environment ‘‘matches’’ the listening environment the auditory experience can be strikingly realistic. To quantify this phenomenon, binaural recordings of ‘‘everyday’’ sounds were made through the KEMAR manikin in each of three real rooms (16 to 194  m3). A 3×3 between-subjects design was used; each untrained subject listened to one of the three recordings over headphones while seated with eyes open in one of the three rooms (the recording might or might not match the room). The effects of recording room size and listening room size on objective measures of 3-D localization accuracy (azimuth, elevation, and distance) and on subjective measures of the listening experience (sense of presence, sound quality, and vividness) are evaluated. [Work supported by AFOSR F49620-95-1-90106, NRC, and Wright State University.]
Recent advances in monitoring techniques for speech production
Authors: Niimi Seiji , Kumada Masanobu
Published: October 2005

In recent years, various techniques have been developed for monitoring the dynamics of articulatory and phonatory systems. For the articulatory system, x-ray, ultrasound, magnetic field, etc., have been utilized. Among them, MRI is most preferable because of high spatial resolution and safety, though it has to be supplemented by other techniques due to its limited temporal resolution. Recent results on tagged MRI of the tongue will be discussed. For the phonatory system, high-speed monitoring of vocal fold vibration is quite important in the study of voice production in normal as well as in pathological cases. A high-speed imaging technique up to 9000 frames per second has been developed, and is being used in this laboratory for recording fiberscopic images of the vocal fold vibration during running speech, singing, and other vocal activities. Aerodynamic measurement is also important for studying various modes of phonation. The difficulty of measuring the transglottal pressure is being overcome by the use of a miniaturized pressure transducer which can be inserted into the subglottic space. These new techniques provide a lot of new data as well as allow new interpretation of already existing data.
Voice source properties of the speech code
Authors: Fant Gunnar , Kruckenberg Anita
Published: October 2005

A challenge for speech research and technology is to advance the knowledge-based approach. A neglected aspect of the speech code is the dynamics of the voice source and how it integrates in an overall production model. In the past, research into the human voice source has mostly been concerned with individual voice qualities. Knowledge of segmentally and prosodically induced variations and of waveform finestructure is now developing but the picture is still incomplete. An elaborated model of speech production integrating supraglottal, glottal, and laryngeal articulations and pulmonary activity is needed. The status of the art with a preliminary overview of contextual and prosodic factors that influence the voice source within a sentence frame will be discussed. One important factor is the degree of articulatory emphasis and thus of VT–source interaction as found in hypo- versus hyperspeech. Novel data obtained in Stockholm on subglottal pressure variations measured directly through a tracheal puncturing probe have contributed insight into intensity-F0 relations at covarying subglottal pressures. These data apply to global intensity declination within a phrase and to local regions of prominent F0 peaks where the normal rising relation of intensity with increasing F0 is canceled or reversed.
The roles of category boundaries and prototypes in the perception and production of speech and language
Authors: Fujisaki Hiroya
Published: October 2005

One of the fundamental issues in the study of speech perception is to model the cognitive process of phonetic judgment and to infer the underlying mechanism. Experimental results on categorical judgment as well as on judgment of typicality do not attest to the existence of a single prototype for each phonetic category, but rather to the existence of a continuum of exemplars, viz., a region of high typicality, whose shape is not necessarily convex, in the multidimensional space of acoustic/phonetic parameters. The neural mechanism for separating such a region from others is based on thresholds (i.e., boundaries), but not on distances from a single point of reference. On the other hand, the production of a specific phone in a given context by a single speaker displays a very sharp distribution, which indicates the existence of a speaker-specific prototype in speech production. Such an apparent asymmetry of production and perception is shown to exist also in the use of language, and can be explained in the light of a general underlying mechanism for coding and decoding the information. Implications of these results to spoken language processing will also be discussed.
Matched-mode localization using conventional and cumulant-based MUSIC algorithms in a real, complex shallow-water environment
Authors: Pierce David D. , Chiu Ching-Sang , Therrien Charles W. , Miller James H.
Published: October 2005

Matched-field processing (MFP) and matched-mode processing (MMP) are two popular techniques for passively localizing an underwater acoustic emitter in range and depth. One major drawback of these techniques has been their sensitivity to uncertainty concerning the acoustic environment. Several methods of addressing this phenomenon have been proposed in the literature, with varying degrees of success. Achieving high-quality location estimates remains a problem except in simple range-independent experiments or numerical simulations. In this study, an approach is demonstrated for robust, accurate emitter localization in a highly range-dependent real environment using MMP. The main factors contributing to successful localization are: (1) use of the high-resolution multiple signal classification (MUSIC) algorithm, which performs well even when only a few robust modes can be obtained by mode filtering; and (2) use of an acoustic propagation model incorporating mode coupling, which is able to generate accurate replica fields in a strongly range-dependent environment. The use of a cumulant-based MUSIC algorithm for reducing noise effects is also investigated.
Incoherent and coherent aeroacoustic wideband direction finding algorithms for ground vehicles
Authors: Pham Tien , Sadler Brian M.
Published: October 2005

Incoherent and coherent wideband MUSIC algorithms are used to perform high-resolution direction of arrival (DOA) estimation on ground vehicles that emit broadband acoustic signatures (e.g., turbine engine vehicles). The algorithms are analyzed with experimental data collected from a small baseline, circular array [T. Pham and B. Sadler, 8th IEEE SSAP Workshop (1996)]. GPS ground truth location data are used to evaluate and compare performance. The nonstationary nature of the detected signals, due to environmental and source variations, requires adaptive selection of a set of operating frequencies. In the incoherent method [M. Wax, T. Shan, and T. Kailath, IEEE Trans. ASSP32 (1984)], MUSIC is processed individually at each narrow-band frequency, and results averaged to estimate the DOAs. In the coherent method [H. Wang and M. Kaveh, IEEE Trans. ASSP 33 (1985); also J. Krolik, Adv. Spect. Anal. Array Proc., Vol. II (P-H, 1991)], correlation matrices at different frequencies are combined at each look direction to form a ‘‘focused’’ correlation matrix, and MUSIC is performed on the focused correlation matrix to estimate the DOAs. Processing gain is evident for both methods compared to narrow-band MUSIC. Computational complexity is dominated by eigendecomposition; it is a function of the number of frequency components for the incoherent case, and the number of look angles for the coherent case.
Neural network classification of air and land vehicle acoustic signatures
Authors: Brooks Lowell W. , DeMetz Fred C.
Published: October 2005

The temporal and spectral features of the acoustic signatures of aircraft and highway vehicles are used to train two types of neural networks. The detection and classification performances of the networks are then evaluated using an independent set of data. The types of networks evaluated are (i) feedforward with backpropagation training, and (ii) probabilistic nets with maximum-likelihood training. The results demonstrate accurate classification as to type of vehicle during aircraft takeoff or under load conditions for heavy trucks, depending on the signal-to-noise ratio. Neural nets appear to offer a promising vehicle classifier for monitoring airport and highway traffic for compliance with noise regulations.