bluuebunny/crossref_metadata_2025 huggingface dataset:
Frequency dependence of phase shifts in human calcaneus in vitro.
Authors: Wear Keith A.
Published: March 2010

If dispersion in a medium is weak and approximately linear with frequency (over an experimental band of frequencies), then it can be shown that the constant term in a polynomial representation of phase shift as a function of frequency can produce errors in measurements of phase velocity differences in through-transmission, substitution experiments. A method for suppressing the effects of the constant phase shift in the context of the single-wave-model was tested on measurements from 30 cancellous human calcaneus samples in vitro. Without adjustment for constant phase shifts, the estimated phase velocity at 500 kHz was 1516 +− 6 m/s (mean +− standard error), and the estimated dispersion was −24 +− 4 m/sMHz (mean +− standard error). With adjustment for constant phase shifts, the estimated mean velocity decreased by 4–9 m/s, and the estimated magnitude of mean dispersion decreased by 50%–100%. The average correlation coefficient between the measured attenuation coefficient and frequency was 0.997 +− 0.0026 (mean +− standard deviation), suggesting that the signal for each sample was dominated by one wave.
Computing bone fragility using transient ultrasonic waves.
Authors: Gilbert Robert P. , Guyenne Philippe
Published: March 2010

In this paper, we determine bone fragility of cancellous bone in vivo by using a simple one-dimensional model of a muscle-cortical bone-cancellous bone sample, by measuring the refracted sound when the sample is exposed to impulses from a transducer. From these data, the bone parameters can be determined. Such a procedure is referred to as solving an inverse problem. A transducer is placed on one side of the skin of the member to be interrogated and a receiver is place diametrically opposed in contact with the member. The muscle will be modeled as an elastic material, whereas the the corrical and cancellous bone may be modeled by modified Biot equations albeit with different porosities. However, because of the known frequency dependence of the viscous term, we propose that the standard Biot equations, which are not well posed, be replaced by adding a convolution term. The form of the kernel b(t) in the convolution integral is suggested by frequency domain representation of the dissipation term in the Biot–Johnson–Koplik–Dashen model.
Electret condenser microphones.
Authors: West James E.
Published: March 2010

For more than 4 decades electret microphones have dominated most transducer applications in the audio frequency range for three main reasons, linearity, simplicity, and they are inexpensive. E. C. Wente invented the condenser microphone in 1916 searching for a replacement for the carbon microphone used in most telephones, but it was impractical until the high-voltage battery was replaced with a permanently charged polymer. Electret microphones have been the workhorse for most microphone applications including microphones for professional sound level measurements, telephones, hearing aids, professional audio recordings, games, and toys. Recent worldwide estimates of the electret microphone market suggest that over 2 billion units are made annually. Electret condenser microphones are available in a variety of sizes from very small (a few millimeters in diameter) to about 2 in. in diameter. The use of multiple microphone elements in arrays and the use of digital signal processing provide interesting prospects for improving signal-to-noise. A variety of microphone arrays and some important historical facts will be discussed in this talk.
Prepolarized measurement microphone properties: 30 years of experience.
Authors: Gramtorp Johan , Olsen Erling Sandermann
Published: March 2010

30 years ago, Brüel & Kjær launched the world’s first prepolarized condenser measurement microphones. These microphones combined the high-performance construction of the normal measurement microphones with a polarized electret layer placed on the back electrode. Since then, prepolarized microphones have gained an increasing share of the measurement microphone market. Prepolarized microphones are flexible in use because they do not need the external polarization, in particular, they can be used with constant current supply preamplifiers (Deltatron, ICP) that are widely used for many purposes in the industry. In order to be used for measurement purposes, the microphones and thereby the electret polarization must be highly stable and reliable. That is indeed achievable, and that is the main emphasis of this paper. A short historical overview of the development of the Brüel & Kjær prepolarized microphone types is given. The properties of prepolarized microphones are discussed and compared to those of externally polarized microphones. These will include stability with respect to time, temperature, humidity and the chemical environment, as well as distortion and dynamic range.
Ringing the changes: The potential for push-pull electret transducers in cell phones.
Authors: Mellow Tim , Kärkkäinen Leo
Published: March 2010

Electret microphones are very common in mobile devices, but are invariably single-ended in configuration. For moderate dynamic conditions this is not a problem, but under high-sound pressure levels, better linearity can be obtained using a push-pull arrangement. Recent progress in nano-porous electret materials also opens up possibilities for electret loudspeakers. Such loudspeakers would offer potential benefits such as high-conversion efficiency, no magnetic field, and an ultra-thin package. However, there are product integration challenges. The authors describe a simulation model of a push-pull electret transducer and discuss the conditions under which it is linear. It turns out that the relationship with a non-electret electrostatic transducer is relatively simple.
Airframe noise flight test measurements using an electret condenser microphone array.
Authors: Humphreys William M. , Lockard David P. , Khorrami Mehdi R.
Published: March 2010

An extensive collection of airframe noise flight test data has been acquired for a regional jet class of aircraft using a ground-based electret condenser microphone phased array system. The measurements were conducted at the Wallops Flight Facility for the NASA Aeronautics Research Mission Directorate Fundamental Aeronautics Program. Gulfstream G450 and G550 aircraft were used as the test beds for the study. The phased array employed for the measurements consisted of a spiral clustering of 167 Panasonic WM-61A quarter-inch, omni-directional electret microphones. The innermost 49 sensors of the array were flush mounted on an aluminum plate with the remaining sensors flush mounted on individual 16-in. circular Plexiglas platters. The aluminum plate and platters were deployed on the overrun area of runway 4 at Wallops. Airframe noise measurements were performed with the aircraft flying a racetrack pattern simulating the approach path to landing as it passed over the array. A distributed data acquisition system was used to acquire simultaneous time histories from the array and was synchronized with instrumentation onboard the aircraft. This presentation will discuss the design of the phased array, the calibration and deployment of the microphones, a sampling of the measurements obtained from the system, and lessons learned from the deployment.
A method for free-field transfer calibration of electret microphones at ultrasound frequencies in air using a stable broad band ultrasound source.
Authors: Campanella Angelo J.
Published: March 2010

A rotating turbulence screen wheel sound source, now in use in the air ultrasound instrument industry for product quality control, provides a steady source of broad band ultrasound. This source is broad band through at least 100 kHz. It is calibrated as a secondary standard, using a one-quarter inch condenser microphone, grid removed, normal incidence, and corrected for free-field conditions per the microphone manufacturer free-field response. Sound pressure amplitude is measured at a distance of 500 mm and from 1 to 100 kHz in 1-kHz FFT bins. Calibration site sound absorption by humid air is calculated to correct the emitted sound pressure values to be reinserted for the user test site air temperature and humidity. Transfer calibrations of two electret condenser microphones are shown. Microphone phase response measurement methods will also be investigated and discussed.
Acoustic field of air-coupled transducers made with electroactive polymer foams.
Authors: Satyanarayan Louis , Berthelot Yves
Published: March 2010

Electroactive films made of soft cellular polypropylene (PP) foams containing electric charges have high piezoelectric coupling coefficient and good matching impedance with air over a broad frequency range. They offer interesting capabilities in the design of new air-coupled ultrasonic transducers. A prototype transducer has been designed and tested in our laboratory. The amplitude displacement on the surface of the transducer was measured by laser vibrometry by vibrating the PP foams with a continuous wave sinusoidal excitation. The transducer was mounted with a needle-type PVDF hydrophone on a high-precision polar C-scan apparatus. The receiving hydrophone was translated along and across the axis of the source transducer and data were recorded. A quantitative characterization of the acoustic field of the transducer was carried out by plotting the on-axis and cross-axis variations in the pressure field, at various separation distances between the source and the receiver in pulsed transmit-receive mode. The results indicate that simple and relatively efficient air-coupled ultrasonic transducers can be made with electroactive polymer foams. [Work supported by CNRS and by the Conseil Rgional de Lorraine.]
Instrumental analysis of extended saxophone techniques for live electronics.
Authors: Braasch Jonas , Van Nort Doug
Published: March 2010

The development of automated music transcription systems focuses predominantly on polyphonic musical instruments. At the same time, the analysis of a monophonic instrument is usually much simpler wherein pitch, loudness, and duration of individual notes may be tracked robustly. When using extended techniques, however, many more parameters than the aforementioned three can be meaningful for the performed music. This paper explores the challenges that extended techniques pose for music recognition systems using the example of the saxophone. The goal is to correctly identify extended techniques over the whole range of the instrument, including subtones, multiphonics, growl, and other voice-enhanced tones, as well as tones where the reed is supported by the lower teeth. The feature analysis is based on cepstrum, spectral moments, pitch, and roughness, among other features. A hidden-Markov model is used to recognize the trajectory of the various extended techniques based on the given feature space. Finally, it is demonstrated how the recognizer can be integrated into an intelligent live electronics system to control its parameters. For example, the characteristics of a virtual acoustic enclosure (room size, reverberation time, etc.) can be adapted this way. [Work supported by NSF 0757454.]
Simulating, mirroring, morphing, and blurring acoustic space as an extended instrumental technique.
Authors: Oliveros Pauline
Published: March 2010

The paradigm for musical performance is the sound of musical instruments in a fixed acoustical space. Ordinarily space does not shift and change during performances. An ideal space for performing classical composers such as Mozart is the Shaffy Theater in Amsterdam with a relatively short reverberation time 1.5 s. Instruments couple to the space, and space is part of the instrument and vice versa. Players develop special relationships with favorite spaces. After experiencing many different types of acoustics this leads to the desire to change the space during performance in order to hear the different qualities that emerge from spatial coupling. Spatial progressions can be made so that rather than moving sound in space, space can be moved around the sound and produce a variety of qualities.
Extending the acoustic ensemble through spectral and temporal transformations in real-time.
Authors: Van Nort Doug
Published: March 2010

The paradigm of live performance mixing acoustics and electronics has predominantly focused on simple background “tape music,” human players performing highly structured sample-based music (e.g., using the ABLETON LIVE software), or reactive systems that respond to player qualities such as timing, pitch, and so on. In this talk I will present my approach to improvised “laptop performance” that focuses on the transformation of acoustic players in real-time. Rather than simply altering the acoustic content in the manner of an effect processor, the goal is to capture notes and phrases in short-term memory and to re-articulate the material so that it presents a new gestural inflection and timbral content that can be completely novel or suggestive of other players’ sound. The system presented utilizes a hybrid system combining spectral analysis and feature extraction with block-based temporal processing and a feedback delay network. The interaction paradigm of “scrubbing” the intermediate time/frequency representation is used to generate the ultimate output. The result in an ensemble context is an extended palette that can “keep up” with the musical dialog while eliciting the subtle textural qualities of acoustic players. [This work was supported by NSF Grant 0757454 and CIRMMT/McGill University Fellowships.]
The digital audio player as aleatory musical instrument.
Authors: Di Angelo Adam
Published: March 2010

The digital audio file player is an ideal instrument for the performance of aleatory music. The near-instantaneous seek time coupled with random track playback yields the possibility of nonlinear musical arrangements. In this is the potential to create unique and unrepeatable listening experiences—music that constantly renews and reinvents itself, free from the static limitations of the recorded medium. This study examines the history of chance as a compositional and performance tool from 18th century musical dice games to contemporary stochastic methods. It presents examples of works which are designed to be structurally random and outlines the process of employing the digital audio player in the production of variable music. Through such considerations, the role of the conventional mp3 player can expand from passive playback device to become an engine integral to a new kind of listening.
The physics of pitch bending on the vibraphone.
Authors: Worland Randy
Published: March 2010

The vibraphone is a member of the keyboard (or mallet) percussion family, with tuned aluminum bars typically spanning three chromatic octaves (F3–F6). Extended performance techniques include the use of harmonics and bowing of bars, as well as the pitch bending effect discussed here. After first striking the middle of a bar with a soft mallet in the normal manner, the pitch bend is obtained by pressing a hard mallet onto the bar at a nodal point and then sliding it away from the node as the note sustains. The audible result is a descending pitch, typically of about one semitone. Experimental data on frequency versus location of the hard mallet along the bar are presented. The mass, hardness, and orientation of the sliding mallet relative to the bar are discussed. Interferograms of the lower frequency modes of the bar are also shown. Results are interpreted using a mass perturbation model.
Image method for acoustic reflections from a wedge.
Authors: Bhatta Ambika , Denis Max , Raspopvic Mira , Thompson Charles
Published: March 2010

In this paper we will present the analysis of reflected waves of a point source from a two dimensional wedge. Exact method of images is used to calculate the position and strength of image sources. It is shown that the traditional fractional order Bessel function series for the total pressure can be replaced by quadratures of elementary functions.
Transition matrix (T-matrix) of circular cylinders embedded in a viscous fluid.
Authors: Reyes-Ayona Edgar , Torrent Daniel , Sanchez-Dehesa Jose
Published: March 2010

The present study concerns the scattering of a plane sound wave by a cylindrical scatterer of circular cross section. An infinitely long cylinder embedded in a viscous fluid is here considered. The linearized equations of viscous fluid and the scalar and vector potentials are employed to obtain the T-matrix for two different types of material cylinders: (i) a rigid cylinder and (ii) a fluidlike cylinder with losses. Both cases are also studied and discussed in the limit of long wavelengths. [Work supported by MICIIN of Spain and ONR.]
A numerical model for acoustic scattering from an inhomogeneous medium.
Authors: Denis Max , Tsui Jing , Piper Jessica , Chandra Kavitha , Thompson Charles
Published: March 2010

In this work, a numerical method for evaluating the scattering of acoustic waves from an inhomogeneous medium is presented. The internal resonant and multiple scattering effects are modeled by the Padé approximants of a perturbation expansion of the contrast properties between the medium constituents. Due to the local divergence in the Padé approximant solution, a local expansion method about any convergent solution is performed. This allows one to extend the numerical model to higher frequencies. The results show good agreement for geometrically simple scatterers. Based on the simulations, analyses are made of inhomogeneous media micro-structural effects on the scattered field. Of particular interest is the frequency dependence relationship.
Single-channel time reversal with Lanczos iterations.
Authors: Waters Zachary J. , Barbone Paul E.
Published: March 2010

Recent investigations of array-based time reversal techniques employing Lanczos iterations have demonstrated convergence properties superior to equivalent methods using power iterations [A. A. Oberai et al., “Efficient time reversal by Lanczos iterations,” J. Acoust. Soc. Am. 123, 3596 (2009)]. In numerical scattering simulations, time reversal techniques developed for a single-channel transducer are applied to study echoes from a thin-walled spherical shell. When power iterations are employed, a narrowband waveform corresponding to the dominant scattering mode of the target is selected. Lanczos iterations converge to this waveform in fewer transmissions and simultaneously identify additional higher-order target scattering features. The performance of both techniques is studied in the presence of varying levels of stochastic and deterministic noise. [Work supported by the Office of Naval Research.]
Energy flux streamlines vs the alternatives for the visualization of energy coupling inside and outside the surface of an ensonified spherical acoustic lens: Preliminary results.
Authors: Dean Cleon E. , Braselton James P.
Published: March 2010

Energy flux streamlines are compared with alternative presentations of the energy flux vector field to visualize energy coupling inside and outside the surface of an insonified spherical acoustic lens. The emphasis is on the energy flux streamline as a natural bridge between necessarily approximate ray solutions and full-fledged wave solutions. The present work is the wave solution for the experimental work performed by Kendez C. Parker and Cleon E. Dean also being presented at this meeting that primarily used ray methods in its theoretical analysis; thus a comparison between energy flux representations and ray analysis results is made. Difficulties in the interpretation of the energy flux streamline representation due to the spherical geometry of the scatterer are noted.
Sound control by temperature gradients.
Authors: Sanchez-Dehesa Jose , Angelov Mitko I. , Cervera Francisco , Cai Liang-Wu
Published: March 2010

The acoustic mirage, like the optical mirage, is a consequence of temperature gradients existing in the air above Earth surface. Here, we report experiments showing that temperature gradients can be used to collimate an airborne ultrasonic beam. A very simple system consisting of two glass tubes heated by an internal resistive coil is used to create a temperature gradient producing the desired effect. Numerical simulations are also performed and results are in agreement with the experimental findings [Work supported by MICIIN of Spain.]
Characterization of a lossy elastic plate thanks to the relationships between its reflection and transmission coefficients.
Authors: Derible Serge , Tinel Alain
Published: March 2010

The normal reflection and transmission coefficients of a lossy elastic plate are measured and associated according to the outcomes of resonance scattering theory (RST) and S-matrix theory (SMT). The sum and the difference of those coefficients provide valuable tools, the transition terms, we use to characterize the plate. The absorption in the plate can be expressed via different manners recalled in the following. By means of a dimensionless loss factor, the velocities of the longitudinal and transverse waves of the studied aluminum plate are no more real but complex. The width of a resonance is the sum of two parts: the elastic width and the absorption one. In the vicinity of a resonance frequency, the transition terms obey the Breit–Wigner resonant form which makes it possible an easy determination of the two parts of the resonant width. There is a good experiment/theory agreement when an imaginary part is added to the Lamé coefficients of the aluminum plate. The loss factor is of the order 5.10–4. On the whole, the effect of the attenuation rises with frequency and reaches a maximal importance at the vicinity of a resonance frequency.
Acoustical probe of the density fluctuations in a binary fluid mixture.
Authors: Osterhoudt Curtis F. , Sinha Dipen N.
Published: March 2010

Silicone oil was found to be fully miscible in “segregated” perfluorinated hydrocarbon (Novec fluid, by 3M) above a critical temperature slightly above room temperature and at atmospheric pressure. This constitutes a binary fluid mixture, and these have long been used as examples of continuous phase transitions, especially in regard to the phenomenon of critical opalescence [T. Andrews, Philos. Trans. R. Soc. London 159 (1869) and M. Smoluchowski, Ann. Phys. 21 (1906)]. During a phase transition in these mixtures, the fluids become spatially segregated, and the density fluctuation correlations are known to span a large number of length scales [L. Landau and E. M. Lifshitz, Statistical Physics, 3rd ed., Part 1]. This behavior is reflected in massive fluctuations in sound speeds and viscosities in the mixtures near the critical point, and the resulting scattering behavior is probed by actively acoustically interrogating the mixture with techniques including swept-frequency acoustical interferometry and pulse-echo measurements. It is shown that with higher-power acoustical probes, the transition from unary to binary mixture may be prevented or enhanced. [This work was supported by Chevron USA.]
The characteristics of thunder on Titan.
Authors: Petculescu Andi
Published: March 2010

Even though Huygens has not detected acoustic signatures of Titanian lightning, electric discharge processes are thought to be very plausible in Titan’s thick atmosphere dominated by nitrogen and methane. When correlated with electromagnetic signatures, thunder will very likely confirm the occurrence of lightning, both during a probe’s descent and/or once it has landed. The presentation has two parts. In the first part, the mechanisms of Titanian thunder generation are discussed, in the context of the most up-to-date predictions of lightning on Titan, as well as relevant Cassini-Huygens data. In the second part, the propagation of thunder in Titan’s troposphere is addressed. Here, the structure, composition, and ambient conditions of the satellite’s atmosphere will be presented and accounted for. Current Titan lightning models predict cloud-to-ground discharge channels of ∼20 km. Long-range Titanian thunder simulations are developed based on the far-field superposition of approximately 7000 N-waves arising from 3-m “elementary” sources. Two lightning geometries will be considered: linear and tortuous. (Tortuous channels are synthesized as a Gaussian distribution in the orientations of the 7000 elementary sources.) For illustrative purposes, to show the effects of atmospheric filtering, the thunder waveform described above will also be propagated in Venus’ atmosphere.
Effects of training in increment-detection and intensity-discrimination tasks.
Authors: Jesteadt Walt , Patra Harisadhan , Krivohlavek Melissa , Rutledge Cynthia
Published: March 2010

Subjects use different cues when asked to detect a brief increment in a longer duration tone (increment detection) than when asked to discriminate between two tones of the same duration that differ only in level (intensity discrimination). To determine whether experience in one task would generalize or interfere with performance in the other, a group of six subjects completed 4000 trials in an increment detection task, followed by 800 trials of intensity discrimination, and final 800 trials of increment detection. A second group of subjects was tested in the reverse pattern, beginning with 4000 trials of intensity discrimination. The increment or signal was a 90-ms, 4-kHz tone. The pedestal or standard was a 490- or 90-ms, 4-kHz tone presented at 70-dB sound pressure level. A two-interval, forced choice adaptive procedure was used to estimate the level of the increment required for 71% correct. The results showed modest improvement over time for two subjects in increment detection and one in intensity discrimination, but at least one subject showed the opposite effect in each task. There was no generalization or interference between the two tasks and no correlation between increment detection and intensity discrimination across the 12 subjects. [Work supported by NIH.]
The relative contribution of adaptation and temporal integration in forward masking.
Authors: Nagaraj Naveen N. , DiGiovanni Jeffrey J.
Published: March 2010

Forward masking has been studied extensively, but the mechanism underlying forward masking is still being debated. The aim of this study was to understand the relative contribution of adaptation and temporal integration on forward masking. In the first experiment, gap detection thresholds were measured for symmetrical and asymmetrical noise markers ranging from 10- to 30-dB SL in 5-dB increments. Further threshold measurements were made while keeping the level of the noise marker before the gap at a constant 30-dB SL while varying the level of noise marker after the gap between 10- and 25-dB SL. Noise marker sets with gap detection threshold of 20 ms or less were selected for the second experiment. In the second experiment, participants adjusted gap durations within an asymmetrical noise marker to be perceptually equal to that of symmetrical noise markers. The results will be discussed in the context of a temporal integration model, an adaptation model, and a hybrid model including both temporal integration and adaptation components.
Perceived continuity and pitch shifts.
Authors: Plack Christopher J. , Watkinson Rebecca K.
Published: March 2010

If a noise burst is introduced into the gap between two brief complex tone bursts, pitch discrimination improves to the level observed when the bursts really are continuous. In the experiment described here, it was shown that perceived continuity also affects sensitivity to envelope phase across the gap. Two complex tone bursts with unresolved harmonics and five regular envelope peaks (“pitch pulses”) were presented sequentially, separated by a gap of two periods. When the relative phases of the two bursts were varied, such that the inter-pulse interval (IPI) across the gap was varied, the pitch of the whole sequence was little affected, consistent with previous results suggesting that the pitch integration window may be “reset” by a discontinuity. However, when the gap between the two bursts was filled with a noise with the same spectral envelope as the complex, variations in IPI had substantial effects on the pitch of the sequence. It is suggested that the presence of the noise causes the two tone bursts to appear continuous; hence resetting does not occur, and the pitch mechanism is sensitive to the phase discontinuity across the gap.
Frequency-specific measurements of the temporal modulation transfer function using narrowband noise.
Authors: Shen Yi , Lentz Jennifer
Published: March 2010

Interpretation of frequency-specific measures of temporal acuity is often confounded by the coexistence of spectral and temporal cues. The current study presents a new experimental technique to measure the temporal modulation transfer function (TMTF), in which inherent envelope modulations in narrowband noise are used as the detection target. Modulation depths of the stimuli were manipulated using a signal processing algorithm that introduced no change in the stimuli’s power spectra. In this way, spectral cues cannot cue the presence of modulation. TMTFs were measured using these narrowbands of noise [TMTF-NS (no spectral cues)] at four center frequencies (2, 4, 6, and 8 kHz) and bandwidths of 16, 24, 32, 48, 64, 96, and 128 Hz. TMTFs using low-pass-noise modulated puretones (TMTF-NBM) were also measured for comparison. In cases where the spectral cues are limited in TMTF-NBM (at low-modulation rates or at high-carrier frequencies), thresholds are very similar between the TMTF-NS and the TMTF-NBM, providing validation that this new technique measures temporal acuity similar to standard TMTF methods. In conditions where spectral cues are available in TMTF-NBM (high-modulation rates and lower carrier frequencies), higher thresholds are measured for TMTF-NS and a more reasonable estimate of temporal acuity is obtained.
Another indication that information in “off-frequency” filters enhances sensitivity to envelope-based interaural temporal disparities.
Authors: Bernstein Leslie R. , Trahiotis Constantine
Published: March 2010

A recent study, [Bernstein, L. R., and Trahiotis, C. (2009). J. Acoust. Soc. Am. 125, 3234-3242] employing “raised-sine” stimuli centered at 4 kHz revealed three outcomes concerning discriminability of envelope-based interaural temporal disparities (ITDs). First, graded increases in raised sine exponent led to graded decreases in envelope-based threshold-ITDs. Second, threshold-ITDs decreased with increases in modulation depth. Third, when the modulation depth was 25%, increasing the exponent led to especially large decreases in threshold-ITD. Overall, an interaural correlation-based model including stages mimicking peripheral auditory processing was shown to capture quite well most of those outcomes. Left unexplained, however, was the interaction between modulation depth and raised sine exponent. Here, we report new threshold-ITDs obtained while varying, factorially, depth of modulation (25% or 100%), raised-sine exponent (1.0 or 8.0), and modulation frequency (32, 64, 128, or 256 Hz). Those stimulus parameters were chosen specifically to shed light on the unexplained interaction. A quantitative theoretical analysis of the new data suggests that listeners can enhance their sensitivity to envelope-based ITDs by utilizing “off-frequency” filters centered above 4 kHz. [This research was supported by research grant NIH DC-04147 from the National Institute on Deafness and Other Communication Disorders, National Institutes of Health.]
Masker modulation regularity and its effects on comodulation masking release.
Authors: Buss Emily , Hall Joseph W. , Grose John H.
Published: March 2010

Several authors have suggested that comodulation masking release is largest when performance in the narrowband masker condition is poor due to perceptual similarities between masker fluctuations and an added signal. The present study tested this hypothesis by manipulating the regularity of masker modulation between but not within listening intervals. The signal was a 500-Hz tone burst, and the masker was a continuous band of noise, either 25- or 1050-Hz wide. After filtering, the masker was amplitude modulated with one of three envelope patterns. The regular envelope was a smoothed 20-Hz square wave, composed of 10-ms raised-cosine ramps and 15-ms steady states. In the irregular envelope conditions, either the duration of the steady state or the modulation depth was randomly selected prior to each modulation period, with one exception: modulation was not randomly perturbed during listening intervals, when the signal might be presented. There were large individual differences, but masker envelope irregularity tended to elevate thresholds, particularly for the 25-Hz masker bandwidth. Results were consistent with more CMR under conditions for which there is greater informational masking in the narrowband masker condition.
The fluctuating-masker benefit for word identification with spectral or fine-structure distortions.
Authors: Bernstein Joshua G.W.
Published: March 2010

Normal-hearing (NH) listeners presented with speech processed to reduce spectral detail or remove temporal fine-structure (TFS) information show a reduced benefit from momentary dips in the level of a fluctuating masker. This has been interpreted as evidence that related deficits may underlie the reduced fluctuating-masker benefit (FMB) observed for the hearing impaired. However, the reduced FMB for processed stimuli could be attributable to the greater signal-to-noise ratio (SNR) required to yield performance levels equivalent to the unprocessed case for unmodulated noise (the reference condition). NH listeners were tested in the identification of isolated words presented in stationary-noise, interfering-talker, and speech-modulated noise maskers. Different response-set sizes were used to offset performance differences between processed and unprocessed conditions. A validation experiment showed no effect of set size on the FMB when compared at a common stationary-noise SNR. A second experiment compared the FMB for unprocessed, spectrally smeared, and noise vocoded speech. When compared at the same SNR and percent-correct level (but different set sizes), processed and unprocessed stimuli yielded similar FMB. This suggests that for the conditions tested here, spectral or TFS distortions do not directly impair the ability to listen in the gaps of a fluctuating masker. [Sponsored by the Oticon Foundation.]
Phonetic learning in utero.
Authors: Moon Christine , Lagercrantz Hugo , Kuhl Patrica K.
Published: March 2010

The prevailing view is that newborn phonetic perception is tabula rasa because of poor transmission of the acoustic features of phonemes to the fetus. However, vowel information may be at least intermittently clear in utero. We tested 80 neonates (M = 32.8 h old, range 7–75) in the US and Sweden with English and Swedish vowels using an infant-controlled sucking procedure. Sucking activated 1 of 17 stimuli (a prototype and 16 variants) from the same vowel category, either the English /i/ or the Swedish /y/. Infants sampled through all 17 stimuli, presented randomly, one time. The dependent measure was mean number of sucks per stimulus. Results showed that the Foreign Vowel Group had significantly greater means to the prototype than the Native Group. A within-group analysis showed another Foreign-Native Group difference. Infants in the Foreign Group had significantly more sucks to the prototype compared to the variants, whereas the Native Group treated the prototype and variants equivalently. These results require us to re-evaluate assumptions about availability of speech sounds to the fetus and the state of speech perception at birth.
Longitudinal analysis of vowels in infant-directed speech.
Authors: McColgan Kerry E. , Bernstein Ratner Nan , Newman Rochelle
Published: March 2010

Adult-directed conversational speech (ADS) is characterized by a high level of acoustic imprecision which should theoretically cause problems for infant language learning. Whether infant-directed speech (IDS) is clearer than ADS is disputed. However, differences in prosody [Fernald (1989)], utterance length [Cooper (1997)], and acoustic features [Bernstein Ratner (1984) and Malsheen (1980)] have been found. Such differences have the potential to assist children in decoding language input [Fernald (1985)]. In prior research, the child’s age and language ability appear to affect speech clarification [Bernstein Ratner (1996)]. This study examines whether vowel clarification is a function of the infants’ age. More than 75 mother-infant dyads are being followed from 7 months to 2 years of age. Data reported here are from 10 dyads whose data have been fully analyzed. Plots of the first and second formant values for vowels in words spoken in IDS and ADS were compared at 7 and 11 months of age. Vowels to 7-month-old infants appear to show less clarification than those to 11-month infants, although individual patterns of clarification are evident. This study analyzes whether such variability in vowel clarification affects the rate of early infant vocalizations prior to language production. [Work supported by NSF BCS074512.]
Vowel space of 3- and 4-year-old children.
Authors: McGowan Rebecca W. , Denny Margaret , McGowan Richard S. , Nittrouer Susan
Published: March 2010

Children learn to produce speech within the constraints of vocal tracts that are smaller and configured differently from those of adult speakers. To study children’s speech development in relation to their vocal tract growth, the acoustic characteristics of vowels spoken by ten children, recorded at 6 month intervals between the ages of 36 and 48 months and a subset between 18 and 48 months, are analyzed. The first three formant frequencies of each token are measured and tokens are categorized by intended vowel. Formant frequencies are compared both among vowel categories at each age and within categories over time to give an understanding of the development of children’s vowel space. Changes in children’s ability to differentiate among vowels in F1-F2 space over time are investigated. The development of vowel space is considered in the context of growth of and changes in the proportions of the vocal tract and improved control of the articulators. [Work supported by NIDCD-0001247 to CReSS LLC.]
Neural coding of formant-exaggerated speech in infants and adults.
Authors: Zhang Yang , Miller Sharon , Koerner Tess , Carney Edward
Published: March 2010

Speech scientists have long proposed that formant-exaggerated speech plays an important role in phonetic learning and language acquisition. However, there have been very little neurophysiological data on how the infant brain and adult brain respond to formant exaggeration in speech. We employed event-related potentials (ERPs) to investigate neural coding of formant-exaggerated speech sounds. Two synthetic /i/ vowels were modeled after infant-directed speech data and presented in alternating blocks to test the effects of formant exaggeration. The fundamental frequencies of the two sounds were kept identical to avoid interference from exaggerated pitch level and range. For adult subjects, non-speech homologs were also created by using the center frequencies of the formants to additionally test whether the effects were speech-specific. In the infants (6 to 12-month olds), ERP waveforms showed significantly enhanced N250 and sustaining negativity responses for processing formant-exaggerated speech. In adults, enhancement was observed in the N100 component for the speech stimuli but not the homologous non-speech sounds. Collectively, these results provide the first evidence that formant expansion in infant-directed speech enhances neural activities for phonetic encoding, which may facilitate phonetic learning and language acquisition regardless of the age factor [Zhang et al. (2009). Neuroimage 46, 226–240].
Development of vowel production in young children with cochlear implants.
Authors: Nagao Kyoko , Cornaglia Allegra , Bunnell H. Timothy
Published: March 2010

Past research on children with cochlear implants (CIs) focused on improvements in speech perception, but few speech production studies have been reported. The literature on speech development in prelingual children with CIs is particularly sparse. This is partially due to difficulty transcribing young children’s productions. The aim of this study was to examine whether developmental changes in vowels can be captured without phonetically transcribed data. The subject, a prelingually deaf child 22 months old at the start of the study, is one of six children with CIs participating in an ongoing longitudinal study. Audio and video data were collected once or twice per week during auditory/verbal therapy at the Alfred I. duPont Hospital for Children. One author perceptually categorized the child’s vowel-like vocalizations in terms of broad height (HIGH, MID, and LOW) and place (FRONT, MID, and BACK) features. Acoustic features (barkscale cepstral coefficients) were then used to train linear discriminant functions on the same vowel data and used to predict perceptual vowel categories. In preliminary results 62% of the tokens were correctly classified for place and 67% for height. Results on generalizability of the features will be presented. [Work supported by NIDRR Grant H133E080003 to the RERC on Hearing Enhancement.]
Coordination of the first and second formants of the Mandarin triphthong /iau/ revealed by adaptation to auditory perturbations.
Authors: Cai Shanqing , Ghosh Satrajit S. , Guenther Frank H. , Perkell Joseph S.
Published: March 2010

Investigations of auditory feedback control of supra-glottal articulatory movements traditionally have focused on quasi-static speech sounds like monophthongs. Only recently has attention been directed toward time-varying sounds. To study the role of auditory feedback in the formation of formant trajectories in time-varying vowels, we perturbed the second formant frequency (F2) trajectory as 20 native speakers produce the Mandarin triphthong /iau/ and measured their patterns of auditory-motor adaptation. The subjects changed F2 in directions opposite to the perturbations, not unlike adaptive responses seen in previous studies. Surprisingly, concurrent with the F2 compensations, the subjects significantly altered the first formant (F1), the unperturbed formant, in ways that assisted cancellation of the perturbations to the location and orientation of the formant trajectory in the F1×F2 space. Care was taken to rule out that the F1 changes were byproducts of the F2 compensations. These observations indicate that F1 and F2 are planned in a coordinated fashion for the triphthong /iau/, a property of the speech system not yet revealed through studying quasi-static sounds. The implication of this F1-F2 synergy for the spatiotemporal nature of vowel auditory targets will be discussed. The patterns of generalization to a set of unperturbed vowels will also be presented.
Lexical frequency and Japanese vowel devoicing.
Authors: Nielsen Kuniko
Published: March 2010

The current study aims to examine the relationship between Japanese vowel devoicing and lexical frequency in light of the probabilistic reduction hypothesis [Jurafsky et al. (2001)], which predicts that more probable words are (phonetically) reduced more than less probable words. In Tokyo Japanese, high vowels become devoiced when they occur between two voiceless consonants. Similar devoicing processes occur in many languages and are generally considered to be part of the vowel reduction processes [e.g., Dauer (1980); Kohler (1990)]. Although Japanese vowel devoicing has been studied extensively from both phonetic and phonological viewpoints, its lexical property, namely, the effect of lexical frequency, is as of yet unknown. If Japanese devoicing is a phonetic reduction process, we would expect to see a stronger degree of devoicing among words with higher lexical frequency. Twenty-four speakers of Tokyo Japanese recorded 60 words which include various devoicing environments. Contrary to the prediction by the probabilistic reduction hypothesis, our preliminary results revealed a significantly higher rate of devoicing [F(1,22)=24.89, p<0.01] as well as shorter vowel duration [F(1,22)=4.61, p<0.05] among low-frequency words than high-frequency words. The implications of these results on the nature of Japanese vowel devoicing will be discussed.
Comparison of the formant frequencies F3 and F4 on a three-dimensional vowel chart.
Authors: Isei-Jaakkola Toshio , Naka Takatoshi , Hirose Keikichi
Published: March 2010

We have developed a three-dimensional vowel chart that should fit into a quadrilateral IPA vowel chart better and later we have added more visually effective functions to it. On the chart we can see not only the locations of F1 and F2 with two scales of x and y, where x is F1 and y F2, but also F1, F2, and F3 as well, with three scales of x, y, and z, where x is F1, y F2, and z F3. Thus, this enables us to compare the vowel locations of any language (multilingual vowel presentations), vowels uttered by male with those of female (thus, gender difference), vowels of adults with those of children (thus, age difference), foreign language learners with target vowels, or vowels uttered by the same language learners. This time, we focus more on the roles of the respective formants, particularly on the roles of F3 and F4. It has been clarified that F3 is more related to lip-spreading. We want to show that F4 may be more related to lip-protrusion. For this purpose, we have further added a selection of F3 and F4 functions to the chart. [Work supported by JSPS and Chubu University Grant (A).]
Psychometric functions of vowel detection and vowel identification in multi-talker babble.
Authors: Miller Maggie , Walker Jessica , O’Brien Kayla , Liu Chang
Published: March 2010

Psychometric functions of vowel detection and vowel identification were measured in multi-talker babble for young normal-hearing listeners. A four-interval forced-choice procedure was used to examine the accuracy of vowel detection in babble with speech level presented from 0- to +15-dB sensation level relative to vowel detection thresholds obtained with method of limits. The accuracy of vowel detection was significantly influenced by vowel category and sensation level. The threshold of vowel detection for each vowel and each listener was defined as the speech level at which 71% accuracy of vowel detection was reached. Vowel identification was then measured in babble with vowel levels presented from 0- to 12-dB sensation level relative to individual thresholds of vowel detection, using a close-set 12-choice procedure. Results suggest that vowel identification was significantly affected by vowel category and sensation level. Altogether, the results of vowel detection and vowel identification indicate that, given the same signal-to-noise ratio, vowels are not equally audible and identifiable, possibly due to the fact that some vowels are more audible than others, and that the slope of psychometric functions of vowel identification is vowel-dependent.