bluuebunny/crossref_metadata_2025 huggingface dataset:
Localizing noise in rooms via steady-state interaural time differences
Authors: Rakerd Brad , Hartmann William M. , Pepin Eric
Published: October 2013

Noise lateralization experiments with headphones show that the steady-state interaural time difference (ITD) is, or is not, a valuable cue for sound lateralization depending on the height of the peak of the interaural cross-correlation function (cc). To discover whether these headphone results could be extended to the localization of sound in rooms, binaural recordings were made of one-third-octave noises in diverse environments: a dry classroom, a laboratory, and a reverberation room. Sound sources and a KEMAR recording manikin were moved about in each room to vary the cc from 0.2 to 0.8. The recordings, with ITDs digitally controlled, were presented to five listeners by headphones to measure listener sensitivity to ITD variations. It was found that sensitivity was almost entirely a function of cc with little weight attached to the actual room environment. ITD thresholds decreased monotonically with increasing cc. In agreement with earlier headphone studies using computed noises instead of room recordings, typical thresholds were higher by a factor of about 6 for a 2800-Hz band where envelope ITD is used compared to a 700-Hz band where waveform ITD is used. [Work supported by the NIDCD.]
Temporal processes involved in simultaneous reflection masking
Authors: Buchholz Joerg M.
Published: October 2013

Reflection masking refers to the specific masking condition where a test reflection is masked by the direct sound. Employing reflection masking techniques, Buchholz [J. Acoust. Soc. Am. 117, 2484 (2005)] provided evidence that the binaural system suppresses the test reflection for very short reflection delays and enhances the test reflection for large delays. Employing a 200-ms-long broadband noise burst as input signal, the critical delay separating these two binaural phenomena was found to be 7–10 ms. It was suggested that the critical delay refers to a temporal window that is employed by the auditory system to analyze the considered stimulus. In the present study, the dependency of this critical delay on the duration of the input signal (i.e., broadband noise) has been investigated. The psychoacoustical results revealed that the critical delay decreases with decreasing stimulus duration, resulting in a critical delay of about 2–3 ms for 20-ms-long stimuli. Hence, for very short stimuli the temporal window or critical delay exhibits values similar to the auditory temporal resolution as, for instance, observed in gap-detection tasks. It is suggested that the larger critical delay observed for longer stimuli is due to the auditory system focusing on lower signal frequencies.
The influence of broadened auditory tuning on across-frequency integration of speech
Authors: Healy Eric W. , Carson Kimberly A.
Published: October 2013

Previous work has suggested that hearing-impaired (HI) listeners may have a deficit in their ability to integrate speech information across different frequencies. Performance of HI listeners was poorer than that of normal-hearing (NH) listeners on across-frequency tasks, and performance dropped more steeply when the signal was degraded through across-frequency asynchrony [E.W. Healy and S.P. Bacon, J. Speech Lang. Hear. Res. 45, 1262–1275 (2002)]. In the current study, the role of broadened auditory tuning in this deficit was investigated. Broad tuning was simulated in NH listeners using noise-vocoded sentences and manipulating the slopes of the amplitude-modulated carriers. It was found that intelligibility of synchronous carriers was poorest in the broadest slope conditions, suggesting that poor integration of synchronous cues may be at least partially attributable to broad tuning. Performance reductions as a function of asynchrony were also found to be greatest in the broadest conditions, but only when analysis bands matched presentation bands and not when analysis bands were held constant. It is concluded that broadened auditory tuning may account for at least a portion of the across-frequency deficit observed in HI listeners. [Work supported by NIH.]
The effect of marker duration on time shrinking
Authors: Fujishima Yoko , Nakajima Yoshitaka
Published: October 2013

Time shrinking is an illusion of time perception. This illusion takes place in stimulus patterns of three successive sound bursts (markers) of equal duration marking two empty time intervals, the preceding time interval (P) and the standard time interval (S). When S is longer than P, S is often underestimated. In this study, we examined whether the duration of the bursts (in this case, pure tones) could affect this illusion. Time shrinking always took place when S−P was less than or equal to 60 ms, and the marker duration did not affect the amount of underestimation of S in this range. However, the illusion took place only for the shortest marker duration of 20 ms when S−P=80 ms. The results roughly agreed with previous data. [Work supported by JSPS and 21st Century COE, Kyushu University.]
The effects of speaking rate and tonal contexts on target undershoot of mandarin tone 3
Authors: Liao Jia-Shiou
Published: October 2013

An investigation of the effects of speaking rate and tonal contexts on target undershoot of Mandarin tone 3 (low or low-falling-rising) was conducted. Data were from seven male and eight female subjects, all native speakers of Mandarin. The test phrase comprises three /ma/s in a row, which were put in a sentence to provide a natural context. The second /ma/ was always tone 3, preceded and followed by tone 1 (high-level), tone 2 (mid-rising), tone 3 or tone 4 (high-falling), resulting in 16 distinct combinations. Subjects read sentences at two relatively different rates: slow and fast. The fundamental frequency (F0) and duration of each /ma/ were measured at the onset and offset of its vowel. In addition, the duration of the second /ma/ and F0 at its midpoint were measured. Quantitative analysis of the rate of change in F0 suggests, by and large, that the F0 control systems are not always the same in slow and fast speech within the same tonal contexts. Furthermore, undershoot phenomena occur more often statistically across a syllable boundary than within a syllable boundary. Moreover, the F0 control system produces rates with less random fluctuation across the syllable boundary than within the syllable boundary.
Analysis on the effects of tonal coarticulation at word and nonword syllable boundaries of Mandarin based on the tone nucleus model
Authors: Sun Qinghua , Keikichi Hirose , Nobuaki Minematsu
Published: October 2013

As a typical tonal language, Mandarin has four lexical tone types, each of which has its own intrinsic pattern when uttered in isolation. However, when produced in context, the tonal contours present many variations, even if we specifically address the tone nucleus, which is defined as a stable part in a syllable. This study describes how and to what extent the word boundary affects tonal coarticulation in continuous speech. Although several previous works have investigated how the prosodic features of syllables are affected by their surrounding syllable tone types, most of them selected nonsense syllables uttered in a carrier sentence as study targets, which is far from natural situation. In this paper, after preparing a Mandarin speech corpus, in which all tone-type combinations of di-syllables are included with both intraword and interword cases, the degree of concatenation is compared for intraword and interword cases, viewing their tone component’s F0 means and ranges of tone nuclei. All target di-syllables are uttered in a phrase. Results indicate that the degree in the intraword and interword cases differ greatly depends on the tone combination. The results were applied into our tone component prediction system, which is based on the tone nucleus model.
Tone sandhi effect on Chinese speech perception
Authors: Hsieh Li , Yu Ya-Ju
Published: October 2013

Mandarin Chinese has four tones with various sandhi processes. The third tone sandhi process is when a third tone (T3) becomes a second tone (T2) before another third tone (T3). Previous studies on third tone sandhi used connected speech in their experiments. This study examines the third tone sandhi effect on Chinese listeners in a nonconnected speech condition. Full-tone Chinese syllables were recorded in isolation. Ten Chinese listeners participated in tonal discrimination tasks which consisted of combinations of T2 and T3 (i.e., T22, T23, T32, T33). Three interstimulus intervals (ISIs) (i.e., 50, 250, 500 ms) and three types of pair conditions (minimal-, rhyme-, mixed-pairs) were introduced in the experiments. The findings of this study confirm the phonological hypothesis that the third tone sandhi effect has a great impact across all the pair conditions. When discriminating T23, Chinese listeners had longer response times and more errors than other tone pairs. The phonological phenomenon of third tone sandhi persists regardless of connected or nonconnected speech conditions in Chinese spoken language. Differences in processing the tone sandhi pairs (T23) and the nontone sandhi pairs (T32) reveal the psychological reality of the third tone sandhi phonological rule.
A cross-language comparison of the use of stress in word segmentation
Authors: Tyler Michael D. , Perruchet Pierre , Cutler Anne
Published: October 2013

In spite of our illusions to the contrary, there are few acoustic cues to word boundaries in spoken language. While statistical probabilities between adjacent speech units provide language-general information for speech segmentation, this study shows that language-specific information may also play an important role. An f0 excursion, a correlate of stress, was provided on the first or last syllable of each word in an artificial language. As expected, French listeners succeeded with final-syllable stress but not initial-syllable stress. However, Dutch listeners appeared to benefit from both initial and final stress, rather than initial stress only. When the results were split according to the language background of the talker (French or Dutch), it became clear that the Dutch listeners only benefited from initial stress when the talker was Dutch, but not when the talker was French. Given that the majority of Dutch listeners spoke French, it is suggested that they readily adapted their word segmentation strategies when faced with French-sounding input (i.e., French talker or final stress). In support of this, Australian English listeners with no knowledge of French clearly benefited from initial stress only, consistent with English stress patterning. [Work supported by Conseil Regional de Bourgogne, NWO, CNRS, NIHDC00403.]
The effect of segmental structure on the production of Japanese pitch-accent
Authors: Shport Irina A.
Published: October 2013

Japanese pitch-accent is a fall in fundamental frequency (F0) after the accented mora. In current models of Japanese pitch-accent contrast, accented and unaccented pitch patterns are predicted to be similar across syllable structure type. This study provides evidence that the segmental make-up of words affects accentual patterns in Tokyo Japanese in terms of their F0 contours. Six native speakers produced accented and unaccented words that varied in segmental type of the word-medial mora (N=864). It was found that the pitch-accent contrast was produced differently in words with a short vowel in the first syllable, CVCVCV and CVC:V words (V-group), on one hand, and words with a longer voiced portion of the first syllable, CV:CV and CVNCV words (V:/VN-group), on the other hand, p>0.01. In the accented condition, the average F0 fall was 50 Hz for all, but the F0 movement within the first syllable was different between the two word groups. In the unaccented condition, word-level, as well as intrasyllabic differences, were observed in F0 contours. F0 was rising in the V-group, whereas it was flat in the V:/VN-group. These results indicated that pitch patterns varied by segmental structure, more so for unaccented words.
Acoustical characteristics of Chinese stringed instruments and their Asian relatives
Authors: Odaka Akiko , Yoshikawa Shigeru
Published: October 2013

In Asia, there are many relative musical instruments that derive from the same origin or share a similar structure. In spite of their similarities, they show distinct differences from each other in terms of their musical usages, playing techniques, and acoustic effects. This varies with each musical genre, musical structure, social background, and acoustic taste swayed by those elements. People often add special devices to musical instrument or improve them so as to realize their acoustical ideas. For example, the shamisen and biwa, Japanese plucked lutes, have devices to allow strings to vibrate against the neck-wood, creating a reverberant high-frequency emphasis called sawari, which is of special value in each performance. A device that creates a similar effect, called jawari in Hindi, is also found in the sitar, a traditional stringed instrument of India. On the other hand, the Chinese sanxian, one of the relatives of the Japanese shamisen lacks this device. In this presentation, specific features of various musical instruments are introduced from both musicological and acoustical viewpoints. In particular, Chinese stringed instruments are focused on and compared with those of Japan, Korea, and India for investigation of their acoustical effects.
Modeling a magnetorheological elastomer’s transient stiffness parameters
Authors: Lerner Anne-Marie , Cunefare Ken
Published: October 2013

Semiactive vibration absorbers are tunable vibration absorbers whose natural frequencies can change controllably to minimize vibration better than their passive counterparts, which can be used to reduce occupant vibration in cars. The variable natural frequency is achieved by using a dynamically tunable spring. In order to use a material as a dynamically tunable spring, its transient parameters must be understood and controlled, so as not to lead to instability. This work characterizes the behavior of magnetorheological elastomers (MREs), which are a type of ferromagnetic-elastomer composites whose tensile/compressional stiffness increases as an applied magnetic field is increased. If the transition from one stiffness state to another is too slow, the MRE can have a detrimental effect on vibration control. Transient behavior of MREs with different concentrations of ferromagnetic material and different geometric shapes was recorded empirically, and the MRE stiffness change in response to a change in magnetic field was modeled as a function of ferromagnetic content, MRE geometric shape, and the change in magnetic field strength. MREs in this configuration exhibited natural frequencies in the 30–65-Hz range, where MREs were able to change their natural frequency by up to 17%.
The acquisition of speech rhythm by bilingual Spanish- and English-speaking four- and five-year-old children
Authors: Bunta Ferenc , Ingram David
Published: October 2013

This study investigates the acquisition of speech rhythm by bilingual Spanish- and English-speaking children, comparing their performance to that of monolingual age-matched peers in both languages, and to monolingual and bilingual adults. Participants included younger children (between 3 years, 9 months and 4 years, 5 months, 15 days), older children (between 4 years, 6 months, 18 days and 5 years, 2 months), and adults (over 18 years). A sentence elicitation task of 26 sentences per language was used to obtain 64 samples that yielded over 15000 duration measurements of vocalic and intervocalic intervals. The duration measurements were entered into the normalized vocalic and intervocalic pairwise variability index (PVI) measures based on Grabe and Low (2002). The results indicated that younger bilingual children displayed distinct speech rhythm patterns for their target languages and deviated from their monolingual English-speaking peers. Older bilingual children also produced significant separation of speech rhythm, and differences between older bilingual children and their monolingual peers speaking English were also found. Comparisons between younger and older bilingual children indicated an effect for the vocalic PVI measure, but not for the intervocalic PVI, providing partial support for age differences for bilingual children. Bilingual adults showed separation of their languages and performed similarly to their monolingual peers, overall.
STRAIGHT as a research tool for L2 study: How to manipulate segmental and supra-segmental features
Authors: Kawahara Hideki , Akahane-Yamada Reiko
Published: October 2013

A morphing algorithm based on STRAIGHT speech manipulation system [Kawahara et. al., ICASSP (2003)] has a potential importance in L2 study, since it generates stimulus continuum with quality close to natural speech. The morphing procedure yields an equally spaced continuum in an abstract parameter domain, which is spanned by fundamental frequency, STRAIGHT’s time-frequency representation (a variant of smoothed spectrogram), the periodicity map, and the time-frequency coordinate. Instead of manipulating all these parameters at once, it is useful to manipulate each parameter independently. This individual parameter manipulation can also be generalized to coupled manipulation by introducing new parameters defined using a set of parameters. This extension enables us to manipulate segmental and supra-segmental features instead of manipulating abstract physical parameters. For example, coupling energy envelope, fundamental frequency, and temporal axis stretching factor yield access to a prosodic feature that is basically supra-segmental. Manipulation of spectral landmark positions yields access to a phonological feature that is basically segmental. A preliminary test results using the right-light continuum sets generated using a STRAIGHT-based method and the Klatt synthesizer will also be presented.
Synthetic aperture sonar imaging of buried spheres and cylinders
Authors: Kargl Steven G. , Williams Kevin L.
Published: October 2013

During SAX04, synthetic aperture sonar (SAS) was used to detect buried spheres where the nominal grazing angle of incidence from the SAS to the point above a sphere was well below the sediment’s critical grazing angle. SAS images from other measurements below the critical angle on buried spheres and finite cylinders in the NSWC-PC test pond facility have also produced target detections. Numerical simulations will be discussed that accurately predict the observed signal to noise (SNR). Simulations include estimates of reverberation from the rough seafloor, and the subcritical penetration through the seafloor, scattering from a target, and propagation back to the SAS. Required material parameters were obtained from environmental measurements including profiles of the small-scale surface roughness and the superimposed ripple structure. These simulations, the sonar equation results, and the model/measurement comparisons over the frequency range of 10–50 kHz further support sediment ripple structure as the dominant mechanism for subcritical penetration in this range. [Work supported by ONR.]
Design and construction of a small sound testing room in an office building
Authors: Duncan Edward C. , Kaliski Kenneth
Published: October 2013

A small sound-testing room has been constructed in the basement of an office building located within 50 ft. of railroad tracks. The room was designed to be multifunctional with both reverberant and semianechoic settings. The sound isolation of the room was designed with the expectation of conducting active noise reduction tests with sound-pressure levels of 120 dB and higher. The room is also being used to roughly estimate absorption coefficients of various materials. The room is 10.5×9.75×7.5 ft. with a volume of 768 ft3. It is composed of a concrete floor and gypsum walls and ceiling. The walls are isolated from the floor with rubber isolation pads. The ceiling is constructed of 5/8-in. gypsum, 1/2-in. plywood, and a 2-in. layer of sand. The wall construction is 2×4 wood studs, two layers of 5/8-in. gypsum resiliently attached, and 3 in. of fiberglass. The entrance to the room is made of double solid core wood doors with a composite isolation and absorption layer in between both doors. Other design components, actual sound transmission testing results, reverberation times, modal issues, and uses for the room are discussed.
Cross-modal perception of room acoustics
Authors: McCreery Anthony , Calamia Paul
Published: October 2013

The development of perceptually accurate audio-visual systems has been hindered by the fact that traditional scientific investigations into auditory and visual perception in humans have separated the two senses into separate and distinct modalities. It is important to consider the perception of these modalities as critically linked and ultimately inseparable. Research investigating expected levels of reverberance given 2-D visual representations of several environments shows that perception of reverberation is heavily dependent on the visual appearance of a room. For the experiments to be discussed here, subjects were asked to adjust the level of reverberant sound that accompanied video of a person speaking in three different rooms: a library, a playhouse, and a cathedral. The results show that reverberance levels chosen by subjects rarely match those measured from impulse responses recorded in the rooms being presented. Experimental methodology and results as well as implications for the development of future audiovisual systems will be discussed.
Acoustic design of studio to record effect sounds in
Authors: Oda Shinichi , Yokoyama Junko , Mikami Junichi
Published: October 2013

We reconstructed a studio that has a space in which to record effect sounds. In this space, we record walking sounds on wooden floors, tile floors, sand, grass, and so on. This studio was planned on a floor with a light allowable load because the floor was formerly the roof of the building. A recording studio is usually constructed with a floating structure. The thin floating floor causes vibration of floorboards. A thick floor is heavy. In addition, we recorded the sound when we jumped on the floor. The noise from the structure disturbs the effect sounds that we need if the vibration caused by jumping stimulates the studio structure. For those reasons, we planned the following design to record pure effect sounds: The effect sound space is insulated from other spaces with the room structure. A floating ceiling is hung from the fixed ceiling, and parts of floating walls are hung from overhead beams. Only the floating floor and the other parts of floating walls are set upon the fixed floor. Consequently, the floating floor in the effect sound space was made thicker than would be possible otherwise.
Auditory streaming and the spectrotemporal analysis of sound in the cortex
Authors: Shamma Shihab , Elhilali Mounya
Published: October 2013

Auditory streaming is a phenomenon that manifests itself in the everyday ability of humans and animals to parse complex acoustic information arising from multiple sound sources into meaningful auditory streams. For instance, the ability to follow a conversation at a noisy cocktail party or hear the violin in the orchestra both rely on the formation of auditory streams. While seemingly effortless, the neural mechanisms underlying auditory streaming remain a mystery. In this talk, we shall discuss how this perceptual ability may emerge as a consequence of the multiscale spectrotemporal analysis of sound in the auditory cortex. A simplified model of this process is developed to demonstrate how auditory streaming could underlie the separation of complex sounds from two sources (e.g., speech or music). The model reveals that certain perceptual attributes are critical for the simultaneous separation of sounds (pitch and onset sensiticity), whereas others are important for the sequential binding of the segregated sound features (timbre and location).
Evaluation of blood flow velocity vector measurement accuracy: Numerical simulation and flow phantom studies of Rankine’s vortex
Authors: Okada Takashi , Sonoyama Teruyuki , Harada Akimitsu , Ohtsuki Shigeo , Tanaka Motonao
Published: October 2013

A new method to measure the velocity vector of blood flow on the observation plane was recently developed [Ohtsuki and Tanaka, J. Visuali. 9, 69–82 (2006)]. The method is based on the calculation of stream functions. To evaluate the accuracy of this method, we have studied the velocity distribution of Rankine’s vortex in numerical simulation and experiments. The vortex of the simulation model was 50 mm in radius, the forced vortex was 20 mm in radius, and the maximum velocity of the vortex was 50 cm/s. The differences between the velocity profiles of the vortex obtained by this method and by simulation were less than 10%, although the accuracy depended on the angle of the observation plane. Furthermore, we conducted phantom experiments in which the vortex was created in degassed water by using a magnetic stirrer. Color Doppler data of the vortex flow were acquired using a ultrasonic diagnostic system (SSD-6500, Aloka, Japan) and the velocity vector distribution was calculated. The experimental results were in good agreement with the simulation model. We conclude that the reliability of this method is very high. The results suggest great potential of the new velocity vector method for quantitative diagnosis of cardiac functions.
Professor Floyd Dunn’s contributions to ultrasound metrology
Authors: Harris Gerald R.
Published: October 2013

A half-century ago the applications of ultrasound in medicine were, apart from physiotherapy, mostly experimental, and the related bioeffects and toxicity studies were in their infancy. Progress in both of these areas was hampered in part by the lack of acceptable tools for measuring the exposure levels, especially for the intense fields then being used in ultrasound surgical studies. Professor Floyd Dunn’s advancements in ultrasound measurement techniques at the University of Illinois Bioacoustics Research Laboratory afforded accurate means for quantifying these exposures and laid the groundwork for subsequent developments in ultrasound measurement and calibration. In this presentation Professor Dunn’s early work in thermal and radiation force techniques will be described and some recent methods that are being investigated to characterize ultrasound fields, particularly at high intensities, will be reviewed.
The design and evaluation of a noise suppression system for custom-made digital hearing aids
Authors: Yoshizumi Yoshiyuki , Fujii Shigekiyo
Published: October 2013

A noise suppression system, which has the inbuilt feature to adapt to the local environment, for custom-made digital hearing aids has been developed. The signal process is based on spectral subtraction technology with 32 bands in the frequency domain. The signal-to-noise ratio was improved by an average of 7.5 dB in rms level. Its effectiveness was evaluated using the seven-point rating scale method for nine normal-hearing subjects and six hearing impaired subjects. Test speech is presented under three different environments, which consist of a car, a duct, and a crowded place. As a result of the assessment, the same tendency is observed between normal and hearing-impaired subjects and there were two good points that were found. First, the noise suppression system makes them more comfortable to listen to speech with the statistical significance of p=0.05. Second, the noisiness is reduced with the statistical significance of p=0.01. It should be noted that there is a scattering of the plotted data; however, statistically, the data result is quite acceptable.
Hearing effects of nonoccupational sound exposure evaluated by temporary threshold shift
Authors: Takinami Hiroaki , Aono Shoji
Published: October 2013

In the 1960s, the relation between the acoustical characteristics of sound exposure and the sound-induced temporal threshold shift (TTS) of people were carefully investigated by Shoji et al. [J. Acoust Soc. Am. 48 978−987 (1970); 6th ICAA−2−8, pp. A9−A12 (1968)]. Based on these studies, an equation that estimates TTS at various test frequencies was proposed as a function of the TTS−critical−band levels. In this presentation, therelation between the sound exposure and the TTS will be reviewed, and a prototype TTS meter that estimates TTS in real time operation will be introduced. The hearing effects of some nonoccupational sounds at different places (Pachinkoparlor, disco, and livehouse) were evaluated with the TTS levels as the indicator. The sounds from portable musicplayers and music instruments of or chestra were also evaluated. The TTS levels were measured for shorttime sound exposure and compared to long−term estimates using the TTS meter. In some cases, after 2− or 4−h sound exposure, the estimated TTS levels at the test frequency of 4 kHz reached almost 20 dB.
A source of high-level acoustic impulses for the measurement of hearing protectors
Authors: Zera Jan , Mlynski Rafal
Published: October 2013

The paper describes a laboratory source for generating high-level, A-type acoustic impulses. The basic part of the source is a steel cylinder closed at one end with elastic foil. The blast is produced by an abrupt break of the foil caused by an increased pressure of air compressed inside the cylinder. By adjusting the air pressure inside the cylinder the impulses peak SPL may be set within a range of 150–170 dB. As the peak SPL is increased from 150 to 170 dB, the rise time and A duration of impulses decrease respectively from 350 to 14 s and from 1.1 to 0.4 ms. An advantage of the source is that it produces impulses with highly repeatable parameters; for a given value of air pressure inside the cylinder, standard deviation of peak SPL distribution is only 1.3 dB. The source is omnidirectional in the horizontal plane and has a nearly circular directional pattern in the vertical plane. The source is a good alternative for generating impulses with the use of explosives and is particularly useful for measurements made indoors, in the laboratory. [Work supported by the Polish Ministry of Science and Education, Grants T07B00428 and SP03.1].
Performance of sound restoration hearing protection in impulsive noise
Authors: Murphy William J.
Published: October 2013

Hearing protection devices (HPDs) are the principal means of protecting the hearing of a person against harmful levels of noise in highly impulsive noise environments. New HPDs which utilize sound restoration circuitry were measured using a mannequin to assess the performance of these protectors in response to small-arms weapons fire. The results for the peak reduction and attenuation were analyzed for both indoor and outdoor measurements. The performance of the protectors exhibited little dependence with level for impulses between 150 and 170 decibels. The passive performance for a single protector characterized the active performance for these high-intensity impulses. The results were also evaluated using the US Army AHAAH cochlear model. As the peak sound-pressure level underneath the different models of protectors increased, the estimated risk of hearing loss increased. [Work supported by EPA Interagency Agreement DW75921973-01-0.]
Unidentified energy conversions in an oscillating bubble
Authors: Vokurka Karel , Buogo Silvano
Published: October 2013

Acoustic radiation of freely oscillating bubbles has been studied experimentally. The oscillating bubbles have been generated by underwater spark discharges. Recorded pressure waves made it possible to determine the acoustic energy radiated by the oscillating bubbles and the potential energy of the bubbles when attaining the first and second maximum volumes. Comparison of the potential and acoustic energies revealed that about 70% of the potential energy of the bubble available at the first maximum volume has been converted into an unidentified energy. A possible candidate for the energy conversion mechanism seems to be internal converging shock wave in the bubble interior. The presence of this wave can be observed indirectly in a radiated pressure wave. The measured energies are compared with energies computed using a suitable model. This comparison shows that the theoretical values agree with experimental data until the bubble attains the first minimum volume. However, at later times discrepancies are observed. [Work has been partly supported (K.V.) by the Czech Ministry of Education as the research project MSM 46747878501.]
Auditory events in language and music
Authors: Nakajima Yoshitaka , Ueda Kazuo
Published: October 2013

Perceptual organization of both language and music is based on temporal units that can be called auditory events. Speech syllables and musical notes correspond typically to auditory events. The listener categorizes these events in various ways in order to extract information. We are at a preliminary stage of searching for acoustic cues that the auditory system can utilize to determine auditory events. Averaged intensity of a speech signal within a moving Gaussian temporal window of 15 ms gives cues of syllable boundaries, especially when this averaged intensity is compared with the intensity change obtained with a Gaussian temporal window of 200 ms. The change of intensity obtained through a filter of about 600–2200 Hz compared with the intensity change obtained through an almost flat filter also gives cues of syllable boundaries. Thus, intensity cues and spectral cues both work to mark syllables. This holds both in Japanese and in English. Such intensity cues may work when applied to monophonic or homophonic music, but spectral changes do not correspond clearly to musical notes. Thus, the perceptual mechanisms that determine speech syllables and musical notes can be different. [Work supported by JSPS and 21st Century COE, Kyushu University.]
Two issues at the interface between speech and music
Authors: Deutsch Diana , Henthorn Trevor
Published: October 2013

In what ways are speech and music subserved by domain-specific mechanisms, and in what ways by common neural substrates? This issue is addressed with respect to absolute pitch, which has generally been considered to be a musical faculty accessible only to rare individuals. Based on several findings obtained from speakers of tone languages, it is argued that absolute pitch originally evolved to subserve speech, and that it may be acquired readily during the first year of life, if infants are given the opportunity to associate pitches with verbal labels. It is further conjectured, based on other findings, that the early acquisition of pitch accent languages, such as the Tokyo and Osaka dialects of Japanese and the Hamkyeng and Kyengsang dialects of Korean, might also be conducive to the acquisition of absolute pitch. Perceptual relationships between speech and song are next considered. These generally differ in their features; however, a spoken phrase can be perceptually transformed to appear as song rather than speech, simply by repeating it several times over without any alteration in the signal (Deutsch, 2003). This phenomenon is demonstrated, its parameters are explored, and its implications for the mechanisms underlying perception of speech and song are discussed.
Providing new methods for teaching Japanese speech communication for learners of Japanese as L2: Developing teaching materials using MRI-movies and natural speech data
Authors: Hayashi Ryoko , Zhu Chunyue , Sadanobu Toshiyuki , Kaneda Jumpei , Erickson Donna , Campbell Nick , Sugito Miyoko
Published: October 2013

It is critical to provide people learning Japanese as a second language information about natural speech, since there is wide variability of articulation and speaking styles associated with various social attitudes and/or expressions in Japanese. For developing teaching materials for English and Chinese learners of Japanese as a second language, two new methods are demonstrated. One utilizes MRI-movies that dynamically demonstrate differences among vowel articulations in Japanese, English, and Chinese. This approach is effective for teaching good pronunciation, especially with regard to consonant cluster production, since the timing of the articulation for consonant clusters is visibly presented. The other is audio-visual data of natural speech in Japanese that demonstrate several typical expressions, e.g., wry face and strained (laryngealized) voice for asking favors politely. This type of material shows not only variations of speech communication in Japanese but also cultural differences among native speakers of Japanese, English, and Chinese. [Work supported by MEXT.]
Factors affecting the success of non-native vowel perception training
Authors: Nishi Kanae , Kewley-Port Diane
Published: October 2013

Previously, Nishi and Kewley-Port [J. Acoust. Soc. Am. 118, 1900 (2005)] reported the successful outcomes of an intensive vowel perception training protocol for Japanese and Korean learners of English. Their results indicated that vowel training worked best when large numbers of vowels in nonsense words were presented within the training blocks. In this report, additional analyses were performed in order to explore the generalization of training from nonsense to real words that were presented only during pre- and post-training tests. Results showed that, although both groups showed steady and substantial improvement for all vowels in nonsense words, generalization to real words was inconsistent across listener groups, different vowels, and talkers. These results indicated that: (1) native vowel phonology had a considerable influence on the success of our structured, intensive identification training protocol using many vowels; (2) nonsense word training incorporating high natural variability across phonetic contexts and talkers may not be sufficient to achieve the desired generalization to real words. These results are explained in part by the vowel acoustics (formant frequencies and vocalic duration) of the stimulus materials used for training and tests. [Work supported by NIDCD-006313 and 02229.]
French conversations do not show evidence of phonetic accommodation to a non-native interlocutor
Authors: Smith Caroline
Published: October 2013

Although a speech style known as foreigner talk has been described, speech addressed to a reasonably fluent non-native interlocutor may show little or no accommodation by native speakers. This issue is of interest in understanding whether on-line modifications to speech are an important contributor to phonetic variation. Ten native speakers of Parisian French were recorded individually as they performed map tasks, one with another native speaker of French, and another with an American who identified himself as such at the beginning of the conversation, but who spoke French during the task. Previous work has found that French listeners who heard brief extracts from these conversations could not reliably identify which interlocutor was involved. In an attempt to identify other possible differences in speech to natives and non-natives, two prosodic characteristics of the conversations were analyzed, vowel devoicing and the epenthesis of word-final schwas. The hypothesis was that vowel devoicing, which is stylistically marked, would be disfavored in speech to non-natives, while schwa epenthesis, characteristic of a deliberate speech style, would be more frequent. The speakers actually showed similar patterns with both interlocutors, suggesting that variation in these properties is not conditioned by the linguistic status of the interlocutor.
Production of English stressed vowels and interstress intervals within and between words in first and second language acquisition
Authors: Sudo Michiko M. , Kaneko Ikuyo
Published: October 2013

The purpose of this study was to observe the differences between first and second language acquisition in English durational patterns with respect to interstress intervals (ISIs). Production experiments were carried out to investigate the durational patterns used by three groups of speakers: adult native speakers of American English, American third graders, and Japanese learners of English. Two sets of English sentences were devised as the linguistic material, and the sentences in each set differed in the number of nominally unstressed syllables that intervened between a target stressed syllable and the next stressed syllable. These sets of sentences contained two different target stressed vowels, and the ISIs within and between words. The shortest ISIs were produced by adult American speakers regardless of the number of unstressd syllables in the target ISIs. The Japanese learners produced shorter ISIs than the American children in one set of sentences, while in the other set, the American children produced shorter ISIs than the Japanese. Regarding the durations of target stressed vowels, foot-level shortening was observed in the three subject groups, the American children exhibiting the highest percentage of shortening followed by the American adults and then the Japanese.
The speech learning model: English and Japanese voice onset time in mature bilingual talkers
Authors: Riney Timothy J.
Published: October 2013

With reference to the Speech Learning Model (Flege, 1995 version), this study looked for evidence among mature American and Japanese bilinguals that (a) L1 VOT changed over time and (b) despite minimal L1-L2 phonetic differences, L2 VOT was different from L1 VOT. Five L1 English Americans and 5 L1 Japanese read the same lists of English and Japanese words containing word initial /p/, /t/, and /k/. In all, 880 tokens (10 talkers ×2 languages ×11 words ×4 tokens) were examined. For both groups and individuals, based on measures of difference and comparisons with age-matched monolinguals (5 American and 13 Japanese), the findings were (a) slight evidence of L1 VOT change, but (b) robust evidence of new L2 categories being formed. By far the most common bilingual pattern observed was L2 compromise VOT, in which the L1 VOT remained close to monolingual VOT and the L2 VOT was positioned about halfway between that of two age-matched monolingual comparison groups.
Sound source tracking for security camera using sensor array utilizing object-related transfer function
Authors: Usagawa Tsuyoshi , Takada Toshimichi , Chisaki Yoshifumi
Published: October 2013

A frequency-domain binaural model has been proposed as a computer model of human auditory system, and it can estimate the direction of arrival (DOA) of target sound source with only two inputs [Nakashima et al., Acoust. Sci & Tech. 21, 172–178 (2003)]. It utilizes a database of interaural level and phase differences obtained from head-related transfer function (HRTF) for DOA estimation. Since the interaural level and phase differences arise from the diffraction of head, similar ones can be obtained from the diffraction of an object, such as a transparent cover of security camera. In this paper, a source tracking method which utilizes an object-related transfer function is proposed. Three sensors attached on a dome-shaped camera cover are used, and the interchannel level and phase differences obtained from object-related transfer function are used for DOA estimation. The performance of sound source tracking is discussed based on the accuracy of DOA estimation as well as observation latency by the computer simulation. [Research partially supported by the MEXT, Grant-in-Aid for Scientific Research (C), 2006, 18500135 and carried out by the Cooperative Research Project Program of the RIEC, Tohoku University (Grant ♯H16.A02).]
Acoustic source localization with eye array
Authors: Alghassi Hedayat , Tafazoli Shahram , Lawrence Peter
Published: October 2013

A novel signal-processing algorithm and array for sound source localization (SSL) in three-dimensional space is presented. This method, which has similarity to the eye in localization of light rays, consists of a novel hemispherical microphone array with 26 microphones on the shell and one microphone in the sphere center. The microphones on the shell map a geodesic hemisphere called two-frequency icosahedron; hence, each microphone has at least four other orthogonal microphones. A signal-processing scheme utilizes parallel creation of a special closeness function for each microphone direction on the shell in the time domain. Each closeness function cell (lens cell) consists of center microphone, shell microphone, and one of its orthogonals. The closeness function output values are linearly proportional to spatial angular difference between the sound source direction and each of the shell microphone directions. By choosing microphone directions corresponding to the highest closeness function values and implementing a linear weighted spatial averaging on them, the sound source direction is estimated. Contrary to traditional SSL techniques, this method is based on simple parallel mathematical calculations in the time domain with low computational costs. The laboratory implementation of the array and algorithm shows reasonable accuracy in a reverberant room.
Seafloor manipulative experiments during sediment acoustics experiments (SAX99 and SAX04
Authors: Richardson Michael D. , Williams Kevin L.
Published: October 2013

Seafloor manipulations were performed during Sediment Acoustics Experiments (SAX99 and SAX04) in an attempt to test predictions of high-frequency acoustic backscattering models in sandy environments. Manipulations included alteration of seafloor roughness by raking to mimic the effects of sand ripples on acoustic scattering and placement of marbles, aluminum disks, and shells on the seafloor at different densities to mimic the effects of shelled animals and sand dollars on acoustic scattering (discrete surface scattering). In general, the predictions from scattering models and measured acoustic backscatter strengths were in agreement and the experimental results provide insight into natural phenomena. [Work supported by ONR.]
Using buried vector sensors to examine seabed layering in sandy sediment
Authors: Hines Paul C. , Osler John C. , Scrutton Jeff , Chapman David M.F. , Lyons Anthony
Published: October 2013

There is considerable interest within the underwater acoustics community in quantifying the sound-speed dispersion in sandy sediments at low kiloHertz frequencies. To address this, data were collected during the SAX04 experiment (a US Office of Naval Research Departmental Research Initiative) using projectors and directional (vector) receivers buried in the top 1 m of the seabed, in conjunction with a water-borne source directly above the buried sensor field. Although using buried sensors significantly increased the complexity of the experimental setup, it should enable a direct time-of-flight measurement of acoustic wave speed, which simplifies interpretation of the results. However, initial analysis of the time-of-flight results indicates the existence of a buried layer, which increases the complexity of the data. In this paper, the vector sensors are used to steer beams to confirm the existence of the layer and to isolate the direct and reflected arrivals to improve the estimate of sound speed dispersion. [Work partially funded by US Office of Naval Research.]