bluuebunny/crossref_metadata_2025 huggingface dataset:
Results of attenuation calculations for a three-component gas mixture
Authors: Dain Yefim , Lueptow Richard M.
Published: July 2012

Acoustic attenuation in a mixture of gases results from the combined effects of molecular relaxation and the classical mechanisms of viscosity and heat conduction. As a result, the attenuation depends on the composition of the gas, frequency, temperature, and pressure. A model of the relaxational attenuation that permits the prediction of acoustic attenuation is used to predict the effect of composition, frequency, temperature, and pressure on the acoustic attenuation in a three-component gas mixture of nitrogen, methane, and water vapor. The attenuation spectrum is dependent upon the composition through the appearance of peaks in the spectrum related to the relaxation frequency of the particular components and their relaxing complexes. The relaxation peak related to methane dominates the spectrum except at low methane concentrations, where the nitrogen peak, which is dependent upon the water vapor concentration, is evident. Temperature and pressure significantly alter the relaxation frequency and the degree of attenuation, but water vapor plays little role in the attenuation at moderate and high methane concentrations. [This work was funded by the Department of Energy under subcontract from Commercial Electronics, Inc. (Broken Arrow, OK) and by the Ford Motor Company.]
Franssen à la phones
Authors: Hartmann William M. , Lawton Scott R. , Rakerd Brad
Published: July 2012

According to current explanation, the Franssen effect reflects a competition between localization cues from the onset of a tone and cues from the steady-state portion. For a sine tone in a room, the steady-state interaural time and level cues (ITD and ILD) are implausible because of room reflections. Therefore, onset cues dominate, and a listener localizes the tone at its onset position regardless of the true origin of the steady state. To test this explanation we used headphones to simulate the simplest kind of Franssen stimulus: The ITD was fixed throughout the tone. At onset, the ILD was zero. After onset, the ILD grew so as to contradict the ITD in the steady state. Equal numbers of trials were done with abrupt and slow onsets. The listener’s task was to localize the steady-state part of the tone. A Franssen analog was identified by comparing responses to abrupt-onset and slow-onset tones. Results were compared with the standard Franssen effect in a large room. Clear evidence of a Franssen effect, with its dependence on plausibility, appeared in the headphones experiments, but the data were less dramatic than data obtained in the room. [Work supported by the NIDCD.]
Reverberant speech intelligibility for monaural and binaural listening
Authors: Libbey Brad , Rogers Peter
Published: July 2012

Humans listening to speech in a small room are frequently unaware of reverberation. It is unknown if neurological processes remove these echoes or if they are simply disregarded when speech is phonetically processed. In other words, is there a neurological mechanism that is capable of removing echoes to create a clean speech neurological signal before phonetic processing? Or is the brain capable of processing reverberant phonemes? These questions have been explored in the process of investigating the hypothesis; ‘‘Binaural listening does not improve speech intelligibility.’’ Previous experiments have shown that binaural listening does not improve intelligibility for highly reverberant small rooms, T60=1.5 s. The current experiment investigates the intelligibility of words in three reverberant rooms, T60=1.1, 0.7, and 0.4 s. Subjects listen with one or two ears to lists of words taken from a standardized testing procedure, ASA 85. In addition the rooms are simulated using the method of images and the reverberant words are presented in a diotic and binaural format over headphones. Intelligibility results from these rooms with monaural and binaural listening are presented.
A model to simulate human sound localization and detection in presence of a distracter
Authors: Braasch Jonas
Published: July 2012

To simulate human auditory localization performance of a target in the presence of a distracter, existing models focus the analysis of the targets on frequency bands or time gaps where the distracter is not present. However, human localization also works quite well if the frequency ranges of target and distracter are identical and the distracter is played during the complete duration of the target—as long as the onsets of target and distracter are separated in time. This is taken into account in the proposed model. For each frequency band, the interaural time differences of the target are calculated from the difference of the interaural cross correlation (ICC) of the total sound (target+distracter) and the ICC of the distracter alone (ΦT=3DΦTD−ΦD, assuming that target and distracter are uncorrelated). The latter is calculated from the part of the distracter that precedes the target. A similar procedure is used to measure the interaural level differences of the target. Auditory events are simulated assuming an ideal observer and internal noise. The model is able to mimic a number of psychoacoustic effects, including spatial shifts of the auditory events for low SNR situations. [Work supported by DFG.]
Individualized HRTFs using computer vision and computational acoustics
Authors: Duraiswami Ramani , Davis Larry , Shamma Shihab A. , Elman Howard C. , Duda Richard O. , Algazi V. Ralph , Liu Qing-Ho , Raveendra S. T.
Published: July 2012

For a given sound source location and frequency content, the sound received at the two ears of a person is first scattered by his/her torso, head, and pinnae, leading to interaural differences in the intensity and spectral features of the sound received. These features are used as cues to perceive the spatial location of the source, and are encoded in an extremely individual ‘‘head-related transfer function’’ (HRTF) that depends on the person’s external anatomy (structure of the torso, head, and pinnae). This individuality has made it difficult to use the HRTF to render perceptually convincing spatially distributed audio for applications such as entertainment or virtual reality. We have commenced a long-term project that will use numerical methods to compute individualized HRTFs from accurate 3D surface models of a person’s torso, head, and pinnae (obtained using computer vision techniques). These surface models will be used with boundary-element methods to solve the wave equation and compute the HRTFs. An overview of our long-term project, as well as preliminary results that compare the numerical techniques to be used with analytical and experimental results for scattering from simple shapes, will be presented. [Work supported by NSF.]
Silicon micromachined ultrasonic transducers for bulk, Lamb, and Rayleigh waves
Authors: Khuri-Yakub B. T.
Published: July 2012

Capacitive micromachined ultrasonic transducers (cMUT) can match and even outperform piezoelectric transducers in terms of efficiency and bandwidth. With the advent of silicon micromachining, it is now possible to make capacitors with very thin gaps that sustain electric fields of the order of 109 V/m or more. At these levels of electric field, the transformer coupling between the electrical and mechanical parts of the capacitor transducer, and thus its performance, are comparable to that of piezoelectric transducers. We will review the design and performance of cMUTs for airborne, immersion, Lamb and Rayleigh waves’ ultrasound applications. We will show examples of immersion transducers (single element, 1-D and 2-D arrays of elements), air transducers and Lamb wave transducers. We will also present transducer systems (single element, 1-D and 2-D arrays of elements) with dynamic ranges of over 150 dB/Volt/vHz, which are in excellent agreement with the theoretical predictions.
Wireless surface acoustic wave and MEMS-based microsensors
Authors: Varadan Vijay K.
Published: July 2012

The integration of surface acoustic waves (SAW), microelectromechanical systems (MEMS), and required microelectronics and conformal antennas to realize programmable microsensors suitable for many engineering and biomedical applications will be reviewed in this talk. This unique combination of technologies results in novel conformal sensors that can be remotely sensed by a wireless communication system with the advantage of no power requirements at the sensor site (passive sensor). The required features in many of these applications are high precision, wide dynamic range, and wide frequency range. The MEMS-SAW-based devices presented possess typical advantages of MEMS sensors including the additional benefits of robustness, excellent sensitivity, surface conformability, and durability. After a brief overview of SAW sensors and MEMS, the paper is focused on the design and fabrication of MEMS devices for engineering and medical applications, for example: accelerometer and gyroscopes for automobile, inertial navigation; sensors for pressure, torque, temperature, and strain; health monitoring of structures; ‘‘smart tongue and electronic nose;’’ micropump for drug delivery; humidity, temperature, and ethlyne oxide sensor for hospital sterilization system; microsensors for monitoring bone healing; and pollution and chemical sensing.
Solid-state MOSFET-based micro hydrophone array
Authors: Zhu Bei , Ziang J. , Varadan Vijay K. , Varadan Vasundara V.
Published: July 2012

The fabrication of low-cost, high-sensitivity, and miniaturized hydrophone array is highly desirable for application in sonar technology. However, as hydrophone size decreases, it becomes necessary to provide an amplifier or buffer in close proximity to avoid sensitivity loss due to interconnect capacitance. This suggests the concept of hydrpophone integration with microelectronics in an integrated circuit environment. A proptotype of an integrated transducer was thus realized in which a P(VDF-TrFE) film was bonded to the extended gate of a metal–oxide–semiconductor field-effect transistor (MOSFET) which was fabricated on a silicon wafer using standard NMOS process technology. This technique is compatible with IC fabrication processes and enables the integration of active elements with the transducer element on the same chip. The integrated structure eliminates long leads between the transducer and the amplifier and makes the net transducer–amplifier detectivity superior to the ceramic transducer–amplifier system. The devices are then encapsulated with Rho-C rubber which results in good acoustic impedance to that of water. The acoustic sensing performance was carried out in a pulse tube over a frequency range from 4 to 10 kHz. The experimental results along with the theoretical prediction by Mason equivalent circuit will be presented.
Effects of spectral mismatch on sentence recognition for noise-band processors
Authors: Galvin John J. , Fu Qian-Jie
Published: July 2012

For cochlear implant (CI) users, the cochlear insertion depth and limited spatial extent of the electrode array causes a mismatch between the frequency content of speech signals and the place of stimulation along the cochlea. To examine the effects of such a spectral mismatch on HINT sentence recognition in noise, five normal-hearing (NH) listeners were tested using noise-band speech processors designed to simulate a variety of cochlear insertion depths. The speech reception threshold (SRT) was measured for two conditions: (1) analysis and carrier bands matched to the insertion depth, and (2) analysis bands fixed over a wide range while carrier bands varied according to insertion depth. In the matched condition, subjects’ performance remained flat along the entire range of insertion depths; on average, the SRT was 5 dB lower than the mismatched condition, indicating that a frequency-limited, but spectrally matched processor provided more benefit. In the mismatched condition, performance was flat over most of the range of insertion depths and declined only for extremely basal or apical insertions. The results indicate that electrode insertion depth and the resulting spectral mismatch is probably not the limiting factor affecting speech recognition or contributing to the variability in performance observed among CI users.
Adaptive thresholds for spoken sentence materials
Authors: Bell Theodore , Wilson Richard , Enrietto Jane , Morris Carmen
Published: July 2012

Two experiments document the reliability of a speech discrimination test designed for use with an adaptive procedure to measure thresholds for sentences and words. Stimulus words were categorized as either high usage or low usage based on published indices of word usage, and also as either high confusability or low confusability depending on phonetic similarity to other words in the lexicon. The first experiment employed 20 young normal-hearing adults, each presented with two 20-item lists of sentences from each of the four lexical categories. One week later, the testing sequence was repeated. A five-factor ANOVA was employed using word usage, word confusability, list sample, and test–retest delay; an adaptive procedure, converging determined sentence thresholds in dB level. Test–retest thresholds, both immediate and delayed, were within 2 dB for all lists. There was a 1.5-dB difference between high- and low-usage words. A second experiment examined thresholds for groups of hearing-impaired individuals categorized by audiometric configuration and speech discrimination scores. Each subject group was presented two sentence lists varying in lexical characteristics based on the Neighborhood Activation Model. The difference between lexical items varied significantly with speech discrimination scores and audiometric configuration of the listeners. [Work supported by Veteran Affairs R&D Contract V621P-3664.]
Relation of pitch glide perception and Mandarin tone identification
Authors: Gottfried Terry L. , Riester Devon
Published: July 2012

Second language (L2) learners differ considerably in their ability to learn new phonetic contrasts in the L2. For example, Gottfried and Suiter (1997) observed that many native speakers of American English were highly accurate in their identification of Mandarin Chinese tones, while others, with similar years of experience in learning Mandarin, were very poor. The current study compared listeners’ identification of Mandarin tones with their ability to perceive the pitch direction of sine-wave glides. Sine waves of 400 ms with starting frequencies of 250 and 300 Hz were created with 0- to 50-Hz sweeps up or down from the initial frequencies. Listeners who had no experience with Mandarin indicated whether the pitch went up, down, or remained the same. After testing on the sine waves, listeners identified the tone (high-level, mid-rising, low-dipping, and high-falling) of intact and silent-center (with only the initial and final portions) Mandarin /l/-vowel syllables. Analysis revealed that listeners with musical training performed significantly better on intact Mandarin syllables than nonmusic majors, but no difference was observed for silent-centers. Voice majors seemed to have a particular advantage. Significant correlations were also found between tests of glide perception and tone identification (ranging from 0.530 to 0.642).
Auditory robustness and duration of vocalic cues
Authors: Ohala John J. , Yu Alan C.
Published: July 2012

Building on an idea of Stevens [ICPhS (1980)] and Stevens and Keyser [Language (1989)], Lang and Ohala (LO) [ICSLP (1996)] hypothesized that the most commonly used features to differentiate vowels in languages would be those few that can be detected in a relatively short time. Other features could be used to augment the vowel inventory at the expense of requiring more time for their perception. They found support for this idea in a perceptual study of end-truncated English vowels. We conducted a similar experiment on the perception of 18 contrasting vowels in Cantonese including monophthongs and diphthongs and both rounded and unrounded front vowels. The stimuli were vowel fragments starting at 30 ms after an initial [s] and end truncated into pink noise at 20-ms intervals thereafter. Twenty Cantonese heard randomized sets of these stimuli and attempted to associate each with the vowel in 18 different key words symbolized by Chinese characters. The resulting confusion matrices of the shorter fragments revealed not only the expected high confusions between monophthongs and diphthongs but also between rounded and unrounded front vowels. The latter contrast, which is indeed used less in language’s vowel inventories, must require more time to perceive, supporting the LO claim.
Underwater acoustic communication by passive phase conjugation: Theory and experiment
Authors: Jackson Darrell R. , Rouseff Daniel , Fox Warren L. J. , Jones Christopher D. , Ritcey James A. , Dowling David R.
Published: July 2012

A new method for coherent underwater communication called passive phase conjugation is evaluated. The technique takes its name because of conceptual similarities to active phase conjugation methods that have been demonstrated in the ocean [Kuperman et al., J. Acoust. Soc. Am. 103, 25–40 (1998)]. In contrast to active techniques, however, the array in passive phase conjugation need only receive. This makes the method plausible for scenarios where spatially compact sources might be communicating to a distant receive-only array. Compared to other approaches for coherent communication, the computational burden is low allowing the method to be evaluated in the field in nearly real-time. Results from an experiment conducted in Puget Sound near Seattle in May 2000 are reported. Various modulation schemes and array geometries were employed. Measurements were made at several ranges and water depths in a range-dependent environment. [Work supported by ONR.]
Self-equalization communications using a time-reversal mirror
Authors: Edelmann G. , Akal T. , Hodgkiss W. S. , Kim S. , Kuperman W. A. , Song H. C.
Published: July 2012

A time-reversal mirror (TRM) was used to transmit self-equalization communication sequences in both range-independent and range-dependent environments. The TRM consisted of 29 source/receive elements centered at 3.5 kHz. A 2-ms pure-tone pulse was transmitted from a probe source at a range of 10 km from the TRM. A vertical receive array with 32 elements was colocated with the probe source to measure the spatial resolution of the TRM focus and bit error rate. The communication sequences were encoded using BPSK and higher-order phase-shift keying. As a comparison, single-source and broadside communication transmissions were also made.
Analytic performance evaluation of coherent equalizers in time-varying acoustic channels
Authors: Preisig James C.
Published: July 2012

Coherent equalization is one of the primary methods of implementing high-speed acoustic communications systems in underwater environments. This environment is often characterized by extended delay and Doppler spreads with different Doppler characteristics for the different delay taps of the time-varying impulse response. An analytic technique is presented which enables the evaluation of the performance of least-squares-based coherent equalizers in such channels. The technique is based upon state-space formalisms developed in the Kalman filtering literature. Comparisons between predicted and simulated equalization performance for realistic channel models are presented.
Low-frequency impact sound rating of floor systems
Authors: Warnock Alf
Published: July 2012

An extensive series of measurements of airborne and impact sound insulation on joist floors of various types was recently completed at NRC. About 190 floors with different joist types, subfloors, ceiling types, ceiling support systems, and type and thickness of sound absorber were constructed and sound insulation measured to frequencies lower than required by standards. The joist types comprised solid wood, wood trusses, I-joists, and steel joists. Glass, rock, and cellulose fiber sound-absorbing material were used. Three concrete slabs were included in the series and a few of the joist floors had concrete toppings installed. Impact devices used on the floors included the standard ISO/ASTM E492 tapping machine, a walker, a JIS 1419 tire impactor, and two experimental rubber balls. The impact measurements provide an extensive, consistent set for examining alternative impact tests and possible modifications to standard tests using the ISO tapping machine. Correlations among the sound levels from the different impactors show that levels from the ISO tapping machine correlate well with those from a walker at low frequencies, although the spectra are quite different in shape. An overview of the data and the prospects for a new test method and rating system will be presented.
Flanking sound transmission in wood-framed construction
Authors: Nightingale Trevor , Halliwell Robin , Quirt David
Published: July 2012

This paper presents results from the first phase of a project to characterize flanking sound transmission at wall/floor junctions in wood-framed constructions suitable for multifamily residential buildings. This phase of the study focused on the junction between a wood joist floor and a separating wall with two rows of 38×89-mm studs. The objective of the study was to identify details that simultaneously provide good noise control and fire resistance. This paper presents the experimental approach and key results from the acoustical measurements. In addition to comparing how five common fire control solutions for the floor/wall junction affect the airborne and impact sound transmission, several generic retrofit modifications to reduce the structural noise transmission were tested. Parameters studied included the floor-joist orientation (parallel or perpendicular to the separating party wall), and the type of subfloor (plywood or oriented strand board). The study has shown that suitable details can provide both adequate fire resistance and noise control. The technical approach provides a framework for extending beyond established laboratory measurement procedures, to predict performance in actual buildings.
First known study, identification, and characteristics of fin and blue-whale vocalizations in the Northeast Pacific with historical notes and anecdotes
Authors: Cummings William C. , Thompson Paul O.
Published: July 2012

In 1964 we were tasked by the Navy to identify and describe low-frequency, high-level noise mainly in the 1/3rd-octave frequency band containing 20 Hz. Due to AGC and other presently antiquated processing then used at Northeast Pacific sound surveillance stations (SOSUS), acoustical monitoring was often confounded for significant annual periods heightening concern during that era of Cold War. Using data from continuous recordings for Gordon Wenz’s ambient noise studies, we characterized the causative individual sources mainly as trains of approximately 1-sec pulsed pairs and as long series of about 20-sec paired moans. We determined seasonality, noise acceleration, geographic variation, representative source levels, and identity of the sources which turned out to be whales. This paper describes methods and findings of several years’ study to identify and describe this ‘‘noise’’ employing numerous expeditions by sea and aircraft, now a more than 30-year-old precedent effort for which the authors (or their colleagues whose contributions shall be specified) have never been fully recognized by the military or by contemporary scientists. Intermingled we include trivia, such as the fate of the invaluable Wenz recordings, and interesting accounts of simultaneous Northwest Atlantic investigations by Schevill, Watkins, and colleagues including the present first author.
Blue whale calls recorded in the Gulf of Alaska
Authors: Stafford Kathleen
Published: July 2012

Blue whales (Balaenoptera musculus) were known to range throughout the North Pacific Ocean, including the Gulf of Alaska, before whaling activities greatly reduced their numbers. Despite repeated dedicated aerial and shipboard surveys in the Gulf of Alaska and off the Aleutian Islands, very few blue whales have been seen in these waters [NMFS, Mar. Fish. Rev. 61, 3843 (1999)]. Recent studies have shown the utility of acoustic methods for monitoring blue whale calls in the northeastern Pacific [cf. Thompson and Friedl, Cetology 45, 1–19 (1982); Stafford et al., Mar. Mamm. Sci. 15, 1258–1268 (1999); Watkins et al., Oceanography 13, 62–67 (2000)] but none of these have surveyed the Gulf of Alaska. In order to address the question of whether or not blue whales are still found in the Gulf of Alaska, six hydrophones were deployed there in October 1999. The first data set was recovered in May 2000. A preliminary analysis of these data showed that both northeastern [cf. Rivers, Mar. Mamm. Sci. 13, 186–195 (1997)] and northwestern [cf. Thompson and Friedl (1982)] Pacific blue whale calls were recorded on the hydrophones.
Rake, a wave measuring benthic acoustic stress sensor
Authors: Morrison Archie T. , Williams Albert J.
Published: July 2012

The Rake is an acoustic travel time current meter designed to make spatially and temporally dense velocity profile measurements in the continental shelf wave bottom boundary layer. The high levels of bottom shear stress generated in this extremely thin, oscillatory, boundary layer play an important role in the entrainment of sediment into the water column and also enhance the turbulent dissipation of energy in overlying steady flows. The array of acoustic transducers measures the horizontal velocity vector with millimeter vertical resolution at up to 64 heights. A multiplexing transmit/receive circuit permits measurement of the vertical component of velocity with 5 to 10 cm vertical resolution using the same array. Complete velocity profiles can be acquired within a 40 ms window. Measurement error due to velocity averaging over the sample volume of a single acoustic axis has been shown to be negligible to within 1 mm of the bottom both by calculation and experiment. Laboratory calibrations and experiments have been conducted in steady and oscillatory flow. A month-long deployment of a field prototype took place in December of 1996. Data from these analyses and experiments will be presented. [The development of the Rake was supported by ONR and NSF.]
A statistical classifier for hearing protector REAT data
Authors: Murphy William J. , Franks John R.
Published: July 2012

Hearing protector attenuation data have been demonstrated to exhibit bimodal characteristics due to poorly fitting protectors. The data have been modeled with two distributions [Murphy and Franks, J. Acoust. Soc. Am. 106, 2262 (1999)]. The bimodal distribution model was used as the basis of a statistical classifier for an individual’s attenuation data to identify whether or not the data come from a poor or well fit distribution. Training classes were developed from ANSI S12.6 Method B subject fit data [Franks et al., Ear Hear. 21, 218–226 (2000)] for the EAR Classic (n=25), EAR Express (n=20) and Howard Leight MAX (n=25) earplugs. Real ear attenuation at threshold was measured under circumaural headphones for each earplug with a panel of 20 subjects. Preliminary results suggest that data from 500 or 1000 Hz yield the best ability to separate the attenuations.
Whole field vibration measurements with laser-based techniques
Authors: Lal Amit K. , Hess Cecil F.
Published: July 2012

This paper discusses two techniques, a time-domain technique and a space-domain technique, to remotely measure 2-D vibration characteristics of a surface. The first technique is based on using multiple diode laser vibrometers to simultaneously measure the surface vibration at multiple points. The principles of laser Doppler vibrometry are briefly discussed and various architectures of systems to perform landmine detection are presented and discussed. Surface velocity data corresponding to various types of soils shaken by a buried geophone are presented. The second technique is based on laser speckle interferometry using a single digital camera to instantaneously capture quantitative phase information. The phase information of two consecutive frames is then converted to velocity profiles of the vibrating surface. The principles of a new method to perform instantaneous phase shifting interferometry are presented and modal analysis data of turbine blades obtained with an all-digital optical system are discussed.
Far-field radiation from a source in a flat rigid baffle of finite size
Authors: Panzer Joerg W.
Published: July 2012

The far-field radiation of a source in a flat rigid baffle of finite size is derived. The velocity distribution of the source is arbitrary as well as the shape of the source and the baffle. The derivation covers the two-sided as well as the single-sided radiating case. The approach makes use of the Kirchhoff–Helmholtz equation as well as of spatial Fourier transforms of the source and the baffle shape.
Theoretical approaches to speech perception
Authors: Kuhl Patricia K.
Published: July 2012

Theoretical approaches to speech perception have shifted dramatically during the last 50 years. Early theories, such as Alvin Liberman’s ‘‘motor Theory,’’ focused on the psychological mechanisms underlying speech perception rather than listeners’ analysis of the acoustic signal. In the 1970’s, Ken Stevens’ work on the acoustic cues for phonetic categories produced a very different view, an ‘‘invariance theory,’’ that turned theoretical attention to the acoustic signal and the detailed phonetic information that listeners gleaned from its analysis. The last decade has seen another theoretical shift. Strongly influenced by developmental studies, new theories (e.g., Kuhl’s native language magnet model, or NLM) combine the sensory and psychological approaches, arguing that listeners both perform a detailed analysis of language input and psychologically represent that information in a complex way that is unique to each language. The presentation will discuss these shifts and the kinds of data now necessary to produce the next theoretical advance. [Work supported by NICHD HD37954 and Human Frontiers RG0159 to PKK.]
Realistic application of acoustic blind source separation
Authors: Parra Lucas
Published: July 2012

Blind source separation in an acoustic environment is a challenging problem. The reverberation times in realistic rooms require long separation filters. They have to be estimated in very short times as the room responses change drastically with user location. In addition, convolutive blind separation has a number of indeterminations that make the identification of filter coefficients difficult. Decorrelation of changing cross-power spectra of nonstationary signal has been proposed as a separation criterion by various authors. We have presented in particular a frequency domain on-line algorithm that seeks to decorrelate cross-power spectra estimated over multiple times. We will report the performance of the algorithm in a realistic environment. The results will be compared to more conventional adaptive beamforming algorithms. Furthermore, speech recognition results for a command and control application with a distant talking user will be presented.
Relation between intelligibility and listening difficulty using word lists based on word familiarity
Authors: Morimoto Masayuki , Kobayashi Masaaki , Sato Hiroshi
Published: July 2012

Listening tests to assess word intelligibility and listening difficulty were performed to obtain a subjective criterion for speech transmission. It is well known that both intelligibility and listening difficulty depend on the choice of test words. The test words used in the present tests were selected according to word familiarity [Sakamoto et al., J. Acoust. Soc. Jpn. 54, 842–849 (1998)]. The subject was asked to repeat each word and to rate listening difficulty into one of four categories: (0) not difficult, (1) a little difficult, (2) fairly difficult, and (3) extremely difficult. After the tests, the four categories were reclassified into (0) and the others. The listening difficulty is defined as the percentage of the total number of responses except (0). The results demonstrate that: 1) the listening difficulty is highly correlated with intelligibility, 2) the listening difficulty can evaluate the quality of speech transmission more exactly and sensitively than intelligibility, 3) the effects of word familiarity on the listening difficulty and intelligibility do not depend on test sound fields, and 4) STI is not related to either the listening difficulty or intelligibility. Furthermore, the results suggest some other advantages in using listening difficulty as opposed to intelligibility.
Effects of word familiarity on word intelligibility for listeners with presbycusis in noise and reverberation
Authors: Sato Hiroshi , Yoshino Hiroshi , Sato Hayato , Suzuki Yoîti , Amano Shigeki , Kondo Kimihisa
Published: July 2012

Listening tests to assess word intelligibility were performed to estimate the effects of presbycusis under control of the word familiarity in experimental sound fields with noise and/or reverberation. The test words used in the present tests were selected to have similar word familiarities [Sakamoto et al., J. Acoust. Soc. Jpn. 54, 842−849 (1998)]. Moreover, difference/similarity of the effect of word familiarity on speech intelligibility between young listeners with normal hearing and elderly listeners with impaired hearing was also examined. The results demonstrate that: (1) Comparing the intelligibilities of the elderly with that of the young, the normal hearing subjects can tolerate 5 dB A more intensive noise than impaired listeners. (2) The word intelligibility of young subjects shows a 25% higher score than that of the aged. This result is very similar to sentence intelligibility [Sato et al., Proc. Noise Effects ’98 (Sydney, Australia) 168−173 (1998)] and Japanese trisyllable intelligibility [Sato et al., J. Archit. Plann. Environ. Eng. AIJ, 15−20 (1997)]. (3) The effect of word familiarity on word intelligibility is almost equal between the young normal and elderly impaired subjects.
Deep, harmonic moans associated with Bryde’s whales in several locations worldwide
Authors: Barlow Jay , Oleson Erin , McDonald Mark
Published: July 2012

Several sounds have been associated with Bryde’s whales in the Gulf of California [Cummings et al. (1986); Edds et al. (1993)]. Here, we report on harmonically rich tonal calls recorded in the vicinity of Bryde’s whales. In the eastern tropical Pacific, sonobuoys were deployed in the immediate vicinity of Bryde’s whales on 40+ occasions, and in two cases, a 1–2-s tonal signal was recorded with a fundamental of 50–53 Hz and with 3 clear harmonic overtones. In the southern Caribbean, sonobuoys were deployed in the vicinity of Bryde’s whales, and, on 20 occasions, a 2–3-s tonal signal was recorded with a fundamental of either 15–20 or 40–45 Hz and with 3–5 clear harmonic overtones. Off New Zealand, a bottom-mounted hydrophone recorded 4–7-s, 22–28-Hz tonal calls which are seasonally associated with the presence of Bryde’s whales; such calls were previously recorded in this area using a hydrophone with greater bandwidth [Kibblewhite et al. (1967)] and showed harmonic overtones that were similar to those at the other two locations. Bryde’s whales appear to produce a rich variety of calls, including these harmonic moans that had not been previously noted in the published literature.
Acoustic analysis of objects ensonified by a bottlenose dolphin during a cross-modal matching task
Authors: DeLong Caroline M. , Au Whitlow W. L. , Harley Heidi E.
Published: July 2012

A bottlenose dolphin performed a matching task in which he was presented with an object (e.g., tin cup) in one modality (e.g., vision) and then asked to choose the same object from a group of three objects using another modality (e.g., echolocation) or the same modality (e.g., vision). Object sets were presented in two cross-modal conditions (visual sample with echoic choices or echoic sample with visual choices) and two intramodal conditions (visual sample with visual choices or echoic sample with echoic choices). These objects were later ensonified by dolphin-like clicks from several different angles and the signals were captured and digitized (objects were aspect dependent, thus producing different echoes at different angles). Echoes from object sets on which the dolphin performed well were compared with echoes from objects sets on which the dolphin performed poorly to elucidate the aspects of the echo signals that the dolphin used to discriminate among the objects (e.g., echo highlights, spectral characteristics) and the reason(s) why some objects were easier to tell apart than others. Visual analysis of the echolocation behavior and search strategy of the dolphin using video recordings supplemented the acoustic analysis.
Acoustic stop bands for two-dimensional periodic arrays of metallic rods: Square geometry
Authors: Kushwaha M. S.
Published: July 2012

Extensive band structures have been performed for two-dimensional periodic arrays of rigid stainless steel cylinders in air, with Bloch vector being perpendicular to the cylinders. For cylinders 2.9 cm in diameter and period of 10 cm (i.e., the filling fraction f=0.066) there is no acoustic gap for frequencies below 6.4 kHz. However, the density of states reveals prominent minima at 1.7 and 2.4 kHz. These frequencies do agree with those of the first two attenuation maxima observed in the experiment [Nature 378, 241 (1995)] and are indeed related to diffraction from the [100] and [110] planes. Thus even with idealization, Sempere’s sculpture exhibits pseudogaps—not full gaps. It is stressed that, for any value of the period of the system, there is no acoustic gap for f≤30%; magnitude of the gap, for f≥30%, is inversely proportional to the period of the system. Moreover, if the filling fraction exceeds 40%, a second gap, higher in frequency, opens up. In addition, the fabrication of a multiperiodic system in tandem that could create a huge hole in sound within the human audible range of frequencies is proposed. [Work partially supported by CONCyT Grant No. 28110E.]
Characterization of multiple scattering acoustic media using backscattered pulsed ultrasound
Authors: Burkhardt John
Published: July 2012

A technique is presented for the determination of the Legendre moments of the differential scattering cross section per unit volume of a multiple scattering acoustic medium. Acoustic propagation within the medium is modeled using a time-dependent diffusion approximation to the scalar radiative transfer equation. An expression for the backscattered acoustic intensity is derived. Determination of the Legendre moments requires backscatter intensity measurements as a function of azimuthal angle at a fixed polar angle. Monte Carlo simulations are presented which support the proposed technique.
Generalized optical theorem for acoustic backscattering
Authors: Marston Philip L.
Published: July 2012

The usual optical theorem for sound expresses the extinction cross section as proportional to a component of the far-field forward scattering amplitude for three-dimensional scattering [V. Varatharajulu, J. Math. Phys. 18, 537–543 (1977); S. G. Kargl and P. L. Marston, J. Acoust. Soc. Am. 88, 1103–1113 (1990)]. It does not appear to be widely recognized, however, that in the absence of dissipation, a component of the acoustic scattering at other scattering angles may be related to an angular integration involving scattering amplitudes in other directions (related by supplemental conditions to the direction of interest) provided the target has inversion symmetry. This analytical result is confirmed computationally for the imaginary component of the form function for backscattering by spheres described by a partial-wave series where the normalization of Kargl and Marston is used as well as numerical integration. The usual optical theorem is the analytical result for the forward scattering case. [Work supported by the Office of Naval Research.]
Context-dependent neural coding in the chinchilla cochlear nucleus
Authors: Holt Lori L. , Ventura Valerie , Rhode William , Behesta Sam , Rinaldo Alessandro
Published: July 2012

One of the most challenging dilemmas for theories of speech perception is the lack of invariance between acoustic signal and perception. Due to physical constraints upon articulators, there is a good deal of context-dependency both in speech production and in the resulting acoustic speech signal. Consequently, the acoustic pattern most closely related to a given speech sound varies dramatically depending on context. Yet, by some means, the perceptual system perceives these unique acoustic events as linguistically equivalent. This phenomenon is observed experimentally as perceptual context effects whereby adjacent speech can modulate perceived identity of a given speech sound. Recent perceptual results suggest that this phenomenon may be governed by general auditory mechanisms [Holt et al., 1996; Lotto et al., 1997; Lotto and Kluender, 1998; Holt, 1999] rather than speech-specific processes. In the present study, we sought to explore how such context-dependencies might be encoded. We recorded responses of ventral cochlear nucleus (VCN) neurons of anesthetized chinchillas to nonspeech stimulus targets with adjacent context stimuli that varied in spectral content. Results demonstrate context-dependent effects of frequency and intensity.
Individual differences in auditory abilities among normal-hearing listeners
Authors: Kidd Gary R. , Watson Charles S. , Gygi Brian
Published: July 2012

An extended version of the Test of Basic Auditory Capabilities (TBAC) [Watson et al., J. Acoust. Soc. Am. Suppl. 1 71, S73 (1982)] includes 19 subtests that evaluate listeners’ abilities to discriminate and identify auditory stimuli. Stimuli include single tones, tonal sequences, SAM and rippled noise, temporal gaps, nonwords, words, sentences, and environmental sounds. A total of 340 college-student subjects were tested with this battery of tests. A principal components analysis yielded a four-factor solution that accounts for roughly 50% of the variance. The first factor primarily reflects loudness and duration discrimination, the second is associated with sensitivity to temporal envelope variation (SAM noises), the third is associated with identification of highly familiar sounds, whether they be speech or nonspeech, and the fourth factor includes pitch-discrimination and spectral-temporal tasks, suggesting a common ability to discriminate complex patterns. As in earlier studies in this series, measures of spectral and temporal resolving power for laboratory-generated sounds are very weak predictors of individual differences in speech processing. SAT scores and grades in college courses were also examined to assess the relation between auditory perceptual abilities and abilities in language and mathematics. [Work supported by NIH/NIDCD.]
Neural correlates of duplex perception: A whole-head magnetencephalography study
Authors: Mathiak Klaus , Hertrich Ingo , Lutzenberger Werner , Ackermann Herrmann
Published: July 2012

Simultaneous experience of the same stimulus events in two distinct phenomenological modes is referred to as duplex perception (DP). The most widely investigated DP paradigm decomposes the formant pattern of the stop consonant-vowel (CV) syllables /ga/ and /da/ into two complementary portions: the isolated transient of the third formant (‘‘chirp’’) and the remaining sound structure (‘‘base’’). Presentation of the two syllable components to opposite ears results in the perception, first, of the chirp and, second, the fused CV syllable. In order to further elucidate the mechanisms underlying DP, the present study recorded mismatch fields in response to a series of dichotically applied base and chirp components using whole-head magnetencephalography (MEG). This magnetic analog to mismatch negativity is considered a correlate of early cognitive stimulus representation. (a) Under a visual distraction task, the preattentive mismatch fields were stronger in response to contralateral deviants and lateralized to the left hemisphere. (b) Under attention toward the fused percept /da/, left ear deviant chirps yielded an ipsilateral enhanced and posteriorly shifted reaction. These data suggest representations of the phonetic percepts in secondary dominant auditory cortex. Perception tasks may activate these templates for syllable fusion.
Preliminary results of an acoustic barrier test using matched-field tracking
Authors: Stevenson J. Mark , Bucker Homer P.
Published: July 2012

In September 2000, a prototype autonomous seafloor barrier HLA (horizontal line array) will be deployed in the Adriatic Sea. This test apparatus is not designed as a beamforming array, but rather contains six hydrophones nonuniformily distributed over 650 m in a line on the sea- floor. The test objective is to make automatic detections of acoustic sources traveling over the barrier using matched-field tracking [H. Bucker, J. Acoust. Soc. Am. 96, 3809–3811 (1994)]. The tracking algorithm operates in real time in a digital signal processor (DSP) running autonomously inside the apparatus. The procedure is to select N Fourier frequency bins based on S/N ratios after CFAR normalization, and then to determine the best tracks using M-of-N processing. Results will be shown comparing real-time results with those obtained in postexperiment analysis. [Work supported by ONR, Code 321-SS.]
Source localization using the waveguide invariant approach and conventional plane wave beamforming with vertical hydrophone array data
Authors: D’Spain Gerald L. , Kuperman William A. , Murray James J.
Published: July 2012

Broadband source localization, both in range and depth, can be performed using conventional plane wave beamforming with vertical hydrophone array data. Little computation or prior knowledge of the environment is required. An analytical model, based on the application of the waveguide invariant approach to a normal mode formulation of the sound field, is developed to predict the behavior of striation patterns as a function of vertical wave number, range, and frequency. The approach is illustrated through numerical simulation and applied to data collected during the shallow-water (200-m) SWellEx-3 experiment in 1994. Extension to canonical deep-water environments also is discussed. [Work supported by ONR, Code 321US.]
The color of early sound arrivals in an auditorium
Authors: Svensson U. Peter , Torres Rendell R. , Medwin Herman
Published: July 2012

In computational room acoustics, most present-day methods calculate specular reflections complemented by various models for surface scattering. A recent method of edge diffraction calculation [U. P. Svensson et al., J. Acoust. Soc. Am. 106, 2331–2344 (1999)], builds on the BTM technique which has been proven to be accurate in such diverse situations as noise control by barriers and sound scatter from the rough ocean surface. Here, it is applied to a large concert hall. The hall chosen is one of those in an International Round Robin study. It is shown that calculations of edge diffraction give quantitative details of what has until now been approximated and loosely called ‘‘scattering.’’ Here, the early part of the impulse response is considered in order to study aspects such as the frequency content of individual early diffractions and reflections and the low-frequency energy distibution in the room. Only rigid surfaces, and up to second-order edge diffraction and/or specular reflections are taken into account, but the possible extension towards realistic wall impedances and higher-order reflections are discussed. Since the BTM technique is exact and deterministic rather than approximate and statistical, it provides a secure basis for predictions in room acoustical modeling.