bluuebunny/crossref_metadata_2025 huggingface dataset:
A hearing-aid signal-processing scheme based on the temporal aspects of compression
Authors: Carney Laurel H. , Shi Lufeng , Doherty Karen A.
Published: September 2013

Changes in gain associated with the basilar membrane compressive nonlinearity are accompanied by changes in the bandwidth of tuning. Filters with level-dependent bandwidth have level-dependent phase properties. These phase properties result in level-dependent timing of sustained phase-locked responses of auditory-nerve (AN) fibers at low frequencies and level-dependent latencies at high frequencies, where phase-locking rolls off. In the healthy ear, level-dependent temporal aspects of AN responses carry information about stimulus level and spectral properties. Loss of compression with hearing impairment thus results not only in a reduction of amplification, but also in distortion of the temporal response pattern of the AN. The temporal aspects of compression suggest that signal-processing schemes that attempt to correct sounds, or restore normal spatio-temporal response patterns, should include dynamic level-dependent phase shifts. A nonlinear signal-processing scheme will be presented which includes dynamic frequency- and level-dependent phase shifts, based on physiological models of the temporal response properties of AN fibers. Preliminary testing measured listeners preferences for sentences and intelligibility of vowel-consonant syllables with different degrees of nonlinear processing. Hearing-impaired listeners tended to prefer the dynamically corrected stimuli based on improved clarity. Correction also improved intelligibility for some phonemes. [Work supported by NIDCD R21-006057.]
Identification of 20-ms speech samples from normal and neurologically impaired talkers: Do coarticulation differences matter?
Authors: Sussman Joan E. , Tjaden Kris , Yu Ya-ju
Published: September 2013

Prior studies of speech suggest that the extent of coarticulation varies both across and within talkers. Speakers sometime use average, high, or low amounts of coarticulation. The perceptual consequences of coarticulatory variability are not well understood, particularly for the speech of individuals with motor control disorders. In the current study, listeners heard the same 20-ms tokens that were used to calculate the degree of anticipatory coarticulation of the following vowel. Acoustic measures were made from the first 20 ms of stop consonants [k] or [t], or from a point 70 ms preceding the vowel in [s] productions. Healthy talkers, speakers with Multiple Sclerosis, and speakers with Parkinson’s disease produced the stimuli. Speech tokens included three degrees of anticipatory coarticulation: average, high, and low. The listeners’ task was to identify whether the vowels that followed the consonant were originally [i] or [u]. Results showed that identification accuracy was poorer for the current 20-ms tokens than for prior speech samples including all aperiodicity of consonants. Identification was poorest for [s] and best for [k] tokens. The speed of listener response gave some indication of the differing extents of coarticulation. Perceptual validity for the 20-ms measure of coarticulation is discussed.
Phonological processing among good and poor readers
Authors: Wayland Ratree
Published: September 2013

Many researchers believe that a connection exists between phonological processing skills and reading ability, and phonological deficits have often been cited as possible explanation for reading disability among both children and adults. This study will present research findings on phonological processing of various speech sounds among school-aged children who were classified as good and poor readers by standardized tests. These subjects will be administered speech discrimination tests using a variety of speech stimuli. Results of their performance on these tasks will be presented and a relationship between their reading and phonological processing abilities will be discussed.
The effect of noise on the perception of phonetic features in acoustic simulations of cochlear implant speech
Authors: Munson Benjamin , Nelson Peggy B. , Muecke Jill E.
Published: September 2013

Previous research [Munson and Nelson, J. Acoust. Soc. Am. 114, 2360 (2003)] found that adults with cochlear implants (CIs) perceive synthetic /ra/-/la/, /wa/-/ja/, and say-stay continua less accurately and less categorically than listeners with normal-hearing sensitivity (NH) in noise. Considerable variability was found among the CI listeners: some demonstrated performance comparable to that of NH listeners, while others showed considerably poorer performance. One potential reason for the variability in performance is the number of distinct spectral channels available to the CI listeners. Friesen et al. [J. Acoust. Soc. Am. 110, 1150–1163 (2001)], for example, demonstrated that better-performing implant users showed improvements in consonant and vowel recognition when the number of signal spectral channels was increased, with asymptotic performance at seven channels. Poorer-performing users, however, showed no improvement in scores when the number of channels increased beyond four. To examine this hypothesis, NH listeners participated in an experiment in which they listened to 4- and 8-band acoustic simulations of synthetic continua in quiet and in steady speech-shaped noise at a +10 dB SNR. Analyses will focus on the effect of number of channels on identification accuracy and within-group variability.
Vowel formant movement and duration perceived through noise vocoders and cochlear implants
Authors: Iverson Paul , Evans Bronwen G. , Smith Charlotte A.
Published: September 2013

Formant movement and duration have been increasingly shown to be important cues for vowel recognition by normal-hearing adults; individuals enhance formant movement and duration contrasts when speaking clearly, and vowel recognition accuracy declines when these differences are reduced in signal-processed or synthesized speech. This study investigated how these cues contribute to vowel recognition by cochlear implant users and normal-hearing individuals listening to noise-vocoded speech. Individuals were tested on vowel recognition using stimuli with and without formant movement and duration cues, and performed a goodness-rating task that found best exemplars for vowels that had formant movement and duration variation. The results suggested that duration was an important cue to recognition for all listeners; removing duration reduced recognition accuracy by 6–19 percentage points in all conditions. Removing formant movement cues reduced the recognition of diphthongs to chance levels, but had little effect on the recognition of monophthongs. Goodness-rating data demonstrated that all listeners preferred systematic patterns of formant movement and duration variation, as occurs in natural speech. The results suggest that cochlear implant users and normal-hearing individuals give similar cue weightings to vowel formant movement and duration, but that formant movement may have less importance under these conditions than in previous studies.
Optimizing cochlear implant frequency boundary tables for vowel perception: A computer simulation
Authors: Fourakis Marios S. , Hawks John W. , Schwager Amy
Published: September 2013

For cochlear implants, the assignment of frequency bands to electrodes is a variable parameter that determines what region of the acoustic spectrum will be represented by each electrode’s output. Technology will soon allow for considerable flexibility in programming this parameter. In a first attempt to optimize these assignments for vowel perception, a computer program was written to evaluate different assignment schemes for categorization accuracy based strictly on the frequency values of the first two or three formants. Databases [J. Hillenbrand et al., J. Acoust. Soc. Am. 97, 3099–3111 (1995)] of formant measurements from American English vowels as uttered by men, women, and children were used. For this simulation, it was assumed that each formant frequency was associated with only the frequency band its center frequency fell within. Each pattern of frequency bands was assigned a vowel category identification based on the plurality of tokens whose intended identification category fell within that pattern. A range of frequency scaling schemes for 19 and 20 electrode arrays was evaluated, with the best of these fine tuned for minimum error. The results indicate that manufacturer’s default assignments categorize reasonably well, but a bark-scaled scheme yielded the best unmodified classifications.
Usable speech processing: A novel approach to processing speech in degraded environments
Authors: Smolenski Brett Y.
Published: September 2013

One of the main challenges still plaguing speech processing applications is enabling them to work in operational environments where interference and noise abound. The traditional approach has been to use some form of adaptive filtering operation. However, since the speech is nonstationary, it is possible to extract segments from the speech signal that have a large segmental signal-to-noise ratio (SNR) even when the overall SNR is very low. Such high SNR segments frequently occur during voiced speech, and experiments have shown that, using a speaker identification system, these high SNR segments can be correctly identified even when the entire utterance cannot. However, the segmental SNR is not normally known a priori. In this research, a statistical model is first developed for the segmental SNR values for several commonly occurring environments. It is then shown that a modified sinusoidal model and the Teager energy operator can be combined using a context dependent nonlinear regression technique to obtain a low variance estimate of the segmental SNR values provided the frame size is larger than 30 samples.
Intelligibility of an ASR-controlled synthetic talking face
Authors: Siciliano Catherine , Williams Geoff , Faulkner Andrew , Salvi Giampiero
Published: September 2013

The goal of the SYNFACE project is to develop a multilingual synthetic talking face, driven by an automatic speech recognizer (ASR), to assist hearing-impaired people with telephone communication. Previous multilingual experiments with the synthetic face have shown that time-aligned synthesized visual face movements can enhance speech intelligibility in normal-hearing and hearing-impaired users [C. Siciliano et al., Proc. Int. Cong. Phon. Sci. (2003)]. Similar experiments are in progress to examine whether the synthetic face remains intelligible when driven by ASR output. The recognizer produces phonetic output in real time, in order to drive the synthetic face while maintaining normal dialogue turn-taking. Acoustic modeling was performed with a neural network, while an HMM was used for decoding. The recognizer was trained on the SpeechDAT telephone speech corpus. Preliminary results suggest that the currently achieved recognition performance of around 60% frames correct limits the usefulness of the synthetic face movements. This is particularly true for consonants, where correct place of articulation is especially important for visual intelligibility. Errors in the alignment of phone boundaries representative of those arising in the ASR output were also shown to decrease audio-visual intelligibility. [Work supported by the EU IST Project 2001-33327.]
Auditory analysis for speech recognition based on physiological models
Authors: Jeon Woojay , Juang Biing-Hwang
Published: September 2013

To address the limitations of traditional cepstrum or LPC based front-end processing methods for automatic speech recognition, more elaborate methods based on physiological models of the human auditory system may be used to achieve more robust speech recognition in adverse environments. For this purpose, a modified version of a model of the primary auditory cortex featuring a three dimensional mapping of auditory spectra [Wang and Shamma, IEEE Trans. Speech Audio Process. 3, 382–395 (1995)] is adopted and investigated for its use as an improved front-end processing method. The study is conducted in two ways: first, by relating the model’s redundant representation to traditional spectral representations and showing that the former not only encompasses information provided by the latter, but also reveals more relevant information that makes it superior in describing the identifying features of speech signals; and second, by observing the statistical features of the representation for various classes of sound to show how different identifying features manifest themselves as specific patterns on the cortical map, thereby becoming a place-coded data set on which detection theory could be applied to simulate auditory perception and cognition.
Regularized reestimation of stochastic duration models for phone-classification
Authors: Russell Martin J. , Jackson Philip J. B.
Published: September 2013

Recent research has compared the performance of various distributions (uniform, boxcar, exponential, gamma, discrete) for modeling segment (state) durations in hidden semi-Markov models used for phone classification on the TIMIT database. These experiments have shown that a gamma distribution is more appropriate than exponential (which is implicit in first-order Markov models), and achieved a 3% relative reduction in phone-classification errors [Jackson, Proc. ICPhS, pp. 1349–1352 (2003)]. The parameters of these duration distributions were estimated once for each model from initial statistics of state occupation (offline), and remained unchanged during subsequent iterations of training. The present work investigates the effect of reestimating the duration models in training (online) with respect to the phone-classification scores. First, tests were conducted on duration models reestimated directly from statistics gathered in the previous iteration of training. It was found that the boxcar and gamma models were unstable, meanwhile the performance of the other models also tended to degrade. Secondary tests, using a scheme of annealed regularization, demonstrated that the losses could be recouped and a further 1% improvement was obtained. The results from this pilot study imply that similar gains in recognition accuracy deserve investigation, along with further optimization of the duration model reestimation procedure.
The relation between semantics and lexical properties in spoken word production
Authors: Amico Patricia , Charles-Luce Jan , McEldowney Elizabeth
Published: September 2013

Previous research has demonstrated facilitation in speech production when multiple levels are activated. In the present study a time course of facilitation and the contribution of various lexical and semantic properties in facilitating spoken word production was investigated. An experiment was conducted that examined the effects of semantics and lexical properties on acoustic-phonetic duration on spoken word production. Specifically, the primary interest was how duration changed as a function of semantic context and its interaction with the frequency and similarity neighborhood of CVC words. The semantic contexts and targets were presented using a visual naming paradigm. Target words were presented either 100 or 1500 ms after the offset of a simultaneous presentation of a string of three primes. The three primes were either all semantically related to the target, all semantically unrelated to the target, or all nonlinguistic characters neutral to the target. Duration of the target stimuli and reaction times to onset of articulation were measured. The results will be discussed in terms of interactive activation. [Work supported by NIH NIDCD Grant R01 0265801.]
Adaptation to structural modifications of the human vocal tract during speech: Electropalatographic measures
Authors: Aasland Wendi A. , Baum Shari R. , McFarland David H.
Published: September 2013

Structural modifications to the vocal tract force speakers to alter their previously learned articulatory patterns in order to produce perceptually adequate speech. Previous research has shown that acoustic output in the production of alveolar consonants changes during adaptation to structural alterations of the palate, but to date, little is known regarding exactly how these changes result kinematically. The present study examines the adjustments made to tongue–palate contact patterns, measured using electropalatography (EPG), during adaptation to a palatal perturbation for the fricative [s]. Productions of the nonsense word [asa] were elicited in nine subjects at five time intervals, 15 min apart, while speakers wore electropalates modified with a thicker-than-normal alveolar ridge. Between measurement intervals, speakers read [s]-laden passages to promote adaptation. Productions were also elicited with an unperturbed electropalate in place to characterize normal articulation. Electropalatographic analyses revealed a posterior shift in center of gravity of tongue–palate contact, alterations in the width of the medial groove necessary for [s] production, and increased variability in productions, which may reflect the instability of the new motor programs. Results are discussed in relation to the development of adaptive articulatory programs in speech motor control. [Work supported by NSERC and a FRSQ Bourse de Formation.]
Sensorimotor adaptation to acoustic perturbations in vowel formants
Authors: Villacorta Virgilio , Perkell Joseph , Guenther Frank
Published: September 2013

The goal of this research is to study the auditory component of feedback control in speech production. This experiment investigates auditory sensorimotor adaptation (SA) as it relates to speech production: the process by which speakers alter their speech production in order to compensate for perturbations of their normal auditory feedback. Specifically, the first formant frequency (F1) was shifted in the auditory feedback heard by naive adult subjects as they produced vowels in single syllable words. Initial results indicate that subjects demonstrate compensatory formant shifts in their speech. This compensation was also present after training when acoustic feedback was masked by noise. This suggests that internal models used in the control of speech movements can be constantly updated by auditory feedback. These results in voiced speech are consistent with results from Houde and Jordan [Science 279, 1213–1216 (1998)], which demonstrated SA in whispered speech. A second study, currently underway, investigates perceptual discrimination of vowel stimuli differing in F1 frequency, using the same subjects as in the SA studies. Cross-subject relations between discrimination scores and extent of compensation will be presented and discussed. [Work supported by NIDCD Grant R01-DC01925.]
Statistical comparison of similarity tests applied to speech production data
Authors: Kollia H. , Jorgenson Jay , Saint Fleur Rose , Foster Kevin
Published: September 2013

Statistical analysis of data variability in speech production research has traditionally been addressed with the assumption of normally distributed error terms. The correct and valid application of statistical procedure requires a thorough investigation of the assumptions that underlie the methodology. In previous work [Kollia and Jorgenson, J. Acoust. Soc. Am. 102 (1997); 109 (2002)], it was shown that the error terms of speech production data in a linear regression can be modeled accurately using a quadratic probability distribution, rather than a normal distribution as is frequently assumed. The measurement used in the earlier Kollia–Jorgenson work involved the classical Kolmogorov–Smirnov statistical test. In the present work, the authors further explore the problem of analyzing the error terms coming from linear regression using a variety of known statistical tests, including, but not limited to chi-square, Kolmogorov–Smirnov, Anderson–Darling, Cramer–von Mises, skewness and kurtosis, and Durbin. Our study complements a similar study by Shapiro, Wilk, and Chen [J. Am. Stat. Assoc. (1968)]. [Partial support provided by PSC-CUNY and NSF to Jay Jorgenson.]
Nonlinearity between acoustics and articulation in Hungarian transparent vowels
Authors: Benus Stefan , Kirke Karen D. , Gafos Adamantios I.
Published: September 2013

We present novel results from the acoustic and articulatory investigation of the production of the transparent vowels (TVs) /i/, /i:/, /e:/ in Hungarian (colon denotes length). The acoustic measurements of the front–back distinction (second formant, the difference of the first and second formants [Ladefoged, 1993]) show that the effect of adjacent back vowels on the front quality of the TVs is only weakly significant. The articulatory measurements of the same data, however, show that adjacent back vowels cause highly significant retraction of the tongue body during the production of the front TVs. The significance of this finding lies in its relevance to the relationship between phonetics and phonology. Our results demonstrate that minor phonetic differences in articulation, impossible to access by traditional theory, correlate with full-fledged phonological alternation of suffix selection in Hungarian. Traditional phonological accounts predict no effect of continuous phonetic details on discrete phonological generalizations. This is supported in our acoustic data but contrasts with our articulatory findings. In the paper we propose a dynamic model where phonological transparency is directly related to nonlinearity between acoustics and articulation [Stevens, 1989; Wood, 1979]. [Work supported by NIH.]
Theory of acoustic design of opera house and a design proposal
Authors: Ando Yoichi
Published: September 2013

First of all, the theory of subjective preference for sound fields based on the model of auditory-brain system is briefly mentioned. It consists of the temporal factors and spatial factors associated with the left and right cerebral hemispheres, respectively. The temporal criteria are the initial time delay gap between the direct sound and the first Reflection (Dt1) and the subsequent reverberation time (Tsub). These preferred conditions are related to the minimum value of effective duration of the running autocorrelation function of source signals (te)min. The spatial criteria are binaural listening level (LL) and the IACC, which may be extracted from the interaural crosscorrelation function. In the opera house, there are two different kind of sound sources, i.e., the vocal source of relatively short values of (te)min in the stage and the orchestra music of long values of (te)min in the pit. For these sources, a proposal is made here.
Magnetoencephalographic responses in relation to temporal and spatial factors of sound fields
Authors: Soeta Yoshiharu , Nakagawa Seiji , Tonoike Mitsuo , Hotehama Takuya , Ando Yoichi
Published: September 2013

To establish the guidelines based on brain functions for designing sound fields such as a concert hall and an opera house, the activities of the human brain to the temporal and spatial factors of the sound field have been investigated using magnetoencephalography (MEG). MEG is a noninvasive technique for investigating neuronal activity in human brain. First of all, the auditory evoked responses in change of the magnitude of the interaural cross-correlation (IACC) were analyzed. IACC is one of the spatial factors, which has great influence on the degree of subjective preference and diffuseness for sound fields. The results indicated that the peak amplitude of N1m, which was found over the left and right temporal lobes around 100 ms after the stimulus onset, decreased with increasing the IACC. Second, the responses corresponding to subjective preference for one of the typical temporal factors, i.e., the initial delay gap between a direct sound and the first reflection, were investigated. The results showed that the effective duration of the autocorrelation function of MEG between 8 and 13 Hz became longer during presentations of a preferred stimulus. These results indicate that the brain may be relaxed, and repeat a similar temporal rhythm under preferred sound fields.
A scale value for the balance between stage and pit and inside an historical opera house
Authors: Prodi Nicola , Velecka Sylvia
Published: September 2013

Despite its recognized importance, the balance between the singer and the orchestra inside an opera house has received minor attention in the past. In fact, after the fundamental work of Meyer [J. Meyer, ‘‘Some problems of opera house acoustics,’’ Proceedings of 12th I.C.A., Vancouver, 1986, pp. 13–18], who explained why the solo singing voice can compete with the orchestra, only partial results were reported on this perceived attribute. In this work a reference scale to assess the balance inside an historical opera house is achieved by means of listening tests inside a controlled room. Two scaling experiments were performed based on the acoustical data measured inside an historical opera house, the Teatro Comunale di Ferrara, Italy. By doing so all of the relevant acoustical characteristics of a typical Italian-style opera house could be exactly reproduced. Acceptable values do not differ much in the stalls and in the boxes and are within 2 dB(A) to +2.3 dB(A). The transfer of the findings to other types of opera houses is discussed, too.
Song variation and environmental auditory masking in the grasshopper sparrow
Authors: Lohr Bernard , Dooling Robert J. , Gill Douglas E.
Published: September 2013

Some grassland bird species, in particular grasshopper sparrows (Ammodramus savannarum), sing songs with especially high mean frequencies (7.0–8.0 kHz). Acoustic interference is one potential explanation for the evolution of high frequency vocalizations, particularly in open habitats. We tested predictions from a model of effective auditory communication distances to understand the potential effects of vocal production and environmental auditory masking on vocal behavior and territoriality. Variation in the spectral structure of songs and the size and shape of territories was measured for grasshopper sparrows in typical grassland habitats. Median territory areas were 1629 m2 at a site in the center of the species range in Nebraska, and 1466 m2 at our study site in Maryland, with average territory diameters measuring 20.2 m. Species densities and sound pressure levels also were determined for stridulating insects and other noise sources in the habitat. Based on current models of effective communication distances, known noise levels, and information on hearing abilities, our results suggest that auditory sensitivity and environmental noise could be factors influencing the mean frequency and spatial dynamics of territorial behavior in grassland birds. [Work supported by NIH and the CRFRC.]
Changing the average frequency of contact calls is associated with changes in other acoustic parameters in the budgerigar (Melopsittacus undulatus)
Authors: Osmanski Michael , Dooling Robert
Published: September 2013

The most-often produced vocalization of the budgerigar, a small parrot native to Australia, is the short (100–150 ms) frequency-modulated contact call. These calls play a role in maintaining flock dynamics and are believed to act as vocal signatures in these birds. Previous findings in our lab have shown that budgerigars can control the intensity of their vocal behavior and exhibit a robust Lombard effect (Manabe et al., 1998). Recently, we have shown that there is a high degree of stereotypy in contact calls across a number of acoustic parameters (Osmanski and Dooling, 2004). Questions arise concerning the limits of plasticity in these calls and the relation or interdependence among the various parameters. As a first approach to answering these questions, four budgerigars were trained using operant conditioning methods to change the average peak frequency of their contact calls (both upward and downward in frequency) to obtain access to a food reward. Results show that these birds can both increase and decrease the average frequency of their contact calls. Such changes are associated with modifications in a number of other acoustic parameters, suggesting constraints on vocal plasticity. [Work supported by NIH DC-00198 to RJD and NIDCD Training Grant DC-00046.]
An audiometric comparison of primate audiograms
Authors: Coleman Mark N.
Published: September 2013

Audiogram data for 18 species of primates were collected from the literature and analyzed by measuring 13 audiometric variables: frequency and threshold of the primary peak, frequency and threshold of the secondary peak, frequency and threshold of the notch between peaks, low-frequency cutoff, high-frequency cutoff, total area of the audible field, low area, middle area, high area, and total audible range in octaves. All areal measurements were made using IGOR PRO 4.04 wave measurement software. Platyrrhines were found to have significantly better low-frequency sensitivity than like-sized lorisoids with an average of 15-dB difference between the means for the two groups. This difference remains significant even when interindividual variation is considered. Callithrix jacchus and Erythrocebus patas have unusual hearing patterns for primates of their size with marmosets showing a reduction in high-frequency sensitivity, while patas monkeys show a reduction in low-frequency sensitivity. It was also noted that chimps have a notch in sensitivity that falls within the range of greatest sensitivity for humans. These findings are discussed in relation to the morphological adaptations that appear to influence these hearing patterns and the evolutionary significance of such patterns for group communication and predator–prey interactions.
Wave-theory modeling of oceanic T-phase coupling at continental margins and seamounts
Authors: Schmidt Henrik , Baggeroer Arthur B. , Sperry Brian J.
Published: September 2013

The role of seismo-acoustic seabed scattering as a mechanism for coupling of seismic energy into oceanic teleseismic waves or T-phases is investigated using a new versatile modeling capability for seismo-acoustic propagation in laterally inhomogeneous or range-dependent ocean waveguides. The Virtual Source Approach (VISA) uses a local Rayleigh–Kirchhoff approximation to handle the transmission and reflection of plane waves at the vertical interfaces separating horizontally ocean stratified sectors. Combined with the wavenumber integration approach which inherently computes the plane-wave decomposition of the seismo-acoustic field in stratified fluid-elastic waveguides, this approach provides a robust approximation to the seismo-acoustic coupling phenomena in shallow and deep ocean waveguides. The VISA approach has been implemented in the OASES seismo-acoustic modeling framework and used to investigate the role of seismo-acoustic conversion and scattering by seabed topography and roughness in generating oceanic T-phases at continental margins and seamounts. It is demonstrated that the excitation of the oceanic T-phases can be explained by the coupling of crustal shear body-waves into seismic interface waves, or seabed Scholte waves, which then subsequently scatter into the waterborne modal spectrum. This wavenumber conversion mechanism implies that the excitation of the T-phases will be significantly stronger by earthquakes producing crustal SV-waves than those producing predominantly P-waves. This in turn suggests that earthquakes associated with dip-slip failure modes excite significantly stronger T-phases than buried explosive sources.
The use of regional hydrophone datasets for statistical studies of mid-ocean ridge earthquake populations
Authors: Bohnenstiehl DelWayne R.
Published: September 2013

Seismically generated tertiary (T-) waves can be used to detect and locate the shallow hypocenter earthquakes associated with mid-ocean ridge spreading centers and oceanic transforms. During the last several years, regional seismic catalogs, which contain estimates of T-wave source regions, origin times and acoustic source levels, have been generated from continuous hydroacoustic monitoring along the northern Mid-Atlantic Ridge and equatorial East Pacific Rise. As the mechanics of T-wave generation remain poorly understood, critics of these datasets have questioned the correlation between T-wave source regions and seismic epicenters, as well as the usefulness of acoustic source level as a proxy for earthquake size. Recent analyses, however, suggest that these data provide a reasonably complete record of seismicity within the mid-ocean ridge environment, yielding significant improvements in location accuracy and catalog completeness level, relative to global seismic catalogs. Most importantly, it appears that many fundamental and well-established properties of seismic distributions can be studied using T-wave catalogs, including the modified Omori decay law of aftershocks, a power-law size frequency distribution (Gutenberg–Richter law), a fractal time-clustering behavior (1/f noise) and fractal spatial distribution of epicenters. These observations show that T-wave derived earthquake catalogs can be used for quantitative seismo-tectonic studies in the ridge setting.
The physical acoustics of HIFU exposure: A suite of in vitro studies
Authors: Roy Ronald A. , Holt R. Glynn , Cleveland Robin O.
Published: September 2013

Exposing tissue and blood media to high-intensity focused ultrasound leads to a broad range of physical and biological effects. In addition to tissue heating from thermo-viscous absorption, one encounters nonlinear and second order phenomena such as wave-front shocking, radiation stress, acoustic streaming, bubble nucleation and cavitation. All of these processes impact one’s ability to effect controlled tissue heating. This paper reviews a suite of studies under way at Boston University that focus on developing a quantitative understanding of the physical acoustics of high-intensity focused ultrasound. We describe research into HIFU heating and both uniform and flow-through tissue phantom and address the potentially critical roles played by bubbles and acoustic cavitation. [Work supported by the U.S. Army and the Center for Subsurface Sensing and Imaging Systems via NSF ERC Award No. EEC-9986821.]
The relative effects of cavitation and nonlinear ultrasound propagation on HIFU lesion dynamics in a tissue phantom
Authors: Khokhlova Vera A. , Bailey Michael R. , Reed Justin , Kaczkowski Peter J.
Published: September 2013

The relative importance of the effects of acoustic nonlinearity and cavitation in HIFU lesion production is studied experimentally and theoretically in a polyacrylamide gel. A 2-MHz transducer of 40-mm diameter and 45-mm focal length was operated at different regimes of power, and in cw or duty-cycle regimes with equal mean intensity. Elevated static pressure was applied to suppress bubbles, increase boiling temperature, and thus to isolate the effect of acoustic nonlinearity in the enhancement of lesion production. Experimental data were compared with the results of simulations performed using a KZK acoustic model combined with the bioheat equation and thermal dose formulation. Boiling and the typical tadpole-shaped lesion shifting towards the transducer were observed under standard atmospheric pressure. No boiling was detected and a symmetric thermal lesion formed in the case of overpressure. A delay in lesion inception time was registered with overpressure, which was hypothesized to be due to suppressed microbubble dynamics. The effect of acoustic nonlinearity was revealed as a substantial decrease in the lesion inception time and an increase in the lesion size for high-amplitude waves under both standard and overpressure conditions. [Work supported by ONRIFO, NASA/NSBRI, NIH Fogarty, and CRDF grants.]
Experimental verification of enhancement of HIFU-induced heating of tissue mimicking phantoms due to acoustic nonlinearity
Authors: Sinilo Tatiana V. , Khokhlova Vera A. , Sapozhnikov Oleg A.
Published: September 2013

A possibility of enhancement of heat deposition of high-intensity focused ultrasound (HIFU) in tissue mimicking phantoms due to acoustic nonlinearity is experimentally studied. A gelatin sample of high concentration was used to model medium with acoustic properties similar to those of biological tissue. Several sets of pulse-periodic regimes with the same mean power but different pulse amplitudes and durations were used. In regimes with higher amplitudes (lower duty cycles), the waveform in the focal region was more strongly distorted because of acoustic nonlinearity. In the linear medium all the regimes of one set would give the same heating. Optical shadow pictures of the HIFU-heated region were taken in the regimes with different duty cycles but the same mean power. The heated region size increased for lower duty cycles, but the shadow shape remained regularly cigar-like in all the regimes, even when the ultrasound waveform was shocked. The transmitted signal measured by a hydrophone was stable. This showed that cavitation was not pronounced. Thermocouple measurements and theoretical modeling of gelatin temperature in the focal region of the ultrasound beam showed significant increase of HIFU heat deposition in the presence of shocks in the waveform. [Work supported by CRDF and RFBR.]
Resonant and nonresonant sound radiation from a guitar
Authors: Lyon Richard H.
Published: September 2013

When forces are applied to a flexible structure, a worker in structural acoustics who is concerned with the radiated sound will normally be interested in how much of the sound is arising from the resonant response of the structure compared to the nonresonant response. Lightweight panels below their critical frequency are likely to have a significant amount of sound arising from nonresonant response due to mass inertial response localized around the location of force application. Since a guitar top is such a structure, it is natural to inquire about the comparison between resonant and nonresonant radiation from a guitar over its normal range of frequencies. Measurements have been made on a student Yamaha and a concert quality Modranas/Connor guitar. The results of these measurements and their interpretation will be discussed.
The application of cochlear audio analysis techniques to electro-acoustic music
Authors: Mills John A.
Published: September 2013

Electro-acoustic music has been ignored by music theorists for years. One theory suggests that this deficiency is due to the lack of an objective visual representation of this type of music. This research is focused upon the audio analysis involved in the creation of a ‘‘pseudoscore’’ for electro-acoustic music. Computer analysis of music is a complicated task. The ‘‘holy grail’’ of automatic computer music analysis has most often been the translation of a continuous pressure variation into traditional Western musical notation. Since electro-acoustic music is rarely able to be transcribed into this type of notation, the automatic analysis of electro-acoustic music confounds many previous algorithms. A top-down approach is suggested in order to extract acoustic and musical information from recordings of electro-acoustics music. A top-down approach has already proven successful in extraction of musical tempo [E. Scheirer, M.I.T. Ph.D. thesis (2000)]. Since humans are able to translate a continuous pressure variation into useful acoustic information, when using a top-down approach to automatic music analysis, a cochlear model is used as a front end. Although this model apparently encodes some redundant information, cross- and auto-correlation techniques allow easier extraction of some acoustic information. This presentation will detail the current state of this research.
Low-cost coding of directivity information for the recording of musical instruments
Authors: Braasch Jonas , Martens William L. , Woszczyk Wieslaw
Published: September 2013

Most musical instruments radiate sound according to characteristic spatial directivity patterns. These patterns are usually not only strongly frequency dependent, but also time-variant functions of various parameters of the instrument, such as pitch and the playing technique applied (e.g., plucking versus bowing of string instruments). To capture the directivity information when recording an instrument, Warusfel and Misdariis (2001) proposed to record an instrument using four channels, one for the monopole and the others for three orthogonal dipole parts. In the new recording setup presented here, it is proposed to store one channel at a high sampling frequency, along with directivity information that is updated only every few milliseconds. Taking the binaural sluggishness of the human auditory system into account in this way provides a low-cost coding scheme for subsequent reproduction of time-variant directivity patterns.
Exploring a Van der Pol woodwind
Authors: Barjau Ana , Genesc Meritxell
Published: September 2013

The behavior of woodwinds can be studied from a theoretical point of view with a variety of mathematical models, ranging from very realistic to fairly simple ones. All of them have in common a nonlinearity responsible for the self-sustained oscillation. In simple models, this nonlinearity is represented through an algebraic equation, while in realistic ones it may be split into different differential equations representing the dynamics of the mechanical device (reed, lips...) and of the air flowing through it. In this paper, an intermediate model is explored. A Van der Pol oscillator is used to represent just the nonlinear device of a woodwind (reed valve, lips...), while the air column is simplified into a cylindrical or conical one, and its behavior is described through an impulse response. The global system is thus represented mathematically through two coupled differential equations, a linear one containing delayed variables and a nonlinear one. Their integration leads to different sounds, some of which are close to those produced by real instruments.
Spatial representation of soundscape
Authors: Boubezari Mohammed , Bento Coelho Jos-Luis
Published: September 2013

For the last 30 years the concept of soundscape has been largely adopted in many scientific disciplines and by the urban experts for the benefit of a better comprehension and management of the sound environment. However, the spatial representation of the soundscape as a simple tool for the description, management or composition of sound environment is always needed. In this article a method is presented for the spatial sound representation with differentiated sources. The first results are shown. This method gives an account of the soundscape as close as possible to the way it can be perceived by the listener in each location. This method generates qualitative sound maps in a reduced urban scale, based on in situ measurements and on the implication of the measuring subject perception. The maps are sufficient enough to isolate many sound sources of the overall sound field. In this manner, sound quality refers to the sound attribute of a perceived object. It is neither an aesthetic judgment nor traditional psychoacoustics criteria. Concrete examples of application to squares in the city of Lisbon will be shown and discussed. The limits and the prospects of such a qualitative representation will also be presented and discussed.
Assessing noise pollution in sensitive areas: Soundscape analysis in an alpine valley by psychoacoustic means
Authors: Lercher Peter , Genuit Klaus , Reichart Urs , Heimann Dietrich
Published: September 2013

Alpine valleys are sensitive areas due to topography, meteorology, housing, and land-use pattern, that modify noise propagation and make protection against noise pollution rather difficult. The ‘‘amphitheater’’ effect was mentioned as explanation for deviating noise-annoyance curves and health effects observed at lower sound levels. However, detailed empirical analyses are lacking. In this study a series of simultaneous, binaural sound recordings was carried out in several cross sections of the Wipp Valley along the central-European transportation route to the Brenner pass. During 6 weeks a wide variety of day- and week times was sampled with variable wind (‘‘Foehn’’) and weather conditions (dry, rain, snow). Sound recordings were paralleled by meteorological recordings near the source and on the slope. First analyses have revealed several facts. (1) The assumption of linear sound propagation to the slope is seriously in error. (2) Tonal components from gearboxes are a significant feature in the slope recordings. (3) Low-frequency modulations make the sound more intrusive on the slope—while near the source this feature is better masked. (4) Low background sound levels (<30 dB,A) on the slopes are in sharp contrast with incoming sound levels (52 dB,A,Leq about 1200 m from the source). (5) Meteorology leads to substantial changes in measured sound levels.
Computer models of possible physiological contributions to low-level auditory scene analysis
Authors: Meddis Ray
Published: September 2013

Auditory selective attention is a general term for a wide range of phenomena including grouping and streaming of sound sources. While we know a great deal about the circumstances in which these phenomena occur, we have little understanding of the physiological mechanisms that give rise to these effects. Because attention is sometimes under conscious control, it is tempting to conclude that attention is a high-level/cortical function and beyond our current understanding of brain physiology. However, a number of mechanisms operating at the level of the brainstem may well make an important contribution to auditory scene analysis. Because we know much more about the response of the brainstem to auditory stimulation, we can begin some speculative modeling concerning their possible involvement. Two mechanisms will be discussed in terms of their possible relevance: lateral and recurrent inhibition at the level of the cochlear nucleus. These are likely to contribute to the selection of auditory channels. A new approach to within-channel selection on the basis of pitch will also be discussed. These approaches will be illustrated using computer models of the underlying physiology and their response to stimuli used in psychophysical experiments.
Auditory distance perception in fixed and varying simulated acoustic environments
Authors: Schoolmaster Matthew , Kopčo Norbert , Shinn-Cunningham Barbara G.
Published: September 2013

Listeners must calibrate to the acoustic environment in order to judge source distance using reverberation. Results of a previous study [Schoolmaster, Kopčo, and Shinn-Cunningham, J. Acoust. Soc. Am. 113, 2285 (2003)] showed that distance perception is more accurate when the simulated acoustic environment is consistent rather than randomly chosen from trial to trial; however, the environments were very different (anechoic versus a large classroom). The study was replicated here using two rooms that were much more similar. Each subject completed two series of trials. In the fixed-room series, the room (large or small) was fixed within a trial block. In the mixed-room series, the room was randomly chosen from trial to trial. Half of the subjects performed the mixed and half performed the fixed series first. Differences between subject groups were smaller in the current study than when anechoic and reverberant trials were intermingled; performance in the two subject groups was similar for (1) fixed trials and (2) trials simulating the large room. However, there was an interaction between subject group and room. Results suggest that listeners calibrate distance judgments based on past experience, but that such calibration partially generalizes to similar environments [Work supported by AFOSR and the NRC.]