bluuebunny/crossref_metadata_2025 huggingface dataset:
Automated categorization of bioacoustic signals: Avoiding perceptual pitfalls
Authors: Deecke Volker B. , Janik Vincent M.
Published: October 2013

Dividing a species acoustic repertoire into biologically relevant categories presents a widespread problem in the study of animal sound communication. Such categorization is fundamental to any attempt to compare repertoires between contexts, individuals, populations or species. Automated procedures allow rapid, repeatable and objective categorization, however, in the past they have often performed poorly at detecting biologically meaningful sound classes. Arguably this is because automated methods have often failed to address the nonlinearities of animal sound perception and a method that incorporates dynamic time warping and an adaptive resonance theory (ART) neural network to avoid some of the problems is presented here. The method was tested on 104 randomly chosen whistles recorded from four captive bottlenose dolphins (Tursiops truncatus). This data set includes known biologically meaningful categories in the form of 42 distinctive stereotyped whistles produced when each individual was kept in isolation. The automated procedure correctly grouped all but two of the stereotyped whistles into their respective categories, and thus performed as good as human observers. However, compared to humans, the automated system provided finer categorization of the remaining non-stereotyped whistles. These results suggest that this methodology provides a repeatable and objective means of splitting bioacoustic signals into biologically meaningful categories.
1 kHz vibration increases proteoglycan production in ATDC5 chondrocytes
Authors: Argadine Heather M. , Kinnick Randall R. , Greenleaf James F. , Bolander Mark E.
Published: October 2013

In vitro studies have shown that treatment with 1.5 MHz ultrasound signal (160 mW/cm2) at a 200 μs tone burst repeating at 1 kHz increases proteoglycan synthesis in chondrocytes [J. Parvisi et al., J. Orthop. Res. 17, 488–494 (1999)]. It was hypothesized that a continuous 1 kHz signal would be similar to the pulsed 1.5 MHz signal in stimulating chondrocytes to produce proteoglycan, which may cause accelerated fracture healing. In vitro experiments were performed with ATDC5 cells, a chondrogenic clonal cell line, plated in 6-well plates for 3 to 7 days before receiving ultrasound treatments. Cells were treated with either 1.5 MHz pulsed signal or 1 kHz signal for 20 minutes per day for 9 to 11 days. The signals were calibrated so that the bottom of the 6-well plate moved 10 nm for each condition. After the final treatment, cell layers were stained with Alcian blue, which stains cartilage nodules providing a measure of chondrogenesis. Both 1.5 MHz and 1 kHz led to a highly significant increase in chondrogenesis compared to control. Quantitative image analysis of stained wells showed that treatments with either signal increased number of nodules 2.3-fold (p<0.02) and total area of nodules 3-fold (p<0.02) compared to controls.
A theoretical study of the feasibility of acoustical tweezer: Ray acoustics approach
Authors: Lee Jungwoo , Shung Kirk
Published: October 2013

Optical tweezer has been found to have many biomedical applications in trapping macromolecules and cells. For the trapping mechanism, there has to be a sharp spatial change in axial optical intensity and the particle size must be much greater than the wavelength. Similar phenomenon may exist in acoustics. This work was undertaken to demonstrate theoretically that it is possible to acoustically trap particles near the focal point if certain conditions are met. Acoustic force exerted on fat tissue in ultrasonic fields is analyzed in ray acoustics regime where the wavelength of acoustic beam is much smaller than the size of the particle. In this paper, the analysis is therefore based on the field pattern produced by a strongly focused 100 MHz ultrasonic transducer with Gaussian intensity distribution. The magnitude of force and Fresnel coefficients at various positions are calculated. According to the simulation results, acoustical tweezer works particularly when the beam width at focus is one wavelength and the tolerance of acoustic impedance mismatch between two media lies within 6.7%. [Work supported by NIH Grant P41-EB2182.]
Microbubble dynamics in contrast agent shell rupture
Authors: Ammi Azzdine Y. , Bridal S. Lori , Cleveland Robin O. , Mamou Jonathan , Wang Grace I. , O’Brien William D.
Published: October 2013

A passive cavitation detector (PCD) was used to study the features of insonified contrast agent microbubble rupture. When Optison microbubbles ruptured, the PCD signal received from single microbubbles showed the presence of post-excitation echoes identified as inertial cavitation and rebounds. Using these signals, the collapse threshold was evaluated as a function of insonation frequency, pulse duration, and pressure amplitude. For each excitation signal, 128 received time traces were recorded (200 MHz sampling, 12-bit), and frequency-domain analysis of the received signals was performed to characterize microbubble collapse. The peak rarefactional pressure threshold for detection of these emissions increased with frequency (e.g., 0.53, 0.87, and 0.99 MPa for 0.9, 2.8, and 4.6 MHz, respectively; 5-cycle pulse duration) and decreased with pulse duration. The modified Herring equation was used to model the bubble response at experimentally determined threshold conditions. These predicted microbubble dynamics were used to determine microbubble rupture indices based on: radial strain, peak velocity, and peak acceleration. This work provides a clearer understanding of the role of contrast agent dynamics in bubble destruction. [Work supported by the cooperative project for biomedical engineering between the University of Illinois at Urbana-Champaign, USA and the Centre National de la Recherche Scientifique, France.]
The effect of size on the acoustic response of polymer-shelled ultrasound microbubble contrast agents
Authors: Bloch Susannah H. , Ferrara Katherine W. , Wisner Erik R.
Published: October 2013

Hard-shelled microbubble contrast agents differ from lipid-shelled agents in that the extra stability provided by their shell prevents dissolution of small gas bubbles: consequently, stable agents can be manufactured with diameters below 1 μm. Near-micron and submicron-sized contrast agents may enable new biomedical applications because they can traverse narrow barriers such as the lymphatic endothelium and the permeable vascular endothelium found in actively growing solid tumors. However, several studies have suggested fundamental differences in the mechanisms of acoustic response of ‘‘soft’’ lipid-shelled ultrasound microbubble contrast agents and ‘‘hard’’ albumin- or polymer-shelled agents. In particular, hard-shelled contrast agents exhibit strong echoes only after the shell is damaged and the enclosed gas is allowed to oscillate freely. Here three polymer-shelled agents with mean diameters of 0.74 μm, 0.91 μm, and 1.33 μm (POINT Biomedical Corp.) are investigated. Significant differences among the agents exist in the probability of response to acoustic interrogation, but not the amplitude or frequency of their acoustic response. These results are explained by a hypothesis of a pressure-dependent threshold diameter above which single hard-shelled agents respond to acoustic interrogation. The implications of this hypothesis for the design of imaging methods and clinical protocols using these agents will be presented.
Optimized interpolations and nonlinearity in numerical studies of woodwind instruments
Authors: Skouroupathis Apostolos
Published: October 2013

The impedance spectra of woodwind instruments with arbitrary axisymmetric geometry are studied. Piecewise interpolations of the instruments’ profile are performed, using interpolating functions amenable to analytic solutions of the Webster equation. Our algorithm optimizes on the choice of such functions, while ensuring compatibility of wave-fronts at the joining points. Employing a standard mathematical model of a single-reed mouthpiece, as well as the time-domain reflection function which is derived from our impedance results, the Schumacher equation is solved for the pressure evolution in time. Analytic checks are made to verify that, despite the nonlinearity in the reed model and in the evolution equation, solutions are unique and singularity-free.
Use and effectiveness of hearing protection in construction
Authors: Neitzel Richard , Seixas Noah
Published: October 2013

This paper presents an analysis of data on noise exposure and hearing protection device (HPD) use among construction workers drawn from several large datasets covering nine construction trades. A unique combination of one-minute dosimetry noise exposure level and simultaneous self-reported HPD use data were evaluated, as were occupational and non-occupational HPD use data collected by questionnaire as part of a longitudinal noise and hearing loss study among apprentices. Direct HPD attenuation measurements were also made on workers at their worksite. Workers reported using HPDs less than one-quarter of the time they were exposed above 85 dBA, the NIOSH Recommended Exposure Limit. Workers who reported always using HPDs in high noise were found to wear them only one-third of the time their exposures exceeded 85 dBA. Direct attenuation measurements indicated that workers achieved more than 50% of the rated attenuation of their HPD on average, but that the variability in achieved attenuation was large. Lastly, when the measured HPD attenuation level and use time data were combined, the effective protection afforded by HPDs was less than 3 dB on average, though there was substantial variation among the different trades. These results demonstrate the need for better hearing conservation programs in construction.
In-situ personal assessment of hearing protector performance–recommendations for an updated standard
Authors: Voix Jérémie , Laville Frédéric , Zeidan Jean
Published: October 2013

Standardized method currently used to assess Hearing Protection Devices (HPD) performance have little correspondence with protection achieved by users in practice. To overcome this problem new measurement method are being proposed. They no longer rely on statistically-based, population-based estimate, such as the Noise Reduction Rating (NRR), but on an individual assessment of the performance of the HPD. These new methods are highly beneficial for effective Hearing Conservation Programs as they provide individual attenuation values for noise-exposed workers in lieu of the statistically-based laboratory performance estimate, but their implementation will necessitate complete revision of existing standards. Among the issues to be addressed, are the aspect of the accuracy of the measurements performed by those methods and associated devices and also the replacement of any single number rating of HPD (such as the NRR) in existing standards. Solutions to address theses issues and some general guidelines for an updated standard will be presented.
Finite-element modeling of acoustic scattering from realistic elastic structures
Authors: Burnett David S.
Published: October 2013

Over the last twenty years the author has developed several finite-element codes for modeling acoustic scattering from complicated elastic structures. All the codes have employed 3-D continuum mechanics throughout the computational domain. Thin structural components, such as plates and shells, are modeled with 3-D elasticity rather than plate or shell theories. The codes are therefore fully 3-D, i.e., in both physics and geometry. The talk will first explain the rationale for this approach and touch briefly on essential aspects of previous codes. It will then focus in more detail on the current code under development, which employs relatively inexpensive commercial software well-suited for this fully 3-D approach.
Scattering by tilted plastic cylinders having curved ends and truncated plastic cones
Authors: Espana Aubrey , Baik Kyungmin , Marston Philip L.
Published: October 2013

In prior research an acoustic backscattering enhancement was demonstrated for a bluntly truncated plastic cylinder caused by a merged caustic [F. J. Blonigen and P. L. Marston, J. Acoust. Soc. Am. 107, 689–698 (2000)]. This was confirmed with analogous light scattering experiments [P. L. Marston, Y. B. Zhang, and D. B. Thiessen, Appl. Opt. 42, 412–417 (2003)]. In recent work a different backscattering enhancement associated with a caustic was identified for tilted plastic cylinders having curved ends. When the cylinder is tilted so as to focus a shear wave at the point of internal specular reflection, the curvature of the outgoing acoustic wavefront vanishes orthogonal to the meridional plane. This was verified with analogous light scattering experiments. The flatness of the outgoing wavefront enhances the scattering. Backscattering by truncated plastic cones as a function of tilt also shows enhancements associated with the composition of the target. The time dependence of the backscattering envelope as a function of tilt reveals different features depending on whether the top or bottom of the cone is illuminated by tone bursts. [Work supported by the Office of Naval Research.]
Binaural models for the detection of interaural incoherence
Authors: Goupell Matthew J. , Hartmann William M.
Published: October 2013

Human listeners have a remarkable ability to detect small amounts of interaural incoherence in bands of noise. Experiments using frozen noise, all with the same value of interaural coherence (0.9922), show that when the bandwidth is large the ability to detect incoherence is essentially determined by the value of coherence itself. However, when the bandwidth is narrow, e.g. 20 Hz, detection depends on some function of the moment-to-moment fluctuations in interaural phase differences and interaural level differences. Experiments using hundreds of different frozen noises and two bandwidths were performed to find the best binaural model for predicting detection performance for such narrow bands. Successful models incorporated known elements of binaural hearing—temporal averaging and compression. Models with independent processing of interaural phase fluctuations and interaural level fluctuations were marginally more successful than models based on fluctuations in lateral position. The most successful models correlated well with average human responses, r=0.89. It seems likely that such individual fluctuations remain dominant in masking level difference experiments in the narrow band limit. [Work supported by the NIDCD grant DC 00181.]
Binaural effects in simultaneous room reflection masking
Authors: Buchholz Joerg M.
Published: October 2013

Masked thresholds (MT) for a single test reflection masked by a direct sound (200 ms long broadband noise) were measured dependent on the time delay of the reflection for diotic as well as dichotic stimulus presentation. In the diotic case, the direct sound and the test reflection were presented equally to both ears via headphones. In the dichotic case, an ITD of 0.5 ms was added to the test reflection. In order to focus on simultaneous masking effects, the reflection was truncated in such a way that it formed a common offset with the direct sound. For the diotic case, the resulting data showed a MT increase with increasing reflection delay and for the dichotic case a MT decrease with increasing reflection delay, producing an intercept between both curves at a reflection delay of 6–8 ms. Hence, negative BMLDs (up to −8 dB) were found for very early reflections and positive BMLDs (up to +8 dB) for later reflections, suggesting a binaural mechanism that suppresses very early reflections and enhances later reflections. The measurement results are discussed in the background of different auditory models.
How the pharyngeal resonator is controlled articulatorily
Authors: Esling John H.
Published: October 2013

Indigenous languages of North America, especially west coast Vancouver Island Nuuchahnulth and Interior BC Salish, illustrate an extensive range of use of the pharyngeal resonator. The pharynx is classified as part of the laryngeal articulator (rather than lingual) because the laryngeal constrictor mechanism (controlling changes from the glottis to the aryepiglottic folds) is the principal articulator whose movements determine the shape of the pharynx. The Nuuchahnulth (Nootka) dialects (e.g., Ahousaht) and Thompson, Moses-Columbia, and Spokane/Kalispel Salish varieties all close the larynx completely for the speech sound epiglottal stop. With the airway optimally shut, the pharynx is small; the aryepiglottic folds are pressed against the epiglottic tubercle, the tongue is retracted, and the larynx is raised. There are good indications that the thyroarytenoid, aryepiglottic, thyroepiglottic, hyoglossus and hyothyroid muscles participate in this buckling maneuvre. These languages also contain either a pharyngeal fricative or approximant (at the same place of articulation but with less stricture) and, in Thompson, also pharyngealized uvulars (with even less stricture). Pharyngeal resonance in all of these cases is dependent initially on the shortening of the supraglottic tube and subsequently on the combined effect of tongue retraction and larynx raising reducing the volume immediately above the supraglottic tube.
Coronal ejectives in Ahtna Athabaskan
Authors: Tuttle Siri
Published: October 2013

Ahtna, a non-tonal Alaskan language, is one of few Athabaskan languages that has retained word-final ejectives. The loss of stem-final glottal stops and ejectives in the other languages is correlated with tonogenesis. Hargus (2004) finds voice quality distinctions (creakiness, variability, and increased energy in higher harmonics) preceding stem-final glottal consonants in Witsuwiten, another nontonal Northern language. The present study considers the acoustics of stem-initial and stem-final [t’] and [ts’] in Central and Lower Ahtna. Stem-initial ejectives are canonically shaped, with defined silent period following oral release. In stem-final position, as predicted by Kari (1990), variable non-ejective realizations are found in both word-final and intervocalic (suffixed) position. These realizations are accompanied by variable voice quality effects in preceding vowels, more frequent and pronounced than those found variably in vowels following stem-initial ejectives. Stem-final [t’] is frequently pronounced as glottal stop, and final [ts’] as [’s] where <,> stands for glottal stop. Strikingly, non-ejective realizations of stem-finals are found in suffixed and non-suffixed pairs with the same stem, suggesting that stem-final effects generalize to all tokens of stems. These findings support Leers (1979) picture of spirantization and suprasegmentalization in Athabaskan tonogenesis, as glottal effects gravitate to nucleus position.
Phonetic structures of Washo
Authors: Midtlyng Patrick J. , Yu Alan C. L.
Published: October 2013

Washo is a severely moribund language spoken in several townships near the California-Nevada border southeast of Lake Tahoe. The Washo language is now used only by approximately 13–20 elderly speakers with very few fluent speakers under 60. Recordings were collected in Dresslerville, Nevada from two native speakers. Recordings collected by William Jacobsen Jr. in the 1960s provided additional data for comparison. Analysis showed that all the stops were either voiceless unaspirated, voiceless aspirated, or ejectivized. Vowel analyses showed the nature of the six long and short vowels. Special attention was paid to the nature of vowels and consonants in and around the stressed syllable. Post-tonic sibilants and sonorants are lengthened intervocalically even though consonantal length is not contrastive in Washo. Vowel length contrasts only in the stressed syllable. Stressed vowels are always long before an intervocalic voiceless unaspirated stop, but vocalic length contrast remains before aspirated stops and ejectives.
Eliminating clutter by coordinate zeroing
Authors: DeFerrari Harry , Rodgers Andrew
Published: October 2013

Shallow water doppler sonar is limited by clutter that masks the presence of slow targets. Clutter arises from scattering from irregularities in the bottom or from fish schools. If the source and receiver are fixed in a bi-static sonar configuration, then the bottom scatterers (and sometimes the fish) have only a zero doppler component. Clutter arises from the doppler leakage resulting from sonar signals that are inherently short in time and therefore cannot resolve bottom reverberation in frequency. Here long continuous pulse compressions signals are considered (m-sequences). Ambiguity diagrams are compared with simple CW pulse and PCM pulses. The unique correlation properties of M-sequences lead to a signal sampling method that resolves sonar returns in complete ortho-normal (CON) data sets. Zero doppler returns are identified as either direct arrivals or reverberation returns and then their particular coordinate can be set zero (coordinate zeroing) without affecting other data points. In this way, the zero-doppler bottom reverberation and all doppler leakage is eliminated. As an example, the return from a slow doppler target is imbedded in a very large number of reverberation returns that have higher signal level. As the zero doppler contributions are removed by coordinate zeroing the target becomes detectable.
High-frequency Acoustic Recording Package (HARP) for long-term monitoring of marine mammals
Authors: Wiggins Sean , Garsha Chris , Hardy Kevin , Hildebrand John
Published: October 2013

Advancements in low-power and high-data-capacity computer technology during the past decade have been adapted to autonomously record sounds from whales over long time periods. Acoustic monitoring of whales has advantages over traditional visual surveys including greater detection ranges, continuous long-term monitoring in remote locations under various weather conditions, and lower cost. One currently used tool for providing long-term acoustic monitoring of marine mammals is an autonomous acoustic recording package (ARP) which uses a tethered hydrophone above a seafloor-mounted instrument frame. Since 2000, ARPs have been deployed to record baleen whale sounds in the Bering Sea, in the Beaufort Sea, in the Gulf of Alaska, off the coast of southern California, around Antarctica and near Hawaii. ARP data have provided new information on the seasonal presence, abundance, call character and patterns of calling whales. The need for a broader-band, higher-data capacity system capable of recording odontocete whales, dolphins and porpoises for long time periods has prompted the development of a High-frequency Acoustic Recording Package (HARP). The HARP design is described and data analysis strategies are discussed using examples of HARP broad-band (sample rates up to 200 kHz) recorded data.
High frequency operation of thermoacoustic coolers and prime movers
Authors: El-Gendy Husam , Kwon Young Sang , Symko Orest G.
Published: October 2013

By operating thermoacoustic engines at high frequencies, 4 kHz and higher, the devices have characteristics which are important for many applications. Since they are resonant systems, the power density increases with frequency. Reduction of device size provides quick thermal response time in both the cooler and the prime mover. Moreover, small device size makes it practical to incorporate them into arrays, which can handle large powers. Most important is the fact that small devices make it simple for operation at high pressures in working gas without exceeding strength of materials limitations. This leads to high power densities. Results will be presented to illustrate how the above features affect device performance for the frequency range of 4 kHz to 21 kHz. Measurements using Particle Image Velocimetry of streaming, instabilities, and resonator mode interactions will be discussed for this high frequency range. Ultimately as the operating frequency is raised, device efficiency is limited by heat conduction along the stack and working fluid. [Work supported by the Office of Naval Research and the State of Utah.]
Effect of frequency on the relationship between intensity discrimination and the detection of amplitude modulation
Authors: Millman Rebecca E. , Bacon Sid P.
Published: October 2013

Wojtczak and Viemeister [J. Acoust. Soc. Am. 106, 1917–1924 (1999)] found a clear relationship between the detection of intensity increments and the detection of amplitude modulation (AM) for continuous 1-kHz sinusoids. In the present study AM detection thresholds and intensity difference limens (DLs) were measured for 1-s gated sinusoids with a frequency of 125 Hz or 1 kHz. The sinusoids were presented at levels between 10 dB SL and 90 dB SPL. Modulation detection thresholds [20 log(m)] improved as the carrier level was increased; for a comparable range of levels (45 dB), the improvement was similar at the two frequencies. Intensity DLs [10 log(ΔI/I)] generally improved as the level of the reference sinusoid was increased, although the improvement (‘‘near miss’’ to Weber’s Law) was smaller at 125 Hz than at 1 kHz over the same 45-dB range of levels. For the 1-kHz sinusoids, the relationship between intensity DLs and modulation detection thresholds was consistent with that shown by Wojtczak and Viemeister (1999). This same relationship apparently does not hold at 125 Hz. Because the relationship between intensity discrimination and modulation detection depends upon frequency, the mechanisms underlying the two tasks may be somewhat different. [Work supported by NIDCD.]
Temporal integration functions of amplitude modulation detection and amplitude modulation depth discrimination
Authors: Lee Jungmee , Long Glenis
Published: October 2013

Previous studies (Lee and Green, 1994; Lee and Bacon, 1997) suggested that both AM rate and AM depth discrimination were influenced by the number of AM cycles, instead of the duration of stimuli. AM detection and AM depth discrimination (standard depth=0.1) were measured as a function of the number of AM cycles for modulation rates of 10, 20, 40, 80, 125, 160, and 320 Hz. Different numbers of modulation cycles were used for each modulation rate: 2, 4, or 8 for 10 Hz; 2, 4, 8, or 16 for 20 Hz; 2, 4, 8, 16, or 32 for 40 Hz; 2, 4, 8, 16, 32, or 64 for 80, 125, 160, and 320 Hz. The carrier was a broadband-noise (10 kHz lowpass), and the carrier was either gated with the modulator or presented 250 ms earlier and 250 ms later than the modulator. The overall level of each presentation was randomized within 6-dB range from 65 dB SPL. The results suggest that there might be different temporal integration processes for AM detection and AM depth discrimination. The pattern is different for lower and higher modulation rates. [Work was supported by NIDCD Grant No. R03 DC06605-01.]
Coherent modulation enhancement: Improving performance in noise for hearing aids and cochlear implants
Authors: Souza Pamela , Atlas Les , Schimmel Steven , Rubinstein Jay , Drennan Ward , Won Jong Ho
Published: October 2013

Difficulty hearing in noise is a pervasive problem for hearing-impaired listeners. One approach is to digitally reduce noise using a modulation filter, which can selectively modify the time envelope of selected spectral bands. Previous approaches to modulation filtering, such as those based upon a Hilbert transform magnitude, increased signal distortion. A new coherent approach was used to determine a fixed and an optimal adaptive modulation filter. The optimal adaptive filter used information from the target speech to design the modulation filter. Threshold signal-to-noise ratio was measured adaptively using a spondee-in-noise task [Turner et al., J. Acoust. Soc. Am. 115, 1729 (2004)]. Subjects selected the spondee heard in a forced-choice task with a two-talker babble background. The spondee level was fixed at the listeners most comfortable level and noise level adjusted adaptively using a 2 dB step size, with threshold based on 10 reversals. Sensorineural hearing-impaired listeners completed three conditions: unprocessed speech; fixed modulation filter; and optimal adaptive modulation filter. Normal-hearing subjects heard a parallel set of conditions processed to simulate a 6-channel cochlear implant. There was significant improvement in threshold signal-to-noise ratio with the optimal adaptive filter for both groups. There was no improvement for the fixed modulation filter relative to unprocessed speech.
The spiral model of pitch: Interrelations with musical and psychoacoustic scales and with cochlear parameters
Authors: Miller James D.
Published: October 2013

A spiral model of pitch can represent pitch chroma and pitch height. Its arclength provides the frequency-position map of the cochlea, and, when scaled by the number of auditory-ganglion cells, provides the Mel scale. Pitch-like scales fall into three groups. Group I is typified by the Cent scale, which can be expressed as a pitch spiral. Other Group-I scales are musical notation, logarithmic scales of frequency, Fechners Law for complex tones, and the pitch helix. Group II is typified by Greenwoods cochlear map, which is simply related to the arclength of the pitch spiral. Other Group-II scales are the sums of the number of (a) equivalent-rectangular bandwidths (NERBs); (b) difference limens for frequency (NDLFs); and (c) inner-hair cells (NIHCs) and are consistent with critical ratios. Group III is typified by the Mel Scale, which can be derived from the pitch spiral and the distribution of auditory-ganglion cells. Other Group-III scales are the sums of the number of (a) auditory ganglion cells (NSGCs), (b) frequency-modulation-difference limens (NFMDLs), and (c) Barks (NBARKs). The pitch spiral seems a basic concept from which other pitch-like scales can be derived.
Influence of beats on mistuning detection in consonant and dissonant musical intervals
Authors: Lewiston Craig E. , Oxenham Andrew J.
Published: October 2013

This study investigated the mechanisms underlying the detection of mistuning in musical intervals. Musical consonance is thought to be defined in part by an absence of cochlea-generated beats. Dissonant intervals, and mistuned consonant intervals, produce a percept of beats. Thus, mistuning of consonant intervals might be detected through the presence of beats, whereas mistuning of dissonant intervals might be detected by a change in the pattern of beats. Alternatively, mistuning might be detected by a direct comparison of the two pitches in each interval. In our experiment the presence of beats was controlled by presenting the two tones to both ears or to separate ears, by using either pure tones or complex tones, and by presenting the tones sequentially or concurrently. Results using highly trained musicians showed large threshold elevations (poorer performance) in the absence of cochlea-generated beats, suggesting that the mistuning of concurrent sounds in normal circumstances is mediated by beat detection. However, even in the absence of beats, mistuning detection thresholds for dissonant intervals, such as the tritone or minor second, were poorer than for consonant intervals, such as the octave or fifth. [Work supported by NIH grants T32DC00038 and R01DC05216.]
Further explorations of the contribution of a nonsimultaneous mistuned harmonic to residue pitch
Authors: Gockel Hedwig E. , Carlyon Robert P. , Plack Christopher J.
Published: October 2013

Ciocca and Darwin [V. Ciocca and C.J. Darwin, J. Acoust. Soc. Am. 105, 2421–2430 (1999)] reported a surprising finding: The shift in residue pitch caused by mistuning a harmonic was the same when the mistuned harmonic was presented after the remainder of the complex as when it was simultaneous. The present study tried to replicate this result, and investigated the role of the presence of the nominally mistuned harmonic in the matching sound. Subjects adjusted a matching sound so that its pitch equaled that of a subsequent 90-ms complex tone (12 harmonics of a 155-Hz F0), whose mistuned (±3%) 3rd harmonic was presented either simultaneously with or after the remainder. In experiment 1, the matching sound was a harmonic complex whose 3rd harmonic was either present or absent. In experiment 2, it was a sinusoid. In experiment 3, the target and matching sound contained non-overlapping harmonics. In all experiments, a non-simultaneous mistuned component produced significantly smaller pitch shifts than a simultaneous one. In the absence of the nominally mistuned harmonic in the matching sound, the pitch shift with simultaneous presentation was about five times larger than that with non-simultaneous presentation of the mistuned component. [Work supported by EPSRC Grant GR/R65794/01.]
Structural vibrations and sound radiation fields of classical guitars
Authors: Richardson Bernard E.
Published: October 2013

In its few lines, Federico Garcia Lorca’s poem ‘‘Las Seis Cuerdas’’ (the six strings) invokes a beautiful image of sound production on the guitar. It is the player’s interaction with the six strings which makes the music, but were it not for the body of the guitar, the instrument would have no voice. Most studies of the acoustical function of guitars have concentrated on measuring the structural vibrations or the sound radiation fields in isolation of the strings. These studies fail to encapsulate the important aspects of string-body coupling, which has a marked influence on the decay rates of string vibrations, and the radiativity of individual body modes. By measuring both input admittance at the bridge and sound-field topology for individual modes of the guitar, it is possible, through psychoacoustical experiments, to investigate the relative importance of the various acoustical parameters which define the response of the instrument. This paper will discuss experimental techniques and present data on ten classical guitars which go some way to identifying key components in the low- to mid-frequency mechanical and acoustical actions of the guitar. [Work originally supported by the Leverhulme Trust.]
Study of effect of speaker variability and driving conditions on the performance of an automatic speech recognition engine inside a vehicle
Authors: Kadambe Shubha
Published: October 2013

Spoken dialogue based information retrieval systems are being used inside vehicles. The user satisfaction of using such a system depends on how an ASR engine performs. However, the performance of an ASR is affected by speaker variability, driving conditions, etc. Here, we report the study that we performed to analyze these effects of speaker variability, different driving conditions and the effect of driving task on the ASR performance. This study consists of experimental design, data collection and systematically testing an ASR engine using this data. From the obtained results, it can be observed that (I) the ASR performance exhibits (a) significant speaker variability since the stress of driving task varies from speaker to speaker, (b) significant performance degradation across driving conditions since the noise type and level varies and (c) significant effect of driving task on recognition performance, and (II) the effect of live noise on recognition performance is not same as adding car noise to the pre-recorded speech data. The former observation is important since by just training an ASR engine on lots of speech data will not help and it is essential to include stress factors and cognition load in ASR engines to improve its performance.
Synthesizer software for modeling voice quality
Authors: Antonanzas-Barroso Norma , Gerratt Bruce R. , Kreiman Jody
Published: October 2013

This poster presents a formant synthesizer that is designed especially for detailed modeling of voice quality. Users may interactively manipulate source pulse shapes and/or spectral characteristics of the voice source, the noise spectrum and noise-to-signal ratio, jitter and shimmer, vocal tremor rate and extent, and the vocal tract transfer function (formants and bandwidths), all in near-real time. Pitch contours can be modeled directly or imported. The synthesizer may be utilized in method-of-adjustment tasks, or sequences of stimuli may be created for use in other experimental paradigms. An interactive audio-visual display makes this a useful tool for teaching voice acoustics as well. Both an executable version of the program and the underlying code will be available during the conference. Copies of software that supports the voice modeling effort, including an interactive inverse filter, will also be distributed. [Work supported by NIDCD grant DC01797.]
Capacity of the oceanic waveguide
Authors: Higley W. J. , Roux Philippe , Kuperman W. A.
Published: October 2013

Capacity between two multiple-input/multiple-output (MIMO) arrays can be computed by decomposing the Green’s function matrix into orthogonal functions, which can be thought of as independent subchannels. In a waveguide with sufficient array sampling, the modes are the independent subchannels, and capacity can easily be calculated for given waveguide parameters. Capacity is calculated using the optimal power allocation strategy of ‘‘water-filling.’’ Water-filling is simultaneously done across modes and frequency. Theoretically computed capacities for an oceanic waveguide between two vertical arrays for various waveguide parameters, along with physical explanation, will be presented.
Backpropagation image analysis of broadband decomposition of the time reversal operator (DORT) data from TREX-04
Authors: Fromm David M. , Gaumond Charles F. , Menis Richard , Calvo David C. , Lingevitch Joseph F. , Edelmann Geoff F. , Kim Elisabeth
Published: October 2013

Single-frequency DORT is a method of isolating scatterers with a multiple-source/multiple-receiver system [C. Prada et al., J. Acoust. Soc. Am. 99, 2067–2076 (1996)]. Broadband DORT overcomes problems of frequency fading and dispersion by exploiting the response of singular values across the entire signal bandwidth. In spring 2004, the Time Reversal Experiment, TREX-04, collected data for the evaluation of the broadband DORT technique. An 80 m vertical source/receiver array with 64 hydrophones was deployed in 100 m deep water in an area south of Hudson Canyon off the New Jersey coast. Sets of four, five, or six beams were transmitted at small angles from horizontal and the response from an echo repeater was recorded on 64 channels. Data was collected in 500 Hz bands between 500 and 3500 Hz with the echo repeater at ranges from 0.5 to 5.0 km and at mid-water column and near-bottom depths. Applying the broadband DORT algorithm, the isolation of the echo repeater signal in the singular vectors and the robustness of broadband backpropagation images will be presented as a function of frequency band, signal coherence, echo repeater range and depth, noise levels, and receiver aperture. [Work supported by the ONR.]
Performance analysis of ambiguity function based broadband chirp direction of arrival estimation
Authors: Ma Ning , Thiam Goh Joo
Published: October 2013

When estimating the direction of arrival (DOA) estimation of broadband sources with an array, the source is usually modeled as white noise. However, when the source signal has certain time-frequency structure, studies have shown that the DOA estimation performance can be improved by using the information about its structure. In our previous work [Ning Ma and Joo Thiam Goh, ‘‘DOA Estimation for Broadband Chirp Signals,’’ Proc. of ICASSP 2004, Vol. II, pp. 261–264], we have proposed two new methods for broadband chirp signal DOA estimation based on the ambiguity functions, namely Incoherent Broadband Chirp DOA estimation (CBD-I) and Coherent Broadband Chirp DOA estimation (BCD-C). Besides improving the accuracy of the DOA estimates, these algorithms work with signal frequencies that are higher than the array design frequency. The number of sources are also not constrained by the number of sensors. The performance of the proposed DOA estimation methods is analyzed theoretically and numerically in this work. The analysis results show that these methods can improve the output SNR and therefore the DOA estimation performance when input SNR is higher than certain level.
Mean and variance of the 3-D field forward propagated through random internal waves in continuously stratified deep and shallow water ocean waveguides
Authors: Chen Tianrun , Ratilal Purnima , Makris Nicholas C.
Published: October 2013

A general modal solution for the mean and covariance of an acoustic field propagating through 3-D random inhomogeneities in an ocean waveguide [Ratilal and Makris, J. Acoust. Soc. Am. 114, 2428 (2003)] is applied to propagation through a general 3-D internal wave field. Our previous two-layer internal wave parameterization is extended to continuous stratification and applied in both continental shelf and deep ocean waveguides. Physical and statistical properties of the internal waves are modeled with the Garret-Munk and other theoretical and empirical formulations. The mean and variance of the forward field propagating through internal waves is expressed in terms of the stochastic moments of the medium’s scatter function density. This is formulated with the Rayleigh-Born approximation to Green’s theorem in terms of the medium’s compressibility and density fluctuations. In this way, the effect of both internal-wave sound speed and density variations on the forward propagated field are quantified. Density variations are shown to greatly affect high order modal propagation in some cases. Two-dimensional models of propagation through random internal wave fields are shown to become inaccurate after exceeding threshold ranges where 3-D scattering becomes important.
The soundscape as a tool for urban planners and town designers
Authors: Semidor Catherine
Published: October 2013

Urban planners and other town designers need information to improve the environmental quality of cities, for instance the components of the urban soundscape. To use the latter as a decision-making tool in a planning process it is necessary first to link objective and subjective acoustic criteria with architectural parameters and then to transmit the observed characteristic relations in a way which designers will understand. One of these ways is to make use of spatial references (streets’ shape, facades’ material, etc.) to explain the impact of the urban morphology, for example, on the sound environment. That is why, for many years, surveys were conducted in a lot of different areas. This paper deals with some comparative results obtained in different European towns.
A new acoustic bat detector
Authors: Heavner Matthew
Published: October 2013

A new small, low-power, long-duration, field-deployable computer for the acoustic monitoring of bats in Southeast Alaska is being developed. The most recent study of bats in the region, reported by Parker et al., (1997), found the geographic range of four bat species to have northern distribution limits in Southeast Alaska. (A fifth species, E. fuscus, goes much farther north.) The relationship of the bats to forest management practices, the habitat usage of the bats, and population size and trends are all very poorly known for bats in Southeast Alaska. Long duration monitoring of several different types of area (such as old-growth versus recently logged forest) will provide knowledge to improve management practices in regards to bat ecology in Southeast Alaska. With the motivations just described, the hardware, development methods, and analysis software designed to develop an improved detector are presented. [Work supported by Alaska Department of Fish & Game.]
Testing acoustic fish deterrents for use under-ice in arctic lakes
Authors: Racca Roberto G.
Published: October 2013

Acoustic deterrent technologies can be used in aquatic settings in lieu of physical barriers to keep fish away from potentially harmful industrial operations. The goal of this study was to determine the efficacy of portable, temporary acoustic deterrents as a means of excluding fish from the neighborhood of sub-bottom detonation activities associated with seismic exploration under ice in arctic lakes. In October 2003 trials were conducted in Dolomite Lake near Inuvik, Northwest Territories, Canada on indigenous fish species. Groups of fish were equipped with orally inserted ultrasonic tags, placed in a large experimental net pen and monitored using an acoustic tracking system that produced a detailed three-dimensional swimming pattern for each subject, thereby revealing any behavioral responses. A flex-tensional broadband sound projector driven by digitally synthesized signals was tested as deterrent, and real-time monitoring at two hydrophone sites was used to estimate the local level of insonification throughout the pen volume. Although the study did not identify an overall effective deterrent, sufficient indications of response were observed to support the future testing of a louder projector capable of emitting a tonally modulated sound pattern at frequencies from about 100 Hz to a few kHz. [Work supported by ESRF (esrfunds.org).]
Nonlinear model for detection of bubble activity from pulse-echo ultrasound
Authors: Ebbini Emad S.
Published: October 2013

Cavitation and transient cavitation plays an important role in modern application of biomedical ultrasound, both in therapeutics and diagnostics. Special imaging and signal processing methods have been recently introduced for the detection and localization of cavitation and other nonlinear phenomena for improving image quality and/or characterization of therapeutic outcome of the ultrasound-based treatments. We have recently developed a model-based approach for separating the linear and nonlinear components of the pulse-echo data. In its simplist form, the approach separates the linear and quadratic signal components from the echo data. The latter is related to the sum and difference frequency interactions in the nonlinear medium. When used in conjunction with elongated coded transmit waveforms (e.g., chirp signals), the approach provides significant separation of bubble activity from tissue nonlinear components. Application to ultrasound contrast-agent imaging and detection of HIFU-induced lesions will be discussed.
Acoustic Casimir pressure for arbitrary media
Authors: Esquivel-Sirvent Raul , Reyes Luis , Barcenas Jeffrey
Published: October 2013

We derive a general expression for the acoustic Casimir pressure between two parallel slabs made of arbitrary materials and whose acoustic reflection coefficients are not equal. The formalism is based on the calculation of the local density of modes using a Green’s function approach. The results for the Casimir acoustic pressure are generalized to a sphere/plate configuration using the proximity theorem.
Hearing space: Identifying rooms by reflected sound
Authors: Rosenblum Lawrence D. , Robart Ryan L.
Published: October 2013

The acoustic engineering literature has shown that listeners are sensitive to the sound reflecting properties of spaces so as to judge the general dimensions and acoustic quality of the space [e.g., J. S. Bradley, R. D. Reich, and S. G. Norcross, J. Acoust. Soc. Am. 108, 651–661 (2000)]. There is also speculation on the perceptual processes listeners use to apprehend properties of spaces [e.g., R. K. Clifton, R. L. Freyman, and J. Meo, Percept. Psychophys. 64, 180–188 (2003)]. Still, there has been little research testing whether listeners can simply recognize different types of rooms based on reflected structure. To examine this issue, five different sound sources were binaurally recorded in four different acoustic spaces. The sources ranged from human speech to white noise bursts. The acoustic spaces included a large indoor gymnasium, a moderately-sized classroom, a public restroom, and a small laboratory room. Untrained listeners heard each sound over headphones as they looked at photographs of the four rooms. They were asked to choose the room in which each sound was recorded. Overall, listeners were quite good at the task, with accuracy dependent on sound source. The results indicate a sensitivity to complex ambient structure not often addressed in the psychoacoustics literature.