bluuebunny/crossref_metadata_2025 huggingface dataset:
Target reconstruction from broadband acoustic backscatter
Authors: Huynh Quyen Q. , Wilbur Jo E. , Intrator Nathan , Neretti Nicola
Published: October 2013

Broadband acoustic backscatter data are examined for mine reconstruction using LFM and dolphin clicks. Broadband backscatter data from free-field and bottom mines were collected during tests conducted in ARL/UT’s sand tank and sponsored by ONR. The data set includes multi-aspect broadband LFM (50–200 kHz) acoustic backscatter returns from a collection of targets. Both free-field and bottom target data were collected. Bottom target data sets include returns from two different levels of bottom roughness. Each data file is sampled over an interval that is large relative to the echo duration with sufficient sampling both prior to the echo onset and post-echo return to enable meaningful target characterization. Data collected over 360 deg in 2-deg increments are processed to generate a descriptive target pattern. The data are pulse compressed and system artifacts are removed using adjacent channels over the entire frequency band and within subbands. The target pattern is generated by applying the new adaptive wavelet filters to the scattered signals. In particular, the multi-aspect data that have several targets are tested to determine the ability of this new adaptive wavelet transform to reconstruct the external and internal structures of mines. Comparisons are made to previous results generated from broadband dolphin clicks.
More on the nonhomogeneous Sturm–Louiville problem and its application to inclusions in a waveguide and the extraction of target properties
Authors: Werby M. F.
Published: October 2013

The second order solution of the wave equation is useful in adopting a method for solving the related inhomogeneous version in which the inhomogeneous term comes from a known target or an unknown object in the waveguide. Formulations are presented for these cases with examples.
Determination of the absorbing and scattering properties of shallow-water sediment using wideband signals
Authors: Katsnelson Boris G. , Grigoryev Valery A. , Petnikov Valery G.
Published: October 2013

Sound propagation in a shallow-water region is studied for bottom parameter estimation within the framework of MFP (matched field processing) method. The experiment was conducted in the Barents Sea at the stationary acoustic track (about 14 km length). Basic bottom parameters are determined on the basis of comparison between experimental and theoretical frequency spectra of a signal received from a wideband source (20–100 Hz). A shallow-water waveguide model is developed on the basis of measured bathimetry, sound-speed profile, and two-component model of the bottom sediment. Porosity and coefficients of absorption and volume scattering are considered as the parameters, characterizing bottom sediment. The connection between porosity and mentioned coefficients is taken from Hamilton’s data [J. Acoust. Soc. Am. 59, 528–535 (1976)]. Porosity (correspondingly, density and sound speed in sediment) is obtained, and a conclusion is made about relative contribution of different mechanisms of losses (absorption and volume scattering) in the seafloor at different sound frequencies. [Work supported by CRDF and ME RF, Award No. VZ-010-0.]
Acoustic radiosity for computation of sound fields in diffuse environments
Authors: Muehleisen Ralph T. , Beamer C. Walter
Published: October 2013

The use of image and ray tracing methods (and variations thereof) for the computation of sound fields in rooms is relatively well developed. In their regime of validity, both methods work well for prediction in rooms with small amounts of diffraction and mostly specular reflection at the walls. While extensions to the method to include diffuse reflections and diffraction have been made, they are limited at best. In the fields of illumination and computer graphics the ray tracing and image methods are joined by another method called luminous radiative transfer or radiosity. In radiosity, an energy balance between surfaces is computed assuming diffuse reflection at the reflective surfaces. Because the interaction between surfaces is constant, much of the computation required for sound field prediction with multiple or moving source and receiver positions can be reduced. In acoustics the radiosity method has had little attention because of the problems of diffraction and specular reflection. The utility of radiosity in acoustics and an approach to a useful development of the method for acoustics will be presented. The method looks especially useful for sound level prediction in industrial and office environments. [Work supported by NSF.]
Infrasounds and biorhythms of the human brain
Authors: Panuszka Ryszard , Damijan Zbigniew , Kasprzak Cezary , McGlothlin James
Published: October 2013

Low Frequency Noise (LFN) and infrasound has begun a new public health hazard. Evaluations of annoyance of (LFN) on human occupational health were based on standards where reactions of human auditory system and vibrations of parts of human body were small. Significant sensitivity has been observed on the central nervous system from infrasonic waves especially below 10 Hz. Observed follow-up effects in the brain gives incentive to study the relationship between parameters of waves and reactions obtained of biorhythms (EEG) and heart action (EKG). New results show the impact of LFN on the electrical potentials of the brain are dependent on the pressure waves on the human body. Electrical activity of circulatory system was also affected. Signals recorded in industrial workplaces were duplicated by loudspeakers and used to record data from a typical LFN spectra with 5 and 7 Hz in a laboratory chamber. External noise, electromagnetic fields, temperature, dust, and other elements were controlled. Results show not only a follow-up effect in the brain but also a result similar to arrhythmia in the heart. Relaxations effects were observed of people impacted by waves generated from natural sources such as streams and waterfalls.
Dragline noise survey
Authors: Vipperman Jeffrey S. , Bauer Eric R.
Published: October 2013

It is estimated that 70%–90% of miners have enough noise induced hearing loss (NIHL) to be classified as a disability (NIOSH, Publication No. 76-172, 1976; Franks, NIOSH Internal Report, 1996). In response, NIOSH is conducting a cross-sectional survey of the mining industry in order to determine the sources of mining noise and offer recommendations on how to mitigate high noise levels, and bring mining operations into compliance with the recent mining noise regulation: 30CFR, Part 62. This paper will outline the results from noise surveys of eight draglines which operate in above-ground coal mining operations. The data recorded include noise dosimetry in conjunction with time-at-task studies and 1/3-octave sound level (Leq, Lmin, and Lmax) measurements. The 1/3-octave band readings were used to create noise contour maps which allowed the spatial and frequency information of the noise to be considered. Comparison of Lmin and Lmax levels offer insight into the variability of the noise levels inside the dragline. The potential for administrative controls is limited due to consistently high noise levels throughout the deck. Implementation of engineering controls is also hindered by the size and number of the noise sources and the frequency content of the noise.
Cochlear filter nonlinearity between 500 and 8000 Hz in listeners with normal hearing
Authors: Lopez-Poveda Enrique A. , Plack Christopher J.
Published: October 2013

Human nonlinear cochlear responses were estimated in three normal-hearing listeners at characteristic frequencies (CF) of 0.5, 1.0, 2.0, 4.0, and 8.0 kHz, as a function of frequency and level. Cochlear responses were derived from fixed-probe-level isoresponse temporal masking curves (TMCs) at signal frequencies (Fs) equal to CF. Signal level was fixed at 14 dB SL. The level of the forward maskers was adjusted so as to find the masker level at threshold as a function of the time interval between the masker and the signal. Masker frequencies (Fm) of 0.5, 0.7, 0.9, 1.0, 1.1, and 1.6 Fs were used. Across CFs, results for Fs≪Fm show a single slope consistent with linear cochlear response. For Fs=Fm, TMCs show a two- or three-stage curve consistent with cochlear nonlinearity. A trend was observed in the compression slopes of derived cochlear IO functions as a function of CF, going from 0.8 dB/dB at 0.5 kHz, to 0.27 dB/dB at 8.0 kHz. Although shifts in the best frequency of the 0.5-, 1.0-, and 4.0-kHz cochlear filters were observed at moderate input levels, they were absent at higher levels. The data may be of use to tune models of the human basilar-membrane response.
Informational masking: Toward improved understanding
Authors: Durlach Nathaniel I. , Kidd Gerald , Mason Christine R. , Arbogast Tanya L. , Colburn H. Steven , Shinn-Cunningham Barbara
Published: October 2013

This paper is concerned with simultaneous informational masking of tones by tone complexes. Past work distinguishes between energetic (peripheral) masking, which occurs when target and masker overlap in the peripheral frequency channels, and informational (central) masking, which can occur even when there is no such overlap but there is uncertainty in the frequency spectrum of the stimulus. We suggest that this distinction is (1) applicable to domains other than the frequency domain (e.g., the spatial domain) and (2) relative to the level of processing in the auditory system considered. We further suggest that this distinction can be rigorously defined for any domain and any level by comparing performance of the ideal detector operating on the inputs at that level to human psychophysical performance. We also argue that informational masking is strongly influenced by target-masker similarity and segregation difficulty, not only by stimulus uncertainty. Experiments designed to probe this issue are discussed in a companion paper (Mason et al.). Finally, ideas relevant to the large intersubject differences observed in informational masking (focused on the distinction between trying to hear the target and trying to not hear the masker) are used to make new predictions about these differences. [Work supported by NIH/NIDCD.]
Intensity-difference limens for very brief Gaussian-shaped tones at 500 Hz and 6.5 kHz
Authors: Nizami Lance , Jesteadt Walt
Published: October 2013

The curve of intensity-difference limen (DL) versus level shows a midlevel hump for 6.5-kHz tones having 5-ms cosine-squared ramps and a 26-ms plateau [Carlyon and Moore, J. Acoust. Soc. Am. 76, 1369–1376 (1984)]. DLs were presently obtained for that tone, as well as for 6.5-kHz tones having Gaussian envelopes, for five subjects. Pedestal levels covered 30- to 90-dB SPL and equivalent rectangular durations D were 0.314, 0.627, 1.25, 2.51, 10.03, and 20.1 ms, where D is the Gaussian-envelope duration divided by 3.19. The DLs for the tone of Carlyon and Moore (D∼30) show a midlevel hump smaller than found earlier. The hump grows noticeably with smaller D (i.e., the Gaussian-shaped tones), reaching 7 dB for D=0.627. The peak drops, however, for D=0.314, causing the curve of DL versus duration to peak for D=0.627 for tone levels of 40–60 dB SPL. DLs were also obtained from seven subjects for 500-Hz tones of 30–90 dB SPL for D’s of 1.25, 2.51, 5.02, and 10.03 ms. A midlevel hump at D’s of 1.25 and 2.51 gives way to a near-miss at D’s of 5.02 and 10.03. A mid-duration peak is seen at D=5.02 for levels of 30–60 dB SPL.
Temporal interference in amplitude modulation perception
Authors: Koopman Jan , Houtgast Tammo , Dreschler Wouter
Published: October 2013

We observed that, especially for hearing-impaired subjects, the perceived modulation strength of a signal is systematically reduced by the presence of a preceding modulated signal. This suggested that modulation detection interference (MDI) might not only apply in case of simultaneous signals, but also for nonsimultaneous signals. To further investigate this we performed an experiment with normal-hearing and hearing-impaired subjects, in which modulation detection was measured for a target signal (1 kHz carrier, 625 ms duration, 8 Hz amplitude modulation), preceded by an interfering signal (2 kHz carrier, 625 ms, 100% 8 Hz modulation). The onset difference between interfering signal and target signal was varied over a range of 0–1250 ms, thus from simultaneous (the traditional MDI condition) to clearly nonsimultaneous. Based on pilot experiments we hypothesized that normal-hearing subjects showed a rapid increase of sensitivity from simultaneous to nonsimultaneous presentation (essentially no MDI after effect), whereas for the hearing impaired the interference effect was observed up to long onset differences, including clearly nonsimultaneous presentation. The observed differences in modulation interference between normal-hearing and hearing-impaired subjects might explain the differences shown in speech intelligibility in fluctuating background noise. Results will be used to validate this hypothesis.
Learning interference associated with training temporal-interval discrimination with two stimuli
Authors: Wright Beverly A. , Ortiz Jeanette A.
Published: October 2013

Determining the training regimes that do and do not yield improvements in auditory discrimination provides information about the mechanisms underlying the trained task, and about learning itself. In a previous report, training with one standard stimulus yielded improvements in temporal-interval discrimination that generalized to an untrained frequency at the trained interval [B. A. Wright et al., J. Neurosci. 17, 3956–3963 (1997)]. Here, the impact of training interval discrimination with two standard stimuli, as opposed to one, was investigated. Listeners were trained either with two standard intervals (100 and 350 ms) at one frequency (1 kHz; n=8), or with one standard interval (100 ms) at two frequencies (0.25 and 1 kHz; n=8). On average, listeners in both groups (1) learned less on each trained condition than listeners trained with only one stimulus (n=10), and (2) subsequently performed on the trained interval(s) at an untrained frequency (4 kHz) as though they had trained with only one of the stimuli—in one case, showing no improvement. These results indicate that, at some stage, temporal intervals are processed at frequency-specific sites, and that training one task with two stimuli sharing all but one feature can interfere with both learning and generalization. [Work supported by NIDCD.]
The auditory cross-section (AXS) test battery: A new way to study afferent/efferent relations linking body periphery (ear, voice, heart) with brainstem and cortex
Authors: Lauter Judith
Published: October 2013

Several noninvasive methods are available for studying the neural bases of human sensory-motor function, but their cost is prohibitive for many researchers and clinicians. The auditory cross section (AXS) test battery utilizes relatively inexpensive methods, yet yields data that are at least equivalent, if not superior in some applications, to those generated by more expensive technologies. The acronym emphasizes access to axes—the battery makes it possible to assess dynamic physiological relations along all three body-brain axes: rostro-caudal (afferent/efferent), dorso-ventral, and right-left, on an individually-specific basis, extending from cortex to the periphery. For auditory studies, a three-level physiological ear-to-cortex profile is generated, utilizing (1) quantitative electroencephalography (qEEG); (2) the repeated evoked potentials version of the auditory brainstem response (REPs/ABR); and (3) otoacoustic emissions (OAEs). Battery procedures will be explained, and sample data presented illustrating correlated multilevel changes in ear, voice, heart, brainstem, and cortex in response to circadian rhythms, and challenges with substances such as antihistamines and Ritalin. Potential applications for the battery include studies of central auditory processing, reading problems, hyperactivity, neural bases of voice and speech motor control, neurocardiology, individually-specific responses to medications, and the physiological bases of tinnitus, hyperacusis, and related treatments.
When sound effects are better than the real thing
Authors: Heller Laurie M. , Wolf Lauren
Published: October 2013

Sound effects technicians (‘‘Foley Artists’’) have long exploited the fact that two physically different events can produce perceptually similar sounds, such as squeezing a box of cornstarch to imitate footsteps in the snow. Although some sound effects succeed because they produce acoustic waveforms nearly identical to the sounds they are imitating (their targets), in other cases there are obvious acoustic differences between sound effects and their targets. Those differences may provide information about which acoustic features are essential, and which are extraneous, for auditory recognition of an event. To address this question, twelve pairs of sound effects and their associated target events were recorded. Listeners attempted to identify each sound as it was presented over headphones. Next, the listeners were informed of the target sound (e.g., footsteps in the snow) and were asked to rate the sound’s realism. The sound effects were sometimes judged as more realistic than recordings of the target events. Acoustic differences between the waveforms of the most realistic sound effects and their targets were interpreted in terms of psychoacoustic principles.
Aging of the ear, exposed and not exposed to noise
Authors: Fleischer Gerald , Müller Reinhard , Heppelmann Guido , Bache Thomas , Gao Xia , Wang Feng , Huang Weiguo
Published: October 2013

Hearing in two special groups was compared, using pure-tone audiometry up to 16 kHz. One group consisted of commercial airline pilots in Germany, the other of people living in remote parts of the People’s Republic of China, without electricity, cars or, other technical equipment. The age of the 502 pilots (99% male) ranged from 22 to 65 years, while that of the 644 Chinese (61% male) ranged from 6 to 82 years. In both groups participants with diseases of the ear and with acute acoustic trauma were excluded. Analysis showed that the threshold of hearing in young adults is identical in Germany and in China. It is evident that the effect of aging is reduced in pilots, as compared to people living without technical or industrial noise. Older pilots clearly hear better than people not subjected to technical noise, most prominently at higher frequencies. Together with other observations, it can be concluded that continuous wideband noise reduces the aging of the ear. Long-term exposure to such noise—within limits—appears to be good for the ear.
A case study: Transforming a vaudeville theatre into a 21st century multi-purpose hall
Authors: Talaske Richard H. , Hoffman Ian B. , Laney Jonathan P.
Published: October 2013

Originally constructed in 1927 as an atmospheric, vaudeville theatre, the Coronado enjoyed 50 years as host to a true variety of events. In the 1970s, functional and space limitations became a catalyst for touring events to avoid the Coronado, and the theatre fell into disrepair. In the mid-1990s, a renovation was planned (completed January 2001) to restore the glory of the original theatre and upgrade it to be a viable multi-purpose hall for the 21st century. By annexing two neighboring buildings, many of the space limitations could be overcome in both the front- and back-of-house. The scope included a significantly enlarged stage and stagehouse, performer support spaces, expanded lobby spaces, quiet ventilation systems, and a new music rehearsal room. Acoustically, a primary goal was to upgrade the hall for both reinforced music/theatre performances and unreinforced, orchestral performances. However, historical regulations prevented modification to the room shape and configuration. In order to make the room suitable for unreinforced music, two primary acoustic features were instituted—a semi-custom concert enclosure for the stage and an acoustic enhancement system for the audience chamber. For theatrical and popular music events, a new performance audio system was designed and integrated into the hall.
Age-related hearing loss in sea lions and their scientists
Authors: Schusterman Ronald J. , Southall Brandon , Kastak David , Reichmuth Kastak Colleen
Published: October 2013

Interest in the hearing capabilities of California sea lions (Zalophus californianus) was first stimulated by the echolocation hypothesis and more recently by rising concern about coastal noise pollution. During a series of audiometric tests, we measured the absolute hearing sensitivity of two sea lions and two of their human investigators. Aerial hearing curves for each subject were obtained with a go/no-go procedure and standard psychophysics. Additionally, underwater hearing curves were obtained for the sea lions using the same procedures. Underwater, the older sea lion (22–25 years of age) showed hearing losses relative to the younger sea lion (13–16 years) that ranged from 10 dB at lower frequencies to 50 dB near the upper frequency limit. The older sea lions’ hearing losses in air were consistent with those measured underwater. The older human (69 years) tested also showed losses relative to the younger human (22 years). These differences ranged from 15 dB at lower frequencies up to 35 dB at the highest frequency tested. The results obtained in this study document age-related hearing losses in sea lions and humans. The findings are consistent with data on presbycusis in other mammalian species, showing that maximum hearing loss occurs at the highest frequencies.
Interrelationships between intranarial pressure and biosonar clicks in the bottlenose dolphin (Tursiops truncatus)
Authors: Elsberry Wesley R. , Cranford Ted W. , Ridgway Sam H. , Carder Donald A. , VanBonn William G. , Blackwood Diane J. , Carr Jennifer A. , Evans William E.
Published: October 2013

Three Atlantic bottlenose dolphins (Tursiops truncatus) were given a target recognition biosonar task. During their performance of the task, both acoustic data in the far field and pressure within the bony nasal passages were digitally recorded (Elsberry et al., 1999). Analysis of over 15000 biosonar clicks provided new insights into odontocete biosonar sound production and is consistent with acoustic and pressure data taken from white whales (Delphinapterus leucas) during biosonar (Ridgway and Carder, 1988). Our work provides the first evidence for a minimum intranarial pressure during biosonar click production for any odontocete (11.8±0.5 kPascals over basal pressure). All three subjects exhibited nearly the same minimum tranarial pressure difference during biosonar click production. Clicks produced at or near this minimum intranarial pressure exhibited a wide range of acoustic power values. The acoustic power of a biosonar click was not highly correlated with intranarial pressure (R2=0.116). The radiated acoustic energy in biosonar clicks ranged from 1 to 1370 microJoules. Estimates of mechanical work during pressurization events were produced using a piston/cylinder model and intranarial volume data from prepared specimens and computed tomography scans. Mechanical work during pressurization events ranged from 2.74 to 23.0 Joules, with an average of 10.3 Joules.
Past and present research on gray whale vocalizations
Authors: Ollervides Francisco
Published: October 2013

Eberhadt and Evans first recorded vocalizations of gray whales in 1967 in Laguna Ojo de Liebre, Mexico. Three and a half decades of bioacoustic research on gray whales has followed this groundbreaking work. Gray whales appear more vocal while at the southern breeding lagoons off Baja California and are least vocal at the northern feeding areas of the Bering, Chukchi, and Beaufort Seas. Ten different vocalizations have been identified on the breeding grounds, while only six different vocalizations have been recorded on the feeding grounds. Evans also conducted the first recordings of vocalizations of gray whales in captivity with his work on J.J. II in San Diego in 1974. Since then, the database of captive gray whale vocalizations has expanded with research on GiGi in 1997. From this review, an acoustic repertoire of at least 12 different types of vocalizations is suggested. Nevertheless, not all gray whale calls fall into obvious or distinct categories; thus, correlating specific behaviors with each vocalization type merits further study.
Echolocation signals of foraging killer whales (Orcinus orca)
Authors: Au Whitlow W. L. , Ford John K. B. , Allman Kelly A.
Published: October 2013

Fish eating resident killer whales that frequent the coastal waters of Vancouver Island, Canada have a strong preference for chinook salmon. The whales in Johnston Strait often forage along the steep cliffs that extend into the water, echolocating their prey. Echolocation signals were measured with a four hydrophone symmetrical star array and the signals were simultaneous digitized at a sample rate of 500 kHz using a lunch-box PC. A portable VCR recorded the images from an underwater camera located close to the array center. Only signals emanated from close to the beam axis (1185 total) were chosen for a detailed analysis. Killer whales project very broad band echolocation signals (Q 1.3 to 1.5) that tend to have a bimodal frequency structure. Ninety seven percent of the signals had center frequencies between 45 and 80 kHz with a band-width between 35 and 50 kHz. The peak-to-peak source level of the echolocation signal decreased as a function of the one way transmission loss to the array. Source levels varied between 200 and 225 dB re 1 μPa. Using a model of target strength for chinook salmons, the echo levels from the echolocation signals are estimated for different ranges between whale and salmon.
Evoked potential application to study of echolocation in cetaceans
Authors: Supin Alexander Ya. , Nactigall Paul E. , Pawloski Jeffrey , Au Whitlow W. L.
Published: October 2013

The evoked-potential (EP) method is effective in studies of hearing capabilities of cetaceans. However, until now EP studies in cetaceans were performed only in conditions of passive hearing by recording EP to external stimuli. Can this method be applied to study active echolocation in odontocetes? To answer this question, auditory brainstem evoked responses (ABR) were recorded in a false killer whale while the animal echolocated a target within an experiment in which the animal reported the target present or absent. The ABR collection was triggered by echolocation clicks. In these conditions, the recorded ABR pattern contained a duplicate set of waves. A comparison of ABR wave delays recorded during echolocation with those recorded during regular external stimulation has shown that the first set of waves is a response to the emitted click whereas the second one is a response to the echo. Both responses, to the emitted click and to the echo, were of comparable amplitude in spite of the intensity difference of these two sounds of more than 40 dB near the animal’s head. This finding indicates some mechanisms releasing responses to echoes from masking by loud emitted clicks. The evoked-potential method may be productive to investigate these mechanisms.
Acoustic measurements of the number of naturally produced bubbles resonating at 50 kHz
Authors: Nuetzel Bernd , Herwig Karl-Heinz
Published: October 2013

When the wind is strong enough to form whitecaps, bubbles caused by breaking waves penetrate into the water column and are transported to depths of more than 10 meters by Langmuir circulation, turbulence, and other mechanisms. In the near-surface layer, bubbles cause acoustic scattering and attenuation. Especially in shallow water, where the sound is reflected and scattered at the boundaries (sea bottom and surface), bubbles significantly influence sound propagation in the sea. The results from acoustic near-surface experiments conducted in the North Sea over several years are presented. Data sets of acoustic attenuation were measured inside the bubble clouds for a propagation range of 1 m. Also, a narrow-beam transducer was utilized to measure the acoustic scattering from the near-surface bubbles. Both investigations were done simultaneously for a large variety of environmental conditions. The data sets have been combined to determine the number of resonating bubbles as a function of depth and wind speed for a selected frequency of 50 kHz. A numerical expression is introduced for this special set of parameters. The comparison with results from other investigators using different techniques shows good agreement.
Measuring bubble distribution with multibeam sonar
Authors: Culver R. Lee , Bradley David
Published: October 2013

Bubbles in water are very efficient scatterers of sound. Depending upon the ratio of acoustic wavelength to bubble radius, the scattering mechanism can be Rayleigh, resonant, or geometric. The strength of the scattered signal depends strongly on this ratio. The strength of the scattered signal can be used to estimate the number of bubbles per unit volume, or bubble density, if the size of the bubbles or the size distribution of the bubbles is known. A number of researchers have employed narrow beam sonar to measure the density of bubbles in water. We have used a 250 kHz sonar with 19 narrow beams, integrated into an autonomous vehicle (AUV), to image the wakes of two large surface ships from the underside. An earlier talk focused on the structure of the turbulent wake revealed by the measurements [J. Acoust. Soc. Am. 110, 2642 (2001)]. In this talk we discuss parameters and assumptions which strongly affect the bubble density estimate. These include: values for the transmit and receive beamwidth; the shape of the bubble size distribution spectrum, including possible variation with depth; strong boundary reflections and high sidelobe levels; and the possible influence of multiple scattering. [Work sponsored by ONR Code 321OA.]
Measurements of the time-dependent attenuation in a nonstationary bubble distribution
Authors: Wilson Preston S. , Roy Ronald A. , Carey William M.
Published: October 2013

In the ocean, natural and artificial processes generate clouds of bubbles which scatter and attenuate sound, and the structure of these clouds (i.e., the space-dependent bubble size distribution and void fraction) evolves over time. Measurements have shown that at the individual bubble resonance frequency (IBRF), sound propagation in this medium is highly attenuated and dispersive. Theory to explain this behavior exists in the literature and is experimentally verified away from resonance. However, due to excessive attenuation near resonance and the high degree of sensitivity to bubble population parameters, little adequate experimental data exists for comparison and theory remains largely unverified. Using an impedance tube, the time-dependent attenuation has been measured in a nonstationary laboratory bubbly fluid at IBRF, along with measurements of the accompanying time-dependent bubble population parameters. The relative variability of the measured maximum attenuation and the frequency at which it occurred agreed closely with existing theory. [Work supported by ONR.]
Audio recording and reproduction in CARROUSO: Getting closer to perfection?
Authors: Teutsch Heinz , Spors Sascha , Buchner Herbert , Rabenstein Rudolf , Kellermann Walter
Published: October 2013

State-of-the-art systems for spatial audio reproduction utilize two to six discrete playback channels. A problem inherent to these systems is the relatively small area where the listener is able to experience a true 3-D sound sensation. This so-called ‘‘sweet spot’’ can be significantly enlarged by using loudspeaker arrays in combination with wave field synthesis (WFS) technology, initially developed at Delft University. By following this approach, actual sonic spaces can be reproduced in their entirety and not only discrete multichannel representations thereof. While loudspeaker arrays can be used to reproduce sound fields, microphone arrays can be used for sound field capture and analysis. Having high-quality audio reproduction in mind, microphone array designs are presented that need to fulfill stricter requirements than what has been traditionally considered for microphone array applications. Information on acoustic source position is essential for WFS-based rendering techniques. As will be shown, joint audio–video object tracking proves to be efficient for this task. Moreover, full-duplex applications based on WFS technology, like high-quality teleconferencing or remote music teaching, call for sophisticated multichannel acoustic echo cancellation algorithms. The European project ‘‘CARROUSO’’ aims at developing, integrating, and building a real-time system that embraces all previously described technologies in an MPEG-4 context.
The structural implications of interactive creativity
Authors: Chadabe Joel
Published: October 2013

The functioning of any particular electronic musical instrument can be placed somewhere along a line that extends from deterministic to indeterministic. Although deterministic instruments may offer more powerful controls than traditional instruments, they typically put a performer in the traditional situation of making a gesture and expecting a predictable effect. Indeterministic instruments, on the other hand, put a performer in an interactive role of improvising relative to an unpredictable output. The unique advantage of such interactive instruments is that they foster ‘‘interactive creativity.’’ The design of a traditional instrument is fundamentally different from the design of an interactive instrument. A traditional instrument is structured as a single cause and effect, articulated as a synchronous linear path through a hierarchy of controls from a performer operating an input device to the multiple variables of a sound generator. An interactive instrument, on the other hand, is structured as a network of many causes and effects at various levels of importance, with a performer’s input as only one of the causes of the instrument’s output in sound. The author will present several historical examples of interactive electronic musical instruments and offer some speculations on the future.
Music scene description: Toward audio-based real-time music understanding
Authors: Goto Masataka
Published: October 2013

Music understanding is an important component of audio-based interactive music systems. A real-time music scene description system for the computational modeling of music understanding is proposed. This research is based on the assumption that a listener understands music without deriving musical scores or even fully segregating signals. In keeping with this assumption, our music scene description system produces intuitive descriptions of music, such as the beat structure and the melody and bass lines. Two real-time subsystems have been developed, a beat-tracking subsystem and a melody-and-bass detection subsystem, which can deal with real-world monaural audio signals sampled from popular-music CDs. The beat-tracking subsystem recognizes a hierarchical beat structure comprising the quarter-note, half-note, and measure levels by using three kinds of musical knowledge: of onset times, of chord changes, and of drum patterns. The melody-and-bass detection subsystem estimates the F0 (fundamental frequency) of melody and bass lines by using a predominant-F0 estimation method called PreFEst, which does not rely on the F0’s unreliable frequency component and obtains the most predominant F0 supported by harmonics within an intentionally limited frequency range. Several applications of music understanding are described, including a beat-driven, real-time computer graphics and lighting controller.
Analytic modeling of monolithic, laminated, and insulating glass sound transmission loss
Authors: Tocci Gregory
Published: October 2013

The sound transmission loss of single lights of glass, and of insulating glass configurations using various combinations of monolithic and laminated glass, have been computed using Mathcad, and formulations of sound transmission loss as described by Beranek and Ver, and Au and Byrne. The importance of damping in these formulations, as seen by comparing estimated and measured TLs, is discussed. This discussion is extended to demonstrate the need for unifying methods for mounting glass and window samples in laboratory test openings, and to suggest ways to do so. [Work supported by Solutia, Inc.]
Noise dimension in the design of urban spaces: Sonic instrumentation of public places
Authors: Rabah Derbal Cobis
Published: October 2013

The sonic environment dimension is not often taken into account in the design of urban spaces. This is not the same for internal spaces inside which a certain control of the quality of the sonic conditions is henceforth possible. However in the case of urban spaces, one is often confronted with situations complex and difficult to handle. This always leads to an impredictible situation in terms of noise and urban sonic environment. Underway studies have shown the importance of the quality of the urban sonic environment in the daily use of public places. The latter, as elements that mark strongly the degree of dynamism of the urban structure of our cities, are often subjected to a very complex situation of noise pollution. Introducing a source of noise that might have an effect of recomposing positively the sonic environment of the public place was the main approach of the study. It can be used as well to restore the sound in its utilitarian and practical function and to articulate the perception of the spatial and temporal aspects of the place. It is a sort of a sonic instrumentation of urban spaces that might recompose or recharacterize urban and architectural environments.
An eigenfunction expansion for low-frequency acoustic propagation in a downward-refracting stratified medium over a complex impedance plane
Authors: Waxler Roger
Published: October 2013

The well-known eigenfunction expansions associated with self-adjoint operators cannot be used to describe sound propagation over a complex impedance plane since, if the impedance has a real part, the resulting impedance boundary condition is not self-adjoint. Instead, a less widely known eigenfunction expansion associated with non-self-adjoint operators can be used. This eigenfunction expansion is applied to a vertically stratified model for downward refraction in the nighttime boundary layer. As in the self-adjoint case the propagation separates into a ducted part, expressed as a sum of modes which decay exponentially with height, and an upwardly propagating part, expressed as an integral over modes which are asymptotically (with height) plane waves. The eigenvalues associated with the ducted modes are complex, the imaginary parts being related to the acoustic attenuation. An efficient method for finding the complex eigenvalues is introduced from which a physically intuitive form of the attenuation coefficients is obtained. For low-frequency propagation (100 Hz or less) the number of modes is small, making this a simple way to model the ducted part of the propagation.
Ultrasonic b-scan imaging with adaptive beamformation using aberration correction
Authors: Pilkington Wayne C. , Lacefield James C. , Waag Robert C.
Published: October 2013

The effectiveness of adaptive beam formation using aberration correction has been demonstrated in ultrasonic b-scans of liver-mimicking scattering phantoms imaged with and without an intervening aberrator that produced wavefront distortion comparable to that of abdominal wall. Images of 4 mm diam spherical features (either positive or negative contrast lesions or scatterer-free cysts) in the uniform scattering background of the phantoms were produced at 3.0 MHz with a two-dimensional (80×80-element) array transducer system. Time-shift aberration was estimated from the scattering data and used to compensate both transmit and receive waveforms. Image improvements were assessed by comparison of feature contrast with and without aberration correction in individual images and by comparison of intensities in averages of independent, statistically identical images. Feature contrasts and borders were visibly and measurably improved, sometimes to near the water path results, using aberration correction, particularly when both transmit and receive corrections were applied. An efficient implementation of aberration correction was achieved by correction of multiple image scan lines with a single aberration estimate. Aberration correction using estimates from one-seventh the number of scan lines in 8 mm wide images produced improvements comparable to those achieved by individually estimating and correcting aberration in every scan line.
Statistical estimation of propagation path parameters for ultrasonic aberration correction
Authors: Waag Robert C. , Astheimer Jeffrey P.
Published: October 2013

Coefficients of a finite impulse response linear filter model for ultrasonic propagation are found using a statistical estimation. The model employs a Green’s function to describe scattering received by a two-dimensional array from a random-medium volume illuminated by a transmit beam. The frequency response of this Green’s function is factored into a homogeneous-transmission term and a path-dependent aberration term. Relative amplitude and phase of the aberration response are estimated over the frequency band of the transmit–receive system by using scattering from closely situated volumes and assuming that the aberration is the same for each volume. The amplitude and phase are obtained from the power and cross-power spectra of signals at positions throughout the receive aperture. The homogeneous response is estimated by averaging and is removed to isolate the aberration response. Propagation path aberration parameters calculated from pulse-echo measurements of random scattering through a tissue-mimicking aberration phantom are similar to corresponding parameters calculated for the same aberrator and array position by using echoes from a point-like reflector. The results indicate the approach is capable of describing, in addition to time-shifts, changes in waveform amplitude and shape produced by propagation through a distributed aberration.
Behavioral estimates of cochlear nonlinearity and its effects on normal and impaired hearing
Authors: Oxenham Andrew J. , Plack Christopher J.
Published: October 2013

Recent physiological studies of basilar-membrane motion have clarified many aspects of normal and pathological cochlear processing. The compressive input–output function in response to tones around the characteristic frequency and the sharp tuning at low levels are examples of basilar-membrane properties that are thought to be important for hearing, but that are also highly vulnerable to cochlear damage. Aspects of auditory perception influenced by cochlear nonlinearity include loudness and dynamic range, temporal processing, and frequency selectivity. Functional models have assisted us in understanding the perceptual consequences of peripheral nonlinearity in a wide variety of psychoacoustic tasks. In addition, many effects of cochlear hearing loss can be simulated within such a model simply by reducing or eliminating the nonlinearity within the model. A number of behavioral measures of cochlear nonlinearity in humans have been developed in recent years. These techniques offer insights into human cochlear processing in general, and may also provide diagnostic information about cochlear function on an individual basis. By gaining a better understanding of the changes in cochlear processing associated with hearing loss, it may be possible to design sound-processing algorithms for hearing aids that better compensate for the effects of cochlear damage. [Work supported by NIH Grant R01DC03909.]
JNDS of interaural time delay (ITD) of selected frequency bands in speech and music signals
Authors: Aliphas Avner , Colburn H. Steven , Ghitza Oded
Published: October 2013

JNDS of interaural time delay (ITD) of selected frequency bands in the presence of other frequency bands have been reported for noiseband stimuli [Zurek (1985); Trahiotis and Bernstein (1990)]. Similar measurements will be reported for speech and music signals. When stimuli are synthesized with bandpass/band-stop operations, performance with complex stimuli are similar to noisebands (JNDS in tens or hundreds of microseconds); however, the resulting waveforms, when viewed through a model of the auditory periphery, show distortions (irregularities in phase and level) at the boundaries of the target band of frequencies. An alternate synthesis method based upon group-delay filtering operations does not show these distortions and is being used for the current measurements. Preliminary measurements indicate that when music stimuli are created using the new techniques, JNDS of ITDs are increased significantly compared to previous studies, with values on the order of milliseconds.
Localization of noise bursts in the median plane: Effects of duration and level
Authors: Yu Gongqiang , Litovsky Ruth Y. , Wightman Frederic L.
Published: October 2013

Previous research has revealed that human sound localization in the median sagittal and in the front/back dimension may depend on the listener’s ability to reliably extract information regarding characteristic peaks and notches of the source spectrum upon reaching the ears. While long-duration sounds are well localized using spectral cues, the processing of spectral cues for sounds of short duration appears to be degraded. The mechanisms involved in ‘‘image formation’’ in the localization process, and the relative importance of sound duration and level are not well understood. In the present study, the effect of signal duration and level on localization performance, with an emphasis on the median-sagittal plane were systematically explored. Three types of stimuli, clicks, noises, and trains of frozen and random noise bursts encompassing the same time period as the noises, with different duration and at various levels, were randomly presented to the loudspeakers. Subjects were blindfolded, but were instructed to move their heads as necessary, to maximize their confidence regarding source position judgment. Results will be discussed in relation to possible mechanisms employed by the auditory system to utilize spectrally shaped cues in the localization process. [Work supported by NIDCD Grant Nos. R29 DC03083 and P01 DC 00116.]
A chinchilla nonlinear cochlear filterbank
Authors: Lopez-Najera Alberto , Lopez-Poveda Enrique A. , Meddis Ray
Published: October 2013

A dual-resonance nonlinear (DRNL) filter [Meddis et al., J. Acoust. Soc. Am. 106, 2852–2861 (2001)] was fitted to model chinchilla cochlear responses to tonal stimuli at individual sites along the basilar membrane (BM) with best frequencies (BF) of 0.8, 5.5, 7.25, 9.75, 10.0, 12.0, and 14.0 kHz. At each BF, parameters were obtained for the DRNL filter to reproduce input/output and tuning curves. The match between the model and the experimental data is almost perfect for frequencies near BF. Quantitatively, the model response gets worse (but is still reasonable) for frequencies well below and well above BF. These discrepancies are discussed in terms of the middle-ear function, which proves critical. The model responses to clicks, AM, multicomponent, and Schroder-phase stimuli were also compared against experimental data. Results show that the architecture of the DRNL filter seems suitable to reproduce this wide range of phenomena. Strategies are discussed for developing a chinchilla nonlinear cochlear filterbank from current parameters. [Work supported by the Consejería de Sanidad of the Junta de Comunidades of Castilla, La Mancha.]