bluuebunny/crossref_metadata_2025 huggingface dataset:
Investigating the precedence effect for noise bursts of different bandwidths
Authors: Braasch Jonas
Published: October 2013

The fact that the precedence effect partly fails for signals with a very narrow bandwidth raises the question of if this phenomenon requires an across-frequency-band interaction. To investigate this, a psychoacoustic experiment was conducted, in which the perceived lateralization of a noise burst (ITD: 300 μs) in the presence of a reflection (ITD: −300 μs) was measured. The parameters in this experiment were the inter-stimulus interval (ISI: 0.0 ms–4.0 ms) and the bandwidth (100 Hz, 400 Hz and 800 Hz). The data of the listeners clearly show that the localization dominance becomes more stable, especially in regard to the dependence on the ISI, when the bandwidth is increased. At the narrowest tested bandwidth, localization dominance could not be observed for some of the listeners, while others perceived at least the sound coming from the side of the direct sound source, although the position of their auditory events varied strongly with the ISI. A signal analysis on the basis of a modified Lindemann algorithm reveals that, in the first case, the precedence effect does not seem to have any significant influence. The signal analysis also indicates that it is not necessary to assume an interaction across the involved frequency bands and that it is sufficient to simply average over the model outputs of those frequency bands.
Level effects in monaural and interaural intensity discrimination
Authors: Stellmack Mark A. , Viemeister Neal F. , Byrne Andrew J.
Published: October 2013

Discrimination of interaural level differences (ILDs) is often assessed using a two-interval task in which ILD changes between intervals. In such a situation, overall level is usually roved in order to minimize monaural cues. This roving makes it difficult to assess level-dependent effects. In the present experiment, monaural intensity discrimination in a 2IFC task was compared to ILD discrimination in a single-interval task, thus producing an analogous situation in terms of number of ‘‘observations’’ and eliminating the need for an overall level rove. Monaural intensity DLs and ILD discrimination thresholds were measured as a function of level for 4-kHz tones and for broadband noise. The Weber functions (10logΔI/I vs I in dB) in the monaural and binaural conditions were parallel. For the noise the Weber functions had slopes close to zero (Weber’s law); for the tones the slopes were −0.082 (near-miss to Weber’s law). Overall, the binaural thresholds showed a small, approximately 2 dB, advantage over the monaural thresholds. The important aspect, however, is that the level effects seen monaurally are also seen binaurally. This suggests that the basic processes responsible for Weber’s law and the near-miss occur prior to binaural interaction. [Work supported by NIDCD DC05343 and DC00683.]
Online estimation of room reverberation time
Authors: Ratnam Rama , Jones Douglas L. , Wheeler Bruce C. , Feng Albert S.
Published: October 2013

The reverberation time (RT) is an important parameter for characterizing the quality of an auditory space. Sounds in reverberant environments are subject to coloration. This affects speech intelligibility and sound localization. State-of-the-art signal processing algorithms for hearing aids are expected to have the ability to evaluate the characteristics of the listening environment and turn on an appropriate processing strategy accordingly. Thus, a method for the characterization of room RT based on passively received microphone signals represents an important enabling technology. Current RT estimators, such as Schroeder’s method or regression, depend on a controlled sound source, and thus cannot produce an online, blind RT estimate. Here, we describe a method for estimating RT without prior knowledge of sound sources or room geometry. The diffusive tail of reverberation was modeled as an exponentially damped Gaussian white noise process. The time constant of the decay, which provided a measure of the RT, was estimated using a maximum-likelihood procedure. The estimates were obtained continuously, and an order-statistics filter was used to extract the most likely RT from the accumulated estimates. The procedure was illustrated for connected speech. Results obtained for simulated and real room data are in good agreement with the real RT values.
Reduced order modeling of head related transfer functions for virtual acoustic displays
Authors: Willhite Joel A. , Frampton Kenneth D. , Grantham D. Wesley
Published: October 2013

The purpose of this work is to improve the computational efficiency in acoustic virtual applications by creating and testing reduced order models of the head related transfer functions used in localizing sound sources. State space models of varying order were generated from zero-elevation Head Related Impulse Responses (HRIRs) using Kungs Single Value Decomposition (SVD) technique. The inputs to the models are the desired azimuths of the virtual sound sources (from minus 90 deg to plus 90 deg, in 10 deg increments) and the outputs are the left and right ear impulse responses. Trials were conducted in an anechoic chamber in which subjects were exposed to real sounds that were emitted by individual speakers across a numbered speaker array, phantom sources generated from the original HRIRs, and phantom sound sources generated with the different reduced order state space models. The error in the perceived direction of the phantom sources generated from the reduced order models was compared to errors in localization using the original HRIRs.
The effect of stimulus bandwidth and subject position on horizontal-plane localization with virtual source images
Authors: Grantham D. Wesley , Ashmead Daniel H. , Wall Robert S. , Frampton Kenneth D. , Willhite J. Andrew
Published: October 2013

In an anechoic chamber normal-hearing subjects performed a localization task in the frontal horizontal plane. The stimulus was a 200-ms burst of filtered noise. Within a block of trials, half of the presentations (randomly determined) were ‘‘real’’—presented from single loudspeakers—and the other half were ‘‘phantoms’’—produced by the simultaneous activation of two loudspeakers at ±30° using a virtual source imaging technique [Takeuchi et al., J. Acoust. Soc. Am. 109, 958–971 (2001)]. Both phantom and real sources spanned the azimuthal range ±80°. When the stimulus was a 4 kHz low-pass filtered noise, rms error was only slighly higher for phantom (D̄=7.1°) than for real (D̄=5.5°) sources. For 8 kHz low-pass filtered noise, performance remained about the same for real sources, but increased for phantom sources (D̄=11.5°). Data will also be reported for conditions in which the subject’s position is systematically varied outside the ‘‘sweet spot.’’ Results will be discussed in terms of robustness of the virtual imaging technique to stimulus and position factors and its potential usefulness as a tool for the investigation of human auditory spatial perception in static and dynamic environments. [Work supported by NIDCD.]
The effect of background noise on the perception of auditory distance
Authors: Brown Christopher A. , Whitmer William M. , Yost William A.
Published: October 2013

Five-ms broadband noise bursts were played and recorded through a KEMAR manikin from 6 different distances in a reverberant room. Distances ranged from 4 to 9 feet, and all recordings were 500 ms in duration. A single loudspeaker sat atop a table, and was moved to each distance prior to recording. During testing, stimuli were presented over headphones to listeners seated in a sound-attenuating chamber. In one condition, the ambient sound of the room was mixed with the stimuli, and served to fill in the silent gaps between stimulus presentations. Previous results have shown that distance judgments improve significantly when the silent gaps are filled with ambient noise. In the other conditions on-line noise, which was filtered to approximate the spectrum of the ambient noise, was added instead. Three levels of attenuation were used, −5 dB, 0 dB, and +5 dB, relative to the background noise level. The task was to judge the distance of the sound. A repeated measures design was employed, and listeners ran in only one condition on a given day. Preliminary results suggest that judgments are significantly improved with ambient-room noise, as compared to on-line noise. Additional findings and significance will be discussed. [Work supported by NIH.]
Statistical estimators of a periodically correlated random process for a voiced speech signal
Authors: Chorna Lesya B.
Published: October 2013

A stochastic model in the form of a periodically correlated random process (PCRP) was applied to a voiced speech signal. In such a signal, the amplitudes and phases of the harmonic components varied randomly; therefore, a stochastic model of the signal was appropriate. A description of the signal in terms of a stationary random process allowed an analysis of the spectral density of the oscillations, but the phase (temporal) structure remained unconsidered. To reveal this structure, a PCRP was used. The speech signal was processed by the co-phase analysis in the time domain and the component analysis in the spectral domain. The algorithms were based on a specific property of the PCRP that the samples selected with a period of correlation formed a series of related stationary sequences. Each sequence was then analyzed using the theory of stationary processes. This technique was used in a study of 70 recordings of the vowel /a/. The recordings were obtained from people with a normal heart rate and provoked arrhythmia. Statistical estimators of the vocal signal calculated on the basis of the PCRP model were statistically different for the normal and arrhythmic heart rates.
Gibbs sampling for multipath arrival time estimation in an uncertain environment
Authors: Picarelli Michele , Michalopoulou Zoi-Heleni
Published: October 2013

Estimation of time delays and amplitudes of multipath arrivals is of great interest in many fields because of the information that can be extracted from these characteristics. In underwater acoustics, multipath features of sound signals can reveal the location of the sound-generating source and properties of the propagating medium; their accurate estimation is, thus, desirable. In this work, a scheme is developed for the estimation of multipath arrival times and amplitudes in a noisy environment considering an unknown signal-to-noise ratio. The number of arrivals at the receiver is considered unknown as well, since, realistically, it depends on properties that would be uncertain a priori. The method is based on the efficient calculation of the number of paths, their amplitudes, and arrival times using Gibbs sampling to calculate a posteriori probability distributions. Results from the new method are compared to those of conventional approaches and show the potential of the new technique for efficient time delay and amplitude estimation in an environment with several unknowns. [Work supported by ONR.]
Audio fine classification using the statistical analysis of acoustic images
Authors: Chilton Edward , Paraskevas Ioannis
Published: October 2013

The fine classification of audio utterances is an important problem because the features that have to be extracted need to be very accurate in order to contribute to effective classification. In this paper, results are presented for a fine classification problem: namely the classification of two groups of different kinds of gunshots. The problem of accurate classification can be divided into two parts: (i) feature extraction and (ii) classification. The more effective the feature extraction, the more effectively the classifier will be able to categorize the various audio samples. In this paper, a novel method for the automatic recognition of acoustic utterances is presented using acoustic images as the basis for the feature extraction. The feature extraction process is based on the time-frequency distribution of an acoustic unit. A novel feature extraction technique based on the statistical analysis of the spectrogram Hartley transform (distribution) and Choi–Williams distributions of the data is reported as well as a brief discussion of the classifier used. The image is compressed using a statistical analysis of the acoustic image formed from the time-frequency distributions of acoustic data. The kurtosis, L-moments and entropy of the distributions, as well as the energy, contrast, etc., of the corresponding co-occurrence matrices of the distributions are calculated and then combined into a feature matrix. These appropriate features are then presented to the classifier. Initial results obtained indicate that the method is capable of accurate discrimination for fine classification.
NFL Films music scoring stage and control room space
Authors: Berger Russ , Schrag Richard C. , Ridings Jason J.
Published: October 2013

NFL Films’ new 200,000 sq. ft. corporate headquarters is home to an orchestral scoring stage used to record custom music scores to support and enhance their video productions. Part of the 90,000 sq. ft. of sound critical technical space, the music scoring stage and its associated control room are at the heart of the audio facilities. Driving the design were the owner’s mandate for natural light, wood textures, and an acoustical environment that would support small rhythm sections, soloists, and a full orchestra. Being an industry leader in cutting-edge video and audio formats, the NFLF required that the technical spaces allow the latest in technology to be continually integrated into the infrastructure. Never was it more important for a project to hold true to the adage of ‘‘designing from the inside out.’’ Each audio and video space within the facility had to stand on its own with regard to user functionality, acoustical accuracy, sound isolation, noise control, and monitor presentation. A detailed look at the architectural and acoustical design challenges encountered and the solutions developed for the performance studio and the associated control room space will be discussed.
Home studio acoustic treatments on a budget
Authors: Haverstick Gavin A.
Published: October 2013

Digital technology in the recording industry has evolved and expanded, allowing it to be widely available to the public at a significantly lower cost than in previous years. Due to this fact, numerous home studios are either being built inside or converted from bedrooms, dens, and basements. Hobbyists and part-time musicians that typically do not have the advantage of a large recording budget operate the majority of these home studios. Along with digital equipment, acoustic treatment has become more affordable over the years giving many musicians the ability to write, record, and produce an entire album in the comfort of their own home without having to sacrifice acoustical quality along the way. Three separate case studies were conducted on rooms with various sizes, applications, and budgets. Acoustical treatment such as absorption, diffusion, and bass trapping were implemented to reduce the effects of issues such as flutter echo, excessive reverberation, and bass build-up among others. Reactions and subjective comments from each individual studio owner were gathered and assessed to determine how effective home studios can be on a personal and professional level if accurately treated acoustically.
Design of a critical listening classroom and studio
Authors: Alach Bob , Clark Lou , Carman William , Case Alex
Published: October 2013

The Sound Recording Technology program at the University of Massachusetts—Lowell required design of a space that could serve three functions: a critical listening classroom for 30 students, a live room for the department’s premier 24 track recording studio, and a 5.1 surround sound mix room. Accomplishing all of the above required a re-evaluation of room shape and treatments and aggressive use of variable acoustics. In addition, all the usual culprits of HVAC noise and vibration, limited space, a constrained budget, and a difficult to renovate reinforced concrete building added to the challenge. This paper reviews the key design and renovation challenges associated with completing this room, opened for use by students and faculty in the Spring 2003 semester.
Acoustic communication in Panthera tigris: A study of tiger vocalization and auditory receptivity
Authors: Walsh Edward J. , Wang Lily M. , Armstrong Douglas L. , Curro Thomas , Simmons Lee G. , McGee JoAnn
Published: October 2013

Acoustic communication represents a primary mode of interaction within the sub-species of Panthera tigris and it is commonly known that their vocal repertoire consists of a relatively wide range of utterances that include roars, growls, grunts, hisses and chuffling, vocalizations that are in some cases produced with extraordinary power. P. tigris vocalizations are known to contain significant amounts of acoustic energy over a wide spectral range, with peak output occurring in a low frequency bandwidth in the case of roars. von Muggenthaler (2000) has also shown that roars and other vocal productions uttered by P. tigris contain energy in the infrasonic range. While it is reasonable to assume that low and infrasonic acoustic cues are used as communication signals among conspecifics in the wild, it is clearly necessary to demonstrate that members of the P. tigris sub-species are responsive to low and infrasonic acoustic signals. The auditory brainstem response has proven to be an effective tool in the characterization of auditory performance among tigers and the results of an ongoing study of both the acoustical properties of P. tigris vocalizations and their auditory receptivity support the supposition that tigers are not only responsive to low frequency stimulation, but exquisitely so.
Perception of species-specific vocalizations by birds
Authors: Dooling Robert
Published: October 2013

The complexity of bird vocalizations has fascinated us throughout the ages and there has long been the suspicion that birds are capable of producing, perceiving, and learning features of their songs that are beyond the capabilities of human hearing. Other investigators point to similarities between song production and perception in birds and special perceptual and attentional processes that are involved in the perception of speech by humans. Evidence will be reviewed that birds are born with early perceptual predispositions for learning species-specific vocalizations and demonstrate unusual efficiency in perceiving species-specific vocalizations throughout adulthood. In spite of these similarities with humans, birds have other auditory perceptual biases that are clearly different from humans as in the perception of tonal patterns. Birds, like humans, show a number of phenomena that aid perception under adverse conditions such as the binaural release from masking and the precedence effect and the detection of vocalizations in noise appears to follow from general principles of masking which has implications for studies of the effect of noise on birds in nature. [Work supported by NIH.]
Songlike vocalizations from a Sumatran rhinoceros calf (Dicerorhinus sumatrensis)
Authors: von Muggenthaler Elizabeth , Reinhart Paul
Published: October 2013

Within the last ten years the Sumatran rhino population has dropped 50%, and only 200–300 individuals exist, with five in captivity. Their native habitat is dense tropical forest and they are solitary, therefore much of their behavior remains unknown. Sumatrans are the smallest living rhino, standing 0.9–1.5 m tall, and are covered in coarse, reddish-brown hair. The first Sumatran rhinoceros born in captivity in 112 years, and the first calf ever recorded, is 17 months old and weighs 448 kg. At the Cincinnati Zoo this male calf was recorded from 1–3 m, using two Statham radio microphones, and one TCD-D8 Sony DAT recorder (9 Hz–22 kHz). Analysis, including power spectrums, spectrographic functions, and cross correlations were performed using National Instrument’s Polynesia. Preliminary analysis indicates that the calf’s vocalizations are similar in structure to adult Sumatran vocalizations, although there are some distinctions. ‘‘Eeps’’ and ‘‘whales’’ that are found in adult repertoires are produced by the calf. However, signals from the calf are higher in frequency, and the calf does not vocalize as consistently as the adults. The calf has yet to produce a ‘‘whistle blow,’’ which is an adult vocalization that has a strong infrasonic component.
Sound fluctuations due to horizontal refraction in the SWARM-95 experiment
Authors: Badiey Mohsen , Katsnelson Boris , Pereselkov Serguey , Lynch James , Siegmann William
Published: October 2013

Fluctuations of the sound field measured in the SWARM-95 experiment are interpreted as manifestations of three-dimensional effects in shallow-water sound propagation. It is shown that features of the received sound signal, along an acoustic track placed at 5° to the crests of propagating internal soliton trains, can be explained by significant horizontal refraction of the sound waves. This refraction leads to a remarkable redistribution of the energy in the horizontal plane as has been recently remarked by other investigators. These features are specifically: (1) a high level of fluctuations of the sound energy (up to 6 dB) which are synchronous in time and have same amplitude over depth of waveguide and (2) anomalies in the arrival times and amplitudes of separate waveguide modes. A theoretical analysis based on the theory of vertical modes and horizontal rays is presented, along with computer modeling of the experiment. Analytical estimations of the observed effects as well as results from numerical modeling are in a good agreement with experimental observations and are consistent with simulations. [Work supported by ONR and US CRDF Award No. VZ-010-00.]
Frequency-dependent anamolies of sound propagation in the SWARM-95 experiment
Authors: Katsnelson Boris , Pereselkov Serguey , Badiey Mohsen , Lynch James
Published: October 2013

Frequency-dependent acoustic propagation is a subject of great interest in shallow-water regions. The phenomenon known as resonance absorption occurs when broadband sound signals propagate across the crests of internal solitary waves (ISWs) if the sound field satisfies the condition of space synchronization. In this paper we show that different anomalies in frequency dependence can take place when sound signals propagate along the wave fronts of ISWs. These anomalies result from the anisotropic properties of the ISWs that can produce frequency-dependent horizontal refraction of acoustic waves in the ocean. Sound propagation of broadband pulses in the SWARM-95 experiment is analyzed from this point of view. The nature of such behavior is connected with the selective (over modes and frequencies) character of thermocline perturbations caused by ISWs. [Work supported by ONR and US CRDF Award No. VZ-010-00.]
Modeling the acoustic receptions at the NPAL array from the Kauai source
Authors: Vera Michael D. , Dzieciuch Matthew A. ,   the NPAL Group
Published: October 2013

Acoustic transmissions from a 75-Hz source near Kauai to a vertical line array near California were recorded as part of the North Pacific Acoustic Laboratory (NPAL) experiment. Extensive environmental measurements were also performed as part of the experiment and were intended to ensure correspondence between numerical simulations and the data. Despite the availability of this information, the process of identifying the recorded arrivals with predictions has not been a simple one. Since the source is near the seafloor at about 800 m depth, and the depth at the receiver is approximately 1800 m, acoustic interaction with the bathymetry has been explored as a possible complication. Ray simulations that allow for specular reflection from the bottom indicate that fully-refracted and bottom-interacting paths can reach the receiver range (about 3900 km) at similar travel times. The simultaneous presence of both kinds of acoustic energy could contribute to the identification difficulties. A series of parabolic-equation simulations have been performed for different geoacoustic parameters in an attempt to correspond more closely to the data. The sensitivity of the predictions to the method used to extract and interpolate the sound speeds has also been investigated. [Work supported by ONR.] a)J. A. Colosi, B. D. Cornuelle, B. D. Dushaw, M. A. Dzieciuch, B. M. Howe, J. A. Mercer, R. C. Spindel, and P. F. Worcester.
Effects of internal solitary waves on the invariance of acoustic intensity striation patterns
Authors: Turgut Altan , Orr Marshall , Rouseff Daniel , Lynch James , Chiu Ching Sang
Published: October 2013

Effects of Internal Solitary Waves (ISWs) on the acoustic intensity striation patterns are studied using a unique data set collected in the South China Sea during the ONR Asian Seas International Acoustics Experiment (ASIAEx). The data set contains measured broadband (270–330 Hz) acoustic fields on a 32-element, ∼400 m aperture horizontal array describing two regimes where the ISW packet being present and not present in a 19 km long propagation track. The geometry of the ISW packet was captured by a Synthetic Aperture Radar (SAR) image when it entered the propagation track at a 50 deg angle. It has been observed that ISWs introduce larger intensity fluctuations as well as frequency-shifts in the striation patterns when they are in the propagation track. However, the slope of the striations seems to be less affected by the presence of ISWs. Results from several broadband simulations are also presented to describe both acoustic intensity fluctuations and variations in the striation patterns in terms of acoustic mode coupling and mode refraction induced by ISWs. [Work supported by ONR.]
HIFU treatment of liver cancer—Successes and failures
Authors: ter Haar Gail , Rivens Ian , Kennedy James , Wu Feng
Published: October 2013

Clinical trials of the HIFU treatment of liver cancer have been underway in the UK at the Royal Marsden Hospital since December 1997, and at the Churchill Hospital since November 2002. Royal Marsden treatments are undertaken using a prototype device known as the Teleson, while those at the Churchill are performed using the machine produced by the Chongqing HAIFU company in China. Both sites have demonstrated the ability to ablate significant volumes of a tumor within the liver. Despite differences in ultrasound exposure delivery, these treatments have highlighted some of the problems associated with the clinical use of extracorporeal HIFU. These problems lie primarily in the areas of targeting, treatment optimization and monitoring of ablation. These problems will be discussed and potential solutions suggested. [Work funded by the UK Department of Health, the Institute of Cancer Research and UTL.]
Large ultrasound phased arrays for noninvasive focal treatments
Authors: Hynynen Kullervo , Clement Gregory , McDannold Nathan , Connor Chris , Vykhodtseva Natalia
Published: October 2013

Ultrasound applicators have become more advanced with the use of large scale phased arrays. These arrays can control the focal spot location and shape and compensate for the ultrasound beam distortions introduced by tissue layers. It has been demonstrated that sharp focus can be formed even through the human skull. However, the exposure control becomes complex with these new arrays, and online monitoring is required to provide clinical safety. For the past few years, magnetic resonance imaging has been tested for imaging the temperature distributions induced by focused ultrasound treatments. Both phantom and animal experiments have shown that the proton resonant frequency is temperature dependent and that maps of the temperature change can be obtained within seconds with phase subtraction methods. Recent clinical treatments have verified the usefulness of large scale phased arrays and temperature mapping during ultrasound surgery.
High intensity focused ultrasound (HIFU) treatment of BPH: results of a multi-center phase III study
Authors: Sanghvi N. , Gardner T. , Koch M. , Bihrle R. , Foster R. , Resnick M. , Seftel A. , Grunberger I. , Stiedle C. , Corchan J.
Published: October 2013

The five centers phase III trial was to show that HIFU can treat prostate tissue thermally for symptomatic relief of BPH and improve flow rates. At five sites, 68 BPH patients were treated with the Sonablate device (Focus Surgery, Inc. Indianapolis, IN). A urethral Foley catheter was inserted into the urethra to aid in positioning and was kept in-situ during the treatment. A cooling device was used to cool the rectal wall. The patients returned home within a few hours after the procedure. The Foley catheter was kept electively to avoid any incidence of acute urinary retention following the therapy. The catheter was removed after 4–5 days. The average treatment time was 38 minutes. The patients were treated without pain, blood loss or complications. At 90 days post treatment, average Qmax and AUA Symptom Scores improved from 8.7 ml/s to 12.66 ml/s (48%) and 23.06 to 11.62 (52%), respectively. Significant prostate tissue changes took place before and after the treatment. 80% of the patients had cavity formation at the site of treatment at the bladder neck and prostate. Nonsurgical HIFU therapy is safe and effective for providing symptomatic relief of BPH symptoms and the treatment can be performed as an outpatient procedure.
Parametric excitation of an acoustic mode
Authors: Denardo Bruce , Smith Derek B. , Varnadore Larry P. , Prather Wayne E.
Published: October 2013

One means of driving an oscillatory mode of a system is to modulate a parameter upon which the natural frequency of the mode depends. This ‘‘parametric excitation’’ occurs most readily if the modulation frequency is twice the frequency of the mode. In addition, the modulation amplitude must exceed a threshold that depends upon the dissipation of the mode. The response amplitude then grows exponentially until it is saturated by a nonlinearity of the system, so large amplitudes are possible. Previous efforts to parametrically excite an acoustic mode will be briefly described, and the theory for a length-modulated tube will be discussed. An analysis of this case indicates that it is feasible to parametrically excite the fundamental mode. Results of an experiment that is currently in progress will be presented. The tube has ambient length 2.0 m and diameter 30 cm, and the length is modulated by low-frequency loudspeakers at the ends. Sulfur hexafluoride gas is employed in order to decrease the required drive frequency and thus increase the maximum drive amplitude. A smooth constriction in the middle of the tube serves to detune the second mode, so that this mode is not directly excited.
Effect of forcing function on nonlinear acoustic standing waves
Authors: Finkbeiner Joshua R. , Li Xiaofan , Raman Ganesh , Daniels Christopher C. , Steinetz Bruce M.
Published: October 2013

Nonlinear acoustic standing waves of high amplitude have been demonstrated by utilizing the effects of resonator shape to prevent the pressure waves from entering saturation. Experimentally, nonlinear acoustic standing waves have been generated by shaking an entire resonating cavity. While this promotes more efficient energy transfer than a piston-driven resonator, it also introduces complicated structural dynamics into the system. Experiments have shown that these dynamics result in resonator forcing functions comprised of a sum of several Fourier modes. However, previous numerical studies of the acoustics generated within the resonator assumed simple sinusoidal waves as the driving force. Using a previously developed numerical code, this paper demonstrates the effects of using a forcing function constructed with a series of harmonic sinusoidal waves on resonating cavities. From these results, a method will be demonstrated which allows the direct numerical analysis of experimentally generated nonlinear acoustic waves in resonators driven by harmonic forcing functions.
Backward displacement of ultrasonic waves reflected from a corrugated interface
Authors: Teklu A. A. , Breazeale M. A.
Published: October 2013

A backward displacement can occur when an ultrasonic beam is reflected from an interface having a superimposed periodicity. This was first observed by Breazeale and Torbett [Appl. Phys. Lett. 29, 456 (1976)]. They observed that at a certain critical angle the reflected beam was shifted at 6 MHz and not at 2 MHz, in agreement with the theory of Tamir and Bertoni [J. Acoust. Soc. Am. 61, 1397–1413 (1971)]; however, the angle predicted by assuming that the shift was caused by a Rayleigh surface wave did not agree with experiment. Recent experiments using the schlieren technique showed backward displacement at an incident angle of 25 deg, in good agreement with the value 23.69 deg calculated using the inhomogeneous wave theory [Briers et al., J. Acoust. Soc. Am. 106, 682–7 (1999) (private communication with N. F. Declercq)]. The inhomogeneous wave theory also predicts a new kind of leaky surface wave at an incident angle of 22.55 deg [N. F. Declercq (private communication)], again in agreement with our measurements. Schlieren photographs of the backward-displaced beams at a water-brass interface are presented and discussed, as is the effect of the presence of an evanescent wave at the interface.
Analysis of particular phononic structures using single integral equations
Authors: Seydou Fadoulourahmane , Duraiswami Ramani , Gumerov Nail A.
Published: October 2013

A fast method for determining phononic (and photonic) bandgaps in composite materials is developed. It is known that in the propagation of waves in a 3D medium containing N scatterers arranged periodically, there exist refractive indices for which such structures have bandgaps, i.e., frequencies for which no waves can propagate inside. Our task is to find the frequencies that generate these prohibited waves. This requires the solution of an eigenvalue problem for the Helmholtz operator. To solve this problem we choose an alternate route which uses boundary integral equations. We derive a single integral equation on each of the interfaces between the outer region and the scatterers, considering a general transmission boundary condition, by using a hybrid method using layer potentials and Green’s formula. This approach reduces the number of unknowns considerably in comparison to other methods, but requires the treatment of large dense matrices and many matrix vector multiplications. To remedy this, we use the Fast Multipole Method. For solving the eigenvalue problem we discuss two methods: the Newton method and a method based on the Cauchy formula. Details of the numerical implementation, and results will be presented for different cases: sound hard, sound soft, impedance and transmission boundary conditions. [Work partially supported by NSF Award 0219681 is gratefully acknowledged.]
Group velocity manipulation in simple acoustic band gap filters
Authors: Robertson W. M. , Baker C. , Bennett C. Brad
Published: October 2013

A simple experimental configuration is presented in which the group velocity of acoustic pulses is adjusted from much slower to much faster than the speed of sound. The experiment is an acoustic analog of two much studied optical phenomena: the superluminal group velocity achieved when optical pulses are tunneled through regions of high absorption or attenuation, and slow (or even stopped) light propagation in the vicinity of strong material dispersion, typically realized using electromagnetically induced transparency in atomic vapors. For the acoustic experiments described here, attenuation and dispersion were created using periodically structured waveguides. The periodicity results in one-dimensional acoustic band gaps, frequency intervals in which propagation of sound is strongly suppressed. The introduction of defects in the perfect periodicity leads to narrow transmission bands within the band gaps. There is strong normal dispersion in the vicinity of these defect modes leading to very slow group velocities. By tuning the carrier frequency of acoustic pulses through the forbidden and defect regions the group velocity can be adjusted from 0.22 to 1.8 times the speed of sound in the straight unstructured waveguide. These results are shown to be consistent with a straightforward theoretical model.
Auditory performance in an open sound field
Authors: Fluitt Kim F. , Letowski Tomasz , Mermagen Timothy
Published: October 2013

Detection and recognition of acoustic sources in an open field are important elements of situational awareness on the battlefield. They are affected by many technical and environmental conditions such as type of sound, distance to a sound source, terrain configuration, meteorological conditions, hearing capabilities of the listener, level of background noise, and the listener’s familiarity with the sound source. A limited body of knowledge about auditory perception of sources located over long distances makes it difficult to develop models predicting auditory behavior on the battlefield. The purpose of the present study was to determine the listener’s abilities to detect, recognize, localize, and estimate distances to sound sources from 25 to 800 m from the listing position. Data were also collected for meteorological conditions (wind direction and strength, temperature, atmospheric pressure, humidity) and background noise level for each experimental trial. Forty subjects (men and women, ages 18 to 25) participated in the study. Nine types of sounds were presented from six loudspeakers in random order; each series was presented four times. Partial results indicate that both detection and recognition declined at distances greater than approximately 200 m and distance estimation was grossly underestimated by listeners. Specific results will be presented.
Speech perception in tight acoustic spaces
Authors: Iyer Nandini , Ratnam Rama , Phatak Sandeep , Lansing Charissa , Feng Albert
Published: October 2013

Acoustic cues for sound localization and speech perception in reverberant environments are more complex than those in anechoic conditions. The ability to utilize these cues deteriorates as a function of age. This study examined the speech perception ability of elderly listeners in well-defined, enclosed spaces and identified signal-processing strategies that might enhance this ability. For this, we performed tests in three environments: a car, a plywood cube, and a conference room. Pre-recorded monosyllabic words (Modified Rhyme Test) were played in quiet or in the presence of an 8-talker babble from a loudspeaker and recorded using two microphones placed at KEMARs ears. These signals were played to 10 young and 10 elderly listeners with normal hearing. We evaluated speech perception abilities of the listeners as a function of the following variables: (1) the location and distance of the loudspeaker from the listener, and (2) the number of reflecting surfaces. We tested the hypothesis that diotic presentations of the signal (better ear signal to both ears, or active steering followed by a presentation of the summed signal) improved speech perception abilities in these situations.
Psychometric functions for informational masking
Authors: Lutfi Robert A. , Kistler Doris J. , Callahan Michael R. , Wightman Frederic L.
Published: October 2013

The method of constant stimuli was used to obtain complete psychometric functions (PFs) from 44 normal-hearing listeners in conditions known to produce varying amounts of informational masking. The task was to detect a pure-tone signal in the presence of a broadband noise and in the presence of multitone maskers with frequencies and amplitudes that varied at random from one presentation to the next. Relative to the broadband noise condition, significant reductions were observed in both the slope and the upper asymptote of the PF for multitone maskers producing large amounts of informational masking. Slope was affected more for some listeners while asymptote was affected more for others. Mean slopes and asymptotes varied nonmonotonically with the number of masker components in much the same manner as mean thresholds. The results are consistent with a model that assumes trial-by-trial judgments are based on a weighted sum of dB levels at the output of independent auditory filters. For many listeners, however, the weights appear to reflect how often a nonsignal auditory filter is mistaken for the signal filter. For these listeners adaptive procedures may produce a significant bias in the estimates of threshold for conditions of informational masking. [Work supported by NIDCD.]
Effect of signal-masker similarity and signal uncertainty on informational masking
Authors: Oh Eunmi L. , Lutfi Robert A.
Published: October 2013

An experiment was conducted to determine the relative importance of signal-masker similarity and signal uncertainty as factors affecting informational masking. The masker was a complex of fixed-frequency tones whose levels varied independently and at random on each presentation. The signal was a complex of fixed-frequency tones whose levels were sampled from one of two Gaussian distributions differing in mean (SD=5 dB). The levels of the signal tones were sampled independently (IS condition) or were the same as determined by a single sample (SS condition). The listener’s task was to select the signal from the distribution with the higher mean. It was expected that if uncertainty were the dominant factor better performance would be obtained in the IS condition—this due to the reduction in uncertainty associated with multiple independent ‘‘looks’’ at the signal. If, however, similarity were the dominant factor better performance would be obtained in the SS condition—this due to the dissimilarity of signal and masker spectra (flat versus irregular). Four of six listeners showed the first result, the remaining two the second result. The remaining two listeners, however, showed results inconsistent with a similarity effect with maskers having flat spectra. [Work supported by NIDCD.]
Acceptance of background noise as a function of speech presentation level
Authors: Franklin Clifford A. , Nabelek Anna K. , Burchfield Samuel B.
Published: October 2013

The acceptance of background noise while listening to speech (ANL) at different speech presentation levels was assessed. Twenty listeners (10 male) between 18–30 years with normal hearing listened to a narrative at speech presentation levels of 20, 34, 48, 62, and 76 dB HL, then adjusted the background noise to the highest intensity level that they would be willing to accept for an extended listening period. The ANL is the intensity of the speech presentation level minus the intensity level of the background noise. The group mean ANLs for presentation levels of 20, 34, 48, 62, and 76 dB HL were 10.60, 14.25, 17.10, 21.80, and 24.55 dB, respectively. The group mean ANLs differ by approximately three and one half decibels between each presentation level. This difference between adjacent speech presentation levels is representative of a linear function. The average MCL was 43 dB HL with a standard deviation of 6.7 dB. The group mean ANL for speech presented at MCL was 15.5 dB with a standard deviation of 7.27 dB. A MANOVA for repeated measures indicated a statistically significant main effect for speech presentation level (F=18.624, p=0.001). [Work partially supported by NIDCD (NIH) R01 DC 05018.]
Relation between measures of speech-in-noise performance and measures of efferent activity
Authors: Smith Brad , Harkrider Ashley , Burchfield Samuel , Nabelek Anna
Published: October 2013

Individual differences in auditory perceptual abilities in noise are well documented but the factors causing such variability are unclear. The purpose of this study was to determine if individual differences in responses measured from the auditory efferent system were correlated to individual variations in speech-in-noise performance. The relation between behavioral performance on three speech-in-noise tasks and two objective measures of the efferent auditory system were examined in thirty normal-hearing, young adults. Two of the speech-in-noise tasks measured an acceptable noise level, the maximum level of speech-babble noise that a subject is willing to accept while listening to a story. For these, the acceptable noise level was evaluated using both an ipsilateral (story and noise in same ear) and a contralateral (story and noise in opposite ears) paradigm. The third speech-in-noise task evaluated speech recognition using monosyllabic words presented in competing speech babble. Auditory efferent activity was assessed by examining the resulting suppression of click-evoked otoacoustic emissions following the introduction of a contralateral, broad-band stimulus and the activity of the ipsilateral and contralateral acoustic reflex arc was evaluated using tones and broad-band noise. Results will be discussed relative to current theories of speech in noise performance and auditory inhibitory processes.
Speech perception in gated noise: The effects of spectral and temporal resolution and auditory streaming
Authors: Jin Su-Hyun , Nelson Peggy B.
Published: October 2013

Hearing-impaired (HI) listeners often report difficulty understanding speech in the presence of background noise, even in the presence of mild degrees of hearing loss. In addition, HI listeners show significantly less benefit from fluctuations in noise than do normal hearing (NH) listeners. Furthermore, HI listeners often show less accuracy of processing frequency and temporal information in acoustic signals and grouping them into a whole meaningful speech. The purpose of this study is to examine differences in performances of NH and HI listeners for speech perception in various types of noise. Three hypotheses will be examined in this study. First, the amount of masking release in HI listeners with mild hearing loss will be smaller than in NH listeners even when speech and noise are presented at intensities sufficient to overcome the hearing loss. Masking release will be measured for consonant recognition and sentence recognition by subtracting the percent correct in steady noise from that in gated noise. Second, the performance of HI listeners on spectral resolution, temporal resolution, and auditory streaming tasks will be significantly poorer than that of NH listeners. Third, the performance differences in these tasks may account for differences in masking release.
Noise suppression algorithm based on the auditory masked threshold in listeners with cochlear hearing loss
Authors: Arehart Kathryn , Rossi-Katz Jessica , Natarajan Ajay , Hansen John
Published: October 2013

This study describes the formulation and evaluation of a new noise suppression scheme, with the goal of improving speech-in-noise perception by hearing-impaired listeners. Arehart et al. [Speech Commun. (2003)] implemented and evaluated a noise suppression algorithm based on an approach that used the auditory masked threshold in conjunction with a version of spectral subtraction to adjust the parameters used in the subtraction process based on the masked threshold of the noise across the frequency spectrum. That original formulation was based on masking properties of the normal auditory system, with its theoretical underpinnings based on MPEG-4 audio coding [Johnston (1988)]. This paper describes details of a new enhancement formulation based on masking characteristic of cochlear hearing loss. The new algorithm improves on previous formulations in two ways. First, the algorithm is implemented with generalized minimum mean-square error estimators, which provide improvements over spectral subtraction estimators. Second, estimation of the auditory masked thresholds and masking spreading functions are adjusted to address elevated thresholds and broader auditory filters characteristic of cochlear hearing loss. In addition to algorithm details, results of the algorithm evaluation (using objective quality measures; intelligibility and quality measures in hearing-impaired listeners) will be presented. [Work supported by the Whitaker Foundation.]