bluuebunny/crossref_metadata_2025 huggingface dataset:
Estimation of vocal-tract area function from lip impulse response based on lossy vocal-tract digital filter model
Authors: Mochida Takemi , Honda Masaaki
Published: October 2013

A method for estimating vocal-tract area function from lip impulse response based on lossy vocal-tract digital filter model is provided. Previous methods of lip impulse response [M. M. Sondhi and J. R. Resnick, 73, 985–1002 (1983)] determines the area function from impulse response measured at lips by using the incident signal excited outside the oral cavity. This method is sensitive to the noise of measured impulse response. An alternative method [M. Honda and T. Mochida, Proc. 1st ETRW-SPM, 167–170 (1996)] is advantageous for accurate estimation under noisy acoustic condition, based on the vocal-tract digital filter model that optimizes the M sections of areas by using the impulse response longer than M samples. In this method, however, the vocal-tract model is based on the transmission line that treats the effect of loss in simplified form. The new method proposed in this study applies a mixture model of the lossless distributed circuit and the lumped circuit accounting for effects such as viscous loss, heat conduction loss, and loss due to wall vibration. The results on acoustic measurements have shown that the new method has better performance over the old method using optimization scheme based on simplified transmission line model. [Work supported by CREST, JST.]
An audition-inspired method for producing high resolution spectrograms
Authors: Dajani Hilmi R. , Kunov Hans , Wong Willy
Published: October 2013

Inspired by the ability of the auditory system to dynamically track frequency and amplitude changes in modulated signals, an algorithm for high resolution time-varying spectral analysis is proposed. The main feature of the algorithm is an auditory-inspired filterbank followed by a peak detector. Mathematical analysis, with various types of synthetic modulated signals, demonstrates that the proposed method correctly demodulates these signals. The resulting time-frequency display, referred to as a fine structure spectrogram, shows the frequency and amplitude modulations in higher detail than is possible with conventional spectrograms. With recorded consonant–vowel syllables, the fine structure spectrogram detects small frequency and amplitude modulations in the initial glides and in the steady vowel formants. These modulations are not clearly seen in conventional time-frequency representations.
Do submerged objects twinkle from gated pulse signals in turbulent wave guides?
Authors: Werby Michael
Published: October 2013

If one illuminates the ocean with a time-gated acoustical signal will an inclusion such as an elastic shell insonfied by the pulsed twinkle or, in the jargon of some researchers, will it scintillate? Twinkling from stars owes its origin to a mildly turbulent atmosphere. The ocean is often much more turbulent and we examine the distortions that one gets from scattered pulse-gated signals.
Extending the legacy of Ed Hamilton in high frequency sediment acoustics
Authors: Thorsos Eric I. , Williams Kevin L. , Jackson Darrell R. , Tang Dajun , Richardson Michael D. , Briggs Kevin B.
Published: October 2013

Ed Hamilton set a high standard for geoacoustic modeling of the sea floor. His comprehensive yet thorough descriptions of sediment properties in forms readily usable by acoustic modelers were seminal contributions to underwater acoustics. Hamilton was able to synthesize a wide diversity of in situ acoustic data within the framework of the Gassmann sediment model in which the pore water and sediment solids move together in response to acoustic fields and with attenuation modeled empirically. During the Sediment Acoustics Experiment in 1999 (SAX99), a wide variety of acoustics measurements were made in the 0.1–1000 kHz range combined with extensive sediment characterization of the medium sand, shallow water site. Some of our acoustic measurements support the hypothesis that relative motion between the pore fluid and sand grains has important consequences for sediment acoustics at the low end of our frequency range. Results from SAX99 will be reviewed which suggest the importance of more complex sediment models for a shallow water sand site. One such model will be described that recovers the spirit of Hamilton’s desire to minimize complexity yet accurately describes the acoustic behavior of the sediment. [Work supported by ONR.]
Ocean sediment high-frequency acoustics
Authors: Chotiros Nicholas P.
Published: October 2013

The upper few meters of ocean sediment is most relevant to acoustics, in the frequency band 10–500 kHz. It is often approximated as a stratified fluid or visco-elastic solid. The acoustics of the sediment, however, is often more complicated than either of these approximations will allow. A symptom of the problem is their inability to match experimental data. The problem seems to be centered about the physics of the grain–grain and grain–fluid interactions. Much of the difficulty lies in the physics of the vast range of structures that lie between the two extremes of a fluidized suspension, such as silt, and a fully cemented porous structure, such as sandstone. The Biot–Stoll equations of sound propagation in a porous medium provide the mathematical framework to represent the acoustics of a porous structure. The definition of pore fluid and skeletal frame is a critical issue that needs further development, in order to accommodate the full range of sediment types. [Work supported by ONR, Ocean Acoustics, Jeff Simmen.]
A comparison of models for water-saturated sand by inversion of reflection loss measurements
Authors: Isakson Marcia J. , McJunkin Sam , Hood Egbert , Chotiros Nicholas
Published: October 2013

Our objective is an accurate model of sediment acoustics that is consistent with all measurable acoustic and geophysical properties. This will lead to a better understanding of acoustic penetration in ocean sediments. In this study, reflection data taken from a smooth water/sand interface are inverted using a broad range of models. The models considered in the inversion are the visco-elastic solid model, the Biot/Stoll model, the composite Biot model, the effective density model, and the Buckingham model. Inversions were performed using a simulated annealing optimization algorithm with OASR as the forward model. Each model is considered for its ability to produce realistic parameters and its fit to the experimental data. [Work supported by ONR, Undersea Signal Processing, John Tague.]
Under-balcony acoustics in concert halls: Vertical energy loss at the listener and acceptable view angles deduced from its effects
Authors: Kwon Young-Min , Shimizu Yasushi
Published: October 2013

Recent investigations into acoustics in under-balcony spaces have found that balcony overhangs reduce the early energy from above as well as the late energy from the main space, and hence cause an acoustic defect in the vertical. Based upon these findings, this research analyzes energy loss under the balcony, when sound energy comes from above through a vertical median plane. Fundamentally, it aims to find acoustically acceptable view angles (Θ) against the balcony opening height (H). Scale-model experiments in an anechoic room were done to obtain the amount of energy reduction (or reduction coefficient) by balcony overhangs as a function of incidence angle. Next, psychoacoustical experiments associated with listener envelopment were conducted under synthetic sound fields with constant lateral energy but with variable vertical energy to determine the appropriate Θ. These experiments, which combined with data from the scale-model experiments, cover paired comparison tests under two sets: with balcony and without balcony conditions. Results of the scale-model experiments show that sound energy decreases greatly within certain incidence angles near the balcony and at lower frequency bands. The minimum vertical angle of view is inferred to be 30–40 deg from psychoacoustical experiments. Overhang depth (D), D/H ratio, and their relationship are also discussed.
Bayesian decay time analysis in coupled spaces using a proper decay model
Authors: Xiang Ning , Goggans Paul M.
Published: October 2013

Acoustically coupled spaces have recently been drawing more and more attention in the architectural acoustics community. Determination of decay times in these coupled spaces from measured Schroeders decay function has become increasingly significant. In architectural acoustics practice, identification of decay times from measurements of room impulse responses in these coupled spaces can often require considerable effort. Recent works [N. Xiang, P. M. Goggans, and D. Li, J. Acoust. Soc. Am. 109, 2283 (2001); P. M. Goggans and N. Xiang, ibid. 109, 2383 (2001); N. Xiang and P. M. Goggans, ibid. 110, 1415–1424 (2001)] have demonstrated that Bayesian parameter estimation and model selection prove to be powerful tools for evaluating decay times in coupled spaces. Following a brief introduction to Bayesian model selection, this paper will discuss further results from decay-time evaluations of measured room impulse responses. Emphasis will be given to the behavior of the decay model selection algorithm.
Characterization of the acoustic response of an auditorium for its recreation in a rehearsal room
Authors: Martignon Paolo , Zanolin Michele , De Stabile Stefano
Published: October 2013

A characterization of the stage acoustic response of the auditorium Niccolò Paganini in Parma (Italy) was carried out. The aim is to reproduce, in a low reverberation room, the response of this auditorium by means of a loudspeakers array system, implementing the image sources corresponding to first reflections and the correct isotropy of the reverberant field [A. J. Berkhout, J. Audio Eng. Soc. 36, 977–995 (1988)]. The original idea is that such a room would become a virtual stage for orchestra rehearsals. Three linear microphone arrays 10 cm apart, arranged as Cartesian axes, were employed to acquire three impulse response arrays using one omnidirectional source. Measurement bandwidth was 20–2000 Hz: within this frequency range and the angles involved, spatial aliasing due to microphone array spacing (10 cm) does not occur (microphone spacing is a little less than the one of speakers array to be employed for reproduction). Data processing gave information about first reflections wave fronts curvature, and consequently their correspondence with architectural elements; after identifying these elements as different contributions, spectral absorption of each was calculated. Reverberation was separately measured in several points with different directions, using a directional microphone (W. P. Bruijn et al., 104th AES Conv., preprint 4690, 1998).
An experimental and theoretical investigation of the high frequency broadband soundfield
Authors: Franzoni Linda
Published: October 2013

A relatively simple theoretical model of a rectangular enclosure with sound sources on one wall, an absorptive end-wall, and two absorptive side-walls is developed. Using a power balance at cross sections, a differential equation for mean-square pressure in the acoustic enclosure is derived. A simple formula for the mean-square pressure as a function of distance from source-wall to absorbing end-wall is obtained. The other variables in the formula include the random incidence absorption coefficients, the cross-sectional area and perimeter, and the source power. This formula is compared to results from numerical simulations, to a revised Sabine prediction for mean-square level, and to experimental results. For the most absorptive cases, however, a more refined theoretical model is needed in order to have reasonable agreement with the experimental data. In the refined model, it is assumed sources are distributed on an end-wall and the resulting sound waves are considered on an angle-by-angle basis. The mean-square pressure is calculated by integrating over all possible angles. Although still a fairly simple result, the effects of angle of incidence on the absorption and the reflection field in the enclosure are retained.
Shear waves in surficial sediments: Revisited
Authors: Richardson Michael D.
Published: October 2013

Numerous measurements of compressional and shear wave speed and attenuation have been made since Ed Hamilton’s classic works on sediment geoacoustic properties (1955–1987). Recent measurements of shear wave speed and attenuation are summarized; regressions between shear wave speed and attenuation and easily measured sediment physical properties are used to demonstrate the predictability of these geoacoustic properties. The effect of overburden pressure (effective stress) on shear wave speed is also discussed. [Work supported by ONR.]
Rapid geo-acoustic characterization from a seismic survey
Authors: Heaney Kevin D. , Sternlicht Daniel , Teranishi Arthur , Castille Brett , Hamilton Michael
Published: October 2013

A recent transmission loss experiment was conducted in Long Beach Harbor for the THUMS Long Beach Company. The objective of the experiment was to measure the range at which the received level was 160 dB for compliance with Marine Mammal regulations. This short experiment provided the opportunity to test the rapid geo-acoustic characterization (RGC) algorithm and perform real-time geo-acoustic inversions from a seismic source. The airgun source transmitted pulses every 20 s corresponding to every 45 m. The water depth was 10–15 m and the water was assumed to be iso-velocity. The data quality was excellent, providing clear striation patterns in the broadband frequency display. The RGC algorithm matches the observed time-spread, striation slope, and TL slope to precomputed values using a normal mode algorithm and parametric geo-acoustic profiles based on Hamilton and Bachman’s model. Precomputation of the acoustic observables, combined with real-time signal processing permits real time geo-acoustic characterization.
Rapid geo-acoustic characterization from a surface ship of opportunity
Authors: Heaney Kevin D.
Published: October 2013

The recent shift from deep water antisubmarine warfare to the littoral has brought about a dramatic change in the requirements of our understanding of the acoustic environment. In particular, shallow water acoustic propagation is dominated by the interaction of sound with the bottom and is therefore very sensitive to the geo-acoustic parameters of the local sea-floor. The rapid geo-acoustic characterization (RGC) algorithm is presented, that determines a geo-acoustic model that best matches the general acoustic propagation parameters. Using data from a passing surface ship, the time spread, TL versus range and striation slope are measured at various frequencies. These are then matched to a simple two-layer geo-acoustic model based on the empirical curves of Hamilton and Bachman. The resulting estimate does reproduce the global features of the acoustic field that are relevant to acoustic prediction. This method is robust, rapid, and produces the relevant acoustic predictions.
Naturalness of voice qualities associated with adjustments in the glottis and lower vocal tract
Authors: Titze Ingo R.
Published: October 2013

Voice qualities such as twang, yawn, ring, pressed, sigh, and creaky have recently been captured by physical modeling of glottal aerodynamics, vocal fold biomechanics, and vocal tract acoustics. This paper focuses on the interactions between source and vocal tract adjustments and their likely co-variations in human voice production. For example, is pressed voice more likely to occur with twang than yawn, and is sigh (of fluty voice) more likely to occur with yawn than twang or ring? The underlying principles may be impedance matching and maximum source-filter interaction. [Work supported by NIH.]
Systematic downgrading for investigating ‘‘naturalness’’ in synthesized singing using STRAIGHT: A high quality VOCODER
Authors: Kawahara Hideki
Published: October 2013

Conceptual simplicity of the classical channel VOCODER provides a powerful means for systematic investigations on perceptual effects of speech related physical parameters when combined with modern computational power and signal processing theories. A modern version of channel VOCODER, STRAIGHT [Kawahara et al., Speech Commun. 27, 187–207 (1999)], which is also an extension to pitch-synchronous analysis and synthesis, generates naturally sounding resynthesized speech from the analyzed smooth time-frequency surface and source parameters such as F0. This high-quality resynthesis enables close investigations on naturalness deterioration as a function of feature modifications in the decomposed parameter domain; for example, detailed shape of a F0 trajectory, underlying parameters to determine F0 trajectory dynamics, group delay alignment of excitation pulses and aperiodicity/periodicity ratio of the excitation source and so on. One of potential advantages of this strategy is based on the fact that our perceptual function is highly nonlinear. The other source of advantage is virtually an independent parameter set which allows precise control of parameter deviations from the original analysis results. An overview of recent findings and modification demonstrations will be presented. [Work supported by CREST grant of Japanese Science and Technology Corporation.]
Performance of electroacoustic hearing protectors
Authors: Murphy William , Little Mark
Published: October 2013

The spectral and temporal response characteristics of five models of electroacoustic earmuffs (Bilsom 707 Impact II, Peltor Tactical 6s, Howard Leight Lightning and Thunder w/ProEars, and Silencio Electronic Low Pro hearing protectors) were measured with noise created by small arms fire from a 0.223 caliber rifle and a 9 mm handgun. Each hearing protector was tested in the fully-amplified (active) and no amplification (passive) conditions. The protectors were measured on an acoustic test fixture designed for high-level noise measurements. The protected and unprotected signals were recorded to digital audio tape and analyzed off-line. The reduction of peak sound pressure levels ranged between 16 and 32 dB for both the active and passive conditions. The time-averaged acoustic waveforms underneath the protectors exhibited little difference between active and passive conditions. The active electroacoustic performance approaches the passive attenuation performance. Auditory damage units were calculated for the unprotected and protected responses using the AHAAH cochlear model [Price and Kalb, J. Acoust. Soc. Am. 90, 219–227 (1991)].
Problems associated with noise measurements in the mining industry
Authors: Bauer Eric R. , Vipperman Jeffrey S.
Published: October 2013

In response to the continuing problem of noise-induced hearing loss (NIHL) among mine workers, the National Institute for Occupational Safety and Health (NIOSH) has been conducting numerous noise- and hearing-loss research efforts in the mining industry. Research is underway to determine worker noise exposure, equipment noise, hearing loss and hearing protection use, and to evaluate engineering controls. Issues that are peculiar to the mining industry have complicated these efforts. A few of the issues that must be overcome to conduct meaningful research include constantly moving equipment, changing work environments, confined space, varying production rates, multiple noise sources, and electronic permissibility of instrumentation. This presentation will address the factors that affect the measurement and analysis of noise in the mining industry and how these factors are managed. In addition, some examples of research results will be included.
Identification and classification of noise sources in a chain conveyor
Authors: Homer John P. , Vipperman Jeffrey S. , Reeves Efrem R.
Published: October 2013

Noise induced hearing loss (NIHL) is one of the most significant disabilities of workers in the mining industry. In response, the National Institute of Occupational Safety and Health (NIOSH) is conducting a study associated with mining equipment. This study outlines the analysis of a chain conveyor. Band-limited accelerometer, sound-intensity, far-field and near-field microphone measurements were taken along the conveyor section. The sound intensity measurements were used to identify areas with high noise as well as to calculate and 1/3-octave sound power levels. The total sound power results were used to classify the dominant noise sources where the 1/3-octave sound power results were used to identify the most contributive frequency bands to the overall noise of the system. Coherence analysis was performed between accelerometer and microphone measurements to identify structure–borne and air–borne noise paths of the system. Summary results from the analysis include recommendations for transmission control and damping devices and their ability to reduce noise to regulatory acceptable levels.
A physiological model for comodulation masking release based on cross-frequency coincidence detection
Authors: Carney Laurel H.
Published: October 2013

Detection thresholds for tones in noise can be significantly improved when the masker is amplitude modulated, or when its various components are comodulated [e.g., Hall et al., J. Acoust. Soc. Am. 76, 50–56 (1984)]. Several stimulus-based signal-processing models have been proposed to explain this phenomenon, referred to as comodulation masking release (CMR); however, no physiologically based models have successfully described the psychophysical data. These models have tested hypotheses that CMR involves either cross-frequency processing strategies or within channel envelope-based processing strategies. Cross-frequency coincidence detection is a physiologically realistic, nonlinear processing mechanism that is consistent with suggestions that information from different frequency channels is combined to explain CMR. Furthermore, cross-frequency coincidence detectors are sensitive to envelope cues, thus this mechanism is also consistent with within-channel envelope-based models for CMR. The sensitivity of coincidence detectors to envelope cues is due to the nonlinear timing of low-frequency auditory-nerve (AN) fibers—changes in amplitude of the stimulus result in systematic changes in the relative times of phase-locked AN fibers tuned to different frequencies [Heinz et al., J. Acoust. Soc. Am. 110, 2065–2084 (2001)]. A cross-frequency coincidence-detection model can explain the psychophysical results for the basic CMR paradigm described in Hall et al. (1984). [Work supported by NIH-NIDCD.]
The effect of audiometric headphone position on the measurements of threshold levels
Authors: Steffani Jovani A. , Gerges Samir N. Y. , Fiorini Ana C.
Published: October 2013

The results of audiometric tests are sensitive to the position of the headphone on the ears. Up to 30-dB differences in all frequency bands can be obtained. In this paper measurements are reported for normal audiometric tests for the same audiometer equipment varying the listener and also for different audiometric equipment for the same listener. These measurements were also carried out using an artificial head with molded external ears to give more parameter control experimental results. All the measurements were carried out in an acoustic room with very low background noise. Nine positions were used for the measurements of SPL (one reference headphone position and four positions in the up, down, left, right and four other positions on the 45 deg), for each position three locations of the headphone were used at 5 mm, 10 mm and 15 mm away from the center), totaling 45 cases. Larger differences obtained at high frequencies. 72% of the measured cases give higher threshold levels and 28% gives lower levels than reference position. The results can give good contribution to the confidence limits and repeatability factor for audiometric testing.
Vaudeville-era renovations from the architect’s point of view
Authors: Filoni Albert L.
Published: October 2013

Interaction between the architect and acoustics consultant on renovation and restoration projects often must reconcile recommendations critical to acoustics goals with uncooperative existing architecture or unwilling landmarks jurisdictions. Through case studies including the renovation of the Pittsburgh Symphony Orchestra’s Heinz Hall, the author will discuss the relationship between the architect and acoustics consultant in resolving the design issues inherent in vaudeville-era buildings.
The renovation of Madison’s Oscar Mayer Theater
Authors: Myers Joseph W. A. , Calamia Paul T.
Published: October 2013

Originally opened in 1928 as the Capitol Theatre, the Oscar Mayer Theatre in Madison, WI underwent substantial renovation in 1980 to support its current performance program. The theatre is now home to the Madison Symphony and the Madison Opera, and is used for ballet, touring shows, and popular concerts as well. As part of the ongoing Overture Project which will transform the theatre’s home, the Madison Civic Center, into the Overture Center, the Oscar Mayer Theatre will be further improved to support its new role. In this paper we will discuss the reasoning behind the upcoming renovation within the context of the Overture Project, we will describe the pending architectural modifications to the theatre, and we will discuss the intended changes in acoustics. Computer modeling results will also be presented for the existing and renovated conditions.
There’s nothing Majestic about burlesque
Authors: Jaffe Christopher , Glosemeyer Robin
Published: October 2013

Turn-of-the-century vaudeville and movie palaces initially seem to lend themselves to use as concert halls or multipurpose theatres. Basic features, including existing structure, appropriate location, and desirable seat count, already exist. But, upon further investigation, these palaces present both acoustical and theatrical limitations. Specifically, shallow stage depth, undersized stage wings, small orchestra pit, deep balcony overhangs, and low ceiling height in stage house and/or audience chamber are common deficiencies which must be addressed in successful conversions. Despite these challenges, converting an existing building is often a viable option for many communities. This paper will investigate several representative palaces which were modified to accommodate modern symphonic and theatrical productions between 1959 and the present.
Interactions between commercial fishing and walleye pollock aggregations
Authors: Stienessen Sarah , Wilson Chris D. , Hallowed Anne B.
Published: October 2013

Scientists with the Alaska Fisheries Science Center are conducting a multiyear field experiment off the eastern side of Kodiak Island in the Gulf of Alaska to determine whether commercial fishing activities significantly affect the distribution and abundance of walleye pollock (Theragra chalcogramma), an important prey species of endangered Steller sea lions (Eumetopias jubatus). In support of this activity, spatio–temporal patterns were described for pollock aggregations. Acoustic–trawl surveys were conducted in two adjacent submarine troughs in August 2001. One trough served as a control site where fishing was prohibited and the other as a treatment site where fishing was allowed. Software, which included patch recognition algorithms, was used to extract acoustic data and generate patch size and shape-related variables to analyze fish aggregations. Important patch related descriptors included skewness, kurtosis, length, height, and density. Estimates of patch fractal dimensions, which relate school perimeter to school area, were less for juvenile than for adult aggregations, indicating a more complex school shape for adults. Comparisons of other patch descriptors were made between troughs and in the presence and absence of the fishery to determine whether trends in pollock aggregation dynamics were a result of the fishery or of naturally occurring events.
Laboratory studies on scattering from bubbles located close to an air–water interface
Authors: Kapodistrias George , Dahl Peter H.
Published: October 2013

Results of laboratory experiments addressing the problem of scattering from bubbles located near an air–water interface are discussed. Well-controlled measurements of scattering from bubbles placed on a fine thread (itself nearly transparent to sound) were made. Depending on the experiment, bubble radii ranged from 500–1000 μm, while insonifying frequencies ranged from 110–130 kHz. The experiments quantified multiple scattering effects, by obtaining scattering data from two bubbles symmetrically arranged at a distance, d, from the acoustic beam axis, and the influence of both a flat and a roughened air–water interface, by measuring backscattering from a bubble located a distance, d, from the interface. Experiments were performed in the kd range 0.2–21, where k is the acoustic wave number. Modeling was achieved by utilizing the multiple scattering series and ray synthesis of the four scattering paths, created by the proximity of the bubbles to the interface, while addressing both bistatic and monostatic conditions. Experimental results are in good agreement with simulations, and they highlight three important features of the problem of acoustic observations from subsurface bubbles: (1) multiple propagation paths, (2) multiple scattering, and (3) rough surface effects. The relation of this work to interpretive models of field data is also discussed.
Characteristics of open ocean bubble plumes from a 3-month inverted echo-sounder record
Authors: Trevorrow Mark V.
Published: October 2013

A self-contained 200-kHz inverted echo-sounder was deployed at Ocean Station Papa (NE Pacific, 1400 km west of Vancouver Is.) from 1 March to 20 May, 1996. The instrument operated continuously, recording calibrated volume scattering from near-surface bubbles with 3-s and 30-cm resolution. Average surface wind and significant wave height during this period were 7.5 m/s and 2.8 m, respectively. The data show the frequent occurrence of bubbles organized into vertical, plume-like structures, presumably drawn downward within the convergence zones of Langmuir circulation. Significant bubble plume activity (defined as average plume depth >3 m) was observed during 61% of the operating period. Average bubble plume penetrations of up to 15 m were observed, with maximum penetrations up to 25 m. Within the plumes, the backscatter cross section exhibited an exponential decay with depth, with e-folding scale in the range 0.5 to 2.5 m. Some implications of this plume-like structure for high-frequency sonar performance will be discussed.
The IMUTUS interactive music tuition system
Authors: Tambouratzis George , Bakamidis Stelios , Dologlou Ioannis , Carayannis George , Dendrinos Markos
Published: October 2013

This presentation focuses on the IMUTUS project, which concerns the creation of an innovative method for training users on traditional musical instruments with no MIDI (Musical Instrument Digital Interface) output. The entities collaborating in IMUTUS are ILSP (coordinator), EXODUS, SYSTEMA, DSI, SMF, GRAME, and KTH. The IMUTUS effectiveness is enhanced via an advanced user interface incorporating multimedia techniques. Internet plays a pivotal role during training, the student receiving guidance over the net from a specially created teacher group. Interactiveness is emphasized via automatic-scoring tools, which provide fast yet accurate feedback to the user, while virtual reality methods assist the student in perfecting his technique. IMUTUS incorporates specialized recognition technology for the transformation of acoustic signals and music scores to MIDI format and incorporation in the training process. This process is enhanced by periodically enriching the score database, while customization to each user’s requirements is supported. This work is partially supported by European Community under the Information Society Technology (IST) RTD programme. The authors are solely responsible for the content of this communication. It does not represent the opinion of the European Community, and the European Community is not responsible for any use that might be made of data appearing therein.
Making the computer ‘‘listen’’ to music
Authors: Raphael Christopher S.
Published: October 2013

A computer system is discussed that provides real-time accompaniment to a live musician playing a non-improvisatory piece of music. Particular attention is devoted to the ‘‘listening’’ process, in which the computer must follow the soloist’s progress through the musical score by interpreting the sampled acoustic signal. The process is complicated by the significant variation and occasional errors from the live player during performance. A hidden Markov model is introduced providing a principled, trainable, and fast solution to the listening problem. The system is capable of assessing its own level of uncertainty about score position, as well as accommodating the sometimes strong signal component from the accompaniment instrument. A live demonstration will be provided.
Practical problems of sound isolation through interior glazing systems
Authors: Rosenberg Carl , Najolia Dorie
Published: October 2013

In a number of recent projects, designers have expanded the concepts of visual openness within a building to incorporate large expanses of glass. At the same time, building occupants expect the glass to achieve similar levels of acoustical isolation as stud constructed partitions. These situations include glass-enclosed conference rooms, clerestory lites between offices and corridors, curtain walls and strip windows, and vision panels that are adjacent to large atrium spaces. Variables impacting the acoustical performance include the size of the vision panels, the type of glass, the airspace between multiple layers, and the framing system that holds the glass. This paper presents a number of case studies and methodologies that address these issues.
A multidisciplinary approach of the problem of noise nuisance in urban environment
Authors: Rabah Derbal Cobis , Hamza Zeghlache
Published: October 2013

More often the problem of noise and sonic pollution, particularly in urban sites, is studied by different disciplines such as physics, the acoustics, the psychoacoustics, the medicine and others. It is independently of each other that these sciences are often approaching this subject. Some studies are carried out in laboratories taking noise as samples cut off their realistic context. Urban noise is studied as well by making an abstraction of the different contextual parameters by idealizing a rather complex sonic environment. The noise, according to this present approach, is suposed to react with surounding space, and it takes the form and the quality of the place by defining and requalifying it. It is found that the contextual aspects such as social, cultural or even symbolic dimensions modulate the listening conditions and the perception quality of the noise and even the living and the daily practice of the urban space. The multiparameter dimension study of the noise in an urban context is necessary to better work out the problem and to try to come up with some practical and efficient solutions. The little amount of studies based on such multidisciplinary approach, confort well our effort to go ahead with this methodological approach.
Quasiwavelet models of sound scattering by atmospheric turbulence
Authors: Goedecke George H. , Ostashev Vladimir E. , Wilson D. Keith , Auvermann Harry J.
Published: October 2013

Quasiwavelet (QW) representations of turbulence are composed of self-similar, localized, eddylike structures. The QW functions are not true wavelets, in that they do not form a mathematically complete basis or have zero mean. Nevertheless, they appear to be very useful for applications involving scattering and propagation of sound waves. In this paper, the QW formulation of Goedecke and Auvermann [J. Acoust. Soc. Am. 102, 759–771 (1997)] is outlined. The QW expressions for the spatial spectra and the corresponding sound scattering cross sections due to the velocity and temperature fluctuations of isotropic homogeneous turbulence are discussed. The spectra for different eddy structures are always similar to the von Karman spectra, and agree with the Kolmogorov spectra in the inertial range. Equations that yield the QW eddy functions in terms of the spectra are derived, and a QW function is found that yields the von Karman velocity spectrum exactly. Some results are presented from a numerical calculation of coherent scattering and temporal spectral broadening due to advecting turbulence modeled by QW eddies flowing with a wind. Future applications to modeling scattering by anisotropic and/or inhomogeneous turbulence are discussed. [Work supported by the ARO under Contract No. DAAD19-01-1-0640 (administered by W. Bach).]
Simulation of ultrasonic focus aberration and correction through human tissue
Authors: Tabei Makoto , Mast T. Douglas , Waag Robert C.
Published: October 2013

Ultrasonic focusing in two dimensions has been investigated by calculating the propagation of ultrasonic pulses through cross-sectional models of the human abdominal wall and breast. Propagation calculations used a full-wave k-space method that accounts for spatial variations in density, sound speed, and frequency-dependent absorption and includes perfectly matched layer absorbing boundary conditions. To obtain a distorted receive wavefront, propagation from a point source through the tissue path was computed. Receive focusing used an angular spectrum method. Transmit focusing was accomplished by propagating a pressure wavefront from a virtual array through the tissue path. As well as uncompensated focusing, focusing that employed time-shift compensation and time-shift compensation after backpropagation was investigated in both transmit and receive and time reversal was investigated for transmit focusing in addition. The results indicate, consistent with measurements, that the breast causes greater focus degradation than the abdominal wall. The investigated compensation methods corrected the receive focus better than the transmit focus. Time-shift compensation after backpropagation improved the focus from that obtained using time-shift compensation alone but the improvement was less in transmit focusing than in receive focusing. Transmit focusing by time reversal resulted in lower sidelobes but larger mainlobes than the other investigated transmit focus compensation methods.
State of the art in perceptual design of hearing aids
Authors: Edwards Brent W. , Van Tasell Dianne J.
Published: October 2013

Hearing aid capabilities have increased dramatically over the past six years, in large part due to the development of small, low-power digital signal processing chips suitable for hearing aid applications. As hearing aid signal processing capabilities increase, there will be new opportunities to apply perceptually based knowledge to technological development. Most hearing loss compensation techniques in today’s hearing aids are based on simple estimates of audibility and loudness. As our understanding of the psychoacoustical and physiological characteristics of sensorineural hearing loss improves, the result should be improved design of hearing aids and fitting methods. The state of the art in hearing aids will be reviewed, including form factors, user requirements, and technology that improves speech intelligibility, sound quality, and functionality. General areas of auditory perception that remain unaddressed by current hearing aid technology will be discussed.
It’s not your father’s recruitment: A new view of loudness growth in cochlear hearing loss
Authors: Florentine Mary , Buus So/ren
Published: October 2013

For over 60 years, one of the most commonly held beliefs in audiology has been that loudness grows more rapidly than normal near the elevated thresholds of listeners with cochlear hearing losses. This belief was based on the untested assumption that loudness at threshold is the same—often incorrectly assumed to be zero—in normal listeners and listeners with cochlear hearing losses. Recent tests of this assumption indicate that loudness at threshold is not zero in either group of listeners. These data also indicate that loudness near threshold grows at a normal rate in listeners with cochlear hearing losses. In other words, recruitment in the sense of a rapid growth of loudness near an elevated threshold does not occur. How, then, can intense sounds have approximately normal loudness in listeners with cochlear hearing losses? The explanation is that loudness at threshold is greater when the threshold is elevated by a cochlear loss than when it is normal. We call this softness imperception. Therefore, listeners with cochlear hearing losses not only have reduced dynamic range of audibility; they also have reduced dynamic range of loudness. Implications for the design of hearing aids will be discussed. [Work supported by NIH/NIDCD grant R01DC02241.]
Biologically inspired binaural hearing aid algorithms: Design principles and effectiveness
Authors: Feng Albert
Published: October 2013

Despite rapid advances in the sophistication of hearing aid technology and microelectronics, listening in noise remains problematic for people with hearing impairment. To solve this problem two algorithms were designed for use in binaural hearing aid systems. The signal processing strategies are based on principles in auditory physiology and psychophysics: (a) the location/extraction (L/E) binaural computational scheme determines the directions of source locations and cancels noise by applying a simple subtraction method over every frequency band; and (b) the frequency-domain minimum-variance (FMV) scheme extracts a target sound from a known direction amidst multiple interfering sound sources. Both algorithms were evaluated using standard metrics such as signal-to-noise-ratio gain and articulation index. Results were compared with those from conventional adaptive beam-forming algorithms. In free-field tests with multiple interfering sound sources our algorithms performed better than conventional algorithms. Preliminary intelligibility and speech reception results in multitalker environments showed gains for every listener with normal or impaired hearing when the signals were processed in real time with the FMV binaural hearing aid algorithm. [Work supported by NIH-NIDCD Grant No. R21DC04840 and the Beckman Institute.]
A cortical network underpinning the perceptual priority for rising intensity and auditory ‘‘looming.’’
Authors: Neuhoff John G. , Bilecen Deniz , Mustovic Henrietta , Schachinger Hartmut , Seifritz Erich , Scheffler Klaus , Di Salle Francesco
Published: October 2013

Relative motion between a sound source and a listener creates a change in acoustic intensity that can be used to anticipate the source’s approach. Humans have been shown to overestimate the intensity change of rising compared to falling intensity sounds and underestimate the time-to-contact of approaching sound sources. From an evolutionary perspective, this perceptual priority for looming sounds may represent an adaptive advantage that provides an increased margin of safety for responding to approaching auditory objects. Here, using functional magnetic resonance imaging, we show that the prioritization of rising contrasted with falling intensity sine-tones is grounded in a specific neural network. This network is predominantly composed of the superior temporal sulci, the middle temporal gyri, the right temporo-parietal junction, the motor and premotor cortices mainly on the right hemisphere, the left frontal operculum, and the left superior posterior cerebellar cortex. These regions are critical for the allocation of attention, the analysis of space, object recognition, and neurobehavioral preparation for action. Our results identify a widespread neural network underpinning the perceptual priority for looming sounds that can be used in translating sensory information into preparedness for adverse events and appropriate action. [Work supported by the Swiss and the American NSFs.]
SFOAE input/output functions elicited by slightly off-frequency suppressors
Authors: Schairer Kim S. , Keefe Douglas H.
Published: October 2013

Basilar membrane (BM) response growth is thought to be related to stimulus frequency otoacoustic emission (SFOAE) input/output (I/O) functions. SFOAEs comprise the response to sinusoidal excitation and provide an estimate of nonlinearity at a tonotopic place on the BM. SFOAE I/O functions at frequency f0 (across levels L0) are measured in the presence of suppressor fs (at level Ls) in subjects with normal hearing. fs is varied below, equal to, and above f0 in the range 0.93<fs/f0<1.07. For fixed suppressor level (Ls=80 dB SPL) and varying L0<Ls, the SFOAE I/O function is predicted to be closely related to the BM I/O function. Measured I/O functions are approximately invariant as fs is varied. Functions saturate at increasing levels with a decrease in some ears at the highest levels. An alternative paradigm in which Ls=L0 are covaried yields SFOAE I/O functions with local minima, but which may have a less direct connection to BM response growth. Based on primaries at fs and f0, SFOAE and DPOAE I/O functions differ. One implication is that SFOAE I/O functions may be used to relate BM nonlinear response growth to perceptual processes, including loudness growth and forward masking. [Work supported by NIH (R01 DC003784, T32 DC00013).]
The effect of ear canal pressure on pure-tone thresholds and middle-ear reflectance and conductance
Authors: Feeney M. Patrick , Hazelbaker Julie , Kelly J. Kip , Graves Amanda
Published: October 2013

The effect of ear canal pressure on pure-tone thresholds and wideband reflectance and conductance was examined in ten human participants. Pure-tone thresholds at 0.5, 1, 2, and 4 kHz were measured in random order using a 2AFC procedure at ambient pressure, and at ear canal pressures of positive and negative 200 daPa. These results were compared to wideband changes in reflectance and conductance using the same probe. Reflectance changes with pressure were in good agreement with recent data on reflectance tympanometry [R. H. Margolis, G. L. Saly, and D. H. Keefe, J. Acoust. Soc. Am. 106, 265–280 (1999)]. Both one-third-octave conductance measures and pure-tone thresholds were found to decrease by around 10 dB at 500 Hz and 1000 Hz compared to a 2- to 4-dB decrease at 2000 and 4000 Hz. These results are consistent with the change in middle-ear sound transmission with ear-canal pressure changes reported in a recent temporal-bone study in Norwegian cattle [M. Kringlebotn, J. Acoust. Soc. Am. 107, 1442–1450 (2000)]. The results of this study suggest that the change in middle-ear conductance may provide an objective measure of conductive hearing loss for the conditions in this experiment. [Work supported by NIH, RO3 DC04129.]
Vibrotactile suppression of tinnitus
Authors: Lenhardt Martin L.
Published: October 2013

At the Society’s 142nd meeting, the efficacy of high frequency bone conducted stimulation in suppressing tinnitus was presented. The hypothesized mechanism was the reprogramming of frequency tuning of auditory neurons in the central nervous system, secondarily to peripheral hearing loss. This mechanism is unlikely in cases of tinnitus in the presence of normal audiometric sensitivity. There is the possibility that hearing loss above 10 kHz can play a role in tinnitus, an association not thoroughly explored. Somatomotor stimulation influencing the quality of tinnitus has been reported, as have interconnections of the auditory and somatosensory systems. There would appear to be an evolutionary advantage of linking the sensorimotor organization of the external ear and the auditory function of the brainstem in sound localization. Thus, stimulation of the pinna and post auricular area may be a means of suppressing tinnitus. To that end a thin aluminum ceramic bimorph was constructed to fit on the inner surface of the pinna. When driven by low (<100 Hz) and high (>10 kHz) frequencies multiplied by MHz carriers, demodulation in the skin resulted in vibrotactile stimulation. Tactile stimulation was an adjunct to the high frequencies resulting in a multimodal suppressive effect in a small pilot study.