bluuebunny/crossref_metadata_2025 huggingface dataset:
The time course of cross-voice speaking rate normalization
Authors: Sawusch James R. , Newman Rochelle S.
Published: October 2013

Individuals vary their speaking rate, and listeners use the speaking rate of precursor sentences to adjust for these changes [Kidd, G. R., ‘‘Articulatory-rate context effects in phenone identification,’’ J. Exp. Biol.: Human Percep. Perfor. 15, 736–748 (1989)]. This speaking rate adjustment is not limited to speech from a single talker [Sawusch & Newman, ‘‘Perceptual normalization for speaking rate II: Effects of signal-discontinuities,’’ Percep. Psychophys. 62, 285–300 (2002)]; if one speaker stops talking and another begins, listeners will use rate information from the first talker to interpret speech in the second. We examined how long this cross-talker effect lasts. A male talker produced a carrier sentence at fast and slow speaking rates. A female talker then produced one or more subsequent words at a constant speaking rate. The final word in the female voice contained a duration-based contrast. We examined how much intervening speech in the female voice was necessary before the male voice’s speaking rate no longer influenced perception of the final target word. Results suggest that listeners require approximately 1 s of intervening speech or of silence before they stop using rate information from the male talker. This suggests that rate normalization may be based on a 1-s temporal window, with any speech information that occurs within 1 s in advance of a target phoneme influencing perception of that phoneme.
Contextual effects on the perception of duration
Authors: Kawahara Shigeto
Published: October 2013

English speakers perceive silent intervals with the same duration as longer when preceded by shorter vocalic intervals [Kluender et al. ‘‘Vowel length differences before voiced and voiceless consonants: An auditory explanation,’’ J. of Phonetics, 16, 153–169 (1988)]. This may be due to sequential contrast effects which exaggerate perceptual differences between neighboring segments. In Japanese, vowels are longer before geminates than before singletons [Han ‘‘Acoustic manifestations of more timing in Japanese,’’ J. Acoust. Soc. Am. 96, 73–82 (1994)]. Reflecting this correlation, Japanese speakers perceive silent intervals with the same duration as longer (as more ‘‘geminatelike’’) when preceded by longer vowels. The same pattern holds when non-speech analogs are used in place of speech stimuli: silent intervals are judged as longer when preceded by longer anharmonic buzzes. These results suggest that the perception of silent duration by Japanese speakers follows the pattern predicted from the acoustic properties of Japanese, but goes in the opposite direction from that predicted if durational contrast determines judgments, as it did for Kluender et al.’s (1988) listeners. This suggests that the segmental properties of the listener’s native language may influence nonspeech temporal judgments [cf. Moreton and Kusumoto ‘‘Native language determines parsing of nonlinguistic rythmic stimuli,’’ J. Acoust. Soc. Am. 102, 3204 (1997)]. English speakers will be tested on the same tasks to determine whether the perception of the temporal properties of nonspeech signals are generally determined by those of the listeners’ native language. [Work supported by NIH.]
Spectral auditory integration and virtual cues to place-of-articulation in stops
Authors: Fox Robert Allen , Smith Marc , Jacewicz Ewa
Published: October 2013

The spectral ‘‘center-of-gravity’’ (COG) hypothesis proposed by Chistovich and colleagues for static vowels suggested spectral auditory integration of two or more formant peaks within 3.5 barks. More recently, studies have examined dynamic COG effects by modifying the amplitude of steady-state formants over time in order to create ‘‘virtual’’ diphthongs. The present study looks at ‘‘virtual frequency change’’ cues to the alveolar-velar place distinction in stops [an extension of Fox and Smith, J. Acoust. Soc. Am. 117, 2423 (2005)]. First, [d≳-g≳] and [t≳-k≳] series were synthesized with a 50-ms F3 transition (with a voiced or voiceless source). Next, matched stimulus series were created with virtual F3 transitions by changing the amplitudes of either a pair of sine waves (in [d≳]-[g≳]) or a pair of narrow band noise signals (in [t≳]-[k≳]) whose frequencies did not change. Another matched [t≳]-[k≳] series was produced with both actual and virtual stops bursts (with no F3 transitions). Identification of these stimulus series by listeners with normal hearing show that a dynamically changing COG in the initial 50 ms of the token can effectively cue place distinctions in stops as do formant transitions and bursts; however, virtual spectral cues are less salient than formant or burst cues. [Work supported by NIDCD R01DC00679-01A1.]
Listener sensitivity to intensity variation in vowels in coarticulatory context
Authors: Jacewicz Ewa , Fox Robert Allen
Published: October 2013

There are two sources of acoustic variation in the relative formant amplitude of a vowel which are unrelated to emphasis-oriented changes in articulatory effort: (1) vowel-specific intrinsic formant frequency pattern, which differs across vowel categories, and (2) the vowel’s immediate phonetic (coarticulatory) context. Listener sensitivity to the latter source of variation is examined here to establish the extent to which distortion in the distribution of acoustic energy across the vowel spectrum affects the perception of vowel quality. Synthetic versions of CVC syllables in symmetrical consonant environments were created (using stops or fricatives differing in place-of-articulation and voicing). Formant frequencies and amplitude values were modeled after those found in a large corpus of CVC acoustic data [E. Jacewicz and R. A. Fox, J. Acoust. Soc. Am. 117, 2623 (2005)]. Initial amplitude values for the first four formants in synthetic CVCs reflected the intensity distribution (and spectral tilt) typical of a vowel coarticulated with these consonants. Subsequent stepwise adjustments to selected formant levels were made to create a range of permissible and not permissible formant amplitude variations in coarticulatory context. Preliminary results indicate that listeners are sensitive to amplitude variation across vowel spectrum as a function of a specific consonant environment. [Work supported by NIDCD R03 DC005560-01.]
The feature [sonorant] in auditory word identification
Authors: Moates Danny R. , Fox Russell , Lee Chao-Yang , Kellgreen Sara
Published: October 2013

The distinctive feature [sonorant] separates resonants and obstruents. Does this feature mark a difference in the perceptual processing of these consonants? Marks et al. [Marks et al., ‘‘Word reconstruction and consonant features in English and Spanish,’’ Linguistics, 40, 421–438 (2002)] showed such an influence in a word reconstruction task. Target words containing an obstruent segment were harder to reconstruct when the obstruent was replaced by a resonant (mismatch condition) than when it was replaced by another obstruent (match condition). Resonant target words showed no such effect. The present study extended Marks et al. (2002) using an auditory priming lexical decision task. Participants heard a nonword prime 50 msec prior to another word or nonword target. The prime differed from the target item in just one consonant which either matched the target segment on the feature [sonorant] or mismatched it. Listeners judged whether the target item was a real word or nonword. The obstruents were divided into fricatives and stops, which were compared with resonants in both cases. Latencies for the fricative target words were significantly slower in the mismatch condition than in the match condition but not for stops. The present study validates the effect found by Marks et al. (2002) with an online task but shows that it is limited to fricatives.
Is nonmodal phonation at the end of utterances a cue for speaker recognition by humans?
Authors: Böhm Tamás
Published: October 2013

Many studies of nonmodal phonation report a high rate of interspeaker variation. These experiments also showed that the rate and type of glottalization are characteristic to the speaker. Still, it remains an open question whether human listeners use this information as a cue for recognizing familiar voices. A listening experiment was conducted to investigate this issue, concentrating on utterance-final irregularities. Short utterances produced by four speakers were copy synthesized with the Klatt synthesizer in two conditions: with modal and with nonmodal voice quality in their final regions. Because two of the speakers were reliable ‘‘glottalizers’’ and two seldom ‘‘glottalized,’’ one of the two synthesis conditions reflected the speaker’s usual utterance-final voice quality and the other did not. These pairs of stimuli were used to test the effect of the presence or absence of utterance-final nonmodal phonation on the identification of voices familiar to the subjects. When asked to choose which one of the two variants is the speaker’s voice, listeners preferred the stimuli with the speaker’s typical voice quality at the end. Results will be discussed in comparison with a well-known perceptual cue on speaker identity, the F0 contour. [Work supported by a Fulbright Fellowship.]
Nonspeech sounds prime acoustically similar words
Authors: Wallace Andrew B. , Blumstein Sheila E.
Published: October 2013

Recent experiments show that nonspeech sounds can shift the perceptual boundaries between the phonetic categories of speech [e.g., Holt, Psych. Sci. 16, 305–312 (2005)]. The current experiment investigates whether nonspeech sounds will also affect the speed of access to words. Subjects received prime-target pairs in which the target was one of the three words deed, dad, and dude, preceded either by tone complexes which matched the frequencies of the first two formants of each target or by a neutral tone midway between the ranges of F1 and F2. Every prime was paired with every target in a word-identification task. Results demonstrated a priming effect of the nonspeech sounds on acoustically similar speech. Significant differences were found in reaction time, with subjects responding faster when targets were preceded by matched primes than when they were preceded by mismatched or neutral primes. These results support a role for general auditory processes in speech perception and lexical access. [Work supported by NIDCD.]
Acquisition of phonological variants in an artificial lexicon: The role of variant frequency
Authors: Pinnow Eleni , Connine Cynthia
Published: October 2013

A series of experiments examined the role of exposure frequency in the acquisition of phonological variants (schwa deletion) using an artificial lexicon. During training participants heard either two- (experiment 1) or three- (experiment 2) syllable stimuli produced in both schwa-present and schwa-deleted forms. Frequency of the schwa-present or deleted form was manipulated with high-frequency forms presented eight times and low-frequency forms presented one time. An orthographic version of the stimuli was presented during training as feedback. A lexical decision task performed after training showed a reaction time advantage for the high-frequency variant form for two syllable stimuli. No frequency effect was found for three-syllable stimuli. Experiment 3 presented the three-syllable stimuli across 2 days of training. On each day of training, high-frequency variants were heard 15 times and low-frequency variants were heard once. A reaction time advantage for high-frequency forms was found on day 1 that was reduced on day 2; both days revealed more accurate identification for high-frequency forms. Results will be discussed in terms of previous findings for recognition of schwa deletion variants and possible mechanisms for the lexicalization of phonological variants.
Estimating sound-speed perturbations with full-wave environmental sensitivity kernels
Authors: Sarkar Jit , Cornuelle Bruce , Kuperman W. A. , Roux Philippe
Published: October 2013

The acoustic field at a receiving array due to an individual source is affected by changes in the intervening sound-speed profile. This sensitivity has been calculated for broadband signals centered at 3500 Hz and 4-km range using a Born approximation to perturbations in the wave equation. These sensitivity kernels have been verified against range-dependent propagation model results. The range of validity for the linear approximation has been explored as part of these tests. Least-squares methods can be used to estimate the environmental changes from the measured differences in the received signal(s) and prior knowledge of the statistics of the variability. Sample estimates and error statistics, including resolution, will be presented.
Time-reversal-based noniterative exact inverse scattering of multiply scattering point targets
Authors: Marengo Edwin A. , Gruber Fred K.
Published: October 2013

This research investigates within exact scattering theory time-reversal-based approaches for the inverse scattering in a known but rather arbitrary embedding background medium of an unknown total scatterer composed of a number of dominant subwavelength (pointlike) scattering centers (targets) that are located at unknown positions and have unknown scattering strengths or reflectivities (characterizing a scattering potential). Of particular interest is the formulation of a new inverse scattering method (comprising both inverse support and inverse medium aspects) based on a two-step procedure. The first step (support inversion) consists of the localization of the targets via signal subspace approaches such as time-reversal MUSIC and a new, high-dimensional signal subspace method that is shown to be linked to maximum likelihood estimation. The second step (nonlinear medium inversion) is a new, noniterative scattering strength inversion formula for the reconstruction of the target scattering strengths or reflectivities. Importantly, the derived method is valid even in the presence of multiple scattering between the targets, which is remarkable since the mapping from the reflectivities to the field data is then nonlinear. The theory is illustrated with several computer examples. [Work supported by AFOSR under Grant FA9550-06-01-0013.]
Application of the acoustic adjoint equation to normal mode propagation in a waveguide
Authors: Thode Aaron
Published: October 2013

This presentation reviews work that applied the acoustic adjoint equation to situations where the propagating acoustic field in an oceanic waveguide can be expressed as a sum of normal modes. The acoustic adjoint equation [Norton, ‘‘Inertive inverse scattering algorithms: Methods of Computing Fréchet derivatives,’’ J. Acoust. Soc. Am. 106, 2653–2660 (1999)], Hurksy et al., ‘‘Adjoint modeling for acoustic inversion, J. Acoust. Soc. Am. 115, 607–619 (2004)] which has a form very similar to the Born approximation, provides an expression for multiple-order derivatives of the received acoustic field with respect to an infinitesimal change in an environmental parameter, which in turn can be used to derive the gradient of an error or ambiguity function without resorting to numerical derivatives. For the situation of a range-independent environment, closed-form analytical expressions can be obtained for pressure derivatives with respect to range-independent perturbations in refractive index and density in the water column and sediment, as well as derivatives with respect to frequency. Expressions have also been derived for derivatives with respect to three-dimensional refractive index perturbations, which permits the evaluation of certain horizontal refraction effects of environmental perturbations without resorting to three-dimensional propagation codes. Current work on obtaining derivatives with respect to perturbations in waveguide boundary conditions (bottom bathymetry or sea surface shape) will also be discussed. [Work supported by the ONR Uncertainty Program.]
The perforated panel resonator with flexible tube bundle
Authors: Lu Yadong , Tang Huide , Tian Jing , Li Hongqi
Published: October 2013

This paper deals with a new type of resonant absorber, i.e., the perforated panel resonator with flexible tube bundle. The acoustical properties of the new sound-absorbing structure are measured and analyzed. From the measurement results and theoretical analysis, the following conclusions can be made. (1) The incorporation of the tube bundle into the resonator absorber is helpful to improve its low-frequency sound absorption. Because the artificial prolongation of air column length inside a hole neck will increase acoustic mass, resonance frequency can be shifted to a lower frequency correspondingly. (2) Compared with traditional perforated panel absorber, the acoustic resistance of the sound-absorbing structure can be increased considerably due to the incorporation of the flexible tube bundle into the resonator absorber, so that higher sound absorption coefficients can be expected. (3) By reasonably utilizing the tube and cavity coupling resonances, the frequency bandwidth of sound absorption of the perforated panel resonator with flexible tube bundle can be expected to broaden. The perforated panel resonator with flexible tube bundle may be used in some cases with strict space limitations and with expectations of increasing the sound absorption at lower frequencies.
Measurements and calculations on microperforated sound absorbers
Authors: Nocke Christian , Hilge Catja
Published: October 2013

The sound absorption of microperforated absorbers can be predicted with a high degree of precision. Microperforated absorbers have been made of various materials in many applications. In recent years different manufacturers have offered a variety of microperforated materials. This paper reviews the first microperforated sound absorbers made of foils introduced some 10 years ago. This paper compares measured sound absorption coefficients of the microperforated foil absorber with the theoretical simulations. Theoretical approaches towards layered sound absorbers with microperforated components will be suggested. Finally, future developments in the field of microperforation in room acoustics will be discussed.
Practical applications of transparent microperforated and microslit absorbers
Authors: D’Antonio Peter
Published: October 2013

It has been over 100 years since absorption was introduced as a quantifiable tool in architectural acoustics. Currently, the architectural acoustic community faces ever-changing challenges that require innovative approaches to absorb sound. Due to the importance of day lighting in sustainable design certification, such as LEED, and the concern about particulate material from fiberglass and mineral wool, there has developed a need for transparent and translucent absorptive materials that do not require the use of fibrous porous absorption. One approach has been the development of microperforated and microslit foils, sheets, and panels with submillimeter openings that provide significant viscous losses when spaced from a boundary. This presentation will review the design theory and absorption mechanisms, describe the various commercial materials now available, and illustrate current projects and potential applications.
An ideal absorbing materials in architectural acoustics: Micro-perforated panel absorbers
Authors: Liu Ke , Tian Jing , Jiao Fenglei , Li Xiaodong
Published: October 2013

Micro-perforated panel absorbers (MPA) can be used to reduce or better the noise in architecture. This absorber was developed in the 1960s, when a robust sound absorber was needed for severe environments, without porous materials. It has a simple structure and the absorption properties can be exactly calculated. The panel can be made of any materials from cardboard to plastic to plywood to sheet metal, for different purposes. Theoretical and experimental investigations on MPA are reviewed in this paper. By reviewing recent research work, this paper reveals a relationship between the maximum absorption coefficient and the limit of the absorption frequency bandwidth. It has been demonstrated that the absorption frequency bandwidth can be extended up to 3 or 4 octaves as the diameters of the micro-holes decrease. This has become possible with the development of the technologies for manufacturing MPA, such as laser drilling, powder metallurgy, welded meshing, and electro-etching to form micrometer-order holes. In this paper, absorption characteristics of such absorbers in random fields are presented and discussed both theoretically and experimentally. This review shows that the MPA has potentials to be an ideal absorbing materials in architectural acoustics in the 21st century.
Loudspeakers and listening rooms, Part I: The physical sound field
Authors: Toole Floyd E.
Published: October 2013

In contrast to the considerable effort invested in concert halls and auditoria for live performances, the design of domestic listening spaces and recording control rooms has evolved with little guidance from science. In this presentation, an examination of the physical aspects of sound in small rooms reveals that traditional acoustical measures, such as reverberation time, are of little value. The reasons have to do with the differentiating facts that loudspeakers are relatively directional, and the rooms are relatively small, relatively absorptive, and furnished with large sound-scattering objects. As a result, at frequencies above about 300 Hz, listeners are in a transitional sound field consisting of direct and early reflected sounds, and a much-diminished late-reflected sound field. Loudspeaker directional properties and the acoustical behavior of the surfaces involved with early reflections are the factors relevant to perceptions. At low frequencies, room modes dominate. Notions of optimum room-dimensional ratios are of value only when loudspeaker and listener locations are known. New solutions, employing multiple subwoofers in configurations that result in selective modal excitation, can yield excellent low-frequency sound for several listeners in both rectangular and nonrectangular rooms.
Surround monitoring challenges in acoustically small spaces
Authors: Berger Russ
Published: October 2013

Critical listening spaces used to monitor and produce surround sound product are usually required to provide an accurate listening environment for more than one listener. Decisions are made not only about source material content and veracity, but also about how best to process and present the multichannel surround event in a manner that translates the aural experience to a wider audience. Oftentimes this audience is listening in other, less-than-ideal acoustical environments. Practical criteria and solutions are presented here to help overcome the limitations of monitoring surround sound in acoustically small spaces, as well as a discussion regarding first reflection control through the application of space coupling diffusors.
Inverse scattering bioacoustics: Under the influence of Fred Lizzi
Authors: Insana Michael
Published: October 2013

A long-standing problem in bioacoustics is the assessment of soft tissue health from an analysis of backscattered echo signals—an inverse problem known as tissue characterization. Pathologists view magnified images of tissue morphology to assess tissue function and grade disease states. Ultrasonic tissue characterization methods are attempts to accomplish the same assessment noninvasively. The goal is not to image the microstructure directly, but rather to diagnose tissues from measurements of statistical moments of the echo spectrum properly normalized to isolate tissue features. Fred Lizzi pioneered this approach and provided an analytical framework used widely. It will be shown how his analysis evolved from classic studies in atmospheric physics to give a clear picture of what is becoming a material science approach to diagnosis using ultrasonics. Dr. Lizzi’s methods have been used by many to diagnosis and monitor treatment of cancer from a very small feature space. Others have expanded these methods to discover the sources of ultrasonic scattering in tissue systems and observe dynamic features of physiology. Coupled with elasticity imaging, inverse scattering approaches provide in vivo tools for studying the influence of molecular signaling on tissue structure during the progression of cancer and atherosclerotic diseases.
Discovery of the ring resonant frequency for noninvasively estimating local elastic modulus of arterial wall
Authors: Zhang Xiaoming , Greenleaf James F.
Published: October 2013

Increased stiffness of arteries has recently gained acceptance as an independent risk factor for cardiovascular and many other diseases. Pulse wave velocity (PWV) is widely used for estimating the stiffness of an artery. However, PWV is an average indicator of artery stiffness between the two measuring points, and therefore does not identify local stiffness. In addition, the thickness of the artery is needed to calculate artery elastic modulus. We have discovered that there is a series of ring resonant frequency in arteries that can be used for estimating local arterial elastic modulus. These resonances can be excited by remote palpation with selectively placed ultrasonic radiation force. Experiments on ring resonance were carried out on an excised artery as well as a rubber tube. This family of resonances occurs at relatively low frequency, around 356 Hz for the artery. Estimation of the Young’s modulus is 135 kPa by the ring frequency 356 Hz without the requirement of the arterial thickness. The estimated modulus is very consistent with the other ring resonant frequency measurements. The theory of ring resonance is developed and verified with the rubber tube experiments.
Violin octet scaling: The practical side
Authors: Spear Robert J.
Published: October 2013

Schelleng’ scaling procedure attempts to place A0 at ∼1.5x and the first corpus bending modes at ∼2.25x, the lowest string pitch. An experiment to move A0 into this position (also influencing A1), using the Stradivari model G as reference, helps produce the characteristic sound of the violin in cases where B1 placement was not optimum. The experiments suggest that the model G worked well for the signature modes of a classic violin, successfully scaling at least four instruments (mezzo, alto, tenor, and a small baritone), but that instruments at the extremes—treble and soprano at one end of the scale and both basses at the other—will likely require a refinement of the model G and placement of modes different from those of the reference violin and from each other. Scaling for acoustics and scaling for size are not the entire answer since ergonomic demands are often at odds with both. Geometrical scaling is essential for aesthetics and the uniform shape of the instruments, but it does not hold up well for rib heights, rib thicknesses, and cavity air volumes. This experiment suggests that significant latitude of air and wood mode placement exists within which good results can occur.
Scaling of reed instruments: The case of the saxophone family
Authors: Dalmont Jean-Pierre , Gazengel Bruno , Kergomard Jean
Published: October 2013

The saxophone family is one of the most extended musical instruments family from the contrabass to the sopranino. The timbre of these instruments is unmistakable: there is a small probability to mix up a saxophone with an oboe or a bassoon. Now a question is: What is so specific in the timbre of the saxophones? Notes of various saxophones are recorded, analyzed, and compared with those of other instruments such as the oboe. More specifically, the transient duration and the formant frequency are determined. Then, the question which physical parameters influence the transient duration and the formants is studied. The influence of geometrical parameters such as the cone angle and the length of the cone truncation will be discussed.
Soundscape measurement in an urban area
Authors: Schulte-Fortkamp Brigitte
Published: October 2013

This presentation is about a demonstration of soundscape evaluation of an urban area with respect to measurements and interviews. In a recent field study in Berlin, indoor and outdoor measurements were carried out in a residential area in which inhabitants were complaining about noise and vibration due to heavy traffic over day and until late at night. In this context, the question of the appropriate positioning and calibration of the measuring devices in indoor and outdoor measurements was very important. The measurement procedure as well as procedures which determine how to conduct interviews will be described and discussed by taking into account how to make progress in measuring soundscapes. Moreover, the focus will be on necessary contributions to a measurement which usually deals with dB(A) and lacks psychoacoustical measuring factors.
Traditional measurement methods for characterizing soundscapes
Authors: Brooks Bennett M.
Published: October 2013

A soundscape may be defined as the acoustical environment at an outdoor location. The physical sound occurring at the location may be from natural or man-made sources, or both. Perception of the soundscape can provide comfort and needed information to the listener, or may be annoying. The existing soundscape may be subject to design modifications for a variety of purposes. The goal could be to protect or enhance some sounds, such as the laughter of children and chirps of wildlife in a park, or the speech in an amphitheater. Conversely, the goal may be to reduce or eliminate unwanted sounds, such as transportation or industrial noise. Engineering and aesthetic soundscape design may proceed based on the characterization of the acoustical environment. The soundscape can be characterized by a number of measured acoustical parameters. Traditionally, these parameters have included A-weighted sound level, spectral content, sound-level statistics and averages, alone or in combination, and sound-level time history. Listener perceptions and nonacoustical parameters, such as time of day or year, community relevance, and listener attitudes may also be applied to soundscape characterization and design. Examples of several different soundscapes, their characterizations, and proposed design modifications will be discussed.
Territoriality and sound delimitation of urban space
Authors: Derbal Cobis Rabah , Zeghlache Hamza
Published: October 2013

Whether the urban space is physically materialized or abstracted it is often delimited by perceptive and sociocultural relations. In the present methodological approach, urban space is supposed to be modulated in its delimitation by qualitative and quantitative criteria of its noise environments. This introduces clearly the concept of sound delimitation and of noise environment territoriality. One can notice that sound phenomena are very present and are marking the social practice and behavior in urban spaces and give them a very specific sonority. Just like the visual or sociocultural aspects, the sonority of the public place or the urban noise environment marks their territoriality and spatiality by creating a whole sound environment characteristic of dimensions and forms of the space. The urban noise can thus be an element of marking, connecting, breaking, or structuring spaces which are morphologically and visually irreconcilable. One can speak about sound delimitation of urban space beyond its morphological limits. That delimitation can well articulate the social practice of urban space to its dimensional aspect. The study put out the concept of acoustic space in opposition to morphological space which it is inevitably necessary to reconcile in order to better practice the urban space.
Recomposition of the urban sonic environment
Authors: Derbal Cobis Rabah , Zeghlache Hamza
Published: October 2013

Acoustics-related considerations are taken to be secondary in most of the design processes of public places. This fact always leads to an unpredictable noise and urban sonic environment around such spaces. Studies have shown the importance of the noise and the quality of the urban sonic environment in the social practice and in the daily use of public places. These places strongly mark the degree of dynamism of the urban structure of our cities and are often subjected to a very complex situation of noise nuisance. The main aim of the study is to introduce a source of noise that might have an effect of recomposing positively a deficient sonic environment of such places. This approach is also to try to restore the sound in its utilitarian and practical function and to articulate the perception of the spatial and temporal aspects of the public place. It could be an element of transforming urban noise into a peaceful soundscape. A fountain turned out to be that sonic source and it can be easily introduced in the urban environment around the public places. It is a sort of urban instrumentation used in order to recompose or requalify an urban noise environment.
Ground-to-ground sound propagation in cloud-topped boundary layers over land
Authors: Waxler Roger , Talmadge Carrick L. , Gilbert Kenneth E.
Published: October 2013

Sound propagation under cloud-covered skies is discussed. Observations are presented which suggest that under dense cloud cover ground-to-ground sound propagation is dominated by a single mode similar to the classic surface mode for outdoor sound propagation. A theory of sound propagation under cloud covered skies has been developed. It is shown that there is a sound duct centered in the cloud layer. Of the modes in this duct a small number (typically one or two for frequencies up to a few hundred Hz) of ground interacting modes, modes whose amplitudes on the ground are significant, are identified. Over land, one of the ground-interacting modes can be identified with the surface mode.
A continuously cascaded conformal mapping for sound propagation over irregular terrain
Authors: Parakkal Santosh , Gilbert Kenneth E. , Xiao Di
Published: October 2013

The so-called earth flattening conformal mapping is a common approach for radio wave propagation over a spherical earth. Applied sequentially (cascaded), it can be used to represent a series of hills, each of which has a constant radius of curvature. We show that the mapping can also be applied to general terrain that has a continuously varying radius of curvature. In the mapped coordinates, the terrain is flat, but the mapped sound speed is c/√J, where c is the actual sound speed and J is the Jacobian of the transformation. The continuously cascaded mapping method is outlined, and numerical comparisons are made with a well-tested nonconformal mapping method.
A multicompartmental cochlear model with piezoelectric outer hair cells
Authors: Lu Shan , Mountain David C. , Hubbard Allyn E.
Published: October 2013

A hydromechanical, multicompartment model of the cochlea, which employed a phenomenological outer hair cell (OHC) cell-body electro-motility, was able to mimic the physiologically measured response of the basilar membrane (BM) [A. E. Hubbard et al., ‘‘Time-domain responses from a nonlinear sandwich model of the cochlea,’’ in Biophysics of the Cochlea: From Molecule to Model, edited by A. W. Gummer (World Scientific, Singapore)]. An improved model that included OHC electro-anatomic parameters and scalar electrical parameters was able to mimic cochlear microphonic (CM) data at low frequencies, but not BM responses at high-frequencies, because the OHC membrane capacitance severely reduces OHC transmembrane potential. To improve the high-frequency performance, we used a piezoelectrical model of the OHC. Consequently, the mechanical loading of the OHC translates bidirectionally with the electrical impedance of the OHC. The new model compares well with physiological data from Gerbil at 40 dB SPL [T. Ren and A. Nuttall, ‘‘Basilar membrane vibration in the basal turn of the sensitive gerbil cochlea,’’ Hearing Res. 151, 48–60 (2001)]. [This work was supported by NIH.]
Searching the whole genome to identify genes contributing to noise resistance in inbred mouse strains
Authors: Tempel Bruce L. , Street Valerie A. , Kujawa Sharon G.
Published: October 2013

Toward understanding the biological basis of noise resistance and eventual formulation of therapies to prevent noise-induced hearing loss (NIHL), the identification of genes contributing to noise resistance is a critic step. To simplify our whole genome search, we chose to study isogenic, inbred strains of mice. When exposed to noise (octave band 8–16 kHz, 103 dB SPL, 2 h) the good-hearing CBA/CaJ strain showed ∼50-dB thresho shift measured 2 weeks postexposure; in contrast 129S6 mice showed ∼10-dB shift. Through directed breeding schemes we showed that the noise resistance trait in 129S6 is recessive to CBA. Screening N2 backcross animal for noise resistance suggested a multigenic pattern of inheritance with 4 to 6 genes contributing. Quantitative trait locus (QTL) mapping of 234 N2 animals reveals 4 different chromosomal regions that contribute significantly to the resistance. Congenic strains carrying only one or two of these QTL regions are currently being bred, with the expectation that these strains will show partial noise resistance. Further studies including ancestral haplotype analysis and differential gene expression array analysis will allow us to narrow the contributing genomic regions and identify specific genes contributing to noise resistance. [Work supported by NIH.]
Is some information perceptually primary?
Authors: Kingston John , Diehl Randy
Published: October 2013

The existence of multiple correlates for phonological contrasts invites the hypothesis that one is perceptually primary and that the others enhance the primary correlate. (Stevens Acoustics Phonetics, MIT Press, 1998) particularly emphasizes the perceptual primacy of acoustic landmarks. Our experiments manipulate the quality or variability of the acoustic information in a landmark versus a neighboring nonlandmark to determine whether listeners rely on the landmark or the most reliable information source. Stimuli are synthetic stop-vowel sequences where the stop’s place of articulation is conveyed by its burst spectrum—landmark—and the formant transitions to the following vowel—nonlandmark. Parameter values in the two intervals fall on opposite sides of, but close to, a category boundary and convey competing place information. Information is degraded in one interval by broadening the bandwidths of the spectral peaks or formants, or it is made more variable by varying the peaks frequencies or the following vowel quality. If landmark information is perceptually primary, then degrading or varying that information should shift place judgments more toward the category conveyed by the nonlandmark interval than vice versa. However, if the quality or consistency of the information matters, then manipulating either interval should produce comparable shifts in place judgments. [Work supported by NIH.]
Enhancing the durational enhancement of a tone contrast
Authors: Gussenhoven Carlos
Published: October 2013

In central and low Franconian dialects of German and Dutch, a lexical tone contrast (accent 1 versus accent 2) exists that is enhanced by a durational difference. While pitch differences vary according to intonation and sentence context, syllables are generally shorter when they have accent 1. In addition, vowels in those syllables are more open, in comparison with vowels in syllables with accent 2. Perception experiments with natural vowels whose acoustic durations had been manipulated revealed that high vowels are perceived as longer by Dutch listeners than midvowels when acoustic durations are equal. It is argued that vowel raising of longer vowels, or vowel lowering of shorter vowels, serves as an enhancement of duration differences. The tendency is widely attested to in the vowel systems of Germanic languages, where short (‘‘lax’’) vowels are opener than long (‘‘tense’’) vowels, and vowel height is used to enhance a phonological duration contrast. In the dialects investigated, vowel height is manipulated to enhance a durational difference, which in turn enhances a tonal contrast. The explanation of the durational illusion of vowel height is the listener’s compensation of the universally shorter duration of higher vowels.
The role of acoustic familiarity in the perception of L2 contrasts
Authors: Vallabha Gautam K.
Published: October 2013

Difficulties in L2 perception are often attributed to assimilation of the L2 contrast by an L1 category or to interference by L1-relevant acoustic cues [Iverson et al. ‘‘Phonetic training with acoustic cue manipulations: A comparison of methods for teaching English /r/-/p/ to Japanese adults,’’ J. Acoust. Soc. Am. 118, 3267–3278 (2005)]. These approaches assume that cue weights change slowly over developmental time and are relatively constant for short time scales. An alternative view is that cue extraction and weighting are dependent on the overall familiarity of the current sound, with more familiar sounds facilitating more robust cue extraction and better discrimination [e.g., McFadden and Callaway, JEP:HPP, 25, 543 (1999)]. This hypothesis was tested with the perception of English sounds by Japanese adults. The Japanese listeners are insensitive to a cue (F3 onset) when it is part of (putatively unfamiliar) English /ra/ and /la/ sounds. The effect of familiarity was evaluated by fixing the range of F3 onsets and manipulating the rest of each sound in two ways: (1) by changing the transition durations to be more suitable for a moraic rhythm and (2) by changing the F1, F2, and amplitude trajectory to be more like the Japanese /r/. Discrimination improved markedly with the latter manipulation, suggesting that cue sensitivity is conditioned upon overall acoustic familiarity. [Work supported by NIH MH64445.]
Adaptive filtering algoritm for time-variant symmetric α-stable signals
Authors: Leth Elmkjær Torsten
Published: October 2013

The existence of non-Gaussian distributed impulsive signals has roots in the generalized central theorem. In particular, the symmetric α-stable (SαS) distribution has been used to model heavy-tailed phenomena encountered in communication, underwater acoustics, and radar. Most previous work in the field of active control uses some of the numerous variants of LMS algorithm extended with the filtered-x scheme. The stable processes, however, do not possess finite second-order moments. Hence, the filter should be based on lp-norm optimization. The p-norm enters the expression for the tap-weight update [p(t)<α̂(t)]. In this work a new adaptive algorithm referred to as regularized normalized least−mean p−norm algorithm [εNLMp(t)] is presented. A running estimate α̂(t) of the time−variant characteristic exponent of the stable signal is obtained. Hence, by adaptively tuning the filter p−norm a more optimal filter performance results in time−variant situations. Simulation analysis provides insight into the adaptive filter performance for various time−invariant and time−variant noisy signals for 1<α(t)≤2. Moreover, it is demonstrated that the filter performance is only marginally degraded compared with LMS algorithm for ordinary time−invariant Gaussian noise signals. Performance of an active control system exposed to time−varying heavy−tailed noise signals is presented.
Data error covariance matrix for vertical array data in an ocean waveguide
Authors: Huang Chen-Fen , Gerstoft Peter , Hodgkiss William S.
Published: October 2013

Information about the data errors is essential for solving any inverse problem. The likelihood function plays a critical role in describing the data uncertainties in geoacoustic inversion. The choice of likelihood function depends on the statistics of the errors (the difference between observed and estimated fields). In all work to date, the likelihood function has been derived based on an assumption of Gaussian errors. Typically, the errors are assumed to be independent, identically distributed with equal variance (referred to as the error variance), and the error variance is estimated as part of the optimization. Recently, there has been interest in estimating a more full data error covariance matrix. To estimate a truly full data error covariance matrix, we adopt a maximum-likelihood approach based on ensemble averages using the observed errors over many inversions. The approach is illustrated using data obtained during the ASIAEX 2001 East China Sea experiment. The parameter uncertainties resulting from incorporating the full data error covariance matrix are compared with those obtained from the simplified data error covariance matrix characterized by the error variance alone.
Model-based blind deconvolution for noisy dynamic channels
Authors: Roan Michael , Gramann Mark
Published: October 2013

It is common in acoustics to measure a signal that has been degraded by propagation through an unknown, noisy channel prior to measurement. While only the degraded measured signal is available, the data of interest are the original signal and the channel parameters. Often, it is desirable to reverse the filtering process by application of an inverse filter to recover the original signal. When neither the input signal properties nor the channel properties are known, this process is known as blind deconvolution (BDC). Typically, BDC algorithms assume noiseless, stationary propagation channels and input sources. These assumptions are usually violated in practical applications (e.g., noisy multipath propagation environments with moving source and receiver). To model these effects, predictive techniques are applied to incorporate a priori information about the system into the existing blind processing framework. The original contributions of this work follow. First, a novel formulation of the extended Kalman filter (EKF) is proposed. This allows incorporation of a priori information into gradient-based blind processing algorithms. This formulation is then applied to the natural gradient (NG) BDC algorithm. Finally, results are presented that suggest significant improvement in signal recovery performance over the NG BDC algorithm for dynamic noisy channels.
The direct path impulse response in bubbly shallow water
Authors: Goodman Ralph , Caruthers Jerald , Ruppel Timothy
Published: October 2013

It is well known that bubbles, even in small numbers, cause significant changes in the acoustic properties of seawater. In shallow water bubbles enter the sea, either by wind-driven breaking waves or by surf-induced rip currents. Both have been observed in acoustic transmission experiments [IEEE, JOE 25, 507–515 (2000); J. Acoust. Soc. Am. 106, 617–625 (1999)]. Wind-driven bubbles were observed, during a period of high winds, to reduce sound speed from the undisturbed sea condition of 1520 m/s to about 1494 m/s. From bubble theory, it is estimated that the average bubble void fraction was approximately 2.6×10−6. For a distribution of bubble radius size, n(a), there will be a frequency-dependent sound speed and absorption related by the Kramers-Kronig relation, from which the change in a pulsed signal can be determined. By using a bubble size distribution of n(a)=Na×10−4, for radii between values of 200 and 10 microns and the void fraction estimated above, the dispersion and absorption are calculated. The impulse response can be calculated as a function of range. [Work supported by the Office of Naval Research.]