bluuebunny/crossref_metadata_2025 huggingface dataset:
Low-frequency ambient noise measurements in the South Fiji Basin
Authors: Marrett R. , Chapman N. R.
Published: October 2005

The effect of wind speed on low-frequency ambient noise has been measured in an experiment carried out in the South Fiji Basin. This site provides an ideal location for studying sea surface noise, owing to the low density of shipping in the basin. The data were recorded using a towed line array over an 11-day period in the winter storm season. The local wind speed was measured at the site, with an observed variation between 0–35 kn. The noise measurements in the frequency band 15–250 Hz were strongly correlated with the variations in the local wind speed. The relationship between wind speed v and noise level N is expressed by N = B + 20 n log(v). The constants B and n have been determined by fitting this model to the data at frequencies from 15–250 Hz. At 250 Hz, the values are 22 dB and 1.5 for B and n, respectively. This value for n is higher than the results of similar experiments carried out in the northern hemisphere oceans where the influence of shipping is greater.
Ambient noise measurements from 100 Hz to 80 kHz
Authors: McConnell Steven O. , Schilt Michael P.
Published: October 2005

Measurements covering a broad frequency range from 100 Hz to 80 kHz have been made in Behm Canal, Alaska. This site represents a fairly deep embayment (400 m) with a very soft bottom (porosity of about 0.8) and, hence, the noise detected at the hydrophones is affected negligibly by multipath contributions except possibly at the very lowest frequencies. Data were gathered over a wide range of wind speeds (0–15 m/s) and during periods of rain and snow. Several unique and noteworthy results were obtained. Foremost was the observation that the wind-generated noise level measured during the winter was 5 dB lower than during the summer for the same wind speeds and air-sea temperature differences (air temperature about the same or colder than the sea surface). The summer data agree well with the most recent published measurements and are approximately 2 dB higher than the standard Knudsen/Wenz reference spectra. Most reported wind noise measurements are restricted to frequencies less than 20 kHz. Those that go beyond this frequency display a noticeable hump above the usual −17 dB/decade power-law slope, and the Behm Canal measurements show that this hump continues to 80 kHz, where the spectrum rejoins the extension of the canonical power-law slope. [Work supported by Battelle Pacific Northwest Laboratories and NORDA.]
Bispectrum of ship-radiated noise
Authors: Hinich Melvin J. , Marandino Davide , Sullivan Edmund J.
Published: October 2005

The bispectrum of ship-radiated noise is estimated. The noise was received on a towed array being towed by the same ship that served as the noise source. The array was beamformed such that the forward end-fire beam pointed toward the towing platform and the broadside beam sampled the environment without the radiated noise of the ship. The results show that there exist frequency-dependent bispectral components in the ship's radiated noise, whereas the ambient noise does not contain any significant bispectral components. Since the existence of a nonzero frequency-dependent bispectrum indicates the existence of nonlinear components in the noise-generating mechanism, it is concluded that the radiated noise of the towing platform contains such nonlinear mechanisms. Also, the bispectrum has the capability of indicating the presence of transients in the data. This property will be briefly discussed, and it will be shown that the data under analysis contain transients. Based on these results, it is concluded that the bispectrum could be used to indicate the existence of transient and nonlinear sources that would normally be hidden in the background noise when the usual spectral estimation procedures are applied.
Where has all the power gone?
Authors: Titze Ingo R.
Published: October 2005

An upper limit on aerodynamic (pulmonary) power produced in shouting and singing is on the order of 1 W. The radiated acoustic power is less than 10% of this aerodynamic power, and in conversational speech can be as little as 0.1%. Major dissipative elements are the vocal folds, the lungs, the soft walls of the vocal tract, and turbulence in the airstream emerging from the glottis. Some analytical calculations and some computer simulation will focus on the distribution of losses along the vocal tract. In particular, difference between the singing mode and the speaking mode will be highlighted. Speculations will be made on the techniques by which a singer can optimize the system for maximum power output. [Work supported by NIH.]
Laryngeal adduction related to characteristics of the flow glottogram
Authors: Scherer Ronald C. , Sundberg Johan , Titze Ingo R.
Published: October 2005

Laryngeal adduction is a basic phonatory property related to voice quality and vocal pathology. In this study, simultaneous inverse filtered airflow and electroglottographic recordings were made for two male subjects phonating at deliberately varied pitches, loudness levels, and qualities. The goal of the study was to relate temporal characteristics associated with adduction with flow characteristics of the flow glottogram. A triangular model of laryngeal flow glottograms predicts that Ûω/Û̇ = k2π(Q0 − Qr), where Û is the peak flow, Û̇ the negative peak flow derivative, Q0 is the open quotient (the duration the flow is above the baseline, divided by the period), Qr is the rise quotient (the duration the flow rises from baseline to peak, divided by the period), and k = 1. Empirical results suggest k ≐ 0.5. These findings will be discussed, and a variety of adduction measures applied to the data will be compared.
Application of Gerchberg's iteration algorithm to noise spectrum restoration
Authors: Wu Mei Q. , Crocker Malcolm J.
Published: October 2005

In noise control engineering, there are many cases where a steady noise is overlapped by an impulse noise. Some information about the noise source can be obtained by analyzing the frequency spectrum of the steady signal. But when the impulse signal is overlapped on the steady signal, the spectrum of the combined signal is usually different for the spectrum of the original steady signal. One way to remove the effect of the impulse is to filter it off in the time domain. By doing this, some time samples of the steady signal are removed also. The spectrum of the filtered signal may still be different from the spectrum of the original steady signal. In this paper, Gerchberg's iterative algorithm was used to restore the lost time samples and hence to restore the spectrum of the steady noise. Experiments were conductd on idealized noise sources where the spectrum of the steady noise was known. From the experimental results, it is seen that the spectrum restored by Gerchberg's iterative algorithm coverges toward the original spectrum. After 25 iterations, the restored spectrum is very similar to the original spectrum. This method only applies to bandlimited steady signals.
An acoustic intensity measurement simulation for evaluating sound power determination standards
Authors: Laville Frédéric , Agnan Jean-Luc , Nicolas Jean
Published: October 2005

Work has been done for several years on the standardization of sound power determination using acoustic intensity measurements. The validation of the standards through testing is difficult because of the large number of measurement conditions to be tested. To help overcome this problem, a computer code simulating the measurement has been developed. Whereas past attempts have been limited to special cases, the developed code is intended to handle cases that integrate more of the industrial reality. The measured noise sources as well as the extraneous noise sources are modeled by any number of monopoles for which the user specifies their quantity, source strength, location, relative phase, and degree of coherence. This approach offers computational simplicity as well as the possibility to model realistic radiation patterns such as plate radiation. The proposed ISO standard (ISO/DP 9614) is evaluated for a plate radiating in the presence of background noise. Recommended corrective actions were found appropriate for farfield measurements and inapropriate for nearfield measurements.
Acoustic radiation from fluid-loaded structures with discontinuities
Authors: Thompson Charles , Sen Rahul
Published: October 2005

The presence of discontinuities such as joints, in a fluid-loaded structure results in farfield acoustic radiation even when subsonic wavenumbers prevail in the main body of the structure. The object of this study is to develop a rational model of this phenomenon using singular perturbation methods. When a flexural wave in a thin structure impinges on a joint, evanescent structural displacement fields are set up in the vicinity of the joint. This paper proposes to show that these fields are associated with rotation-dominated modes that are supported by shear-corrected plate theories of the Timoshenko-Mindlin type flexural mode. The method of matched asymptotic expansions will be used to obtain mode-conversion coefficients, and it will be shown that the spatial wavenumber spectrum of the local field is instrumental in setting up an acoustic farfield in the surrounding fluid. An important mathematical issue is the proper description of evanescent fields; this analysis will be based on results obtained previously in connection with acoustic waveguides.
Experimental facility for fluid/structure interaction research
Authors: Burroughs Courtney B.
Published: October 2005

The boundary layer research facility (BLRF) at Penn State is being modified so that experimental research on the interaction between fluid boundary layers and vibration response of underlying structures can be conducted. Because the BLRF is a closed-loop glycerine tunnel, the boundary layer is thick, which provides space for detailed measurements required for research in fluid/structure interaction. In addition, laminar, transition, or turbulent flow can be generated and maintained. Modifications to the BLRF for fluid/structure interaction research include: (1) the introduction of an elastic and exchangeable structure in contact with the flow, (2) the reduction of background vibration to remove interference of structural response measurements by background vibrations, and (3) an increase in the measurement accessibility to the boundary layer. A description of the modified BLRF is presented along with technical support for several of the key design elements. Measurements to characterize and define the capabilities of the modified BLRF are discussed. [Work supported by M. Reischman, ONR Code 432F.]
Vibratory response of heavily fluid-loaded plates to convecting pressure excitations via impulse response methods
Authors: Ebenezer D. D. , Stepanishen Peter R.
Published: October 2005

A general time domain approach is presented to evaluate the vibratory response of fluid-loaded plates to one-dimensional convecting pressure excitations. The velocity of the plate is expressed as a modal sum of the product of the in vacuo eigenfunctions and time-dependent coefficients. Previously developed [D. D. Ebenezer and P. R. Stepanishen, J. Acoust. Soc. Am. 82, 659–666 (1987)] sets of coupled convolution integral equations for the time-dependent modal velocity coefficients are rearranged to facilitate the numerical solution. In the previous approach, FFT methods were used to evaluate the uncoupled velocity responses. In the present approach, analytical expressions are obtained for the impulse responses of the modal admittances and the impulse responses of the modal radiation impedances are obtained via a combination of analytical and wave-vector-time domain techniques. The known terms in the coupled convolution integral equations are thus very easily obtained and a large number of modes can be included in the analysis. Numerical results are presented for the uncoupled and coupled velocity responses to loads moving across the plate. They clearly show that coupling between modes is significant and that the general characteristics of the vibratory response can be anticipated based on a knowledge of the Mach number and the fluid-loaded modal resonant frequencies. [Work supported by ONR.]
Sound radiation of a finite plate with several types of boundary conditions
Authors: Berry A. C. , Nicolas J. , Guyader J. L.
Published: October 2005

The sound radiation from baffled, thin, rectangular plates has generally been restricted to the simply supported plate analysis, for which simple relation hold. This paper investigates the influence on the acoustic radiation, of a more general type of boundary conditions, namely, edges presenting arbitrary translation and rotation stiffness. For such structures, the in vacuo modal basis is not known a priori. Thus the in vacuo vibrational response of the structure is determined using a polynomial expansion of the transverse displacement. In the limit of a low fluid-structure coupling (air loading), the in vacuo vibrational response may be used to calculate the acoustical perturbation generated in the fluid. Results are shown in terms of quadratic velocity of the plate, farfield acoustic pressure, acoustic power, and radiation efficiency of the plate. The influence of the boundary conditions is examined, and the limiting cases of simply supported, clamped, and free plates are presented.
A simple apparatus for the measurement of the shear modulus of viscoelastic materials over 4 decades of frequency
Authors: Schloss Fred
Published: October 2005

Two identical test samples are sandwiched between the metal ends of two small identical piezoelectric stacks, one acting as a transmitter having a displacement proportional to the applied voltage V, the other receiving a charge Q proportional to the force. Since the ends of the stacks are moving out of phase with zero motion at the center, no external fixtures are necessary. The modulus equals Qt(VS)−1(d33)−2, where t and S are the sample's thickness and cross-sectional area and d33 is a piezoelectric parameter. The useful frequency range is 1–10 000 Hz, which can be extended using smaller parts and different materials for the end plates, and the temperature range is 50–200 °C. The device is limited to low values of dynamic strain.
Inverse acoustic problems in the resonance scattering regime
Authors: Gaunaurd G. C.
Published: October 2005

Solving an inverse scattering problem here means the extraction of physical (viz., composition and shape) information from the active echo returned to an underwater elastic target. This is the general area of target identification. In this sense, solving inverse scattering problems is relatively simple (!) in the low or high spectral regions because there a number of simplifying approximations may apply. However, for the intermediate or resonance region of all scatterers, the above approximations do not hold, and exact solutions or novel types of approximations are necessary, making the actual inversions of scattered waveforms extremely hard. A certain level of success has been achieved by exploiting the presence of certain resonance features that manifest themselves in the resonance region of the scattering cross sections (SCS) of submerged elastic targets. Examples of how and how well one can extract identifying sets of resonance features from an echo have been recently overviewed [G. Gaunaurd, J. Acoust. Soc. Am. Suppl. 1 84, S168 (1988)]. Although the main interest over the years has been the isolation of the resonance features in the SCS either by theoretical (viz., the background subtraction) and/or experimental (viz., samplings of echoes in their tail portions) means, little has been said about what to do with them, once isolated. This paper will show (idealized) examples illustrating how these resonance features can be actually used for unambiguous target characterization.
Three-dimensional inverse scattering tomography using elastic waves
Authors: Johnson Steven A.
Published: October 2005

A 3-D ultrasound method that provides clinically significant (and diagnostically valuable) images of important tissue characteristics is described. This is made possible by an inverse-scattering technique that provides accurate, independent images of speed of sound, density, and absorption (with extension to five viscoelastic constants) at the very high spatial resolution of 0.5 to 1.0 wavelength of the incident ultrasound energy. Image size has increased from 16×16 pixels to 200 × 200 pixels and to 32×32×96 voxels while maintaining accuracy and spatial resolution. No other methods, for example the B-scan mode of diffraction tomography based on the Rytov or Born approximations, have come close to matching this achievement. Further improvement in tissue imaging is possible by using the inverse scattering method in conjunction with a high performance reflection mode imaging method called synthetic focusing. The synthetic focusing method in nearly homogeneous medium produces medium to very large size, high spatial resolution images. The synthetic focus method also can be made to produce excellent images in the moderately high refracting environment of the human body by applying time shift and attenuation corrections obtained from medium resolution inverse scattering images of speed of sound and absorption. Regularization and constrained optimization methods have been used to improve the conditioning and therefore the speed of the inverse scattering methods.
Theoretical development of acoustic agglomeration of aerosol particles
Authors: Song Limin
Published: October 2005

The orthokinetic interaction is recognized as a major mechanism responsible for rapid agglomerations of polydisperse aerosols in a sound field. The acoustic agglomeration theory based on this mechanism was developed by Mednikov in the 1960s. However, this theory has two basic problems that are not answered satisfactorily: How many small particles in an agglomeration volume created by a large particle will actually be collected in one cycle and how many small particles outside will be attracted into the volume so that agglomeration can continue for successive cycles? These problems have been solved to the first-order approximation in this investigation. A general expression for the specific acoustic agglomeration rate is derived by a statistical approach. The efficiency of collison between particles in a sinusoidal wave field is given explicitly. In addition, the analysis of the relative motion between particles was performed by including the effects of acoustic scattering from particles. The results show that a small particle is strongly attracted toward a larger particle once they cross over each other. This transverse attraction provides an efficient mechanism for the fill-up of the agglomeration volume.
The role of rate of transition in the perception of place of articulation
Authors: Zakia Renee A. E. , Kingston John
Published: October 2005

Listeners can reliably identify the place of articulation of bilabial and alveolar stops in severely truncated CV syllables from just the initial 20 ms, but they need 40 ms or more to identify a velar. The need for a longer interval may reflect listeners' expectations that rate of transition should be slower for velars because of the slower opening gesture at this place of articulation [Kewley-Port et al., J. Acoust. Soc. Am. 73, 1779–1783 (1983)]. This paper investigates the perceptual effects of differences in rate of transition in two experiments. First, synthetic CV syllables, where the direction and magnitude of the F1, F2, and F3 transitions matched natural [ba,da,ga] syllables, but where the transition durations were varied in 15-ms steps between 15 and 120 ms were presented to listeners for two-interval, forced-choice discrimination and for identification along a stop—glide dimension. If listeners expect longer transitions with velars, then they should accept stimuli with longer transitions as stops when the formants indicate a velar than when they indicate a bilabial or alveolar. Second, variation in F2 and F3 transitions between values appropriate for [da] and values appropriate to [ga] were combined with variation in the rate of transition; rate of transition was varied in 5-ms steps between 20 and 60 ms. These stimuli were also presented to listeners in two-interval, forced-choice discrimination and identification of place of articulation tasks. It is expected that faster rates of transition will shift listeners' response to stimuli with F2 and F3 transitions near the [d]/[g] boundary toward [d] and slower transitions will shift them toward [g].
Effects of first formant onset qualities on VOT judgments can be explained by auditory processes not specific to humans
Authors: Kluender Keith R.
Published: October 2005

Both F1 transition duration and F1 onset frequency have been proposed to be perceptually significant in categorization of voiced and voiceless syllable-initial stops. Transition duration per se may not, however, explain the fact that, for longer transitions, longer F1 cutback is required in order to perceive a stop as voiceless. Longer transitions result in lower F1 onsets at any duration of cutback greater than zero, and it is possible that the major effect of F1 is determined by its frequency at onset. In this study, F1 transition duration, onset frequency, and slope were varied across four types of F1 transition in which one of the three variables (onset frequency, duration, slope) was held constant while the other two were allowed to vary. Each of the four F1 types was used in syllables with higher formants appropriate for labial, alveolar, and velar places of articulation. By far, the best predictor of identification of these stimuli by human listeners was F1 onset frequency; F1 duration, F1 slope, and place of articulation had little or no effect on labeling boundaries. Six Japanese quail (Coturnix cotunix japonica), trained to respond differentially to voiced versus voiceless stops, evidenced “labeling” behavior almost identical to humans' with the same stimuli. These results are taken to provide strong evidence that F1 onset frequency is the primary determinant of shifts in VOT boundaries across place of articulation, and that an auditory mechanism not specific to humans is responsible.
The perception of stop consonants: Spectral tilt revisited
Authors: Richardson Kevin H. , Sawusch James R.
Published: October 2005

The claim that invariant features involving spectral tilt, as proposed by Lahiri et al. (1984), are used by human listeners when classifying stop consonants was explored. Subjects were presented with stop-vowel syllables representing [b], [d], and [g] with a range of vowels in an identification task. Stimuli were either natural, synthetic (based on the natural set), or modified tokens (synthetic tokens containing a burst which, according to the Lahiri et al. metric, should be misidentified). Results indicate that the change in spectral tilt plays little role in stop perception. Only [g] showed any evidence of a systematic effect. To investigate further the difference found in the identification task, tone analogs mimicking the [b], [d], and [g] CV sets were produced. Both speech and nonspeech groups were run to examine the perceptual locus of any effects of change in spectral tilt. The results of these two groups will be compared to those of the speech groups in the first experiment. The implications for theories of auditory to phonetic coding of stops and possible invariant attributes in stop perception will be discussed.
Spectral shape factors for speaker-independent automatic recognition of stop consonants
Authors: Nossair Zaki B. , Zahorian Stephen A.
Published: October 2005

A series of automatic recognition experiments was conducted with naturally produced English stop consonants /b,p,d,t,g,k/ in syllable initial position. The objectives of the experiments were to investigate in detail the effectiveness of spectral shape factors, as both dynamic and static features for automatic recognition of stop consonants. Spectral shape factors were computed as the discrete cosine transform coefficients (DCTCs) of the magnitude spectra. The database used in these experiments consisted of 2481 CVC syllables spoken in isolation by ten males, ten females, and ten children. In all experiments, 15 speakers were used to train the classifier and the other 15 speakers were used for evaluation. For the case of dynamic features, DCTCs were computed over a 50-ms interval beginning with the burst using 7-ms frames spaced every 5 ms. For the static case, the DCTCs were computed from one 25.6-ms frame beginning at the burst. Automatic classification results, based on the test data, were 87.2% for the dynamic spectral shape features versus 60.1% for the static case. Dynamic features timed to begin with the formant transition area resulted in 44.6% recognition rates. Control experiments with formants resulted in much lower recognition rates under every condition tested. In summary, these results are in agreement with the theory that dynamic spectral shape, spanning an interval of approximately 50 ms beginning at the burst, carries most of the information for initial stop consonants [D. Kewley-Port, J. Acoust. Soc. Am. 73, 322–335 (1983)]. [Work supported by NSF.]
Effects of speaking rate are moderated by phonological structure
Authors: Volaitis Lydia E.
Published: October 2005

Changes in speaking rate can alter the duration of properties that convey segmental information. For example, with an increase in syllable duration, the distribution of VOT values for labial stops, especially the voiceless /p/, shifts upward along the VOT continuum. Perceptual adjustment to this rate effect entails a shift in the category boundary toward longer VOT values for longer syllables and, as recently reported, a change in which stimuli along the VOT continuum are perceived to be best category exemplars [J. L. Miller and L. E. Volaitis, J. Acoust. Soc. Am. Suppl. 1 80, S125 (1986)]. The present two-part investigation focuses on whether the location of best exemplars is solely a function of syllable duration, or depends as well on the phonological structure of the syllable, specifically, the place of articulation of the initial consonant. Results of the production study have confirmed that as place moves from labial to alveolar to velar, VOT increases, and have shown that this occurs across a range of syllable durations. The parallel perception study, currently in progress, addresses whether at any given syllable duration, the location of the best exemplars varies with place, in accord with the production data. Such a result would indicate that the location of the best exemplars is a function of the phonological structure of the syllable, in conjunction with its duration. [Work supported by NIH.]
The effect of position of stress on the durations of stop consonants in American English
Authors: Turk Alice E.
Published: October 2005

It has been documented for American English that /t/ and /d/ are realized as very short alveolars (flaps) following a stressed vowel nonfoot initially: [V́_V(…)] or [V́CV_V(…)] [Zue and Laferrière (1979)]. This study investigates the duration of other stop consonants in the flapping environment to see whether the shortening of /t/ and /d/ is a phonologization of a more general phonetic principle. Stathopoulos and Weismer (1983) report that in nonsense words, stop closure durations for all places of articulation are shorter in the post-stress environment. In a study of two speakers using natural speech, one speaker's consonants shortened significantly (p<0.05) in the flapping environment. However, the fact that the second speaker did not exhibit the same shortening pattern suggests the need to expand the number of speakers; results from such a larger subject pool will be reported.
Fricative perception: Frication and transition cues
Authors: Zeng Fan-Gang , Turner Christopher W.
Published: October 2005

By presenting fricative-vowel tokens across a range of intensities and determining the audibility of various portions of the tokens, the present study investigated the cues for fricative perception. The frication cue was defined as gross spectral-shape differences residing in the steady-state frication noise burst; while the transition cue was defined as dynamic formant changes in the transition portion between the fricative and its adjacent vowel. Both natural and synthetic syllables, with duration and amplitude normalized, were used in closed-set recognition tasks by four normal-hearing subjects. The presentation levels varied from 0%–100% audibility for individual fricatives. Results suggest that the frication burst may be a sufficient cue for the correct perception of fricatives; while the transition cue, although it may not be necessary, plays an important role in helping to identify the place information, particularly at low presentation levels where the frication cue is always not audible. Two other experiments, using truncated stimuli, with frication or transition cues presented independently, further supported our conclusions. [Work supported by Syracuse University Senate Research Grant and Deafness Research Foundation.]
Integrality in consonant perception: Vowel environment influences consonant perception
Authors: Tomiak Gail R.
Published: October 2005

Previous research designed to examine the nature of phonetic processing through the use of noise-tone analogs to speech syllables has demonstrated that the phonetic coding of noise and tone segments as consonant and vowel within a syllable proceeds in a mutually dependent (integral) fashion. However, the auditory coding of these segments is independent (separable). The purpose of the present experiment was to determine whether consonant information is also processed in an integral fashion with respect to vowel information contained in an adjacent syllable. Tone-noise-tone analogs of V_CV disyllables were used in a speeded classification task. The results indicate that the phonetic coding of the noise as a fricative is integral with respect to both the preceding and following vowel. Thus the perceptual coding of consonants seems to proceed with reference to a surrounding vowel environment regardless of syllable boundaries. The implications of these results for the phonetic coding of speech, and recent proposals regarding the priority of vowel information in perception, will be discussed. [Work supported by NINCDS.]
Trainlng Japanese listeners to identify /r/ and /l/
Authors: Logan John S. , Lively Scott E.
Published: October 2005

Native speakers of Japanese learning English generally have difficulty differentiating the phonemes /r/ and /l/, even after years of experience with English. Previous research that attempted to train Japanese listeners to distinguish this contrast using synthetic stimuli showed little success, especially when transfer to natural tokens containing /r/ and /l/ was tested. In the present study, a procedure that differed from these earlier attempts was used. Japanese subjects were trained in an identification paradigm using as stimuli multiple natural exemplars contrasting /r/ and /l/ from a variety of phonetic environments. A pretest-posttest design combined with a test of generalization containing novel natural tokens was used to assess the effectiveness of training. Analysis of pilot data from a small group of subjects showed that the new procedure was more robust than earlier procedures. The results demonstrate the importance of stimulus variability and task-related factors in training second language learners to perceive novel phonemic contrasts that are not distinctive in their native language. [Work supported by NIH.]
An audio-visual investigation of linguistic stress perception
Authors: Rosenblum Lawrence D.
Published: October 2005

The true acoustic correlate for linguistic stress has not yet been found. While it was originally thought that listeners base stress judgments on syllable intensity, it has been shown that intensity, fundamental frequency, duration, and spectral structure can all act as effective information for stress perception [D. Isenberg and T. Gay, J. Acoust. Soc. Am. Suppl. 1 63, S21 (1963)]. Because no single simple acoustic dimension seems to be dominant, it has been proposed that listeners base stress judgments on an articulatory property such as vocal effort. Since such properties can be specified optically as well as acoustically, a study to test this hypothesis was conducted which used conflicting audio-visual presentations of a speaker producing tokens from two noun-verb pairs (CONvict—conVICT and PERmit—perMIT). The prediction was made that if stress judgments are based on perception of articulatory dynamics rather than on simple acoustic parameters, then judgments should be affected by visual as well as auditory information. It is shown that stress judgments are affected by visual information even when subjects (1) are instructed to base their judgments on only what they hear and (2) cannot detect a discrepancy between the audio and visual components. Similar results are also shown for noun-verb tokens distinguished by an auditory dimension that cannot be specified visually (fundamental frequency) indicating that a more general articulatory property, such as vocal effort, might be the basis of stress judgments.
Effects of lip-read information on auditory perception of Japanese syllables
Authors: Sekiyama Kaoru , Tohkura Yoh'ichi
Published: October 2005

The McGurk effect is a phenomenon that demonstrates a perceptual fusion between auditory and visual (lip-read) information in speech perception under visual-auditory discrepancy condition (using dubbed video tapes). This paper examined the relation between the “McGurk effects” and the intelligibility of auditory stimuli. A female narrator's speech was video taped for ten Japanese syllables (/bad,/pad,/mad, /wad,/da/,/tad,/na/,/ra/,/ga,/ka/). The video and audio signals for these ten syllables were combined, resulting in 100 audio-visual stimuli. These stimuli were presented to ten subjects who were required to identify the stimuli as heard speech in both noisy and noise-free conditions. For both conditions, the intelligibility of the auditory stimuli was measured, presenting the auditory stimuli alone. In the noise-free condition, the McGurk effect was small and found only in conditions in which the intelligibility of the auditory stimuli was not 100%. In the noisy condition, the McGurk effect was very strong and widespread. These results suggest that incomplete intelligibility of auditory stimuli is necessary for the McGurk effect.
Application of the finite element method to the modeling of plane wave scattering from compliant tube gratings at oblique incidence
Authors: Hennion A. C. , Bossut R. , Decarpigny J. N. , Audoly C.
Published: October 2005

The modeling of compliant tube gratings has been performed with the help of the finite element method, using the ATILA code [Decarpigny et al., J. Acoust. Soc. Am. 78, 1499–1507 (1985)]. To do this, only the unit cell of the periodic structure, including a small part of the surrounding fluid domain, has to be meshed, due to the use of a classical Bloch-type relation between the displacement components of points that are separated by the grating spacing. Then, the effect of the external fluid domain is accounted for by matching the pressure field in the finite element domain to simple ingoing and outgoing plane-wave expressions. This paper describes results obtained for the scattering of a plane wave from different compliant tube gratings, including internal losses, at oblique incidence. First, it shows that a nice agreement has been found between predictions from an analytical model and finite element results for the insertion loss of an infinite elastic plate, thus validating the method. Then it describes the case of cylindrical tube gratings embedded or not in a viscoelastic layer, and shows the effects of low-stiffness encapsulant on insertion loss curves. Finally, double-layer gratings with different separations are also considered.
Physiological correlates of forward masking in single nerve-fiber and compound neural responses recorded from the auditory nerve
Authors: Relkin Evan M. , Doucet John R. , Smith Robert L. , Turner Christopher W.
Published: October 2005

Masking can be defined as a reduction in the detectability of one sound stimulus, the probe, due to the presence of a second sound stimulus, the masker. Reduction in detectability can be quantified by measuring the increase in threshold for the detection of the probe tone. Most previous studies of the correlates of forward masking in the auditory nerve have measured reductions in averaged responses to the probe produced by the masker. It is not possible to relate reductions in averaged responses to changes in detectability since detectability depends on response magnitude and variance. However, two-interval forced procedures can be used to measure detection thresholds for spike trains of primary, single nerve fibers [Relkin and Pelli, J. Acoust. Soc. Am. 82, 1679–1691 (1987)] and N1 of the compound action potential (CAP) [Relkin and Smith, J. Acoust. Soc. Am. Suppl. 1 83, S98 (1988)]. Relkin and Turner [J. Acoust. Soc. Am. 84, 584–591 (1988)] showed that there are large discrepancies between behavioral thresholds and thresholds for single nerve fibers for forward masking. Particularly at high masker levels, thresholds for the detection of the probe in the activity of single fibers are more sensitive than those measured behaviorally. Several hypotheses that might explain this discrepancy include the effects of spatial and/or temporal processing of the responses of single neurons. These hypotheses are beginning to be tested using both single fiber and compound neural recordings. Forward masking of the CAP is more similar to behavioral forward masking than is forward masking in single neurons, suggesting a spatial summation effect. Methods for measuring peristimulus compound potentials have also been developed to investigate temporal effects such as the relative importance of onset and steady-state responses.
Modeling of acoustic environment for the purpose of noise cancellation
Authors: Mohammed Osama A. , Yen Kang , Raman Vijay , Ramdin Kurt
Published: October 2005

The specific noise-cancellation problem being considered here is the active cancellation of acoustic noise by the use of electronic earphones. Since it is important to achieve cancellation of ambient noise only, and not of warning sounds and speech, an adequate understanding of the noise environment in which the earphones are to be used is required. A key step in the approach taken by these authors is ton initially extract information regarding specific noise environments off-line. This information can then be used to define details of control structure and values for the canceler in that environment. During actual usage of the earphones, the algorithm must adapt to fine-tune itself to the environment at that time. Recordings of ambient noise were made for specific environments. Data-analysis programs were developed and used to construct a model of the stationary noise environment and extract the relevant parameters. These are used to define the structure, parameter range, and computational requirements of the noise canceler. [Work supported by the Florida High Technology and Industry Council.]
Dynamics of musical expression
Authors: Carterette Edward C. , Kendall Roger A.
Published: October 2005

Various aspects of the dynamics of musical expression in terms of both acoustic and perceptual parameters were studied. In one study, duets of soprano orchestral winds performed brief musical passages in unison and in simple diatonic harmony. Perceptual spaces, obtained by scaling, were compared with acoustical analyses. Orderings of instrument pairs in three-dimensional space were influenced by context (unison versus harmonized single notes and melodic fragments) and by such acoustic factors as combinations of excitation mechanisms and tone-hole lattice cutoff frequencies. In a study of musical expression, single instruments, which differed in performer degrees of freedom (piano, violin, trumpet, clarinet, and oboe), were played at three levels of expressiveness. The aim was to explicate the variables used by the performer in successfully communicating expressive intent. Acoustical analyses were compared by means of a theoretical model based on the interactions of pitch, time, and dynamic contour strata.
How to treat dynamic range if there is no dynamic space
Authors: Houtsma Adrianus J. M.
Published: October 2005

The large dynamic range of modern sound equipment is often incompatible with the playback situation. Compact-disc music played at a low level because of sensitive neighbors. or car-radio music played against a background of intense and varying noise, is almost always partly inaudible unless signal levels are constantly adjusted. For a fixed ceiling on sound dynamics or for constant levels of background noise, as is often found in a typical home, use of dynamic compression according to the function y = Ax‴ turns out quite satisfactory, with the compression exponent m between l (no compression) and 2/3 (maximum acceptable compression), and attack and release times equal to 5 and 180 ms, respectively. For treatment of music and speech under conditions of intense and variable noise, as is the case in a moving car, several signal processing schemes are discussed. such as noise-dependent volume control, noise-dependent frequency control, and noise-dependent amplitude compression.
Musical dynamics with keyboard instruments
Authors: Hall Donald E.
Published: October 2005

Keyboard instruments provide the extreme cases of wide or narrow dynamic ranges. The harpsichord mechanism apparently has a fixed dynamic level, aside from registration changes. But performers do claim to have some expressive control, and it may be asked whether measureable physical differences can be demonstrated. The pipe organ uses both swell box shades and registration changes, and what may be typical sound level differences involved will be discussed. The clavichord and piano are both more akin to other instruments in having output strongly dependent on the force with which the keys are struck. Comparisons among these cases raise questions about how much perceived musical dynamics depends on mere loudness, and how much on timbre, articulation, and other aspects of the musical performance.
Distinguishable musical dynamics
Authors: Patterson Blake R.
Published: October 2005

Loudness changes, which composers specify by musical dynamics pp, p, mp, mf, f, and ff, make music exciting. The goal of this study is a usable and rational specification of an “acceptable” dynamic range for musicians. Requiring that a note at one dynamic level be at least s dB more intense than nearby notes at the next-lower dynamic level leads to a minimum solo-instrument per-note dynamic range (k−1) (d+s), where k is the number of dynamic levels and d is the maximum intensity difference between nearby notes at “constant” dynamic level. For six dynamic levels and the bassoon, for which d is about 5 dB, a 1-dB spacing s between dynamic levels gives a minimum dynamic range of 30 dB [“Musical Dynamics,” Sci. Am. 231 (5), 78–95 (November 1974)]. Some related results published by others over the past 14 years are briefly noted. Professional performances having different dynamic ranges are exhibited. A live demonstration of distinguishable dynamics on the bassoon is presented. Much remains to be studied in this rich field, for example, (1) dependence of d on instrument and dynamic level; (2) dependence of (listener-perceived) loudness on musical factors other than intensity, such as pitch, timbre, note duration, and transients; (3) dynamic complexities of ensemble performance: solo levels, heterogeneous ensemble, “chorus” effects, ensemble with soloist, etc.; (4) dependence of musical dynamics on music type, performance environment, and performance objective.
From acoustics to the social sciences: The concept of “sound effects”
Authors: Thibaud Jean-Paul
Published: October 2005

The analysis of sound within context is very complex. It requires an interdisciplinary approach in several fields: physical, architectural, psychological and physiological acoustics, and environmental sociology. The concept of “sound effect” involves concrete links between various forms of data. Sound effect can be defined by three major characteristics. (1) It is not the product of a cause, but rather the product of the relationship between the signal and the context (i.e., Doppler effect). Thus its definition necessitates a recognition of the physical signs in question and of the conditions of propagation and sound perception. (2) It involves a modal or instrumental logic. (3) It permits a general discourse about sound, but requires examples related to a particular situation. The concept of sound effect serves as a tool to uncover information about sound. It allows the researchers to put the sound phenomena in a wider perspective. [Work supported by Centre National de la Recherche Scientifique (CNRS, France.]
Detection and recognition of temporal patterns
Authors: Weber Daniel L.
Published: October 2005

The detection/recognition theorem [S. J. Starr, C. E. Metz, L. B. Lusted, and D. J. Goodenough, Radiology 116, 533–538 (1975)] was used to predict a listener's ability to recognize (identify) one of a pair of temporal patterns based upon the listener's detection performance. The six experimental conditions compared all possible pairs of four 860-ms patterns. The four patterns consisted of either two or four 1-kHz sinusoidal components of either 20- or 200-ms duration. Specifically, the four patterns were: two 20-ms components separated by 820 ms; two 200-ms components separated by 460 ms; four 20-ms components separated by 260 ms; and four 200-ms components separated by 20 ms. All components were square gated on and offer zero crossings. In four of these comparisons, stimuli differed on “one” dimension (either number or duration of components), whereas the other two comparisons involved differences in “two” dimensions (both number and duration of components). (Interstimulus gaps were necessarily confounded with number of components and component duration in all conditions.) Although the results were not uniform for all 11 listeners, comparisons that involve two dimensions generally led to better recognition performance than comparisons that involved one dimension. More surprising was the apparent unimportance of temporal structure for the encoding of these complex stimuli inferred from the general failure to reach predicted recognition performance. [Research supported by AFOSR through WPAFB AAMRL/BBA.]
Temporal window shape as a function of frequency and level
Authors: Plack Christopher J. , Moore Brian C. J.
Published: October 2005

The temporal window is an intensity weighting function that processes the internal representation of auditory stimuli by integrating energy over time. Temporal window shapes were measured using the technique described in an earlier article [Moore et al., J. Acoust. Soc. Am. 83, 1102–1116 (1988)]. Window shapes were measured at four different frequencies (300, 900, 2700, and 8100 Hz) and at three different masker levels covering a 20-dB range at each frequency. The shape of the temporal window was well described by modeling each side as the sum of two rounded-exponential functions. The equivalent rectangular duration (ERD) of the window was roughly constant at about 8 ms for center frequencies from 900 to 8100 Hz, but increased to about 13 ms at 300 Hz. The increase at 300 Hz may be explicable in terms of “ringing” in the auditory filter. The ERD decreased somewhat with increasing level, for example, having a value of about 10 ms at 2700 Hz with a 20-dB masker spectrum level and about 7 ms with a 40-dB masker spectrum level. Temporal window shapes can be used to produce temporal excitation patterns that illustrate the effect of limited temporal resolution on the internal representation of acoustic stimuli. [Work supported by MRC, UK.]