bluuebunny/crossref_metadata_2025 huggingface dataset:
Explanation of direction finding with a vector sound-intensity probe
Authors: Hickling Robert , Wei Wei
Published: October 2005

An explanation is provided of direction finding with a vector sound-intensity probe in the far field of a single sound source. As described earlier [Hickling etal., J. Acoust. Soc. Am. 94, 2408–2412 (1993)], the probe consists of four pressure sensors in the tetrahedral arrangement, intensity being measured with the cross-spectral method. The probe has omnidirectional sensitivity. Direction-finding with an arrangement of pressure sensors makes use either of beamforming or of differences in phase, time-of-arrival or amplitude at two sensors. It is shown that direction finding with the vector probe is based on phase differences. The close spacing of the pressure sensors of the probe relative to wavelength creates a relation between the intensity-vector components and phase that is independent of the ratio of sensor spacing to wavelength, each component being dependent only on the pressure-amplitude squared and the direction of the source. Accuracy of direction finding depends on the accuracy with which the components of the intensity vector can be measured. The accuracy of vector intensity measurement has been demonstrated a number of times. Examples are presented of the accuracy of direction finding with a vector probe in water.
Measurement of the sound-power of a moving source with a vector sound-intensity probe
Authors: Hickling Robert , Wei Wei
Published: October 2005

A vector sound-intensity probe can be used to measure the sound-power of a moving source that has an axisymmetric sound field. The source moves along the axis. The vector probe measures intensity normal to a hypothetical cylinder coaxial with the source path and simultaneously tracks the source. In this way it is possible to determine the intensity (sound-power per unit area) distribution over the surface of a long cylinder enclosing the source. If the cylinder is long enough, the contribution of the ends can be neglected and the intensity distribution can be integrated to obtain the sound power of the moving source. This procedure was applied to a small moving continuous sound source (i.e., a driven Bruel & Kjaer 8105 hydrophone) in a water tank. The vector sound-intensity probe consisted of four hydrophones in the tetrahedral arrangement, intensity being measured with the cross-spectral method [see Hickling etal., J. Acoust. Soc. Am. 94, 2408–2412 (1993)]. Good agreement was obtained between the sound power of the moving source, the sound power of the source when stationary and the sound power of the source estimated from information provided by the hydrophone manufacturer.
The combustive sound source; basic principles
Authors: Wilson Preston S. , Muir Thomas G. , Ellzey Janet L.
Published: October 2005

The combustive sound source (CSS) is a low-frequency underwater sound source that has appeared in several U.S. patents dating back to 1924, but until now, has not undergone extensive study. Combustible gases are introduced into a combustion chamber and ignited with a spark. The ensuing combustion produces a bubble of expanding gas which, in turn produces high-intensity, low-frequency acoustic pulses. It was observed that low-energy combustions produce several bubble pulses, while high-energy combustions produce a high-pressure first bubble pulse, and fewer, lower amplitude bubble pulses. The boundary between the two regimes was observed at 8900 j, for stoichiometric methane-oxygen (which corresponds to a volume of 0.75 liters at STP), ignited in an open-bottomed conical combustion chamber. A high-speed motion picture camera was used to observe the bubble produced at the boundary condition and pressure signatures were recorded. Behavior from both regimes was observed. For the high-energy case, the bubble motion observed on film is correlated with the measured pressure signature. For the low-energy case, bubble periods observed on film are compared to acoustically measured bubble periods. [Work supported by ONR.]
Influence of the expectation of a particular signal duration on detectability
Authors: Dai Huanping , Wright Beverly A.
Published: October 2005

The results of two experiments demonstrated that the detectability of an auditory signal depends on the subject’s expectation about its duration. The signal (a 250-, 1000-, or 4000-Hz tone, or a broadband noise) had one of six durations between 5 and 295 ms. The masker was a continuous, broadband noise (N0=25 dB). Each 2IFC interval was always marked by a fixed-duration visual display whose onset coincided with the onset of all potential signals. When the signal duration was fixed, each signal was correctly detected on about 85% of the trials. in experiment 1, the six subjects were encouraged to expect a particular signal duration (5 or 295 ms) by presenting the signal most often at that duration, and only occasionally at each of the five remaining unexpected durations. Performance for the unexpected signals decreased from 85% to chance as their durations departed from the expected duration. In experiment 2, the signal was randomly presented at each of the six durations with equal probability, and the subjects were so informed. The mean percent correct value was about 79%, with the longest signals being affected the most by the duration uncertainty. The results are analyzed using signal-detection models incorporating the idea of an adjustable temporal-integration window. [Work supported by NIH.]
Does recovery from forward masking reflect two neural channels?
Authors: Nizami Lance , Schneider Bruce A.
Published: October 2005

Recovery rates differ significantly [E. M. Relkin and J. R. Doucet, Hear. Res. 55, 215–222 (1991)] for nerve fiber groups believed to be separate channels for loud and soft sounds [e.g., Winslow etal., in AuditoryProcessingofComplexSounds (Erlbaum, Hillsdale, NJ, 1987), pp. 212–224]. Such differences could produce two distinct portions in the auditory recovery curve. Forward-masked detection thresholds were obtained from three experienced listeners using a two-interval forced-choice task, at masker-target time gaps (recovery times) of 0–10 ms with a resolution of 1 ms or less. The masker, a 97 dB SPL, 2-kHz, 200-ms tone, preceded a 2-kHz Gaussian-shaped tone pip (σ=0.5 ms). A cusp appeared in the curve relating forward-masked thresholds to recovery time at 3–6 ms, indicating that recovery followed a two-part decay curve. Forward-masked thresholds rise over 0–2 ms replicating a pattern found earlier [H. Duifhuis, J. Acoust. Soc. Am. 54, 1471–1488 (1973); M. J. Penner, J. Acoust. Soc. Am. 56, 179–182 (1974)]. [Work supported by NSERC.]
Optimal decision rules for the same-different task
Authors: Dai Huanping , Versfeld Niek J. , Green David M.
Published: October 2005

In the same-different task there are two stimulus alternatives, and the decision maker must decide whether they are the same or different. Two likelihood-ratio decision rules are derived by treating the stimuli as Gaussian random variables with means that depend on the stimulus conditions. They depend on the correlation between the observations. One decision rule is optimum when the observations are independent. This corresponds to experiments in which the standard is fixed. The other rule is optimum when the observations are highly correlated. Such a high correlation is likely to occur when the standard is randomly varied over trials. A COSS (conditional on a single stimulus) analysis provides a convenient way of estimating the rule actually used by a decision maker in a typical decision task. [Work supported by the U. S. AFOSR and the Dutch Ministry of Defence.]
Structural acoustic analysis of structures with frequency-dependent damping treatments
Authors: Everstine Gordon C. , Cheng Raymond S.-C.
Published: October 2005

An approach is described for computing the direct frequency response of structural acoustic systems containing frequency-dependent materials or damping treatments. The frequency dependence of the damping material is treated by approximating its modulus with a second-order polynomial so that the stiffnesses associated with the constant, linear, and quadratic terms can be combined with the stiffness, damping, and mass matrices assembled for the rest of the structure. The mathematically complex stiffness, damping, and mass matrices that result can then be used in coupled finite element/boundary element structural acoustic analyses using a direct frequency sweep.
Targets in nonuniform acoustic environments
Authors: McDaid E. P. , Gillette D. , Barach D.
Published: October 2005

The capability to numerically model the sound scattered from a target in a dispersive environment has been developed. The target has been rigorously coupled to its environment by combining a finite element formulation of the target shell dynamics with a Helmholtz Integral Equation formulation of fluid loading and a normal mode formulation of sound propagation. While the computational cost of obtaining numerical solutions to this problem are awesome, some simple approximations can result in significant savings. In particular, neglect of Fresnel diffraction and boundary-induced multiple scattering permits a particularly simple and tractable reformulation of the problem. The reported work makes the following contribution to solving the problem of target-environment interaction: the errors caused by neglecting Fresnel diffraction terms and multiple scattering are numerically evaluated. The significance of this contribution is that it will provide guidance as to when these useful approximations can be applied to speed calculations while still maintaining satisfactory fidelity.
Numerical estimates—based on ray methods—for the scattered acoustic fields of pairs of targets
Authors: Gaunaurd G. C. , Huang H. , Strifors H. C.
Published: October 2005

The sounds fields scattered by a multiplicity of spherical bodies can be predicted in an exact analytic fashion. This was illustrated in a previous presentation (cf., the paper entitled ‘‘Exact Analysis of the Acoustic Scattering Produced by Arbitrarily Insonified Bodies Near Boundaries,’’ by Huang and Gaunaurd (in this session), which also included the exact analysis for two spheres in free space, at arbitrary frequencies, and accounting for multiple scattering interactions. That work was mainly concerned with the low- and mid-frequency regimes. However, at the high frequencies for which certain asymptotic approximations begin to hold true, one can construct relatively simple-looking expressions for the sound fields scattered by pairs of spherical bodies. These expressions become even simpler looking for bodies separated by distances larger than their sizes. Even when this is not the case, the asymptotic results seem to hold just as well. The final estimates are: (a) evaluated numerically, (b) compared to (now available) exact analytic solutions, (c) graphically displayed, and (d) physically interpreted. These estimates are the acoustical counterparts of analogous ones in the radar literature and they are useful to determine the minimal separation at which remote sensors can actually discriminate two (or more) interacting neighboring targets. [Work partially supported by the NSWC IR Program.]
Linking speech and language processing through prosody
Authors: Ostendorf M. , Veilleux N. , Hendrix M. , Macannuco D.
Published: October 2005

In most spoken language understanding systems today, the interface between speech recognition and natural language processing is little more than a serial connection. The top scoring recognition results (a list of sentence hypotheses) are passed to the natural language system, which typically processes these taking the highest ranking recognition hypothesis that is interpretable and ignoring any acoustic information other than rank. The system can be improved by choosing the utterance interpretation based on a combination of the acoustic likelihood of a hypothesized word string and an associated language score (e.g., parse probability or semantic preference score). However, further gain is offered by incorporating a prosody score as well, since prosody can provide a link between syntactic/semantic structure and certain types of acoustic variability. In this talk, two types of computational models of prosody will be described that integrate acoustic cues with text features based on recognizer sentence hypotheses and associated parser outputs. Also, methods for using prosodic information in automatic speech understanding systems will be described. Experimental results with speech understanding in the ATIS domain will be described for three different aspects of prosody: prosodic phrasing, prominence placement, and cues to disfluencies. [Work supported by NSF and ARPA.]
Computational aids for the study of prosody
Authors: Wightman Colin W. , Talkin David
Published: October 2005

In the past few years, corpus-based methods of inquiry have yielded significant insights into the structure and role of prosody in human speech. Efforts are currently underway to discover the relationship between prosody and syntax and discourse, and to develop automated speech processing systems (both for synthesis and recognition) that take advantage of the information contained in prosody. These efforts, however, are critically dependent upon the availability of large speech corpora in which the relevant prosodic phenomena have been consistently transcribed. If the development of such a corpus is to be cost effective or, indeed, if prosodic cues are to be detected in automated systems, computational tools that facilitate and, where possible, automate the transcription process must be made available. In this paper, some the tools currently available will be presented and their performance and utility reviewed. In particular, a new tool for generating accurate, time-aligned phonetic transcriptions of spontaneous speech will be discussed, and quantitative measures of its performance will be presented. Integration of this tool into a comprehensive prosodic labeling environment will also be discussed.
Acoustic study of verbal hesitations, filled pauses, and unfilled pauses in spontaneous speech
Authors: Daly Nancy A.
Published: October 2005

Extemporaneously generated speech often contains verbal hesitations, filled pauses and unfilled pauses, reflecting the speaker’s uncertainty in formulating sentences, as in ‘‘Where iiiiiis um the nearest bank.’’ This study attempts to describe their acoustic properties using a subset, consisting of 3167 utterances from 66 speakers, of the spontaneous speech voyager urban navigation corpus, based on time-aligned orthographic and phonetic transcriptions. There are 564 verbal hesitations, 2518 unfilled pauses, and 148 filled pauses, and they are concentrated in 49.6% of the corpus utterances. 74.4% of the unfilled pauses occur in isolation, and their durations are longer when they cooccur with verbal hesitations and filled pauses by 46.1% and 363.9%, respectively. Over 70% of the verbal hesitations and filled pauses are followed by unfilled pauses, and they are longer than their isolated counterparts. Thus the results suggest that there may be a mutually enforcing effect among these acoustic events. An attempt has been made to identify verbal hesitations and filled pauses based on relative duration, proximity to silence, and relative mean F0 , using regression tree analyses, and a classification accuracy of approximately 70% on unseen data has been achieved.
Automatic intonation stylization using a model of pitch perception
Authors: d’Alessandro Christophe R. , Mertens Piet
Published: October 2005

Automatic analysis of intonation is a crucial step towards the long awaited automatic transcription of intonation. A new psychoacoustic model of pitch perception for short tones [d’Alessandro and Castellengo, J. Acoust. Soc. Am. 95, 1617–1630 (1994)] is applied to the problem of automatic F0 stylization for syllable-sized units in French. The aim is to compute one or several psychologically justified tonal movements for each syllable. An important property of the model is that it models perception prior to any process of linguistic categorization. The model was implemented in a computer program for tonal analysis. This program contains several procedures: (1) F0 detection and voiced/unvoiced decision; (2) syllabic segmentation, based on loudness and zero-crossings; (3) pitch integration using the model of pitch perception; (4) glissando rate computation, and static/dynamic tone decision; (5) differential glissando rate for complex pitch patterns computation. The analysis procedure described above has been tested through resynthesis. The modelized pitch contours delivered at the analysis stage were taken as input of a linear predictive coding (LPC) synthesizer. The perceptual equivalence between LPC natural speech and LPC modelized speech was demonstrated using a same/different test paradigm, on a corpus of read and spontaneous French.
Radiation-pressure-induced capillary waves
Authors: Glynn Holt R. , Trinh Eugene H.
Published: October 2005

Capillary waves are often observed on the surface of acoustically levitated drops, bubbles, and shells. The onset and parameter dependences of such waves on bubbles and shells have been investigated. Via video and light scattering it has been established that the mechanism for converting the acoustic energy into wave motion is the Faraday instability. Measurements of pattern formation, fluctuation, and transition will be presented, as well as investigations of temporal dynamics. In particular, the transition to spatiotemporal chaos and/or turbulence is investigated. The shell is the primary ‘‘laboratory’’ for turbulence, since, for certain wall thicknesses and levitation pressures, the entire surface becomes covered with apparently turbulent capillary waves, accompanied by a reduction of the gravity-induced pooling. The potential uses of this knowledge to produce more uniform coatings in certain applications will be discussed. [Work supported by NASA.]
Instability of an interface between two fluids in an acoustic field
Authors: Shi Tao , Apfel Robert E.
Published: October 2005

The interfacial instabilities between two fluids under the action of gravity, interfacial tension, and acoustic radiation pressure is examined, with particular attention given to instabilities at a flat interface. As an example of the instability at a curved surface, the break-up of a liquid drop levitated in air in a standing acoustic field has been studied using the boundary integral method. The nature of the instabilities at a flat interface is a more fundamental question. The theory to predict the threshold of the instability has been derived. Unlike results for static electromagnetic fields, the threshold depends not only on the properties of fluids but also on the frequency of acoustic waves. Experiments on the instabilities at the interface of carbon-tetrachloride and water were performed with a PZT disc transducer with a resonant frequency of 2.11 MHz. At a critical ‘‘threshold’’ level, a disturbance on the interface quickly grows into several sharply pointed peaks, a phenomenon similar to ‘‘Taylor’s cones’’ observed in an electric field. [Work supported by JPL through Contract No. 958722.]
Experimental study of streaming in ultrasonic levitators
Authors: Trinh Eugene H.
Published: October 2005

Flow visualization studies of streaming flow fields in 20- and 37-kHz ultrasonic levitators have revealed patterns of circulation of increasing complexity as the sound pressure level is increased. At relatively modest sound pressure level on the order of 145 dB (re: 0.0002 μbar), a straightforward superposition of the background circulation induced by the levitator components and enclosure and of the outer streaming flow associated with a levitated sample, provides a clear explanation for the observed flow field morphology. At the sound pressure level above about 155 dB, the resulting flow morphology cannot be rationalized so easily because the initially laminar background field becomes disordered with imbedded eddies. When a levitated sample is locally heated, the resulting pattern is complicated even at low sound pressure level. A qualitative interpretation of this evidence in terms of the existing boundary layer streaming theory is presented. [Work supported by NASA.]
Acoustic microcavitation prompted by smooth spherical microparticles
Authors: Madanshetty Sameer I.
Published: October 2005

Acoustic microcavitation in water is studied primarily at 1 MHz and 1% duty cycle. To detect cavitation two kinds of acoustic detectors are used. The first one is an unfocused, untuned 1-MHz receiver transducer that serves as a passive detector. The other one is a focused 25-MHz transducer that is used in the pulse-echo mode and is called the active detector. Cavitation itself is brought about by a focused LZT-1, 1-MHz crystal driven in pulse mode. The active detector is arranged confocally with respect to the cavitation transducer. Both the interrogating pulse and the cavitation pulse arrive simultaneously at the common focus, which is the region of cavitation. With the test chamber filled with clean water no cavitation is observed, even when the cavitation transducer is driven to give its peak output. Cavitation is, however, observed when microparticles are added to the host water. How these smooth, spherical, monodispersed, microparticles give rise to cavitation is described with some predictions, and corroborating experiments. It is found that the weak, high-frequency detection field affects the cavitation environment significantly.
Gap detection by human infants
Authors: Werner Lynne A. , Mancl Lisa R. , Constantino Janelle
Published: October 2005

In a previous study [L. A. Werner etal., Child Devel. 63, 260–272 (1992)], it was found that human infants are very bad at detecting gaps in broadband noise, with gap detection thresholds of about 50 ms. An adaptive method was used to estimate thresholds. Here, d′ is estimated in blocks of 30 trials at a fixed gap duration (ranging from 5 to 70 ms) and a psychometric function is constructed for each infant to estimate the gap detection threshold. A 30-dB spectrum level noise was presented continuously. On signal trials, a series of 12 gaps, 500 ms apart occurred; on no-signal trials, the noise continued uninterrupted. An observer-based method was used to test the infants. Three- and six-month-old infants were tested. Infants’ psychometric functions for gap detection were quite variable in slope: some infants had functions with ranges of about 10 ms, while a few infants had ranges of 30 ms. As in the previous work, infant thresholds remained quite poor; there was little change in threshold between 3 and 6 months of age. [Work supported by NIH.]
An asymmetric roex filter model for describing detection of silent temporal gaps as a function of marker frequency separation
Authors: Formby C. , Sherlock L. P. , Forrest T. G.
Published: October 2005

Formby and Forrest [J. Acoust. Soc. Am. 89, 830–837 (1991)] measured temporal gap detection (TGD) thresholds as function of frequency separation between two sinusoids marking a silent temporal gap. Their results were fit well with a roex-filter model. Filter estimates (1) were based on TGD thresholds for post-gap marker frequencies F2≥ pre-gap marker frequencies F1 and (2) assumed symmetry of the roex-filter function for F2≤F1. In a related study, TGD thresholds were measured with narrow-band markers [Formby etal., J. Acoust. Soc. Am. 93, 1023–1027 (1993)]. The resulting thresholds raised doubts about the earlier assumption of filter symmetry. To resolve this issue, TGD thresholds were measured for F2 frequencies presented above and below F1=500 and 4000 Hz for a range of marker and masker presentation levels. Roex functions fitted for conditions of F2≥F1 were typically shallower than those fitted for conditions of F2≤F1. The same qualitative patterns of filter asymmetry have also been found and will be reported for TGD thresholds simulated by single- and multi-channel auditory models [Forrest and Formby; Heinz etal., companion papers at this meeting]. [Research supported by NIH.]
Temporal gap detection thresholds in sinusoidal markers simulated with a single-channel envelope detector model
Authors: Forrest T. G. , Formby C.
Published: October 2005

The temporal envelope detector [N. Viemeister, J. Acoust. Soc. Am. 66, 1364–1380 (1979)] is a good single-channel model for auditory temporal processing. The model has three stages: (1) an initial bandpass filter, followed serially by (2) a nonlinear element, and (3) a lowpass filter. A digital version of the model using a max–min decision rule [Forrest and Green, J. Acoust. Soc. Am. 82, 1933–1943 (1987)] at the model’s output has been used to simulate temporal gap detection (TGD) thresholds for silent temporal gaps in pairs of sinusoidal markers (F1 and F2). Simulated TGD thresholds were measured as a function of marker frequency separation in the same manner as described in psychophysical experiments [Formby etal., companion paper at this meeting]. Simulated TGD thresholds depended directly on marker frequency separation. For a given frequency separation, simulated TGD thresholds were minimized by centering the symmetrical, first-stage roex filter between the two marker frequencies. This single-channel model consistently yielded lower TGD thresholds than did human listeners, but qualitatively simulated an asymmetry in the psychophysical thresholds where the relative change in the TGD threshold was smaller when the frequencies of F2≥F1 and larger for F1≥F2. [Research supported by NIH.]
Time-interval patterns in damped and ramped sinusoids
Authors: Patterson Roy D. , Akeroyd Michael A.
Published: October 2005

When a sinusoid with a temporally asymmetric envelope is reversed in time, the power spectrum of the sound does not change but the timbre often does. When the envelope is a repeated, damped exponential, the forward and backward versions are discriminable over a wide range of carrier frequencies (400–4800 Hz) and half-lives (1–64 ms) [R. D. Patterson, J. Acoust. Soc. Am. 93, 2293 (A) (1993)]. The earlier experiment has been replicated with a fixed number of carrier periods per envelope period to test the hypothesis that the sound of a sinusoid is more closely related to the number of phase-locked time intervals at the carrier-period than it is to the energy in the carrier channel. Time-interval histograms were calculated from auditory-image representations of the stimuli. The ramped stimuli generate more carrier-period activity in cases where the ramped stimulus is identified as more sinusoidal. Much of this activity is in channels well above and below the carrier channel indicating the time-interval models will probably provide a better basis for timbre perception than spectral models. [Work supported by the UK Medical Research Council and DRA Farnborough.]
Discrimination of time-asymmetric-modulated noise
Authors: Akeroyd Michael A. , Patterson Roy D.
Published: October 2005

Modulated noises with asymmetric temporal envelopes have identical long-term power spectra, yet sound very different. A ‘‘ramped’’ noise—constructed by amplitude modulating a wideband noise with a repeating exponential rise function—sounds like a repeating noise, whereas a ‘‘damped’’ noise—constructed with the same modulation function, but time-reversed—sounds like a snare drum struck by brushes. A damped noise has none of the usual timbre associated with a wideband noise. In experiment 1, listeners reliably discriminated a damped noise from a ramped noise, for modulation half-lives from 2 to 64 ms at modulation frequencies of 10 and 20 Hz. At higher frequencies (40 and 80 Hz) performance deteriorated. The maximum noise spectrum level was 52 dB SPL. The experiment was replicated at 32 and 12 dB SPL, and gave essentially the same results. It is proposed that the sounds’ timbre was used by listeners as the decision statistic. [Work supported by an MRC Research Studentship to the first author and by a DRA grant.]
Perception of sinusoid ‘‘ramps’’ and ‘‘damps’’ by the goldfish
Authors: Fay R. R. , Patterson Roy
Published: October 2005

A stimulus generalization paradigm combined with classical respiratory conditioning investigated the perceptual tonal qualities of repeated, asymmetrically shaped tone bursts (‘‘ramps’’ and ‘‘damps’’) in the goldfish. Ramps are tone bursts with a slow rise and a relatively fast fall envelope (raised cosine-shaped). Damps are ramps played backward. Human listeners report that ramps sound more tonelike than damps, in spite of their identical spectra. Groups of animals were given 40 classical conditioning trials to 6 s trains of either ramps or damps (400-Hz carrier) repeated at 20/s, and then tested for generalization to eight continuous pure tones above, below, and including 400 Hz. A relatively large generalization response in the region of 400 Hz indicates a perceptual similarity between the ramp or damp conditioning stimulus and a sinusoid. Differences in the magnitude of generalization between groups conditioned to ramps or damps indicate that goldfish perceive ramps as slightly more tonelike than damps. Implications for the mechanisms underlying human perception of ramps and damps are discussed. [Work supported by the NIH, NIDCD.]
The relationship between the difference limen (DL) in intensity and the masked threshold of sub-critical bandwidth signals
Authors: Sydorenko Mark R. , Allen Jont B.
Published: October 2005

Several studies in the literature have reported that the masked threshold in the presence of a tone masker versus a narrow band of noise of equal intensity can differ by as much as 20 dB. This difference is known to be important in music coders, and will probably turn out to be important in future speech coders. This presentation describes results that show that the difference in masking ability of tone versus noise maskers is due to the difference in acoustic intensity when a probe is added to these maskers. In addition, a series of psychoacoustic experimental results are reported that suggest a fundamental relationship between the just-noticeable difference in intensity (jndI) for a sub-critical bandwidth masker and the masked threshold of a tone. This result is a quantitative demonstration of the relationship between jnds and masking originally hypothesized by Miller [G. A. Miller, J. Acoust. Soc. Am. 60, 115–142 (1947)]. It was concluded that (a) the difference in masking ability of tones versus sub-critical bandwidth noise depends on the phase relationship between the probe and masker and (b) that the masked threshold for a tone in a two-interval forced choice (2IFC) procedure can be predicted from the jndI of the masking stimulus.
Some detection-theory models for relating loudness and intensity discrimination
Authors: Schlauch Robert S.
Published: October 2005

The relation between loudness and intensity discrimination was studied in three normally hearing listeners in quiet and in two levels of broadband noise. Intensity discrimination was measured for gated and continuous standards as a function of level. Loudness balances were obtained as a function of level as well. Forced-choice procedures were used for all measurements. Just-noticeable differences (jnd’s) were elevated in noise conditions, relative to those in quiet, when comparisons were made for equal SPLs. Subjects showed loudness recruitment in the noise conditions. The framework for the models is an extension of the work reported by McGill and Goldberg [J. Acoust. Soc. Am. 44, 576–581 (1968)]. A model based on the subjective analog of Weber’s law predicted smaller jnds in the shifted-threshold conditions than in quiet. By contrast, a neural counting model, based on the assumption that the neural noise that limits discrimination is related to the spontaneous firing of units not activated by the stimulus, predicts the correct relation for all conditions. The neural counting model also predicts Weber’s law for intensity discrimination of broadband noise.
Is ‘‘recalibration’’ of loudness related to short-term auditory fatigue?
Authors: Marks Lawrence E.
Published: October 2005

After listening to tone sequences containing brief, relatively low-SPL signals at one frequency (f1) and higher-SPL signals at another frequency (f2), tones at f1 appear relatively louder than tones at f2. Similar changes in loudness occur after exposure to only the higher-SPL signals. Such changes have been called ‘‘recalibration’’ [L. E. Marks, J. Exp. Psychol.: Hum. Percept. Perf. 20, 1–15 (1994)], and they resemble the temporary threshold shifts of short-term auditory fatigue. But fatigue is widely viewed to involve peripheral processes, whereas recalibration, a complex phenomenon also observed in other sensory modalities, may rely on central processes. Consequently, two experiments examined whether ‘‘recalibration’’ shows interaural transfer. Subjects compared the loudness of 500- and 2500-Hz signals before and after being presented sequences of monaural exposure tones that were themselves either listened to passively or judged for loudness. Both experiments showed substantial recalibration in the ear ipsilateral to the exposure stones, and smaller but still clearly evident recalibration in the contralateral ear. Thus recalibration may involve central mediation and, if so, may then rely on mechanisms differing at least in part from those underlying auditory fatigue. [Work supported by NIH.]
Dichotic loudness of tone-noise complexes
Authors: Hellman Rhona P.
Published: October 2005

The overall loudness of dichotic sounds composed of a tone to one ear and a low- or high-pass noise to the other ear, was determined by absolute magnitude estimation for groups of ten listeners with normal hearing. Within a 30-dB range above 70 dB, the sound pressure levels (SPL) of the tone and noise were independently varied in 5-dB steps. Results showed that total loudness was generally a nonmonotonic function of the overall SPL of the complex. Moreover, the obtained loudness functions were steeper than those typically measured separately for pure tones and broadband noise. Instead of the standard slope of 0.60 (log loudness as a function of SPL), overall slopes ranged from 0.75 to 0.86. For a given complex, the rate of loudness growth was dependent on the relative loudness of the component stimuli. Loudness increased more rapidly with overall SPL when the tone and noise were nearly equal in loudness than when they were markedly unequal. In contrast to earlier results for the same stimuli presented together binaurally [R. P. Hellman, J. Acoust. Soc. Am. 75, 209–218 (1984)], the current findings suggest that dichotic loudness is mainly based on the summed output from each ear without the effects of intraaural interactions. [Work supported by NIH.]
Individual, sex, and ear differences in measures of overshoot and psychophysical two-tone suppression
Authors: Wright Beverly A.
Published: October 2005

Notched-noise overshoot and two-tone suppression were measured at 1000 Hz using standard procedures in 40 normal-hearing, psychoacoustically naive adults, counterbalanced for sex and listening ear. The threshold variability across listeners for the same condition often exceeded 30 dB. In the overshoot conditions, the signal was an average of 7 dB harder to detect when it was presented at the onset of a 500-ms masker (masker notch width: 1000 Hz) than when signal onset was delayed by 400 ms. Across listeners, this overshoot effect ranged from 1 to 17 dB. In the suppression conditions, for some listeners, the forward-masked signal was as much as 15 dB easier to detect when the tonal on-frequency masker (1000 Hz) and suppressor (1150 Hz) were presented together, than when the masker was presented alone, indicating suppression. For other listeners, however, presenting the masker and suppressor together increased thresholds by as much as 13 dB, indicating additional masking (negative suppression). The average amount of suppression was −1.0 dB. ANOVA tests shoed significantly more overshoot and less suppression in females than males, and significantly higher thresholds in left than right ears in the overshoot conditions. Results will be discussed in terms of possible individual, sex, and ear differences in efferent inhibition. [Work supported by NIH.]
The effect of coherent frequency change on signal detectability
Authors: Chen Xiaofen , Cohen Marion F.
Published: October 2005

Coherent frequency change of a signal and masker can result in poorer detectability of the signal [M. F. Cohen and X. Chen, J. Acoust. Soc. Am. 92, 766–772 (1992)]. To further demonstrate the independence of this effect from the influences of spectral or harmonic factors, the following experiment was performed. Detectability of a 125-ms single component signal, of varying frequencies, was measured in the presence of three-component simultaneous masker. The signal and masker were either coherently frequency modulated (downglide), or were unmodulated (sinusoids), and were either harmonically related or not. The data show that the signal having coherent frequency change is the more difficult to detect for both harmonic and nonharmonic stimuli, even though the spectral and harmonic relationship between signal and masker are unchanged by the coherent modulation. In addition, this effect occurs even for signal frequencies spectrally distant from the masker. The experiment was repeated with signal modulating phase reversed to the masker modulator, i.e., an upward glide. The data show that the signal is easiest to detect in this condition. These results suggest that coherent frequency change among stimulus components can affect signal detectability. [Work supported by AFOSR.]
Active control of inner field and radiation of a ducted noise source
Authors: Kang Seong-Woo , Kim Yang-Hann
Published: October 2005

A ducted noise source such as a fan with a duct can either be controlled by sound absorptive material or by a active noise control system that has some advantage to control a rather low frequency of noise. To investigate what is happening inside of a duct when a certain active noise control strategy is applied, an analytic solution has been obtained based on a Green’s function for a duct that has two sources and frequency-dependent termination [Y.-H. Kim and S.-W. Kang, J. Acoust. Soc. Am. 94, 473–490 (1993)]. A number of cost functions are studied for active control of the inner and outer field of the duct. The cost functions investigated include the total acoustic potential energy, total acoustic power output, acoustic absorbing power of the control source, active intensity; that is, power flow, propagating and reflected acoustic potential energy, and pressure at a fixed point. The control effects by minimization of each cost function are investigated for various termination conditions that mean the loading condition of the exterior acoustic field. The location of a control source is also considered to see the effect upon the acoustic characteristics inside and outside of the duct. The numerical results that visualize the sound energy and intensity distribution are demonstrated.
More on flow fields driving a contiguous acoustic field—A simple physical argument in terms of vorticity alone, yielding Möhring’ form
Authors: Powell Alan
Published: October 2005

In J. Acoust. Soc. Am. 36, 830–832 (1964): ‘‘The sound radiation from a slightly compressible [compact] flow can be calculated by estimating the pressure that would exist at an enclosing [spherical] surface just as if the fluid were incompressible, this [hydrodynamic] pressure then being inserted in the appropriate point-multipole expression for the sound radiation in the [surrounding] compressible fluid...the [resultant] far field sound pressure is the same as for the theory of vortex sound,’’ for which the quadrupole strength contribution is the volume integral of yx (ζ×u)″x, where u=velocity, ζ=∇×u=vorticity, y=coordinate in flow, x=far-field point, ( )x≡(vector)⋅x/x, and ( )′≡∂/∂t( ). Now work in terms of induced velocity instead of pressure, starting with the so-called Biot–Savart law. The result is the integral of (1/2)(y×ζ″)x for the dipole contribution and of (1/3)yx(y×ζ‴)x for quadrupoles. These involve only the vorticity and not the velocity. These final results were first given by Möhring [J. Fluid Mech. 85, 685–691 (1978)] and later by Kambe [J. Fluid Mech. 173, 643–681 (1986)], both using more complex methods.
A simple harmonic oscillator teaching apparatus with active velocity feedback
Authors: Keolian Robert M.
Published: October 2005

A tuning fork instrumented with an electromechanical drive and pickup is used to show many of the properties of the simple harmonic oscillator as well as self-maintained oscillations (negative damping) and the active damping of vibrations (positive damping). The apparatus is used both as a teaching laboratory experiment and as a lecture demonstration. The drive and pickup consist of coils of wire taken out of relays, and small but powerful rare-earth magnets available at Radio Shack. An eraser between the tines can be used to vary the passive Q. By amplifying the signal from the pickup coil, compensating for inductively induced phase shifts, and applying this signal back to the drive coil, a force proportional to the velocity of the tines can be used to subtract from or add to the passive damping, resulting in a self-excited tuning fork, a tuning fork with infinite Q, or one with very low Q.
Of flaming tubes and spinning cups
Authors: Everbach E. Carr
Published: October 2005

An enhanced version of the flame tube will be demonstrated to help students visualize compressional standing waves in a pipe. A more surprising demonstration of the ability of acoustic waves to cause cups to spin, and why it works, will also be presented.
Demonstrations of coupled oscillators
Authors: Edmondson Charles A. , Wilbur Thomas M. , Korman Murray S.
Published: October 2005

Coupled oscillators involving a small or large number of elements can be used to study beats, normal modes, wave motion, and dispersion effects. Several experimental systems, each using a mechanical apparatus, will be demonstrated. The motion studies will be enhanced using transducers (such as strain gauges or accelerometers) on various elements. Transducer signals will be FFT analyzed and displayed to reveal the frequency characteristics of the normal modes. The coupled oscillators will involve pendulums and springs or masses on springs resting on an air track. A system of N nonlinear coupled pendulums will be used to demonstrate the motion of a soliton. For the finale, a metallic pingpong ball will oscillate between two oppositely charged parallel metal plates.
Acoustic resonators with variable nonuniformity
Authors: Denardo Bruce
Published: October 2005

The standard method of altering the fundamental frequency of a one-dimensional resonator is to change the effective length. Another method is to introduce or alter the nonuniformity (variation in cross-sectional area), which can be accomplished very simply. If one end of an open-open nonuniform resonator is slapped by hand, the fundamental frequency of the open-closed system is heard. If the other end is slapped, the fundamental frequency is in general different. Furthermore, if the resonator is cut in half along its length, and one set of ends is placed on a flat surface, the fundamental frequency is heard if the halves are clapped together. Because the configuration can be altered by turning either or both halves upside down, as many as four different frequencies can be generated. The dependence of frequency upon nonuniformity can be understood and calculated perturbatively by Rayleigh’s energy method, or more generally by the theorem of adiabatic invariance. For the simplest case of piecewise-uniform resonators, improved accuracy is obtained by matching solutions of the wave equation in the uniform regions. With this theory, resonators have been accurately designed to yield musical intervals. Demonstrations of these ‘‘slappers’’ and ‘‘clappers’’ will be given.
Demonstrations on the nonlinear scattering of crossed focused sound beams in the presence of turbulence
Authors: Korman Murray S.
Published: October 2005

The interaction of two, mutually perpendicular crossed continuous wave ultrasonic beams, overlapping (at a common focal point) and interacting in the presence of turbulence in water is demonstrated by the detection of the far-field radiated sum frequency component. This scattered sum frequency pressure p+ exhibits a time averaged Doppler shift and frequency broadening due to scattering from turbulent eddies (in the overlap region) that are not stationary in time. The angular dependence on the frequency spectrum of the scattered sound makes it possible to measure the mean flow velocity and the so-called higher order turbulent velocity correlations. The audience will be able to hear the frequency demodulation of the sum frequency signal by using an FM radio receiver. The radio receiver is coupled to the acoustic receiving transducer and is tuned to detect wideband FM centered on f+=f1+f2. Here, f1=2.1 MHz and f2=1.9 MHz are the respective primary frequencies. The demodulation signal will change as the scattering angle is changed or the overlap region is moved to a new point in the turbulent flow (which is generated by a submerged jet). [Work supported in part by the Naval Academy Research Council.]
How to attack a piano attack—An application of time-frequency estimation
Authors: Pielemeier William J. , Guevara Rowena , Wakefield Gregory H.
Published: October 2005

Piano signals present both rapid attacks and compact clusters of inharmonic partials and are, therefore, good examples of worst-case signals for many techniques of short-time spectral representation. This paper demonstrates the application of a recently developed time-frequency distribution [Pielemeier and Wakefield, J. Acoust. Soc. Am. 92, 2430 (1992); J. Acoust. Soc. Am. 94, 1806 (1993)] to the analysis of piano and other challenging signals in musical acoustics. A step-by-step analysis of the piano signal using the time-frequency distribution is presented with caveats concerning data interpretation. Both visual representations and estimators for the instantaneous frequency and power of the partials are compared with those obtained from Fourier series and from short-time Fourier analysis. The type and source of artifacts associated with each method are highlighted and issues of computational complexity are discussed. When applied properly, the proposed time-frequency distribution provides better resolution of the temporal and spectral properties of these musical signals than is achieved with either alternative. [Work supported by the EECS Department and by the Ford Motor Co.]
Musical signal spectrum analysis with the matrix pencil method
Authors: Laroche Jean
Published: October 2005

The matrix pencil spectrum analysis method provides an interesting alternative to classical Fourier analysis. The method is based on the assumption that the signal can be represented as a sum of exponentially damped sinusoids whose parameters (frequency, damping factor, amplitude and phase) are estimated by a matrix-based algorithm. In spite of its computational cost, the matrix pencil method achieves extremely good results in the case of highly damped signals, short data records, or close frequencies. The matrix pencil method is first briefly described and compared to other spectrum analysis techniques (Fourier transform, Prony method). Several implementations are proposed and their computational costs evaluated. The good performance of the method is illustrated by analysis examples involving low-frequency beats of piano tones. These beats are decomposed as sums of closely spaced damped sinusoids, which makes it possible to verify some of results on the coupling between orthogonal vibrating modes of strings [G. Weinreich, ‘‘Coupled piano strings,’’ J. Acoust. Soc. Am. 62, 1474 (1977)]. The matrix pencil method is also applied to the analysis of the bridge admittance of a guitar and is used to model the admittance as a set of simple mechanical systems in parallel.