Preceding phonetic context affects perception of nonspeech.
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Engine run-ups are part of the regular maintenance schedule at Vancouver International Airport. The noise generated by the run-ups propagates into neighboring communities, disturbing the residents. Active noise control is a potentially cost-effective alternative to passive methods, such as enclosures. Propeller aircraft generate low-frequency tonal noise that is highly compatible with active control. This paper presents a preliminary investigation of the feasibility and effectiveness of controlling run-up noise from propeller aircraft using local active control. Computer simulations for different configurations of multi-channel active-noise-control systems, aimed at reducing run-up noise in adjacent residential areas using a local-control strategy, were performed. These were based on an optimal configuration of a single-channel control system studied previously. The variations of the attenuation and amplification zones with the number of control channels, and with source/control-system geometry, were studied. Here, the aircraft was modeled using one or two sources, with monopole or multipole radiation patterns. Both free-field and half-space conditions were considered: for the configurations studied, results were similar in the two cases. In both cases, large triangular quiet zones, with local attenuations of 10 dB or more, were obtained when nine or more control channels were used. Increases of noise were predicted outside of these areas, but these were minimized as more control channels were employed. By combining predicted attenuations with measured noise spectra, noise levels after implementation of an active control system were estimated.
A sensitive, large-aperture schlieren optical instrument is applied to observe gas-dynamic phenomena at the exit of a trumpet. Shock waves are seen, especially for loud, high-pitched trumpet notes, and several illustrations are given. Microphone waveforms are given for representative examples. These shock waves arise from the shock-tube-like effect of the performer's intermittent breath pressure driving the cylindrical duct of the trumpet, and are the result of cumulative nonlinear acoustic propagation inside the trumpet bore. They are, however, very weak, traveling only marginally above the acoustic speed. In the 118-124 peak dB(A) range, they are near the weak limit of shock wave visibility by schlieren optics. The schlieren evidence confirms that the frequency of the emitted shock waves corresponds to the frequency of the note being played. Ancillary laminar and turbulent jet phenomena associated with the performer's breath are also visible in the images.
This paper presents a three-dimensional time-domain numerical model of the vibration and acoustic radiation from a guitar. The model involves the transverse displacement of the string excited by a force pulse, the flexural motion of the soundboard, and the sound radiation. A specific spectral method is used for solving the Kirchhoff-Love's dynamic top plate model for a damped, heterogeneous orthotropic material. The air-plate interaction is solved with a fictitious domain method, and a conservative scheme is used for the time discretization. Frequency analysis is performed on the simulated sound pressure and plate velocity waveforms in order to evaluate quantitatively the transfer of energy through the various components of the coupled system: from the string to the soundboard and from the soundboard to the air. The effects of some structural changes in soundboard thickness and cavity volume on the produced sounds are presented and discussed. Simulations of the same guitar in three different cases are also performed: "in vacuo," in air with a perfectly rigid top plate, and in air with an elastic top plate. This allows comparisons between structural, acoustic, and structural-acoustic modes of the instrument. Finally, attention is paid to the evolution with time of the spatial pressure field. This shows, in particular, the complex evolution of the directivity pattern in the near field of the instrument, especially during the attack.
The term informational masking has traditionally been used to refer to elevations in signal threshold resulting from masker uncertainty. In the present study, the method of constant stimuli was used to obtain complete psychometric functions (PFs) from 44 normal-hearing listeners in conditions known to produce varying amounts of informational masking. The listener's task was to detect a pure-tone signal in the presence of a broadband noise masker (low masker uncertainty) and in the presence of multitone maskers with frequencies and amplitudes that varied at random from one presentation to the next (high masker uncertainty). Relative to the broadband noise condition, significant reductions were observed in both the slope and the upper asymptote of the PF for multitone maskers producing large amounts of informational masking. Slope was affected more for some listeners and conditions while asymptote was affected more for others; consequently, neither parameter alone was highly predictive of individual thresholds or the amount of informational masking. Mean slopes and asymptotes varied nonmonotonically with the number of masker components in a manner similar to mean thresholds, particularly when the estimated effect of energetic masking on thresholds was subtracted out. As in past studies, the threshold data were well described by a model in which trial-by-trial judgments are based on a weighted sum of levels in dB at the output of independent auditory filters. The psychometric data, however, complicated the model's interpretation in two ways: First, they suggested that, depending on the listener and condition, the weights can either reflect a fixed influence of masker components on each trial or the effect of occasionally mistaking a masker component for the signal from trial to trial. Second, they indicated that in either case the variance of the underlying decision variable as estimated from PF slope is not by itself great enough to account for the observed changes in informational masking.
Rates of reverberative decay and frequency attenuation are measured within two Australian forests. In particular, their dependence on the distance between a source and receiver, and the relative heights of both, is examined. Distance is always the most influential of these factors. The structurally denser of the forests exhibits much slower reverberative decay, although the frequency dependence of reverberation is qualitatively similar in the two forests. There exists a central range of frequencies between 1 and 3 kHz within which reverberation varies relatively little with distance. Attenuation is much greater within the structurally denser forest, and in both forests it generally increases with increasing frequency and distance, although patterns of variation differ between the two forests. Increasing the source height generally reduces reverberation, while increasing the receiver height generally reduces attenuation. These findings have considerable implications for acoustic communication between inhabitants of these forests, particularly for the perching behaviors of birds. Furthermore, this work indicates the ease with which the general acoustic properties of forests can be measured and compared.
A commercial rheometer (Bohlin CVO120) was used to mechanically test materials that approximate vocal-fold tissues. Application is to frequencies in the low audio range (20-150 Hz). Because commercial rheometers are not specifically designed for this frequency range, a primary problem is maintaining accuracy up to (and beyond) the mechanical resonance frequency of the rotating shaft assembly. A standard viscoelastic material (NIST SRM 2490) has been used to calibrate the rheometric system for an expanded frequency range. Mathematically predicted response curves are compared to measured response curves, and an error analysis is conducted to determine the accuracy to which the elastic modulus and the shear modulus can be determined in the 20-150-Hz region. Results indicate that the inertia of the rotating assembly and the gap between the plates need to be known (or determined empirically) to a high precision when the measurement frequency exceeds the resonant frequency. In addition, a phase correction is needed to account for the magnetic inertia (inductance) of the drag cup motor. Uncorrected, the measured phase can go below the theoretical limit of -pi. This can produce large errors in the viscous modulus near and above the resonance frequency. With appropriate inertia and phase corrections, +/- 10% accuracy can be obtained up to twice the resonance frequency.
The detection of a tone added to a random-frequency, multitone masker can be very poor even when the maskers have little energy in the frequency region of the signal. This paper examines the effects of adding a pretrial cue to reduce uncertainty for the masker or the signal. The first two experiments examined the effect of cuing a fixed-frequency signal as the number of masker components and presentation methods were manipulated. Cue effectiveness varied across observers, but could reduce thresholds by as much as 20 dB. Procedural comparisons indicated observers benefited more from having two masker samples to compare, with or without a signal cue, than having a single interval with one masker sample and a signal cue. The third experiment used random-frequency signals and compared no-cue, signal-cue, and masker-cue conditions, and also systematically varied the time interval between cue offset and trial onset. Thresholds with a cued random-frequency signal remained higher than for a cued fixed-frequency signal. For time intervals between the cue and trial of 50 ms or longer, thresholds were approximately the same with a signal or a masker cue and lower than when there was no cue. Without a cue or with a masker cue, analyses of possible decision strategies suggested observers attended to the potential signal frequencies, particularly the highest signal frequency. With a signal cue, observers appeared to attend to the frequency of the subsequent signal.
A theory for the elastic scattering response from a cylinder insonified by a plane wave was previously derived by Faran. In the present paper, the empirical relationship between Faran's theory and measurements of backscatter coefficient from cylindrical targets using focused transducers is investigated. Experimental measurements of dependence of backscatter coefficient on frequency and diameter for nylon wires are reported. It is found that, under certain conditions (including weak, incoherent scattering), backscatter coefficient measurements from collections of cylindrical scatterers may be meaningfully compared with Faran's model predictions. At low frequencies, the theory and experimental measurements exhibit similar dependences on frequency and diameter, provided that the scatterers are not too densely packed. At higher frequencies, the fine structure of Faran's predictions becomes difficult to reproduce experimentally with a focused transducer. Implications regarding applications to characterization of trabecular bone are discussed.
Auditory scene analysis involves the simultaneous grouping and parsing of acoustic data into separate mental representations (i.e., objects). Over two experiments, we examined the sequence of neural processes underlying concurrent sound segregation by means of recording of human middle latency auditory evoked responses. Participants were presented with complex sounds comprising several harmonics, one of which could be mistuned such that it was not an integer multiple of the fundamental frequency. In both experiments, Na (approximately 22 ms) and Pa (approximately 32 ms) waves were reliably generated for all classes of stimuli. For stimuli with a fundamental frequency of 200 Hz, the mean Pa amplitude was significantly larger when the third harmonic was mistuned by 16% of its original value, relative to when it was tuned. The enhanced Pa amplitude was related to an increased likelihood in reporting the presence of concurrent auditory objects. Our results are consistent with a low-level stage of auditory scene analysis in which acoustic properties such as mistuning act as preattentive segregation cues that can subsequently lead to the perception of multiple auditory objects.
A method is proposed to model the interspeaker variability of formant patterns for oral vowels. It is assumed that this variability originates in the differences existing among speakers in the respective lengths of their front and back vocal-tract cavities. In order to characterize, from the spectral description of the acoustic speech signal, these vocal-tract differences between speakers, each formant is interpreted, according to the concept of formant-cavity affiliation, as a resonance of a specific vocal-tract cavity. Its frequency can thus be directly related to the corresponding cavity length, and a transformation model can be proposed from a speaker A to a speaker B on the basis of the frequency ratios of the formants corresponding to the same resonances. In order to minimize the number of sounds to be recorded for each speaker in order to carry out this speaker transformation, the frequency ratios are exactly computed only for the three extreme cardinal vowels [i, a, u] and they are approximated for the remaining vowels through an interpolation function. The method is evaluated through its capacity to transform the (F1,F2) formant patterns of eight oral vowels pronounced by five male speakers into the (F1,F2) patterns of the corresponding vowels generated by an articulatory model of the vocal tract. The resulting formant patterns are compared to those provided by normalization techniques published in the literature. The proposed method is found to be efficient, but a number of limitations are also observed and discussed. These limitations can be associated with the formant-cavity affiliation model itself or with a possible influence of speaker-specific vocal-tract geometry in the cross-sectional direction, which the model might not have taken into account.
The mandate of ASA Working Group S12/WG11 has been to develop "laboratory and/or field procedure(s) that yield useful estimates of field performance" of hearing protection devices (HPDs). A real-ear attenuation at threshold procedure was selected, devised, tested for one earmuff and three earplugs via an interlaboratory study involving five laboratories and 147 subjects, and incorporated into a new standard that was approved in 1997 [Royster et al., "Development of a new standard laboratory protocol for estimating the field attenuation of hearing protection devices. Part I. Research of Working Group 11, Accredited Standards Committee S 12, Noise," J. Acoust. Soc. Am. 99, 1506-1526; ANSI, S12.6-1997, "American National Standard method for measuring real-ear attenuation of hearing protectors" (American National Standards Institute, New York, 1997)]. The subject-fit methodology of ANSI S12.6-1997 relies upon listeners who are audiometrically proficient, but inexperienced in the use of HPDs. Whenever a new method is adopted, it is important to know the effects of variability on the power of the measurements. In evaluation of protector noise reduction determined by experimenter-fit, informed-user-fit, and subject-fit methods, interlaboratory reproducibility was found to be best for the subject-fit method. Formulas were derived for determining the minimum detectable difference between attenuation measurements and for determining the number of subjects necessary to achieve a selected level of precision. For a precision of 6 dB, the study found that the minimum number of subjects was 4 for the Bilsom UF-1 earmuff, 10 for the E.A.R Classic earplug, 31 for the Willson EP100 earplug, and 22 for the PlasMed V-51R earplug.
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Two experiments investigated whether listeners change their vowel categorization decisions to adjust to different accents of British English. Listeners from different regions of England gave goodness ratings on synthesized vowels embedded in natural carrier sentences that were spoken with either a northern or southern English accent. A computer minimization algorithm adjusted F1, F2, F3, and duration on successive trials according to listeners' goodness ratings, until the best exemplar of each vowel was found. The results demonstrated that most listeners adjusted their vowel categorization decisions based on the accent of the carrier sentence. The patterns of perceptual normalization were affected by individual differences in language background (e.g., whether the individuals grew up in the north or south of England), and were linked to the changes in production that speakers typically make due to sociolinguistic factors when living in multidialectal environments.
Internal and external noise surveys have been carried out around schools in London, UK, to provide information on typical levels and sources to which children are exposed while at school. Noise levels were measured outside 142 schools, in areas away from flight paths into major airports. Here 86% of the schools surveyed were exposed to noise from road traffic, the average external noise level outside a school being 57 dB L(Aeq). Detailed internal noise surveys have been carried out in 140 classrooms in 16 schools, together with classroom observations. It was found that noise levels inside classrooms depend upon the activities in which the children are engaged, with a difference of 20 dB L(Aeq) between the "quietest" and "noisiest" activities. The average background noise level in classrooms exceeds the level recommended in current standards. The number of children in the classroom was found to affect noise levels. External noise influenced internal noise levels only when children were engaged in the quietest classroom activities. The effects of the age of the school buildings and types of window upon internal noise were examined but results were inconclusive.
This study quantifies sex differences in the acoustic structure of vowel-like grunt vocalizations in baboons (Papio spp.) and tests the basic perceptual discriminability of these differences to baboon listeners. Acoustic analyses were performed on 1028 grunts recorded from 27 adult baboons (11 males and 16 females) in southern Africa, focusing specifically on the fundamental frequency (F0) and formant frequencies. The mean F0 and the mean frequencies of the first three formants were all significantly lower in males than they were in females, more dramatically so for F0. Experiments using standard psychophysical procedures subsequently tested the discriminability of adult male and adult female grunts. After learning to discriminate the grunt of one male from that of one female, five baboon subjects subsequently generalized this discrimination both to new call tokens from the same individuals and to grunts from novel males and females. These results are discussed in the context of both the possible vocal anatomical basis for sex differences in call structure and the potential perceptual mechanisms involved in their processing by listeners, particularly as these relate to analogous issues in human speech production and perception.
The West Indian manatee (Trichechus manatus latirostris) has become endangered partly because of a growing number of collisions with boats. A system to warn boaters of the presence of manatees, based upon the vocalizations of manatees, could potentially reduce these boat collisions. The feasibility of this warning system would depend mainly upon two factors: the rate at which manatees vocalize and the distance in which the manatees can be detected. The research presented in this paper verifies that the average vocalization rate of the West Indian manatee is approximately one to two times per 5-min period. Several different manatee vocalization recordings were broadcast to the manatees and their response was observed. It was found that during the broadcast periods, the vocalization rates for the manatees increased substantially when compared with the average vocalization rates during nonbroadcast periods. An array of four hydrophones was used while recording the manatees. This allowed for position estimation techniques to be used to determine the location of the vocalizing manatee. Knowing the position of the manatee, the source level was determined and it was found that the mean source level of the manatee vocalizations is approximately 112 dB (re 1 microPa) @ 1 m.
Sentences spoken "clearly" are significantly more intelligible than those spoken "conversationally" for hearing-impaired listeners in a variety of backgrounds [Picheny et al., J. Speech Hear. Res. 28, 96-103 (1985); Uchanski et al., ibid. 39, 494-509 (1996); Payton et al., J. Acoust. Soc. Am. 95, 1581-1592 (1994)]. While producing clear speech, however, talkers often reduce their speaking rate significantly [Picheny et al., J. Speech Hear. Res. 29, 434-446 (1986); Uchanski et al., ibid. 39, 494-509 (1996)]. Yet speaking slowly is not solely responsible for the intelligibility benefit of clear speech (over conversational speech), since a recent study [Krause and Braida, J. Acoust. Soc. Am. 112, 2165-2172 (2002)] showed that talkers can produce clear speech at normal rates with training. This finding suggests that clear speech has inherent acoustic properties, independent of rate, that contribute to improved intelligibility. Identifying these acoustic properties could lead to improved signal processing schemes for hearing aids. To gain insight into these acoustical properties, conversational and clear speech produced at normal speaking rates were analyzed at three levels of detail (global, phonological, and phonetic). Although results suggest that talkers may have employed different strategies to achieve clear speech at normal rates, two global-level properties were identified that appear likely to be linked to the improvements in intelligibility provided by clear/normal speech: increased energy in the 1000-3000-Hz range of long-term spectra and increased modulation depth of low frequency modulations of the intensity envelope. Other phonological and phonetic differences associated with clear/normal speech include changes in (1) frequency of stop burst releases, (2) VOT of word-initial voiceless stop consonants, and (3) short-term vowel spectra.