bluuebunny/crossref_metadata_2025 huggingface dataset:
Deaf speaker's control of acoustic intensity
Authors: Mahshie James , Brandt Fred , Brunner Barbara
Published: October 2005

Control of acoustic intensity is required for both linguistic and paralinguistic aspects of speech communication. While generally recognized that deaf speakers experience difficulty learning to control acoustic intensity, the characteristics and mechanisms of intensity control among deaf speakers are not well understood. The objectives of the present research were to: (1) study and describe characteristics of intensity control by deaf individuals during speech production; (2) compare deaf and normal-hearing speaker's control of acoustic intensity; and (3) suggest possible mechanisms used by deaf speakers to control acoustic intensity. Deaf and normal-hearing speakers produced sustained vowels at different acoustic intensity and fundamental frequency levels. The average airflow, acoustic intensity, spectral characteristics, fundamental frequency, and characteristics of vocal fold oscillation were measured. Results showed that deaf speakers tended to use greater than normal air flow for all speaking tasks. While normal-hearing speakers tended to increase airflow with increased intensity, the deaf speakers often failed to produce such a distinction. However, certain systematic changes in laryngeal adjustments associated with increased acoustic intensity were observed in both groups of speakers. Possible mechanisms for intensity control will be discussed, along with clinical implications.
Vowel F1 as a function of speaker fundamental frequency
Authors: Syrdal Ann K. , Steele Shirley A.
Published: October 2005

The relationship between speaker fundamental frequency (F0) and first formant frequency (F1) was investigated. A wide F0 range was sampled in each of two sentence positions, using representative high and low vowels (/i/ and /ae/) in two phonetically controlled, interchangeable key words. Data were collected from four women and four men and processed with a 14th-order LPC analysis, using a correlation pitch tracker. The bark-transformed F0 ranges were equivalent for women and men. The relationships between F0 and F1 were similar for both sexes, although women consistently adjusted F1 frequency in /i/ to closely approximate F0 in their higher F0 range, whereas men typically did not. It was found that F1 increased significantly with increasing F0. Although the difference between bark-transformed F1 and F0 decreased significantly as F0 increased, a 3-bark critical distance consistently separated /i/ and /ae/ throughout the F0 range. [Work supported in part by NIH.]
A multibounce, single-scatter ray theoretic model for underice predictions
Authors: Tolstoy A. , Berman D. H. , Wright E. B. , Baer R. N.
Published: October 2005

This study describes the propagated and scattered acoustic field from a point source to a point receiver in an approximate underice environment characteristic of the deep Arctic Ocean. Our model considers both the azimuthal and vertical redistribution of scattered energy caused by a rough surface overlying an upwardly refracting, linear sound-speed profile. The model considers one nonspecular surface scattering per path (connecting the source with the receiver) coupled with any number of specular interactions, each of which suffers losses due to scattering into nonspecular directions, i.e., consistent with conservation of energy. The propagation model is based on ray theory, while the scattering model uses Twersky's scattering cross section for hard or soft planar surfaces roughened by hard or soft, hemispherical bosses. The model calculates intensity as a function of position and direction. Bistatic results at all frequencies show that for hard and soft surfaces the strongest scattered field is produced directly above the submerged source, and that the total scattered energy levels increase as frequency increases. Monostatic geometries and effects of distributions of boss sizes will also be considered. [Work supported by NRL and ONR.]
Elastic wave scattering in a horizontally stratified ocean environment with application to arctic propagation
Authors: Kuperman W. A. , Schmidt Henrik
Published: October 2005

A previously developed boundary perturbation method [W. A. Kuperman, J. Acoust. Soc. Am. 58, 365 (1975)] is extended to treat scattering at a randomly rough interface which separates viscoclastic media. This method is then combined with a full-wave treatment of sound propagating in a stratified ocean described by a system of liquid and elastic layers [H. Schmidt and F. B. Jensen, J. Acoust. Soc. Am. 77, 813–825 (1985)]. The net result of combining the extended boundary perturbation method with the full-wave solution technique is to define a set of effective potentials which, when inserted in the full-wave solution algorithm, yields the coherent components of the compressional and shear wave fields in the Kirchhoff approximation. The fields decay with range due to boundary roughness scattering into both incoherent compressional and shear waves. A natural outcome of this model is also the option to calculate the coherent reflection coefficient for an arbitrarily arranged layered system of liquid/solid media separated by randomly rough interfaces. Numerical examples of reflection coefficients and sound propagation in an ocean described by a stratified waveguide are presented. Emphasis is on arctic propagation demonstrating a new low-frequency attenuation mechanism resulting from scattering into shear waves at the upper and lower rough boundaries of an ice layer.
Rough surface scattering: Exact numerical methods compared with the Kirchhoff approximation
Authors: Thorsos Eric I.
Published: October 2005

Acoustic scattering has been calculated exactly using integral equation methods for deterministic and randomly rough surfaces. The surfaces are invariant in one direction and the Dirichlet (pressure release) boundary condition was used. The scattered field can be found exactly by solving an integral equation for the normal derivative of the pressure on the surface. This was done two ways by using linear integral equations of both the first and second kinds; the solutions were found to agree. Comparisons of the exact surface and scattered fields with those obtained using the Kirchhoff approximation were made. Significant differences in forward scattering were found between the Kirchhoff and exact results. This was true even when the Kirchhoff validity criterion, based on the radius of curvature, was well satisfied. Evidence will be presented to show that these differences are due to multiple scattering. [Work supported by ONR.]
Equivalence of perturbation theories for sound scattering by rough surfaces
Authors: Jackson Darrell R.
Published: October 2005

Some approaches to rough-surface perturbation theory employ the Rayleigh hypothesis, but a method based on the extinction theorem avoids this assumption and gives a series which is apparently different. One might expect the latter method to be superior, but Winebrenner and Ishimaru have shown numerically that the two series give equivalent results for scattering from sinusoidal surfaces [D. P. Winebrenner and A. Ishimaru, J. Acoust. Soc. Am. Suppl. 1 75, S73 (1984)]. We have shown that the two series are the same for arbitrary surfaces. This proof depends upon term-by-term reciprocity of the perturbation series.
Nearfield scattered acoustic waves in shallow-water waveguides with boundary topography
Authors: Chamuel Jacques R. , Brooke Gary H.
Published: October 2005

Experimental results are presented demonstrating the importance of backscattered, forward-scattered, and locally reactive acoustic fields created by shallow-water acoustic waveguide boundary topography. In a perfect waveguide, the transmission of a pulse may create a set of nondispersed waveguide pulses equivalent to the summation of highly dispersive normal modes. The presence of a boundary scatterer acts as a new delayed sound source in the waveguide. This new source produces a second set of delayed waveguide pulses. Each of the topographic features generates a new set of waveguide pulses delayed accordingly. The resulting acoustic field in the waveguide is complex. A technique has been devised to separate the original flat-waveguide over a periodic region creates local standing-wavelike behavior in the nearfield of the scatterers. Although the source-receiver range may be great, the distance between the receiver and the topographic features is small. The importance of the inhomogeneous normal modes increases as the receiver is moved closer to the scatterer. The received signal is greatly influenced by the relative position of the rough waveguide region with respect to both source and receiver. [Work supported by DREP.]
Near grazing forward scatter over a low roughness rigid ocean bottom
Authors: Medwin Herman , Dapos;Spain Gerald L.
Published: October 2005

When low-frequency sound interacts with a low roughness rigid surface at grating or near-grazing angles, a significant coherent boundary wave is generated by repeated forward scatter. The boundary wave extends several wavelengths into the fluid and affects both the phase velocity and the amplitude of the forward propagating sound near the surface. The propagation is subsonic and slightly dispersive and, under certain conditions, the reflection coefficient can be greater than unity. Laboratory model results have not been extended to quantify these boundary wave phenomena in terms of the average height and rms slope of steep-sloped randomly rough surfaces. Predictions are made of the consequences of boundary wave propagation at a rough rigid ocean bottom. [Research supported by the Office of Naval Research.]
Impulse noise repetition rate: Implications for the equal energy hypothesis
Authors: Henderson D. , Hamernik R. P. , Salvi R. J.
Published: October 2005

The equal energy hypothesis (EEH) for impulse noise is based on the assumption that the hearing hazard associated with an impulse noise exposure is a monotonic function of the total A-weighted energy received by a listener. A series of experiments will be described in which the EEH was evaluated using exposure paradigms in which the repetition rate (4/s to 1 every 16 s) and the intensity (107- to 131-dB peak SPL) of the impact noise were varied. Each of the various exposure paradigms had the same total energy. If the magnitude of the hearing loss is determined by the amount of acoustic energy, then all groups should develop the same amount of hearing loss. For equal intensities, the degree of damage increases with increased repetition rate. For example, the level of TTS increased approximately 25 dB when the repetition rate was increased from 1 every 16 s to 4/s. The effect of increasing the repetition rate also resulted in greater PTS and cochlear damage. The results are discussed in terms of the EEH and its appropriateness to impulse/impact noise. [This work was supported by NIOSH-R010H1152.]
Single-microphone method for certain applications of the sound intensity technique
Authors: Chu W. T.
Published: October 2005

Recent developments of the two-microphone intensity technique offer the possibility of improving measurements of acoustic impedance in impedance tube, sound transmission loss, and sound absorption coefficient in reverberation room. In these applications, the cross spectrum Gρ1ρ1 of the two microphone signals taken simultaneously is required. However, with the application of a deterministic source signal such as the periodic pseudorandom sequence [W. T. Chu, J. Acoust. Soc. Am. 76, 475–478 (1984)], Gρ1ρ2 can be obtained using a single microphone to measure the pressure at the two locations sequentially. Some of the inherent difficulties associated with the two microphone method are thus eliminated. The proposed method and some of the preliminary results will be presented.
Normal incidence absorption properties of single layers of elastic porous materials
Authors: Bolton J. S.
Published: October 2005

Recently, a theory has been developed which describes wave propagation in relatively stiff, partially reticulated polyurethane foams, the type most commonly used in noise control applications [J. S. Bolton and E. Gold, J. Acoust. Soc. Am. Suppl. 1 77, S59 (1985)]. A high impedance wave associated with the bulk mechanical properties of the foam matrix is usually significantly excited in these materials. As a consequence, the acoustical performance of finite depth layers of foam of this type is very sensitive to the boundary conditions which apply at the front and rear layer surfaces. Specifically, it will be shown in this paper that the action of a film facing is dependent on how it is attached to the foam layer. In addition it will be demonstrated that a small gap, e.g., 1 mm, separating a foam layer from a hard backing can increase the low-frequency absorption dramatically. A similar effect occurs when a film facing is not bonded directly to the surface of a foam layer but is separated from it by a thin air gap. This work has suggested an arrangement for enhancing the low-frequency absorption of thin foam layers.
The RASTI method, objective measurements of speech intelligibility
Authors: Højbjerg Klaus
Published: October 2005

In recent years, there has been increasing interest in objective measurements of speech intelligibility. For rooms with long reverberation time (also those in which sound systems have been installed), it is very useful for control purposes to have a fast objective way of measuring speech intelligibility. RASTI (rapid speech transmission index) is a new standardized method of measuring speech intelligibility, which provides a rating in less than 10 s. The theory and principles of operation will be presented. Measurement results from the “Grundtvigskirken,” which is a Danish church having a very long reverberation time and a recently installed sound system, will be reported.
Comparison of amplification systems in a small and large room with normal-hearing and special listeners
Authors: Nábělek Anna K. , Donahue Amy M.
Published: October 2005

Listeners with either hearing loss or perceptual problems require a good speech-to-noise ratio and short reverberation for their best speech discrimination. These conditions are sometimes difficult to achieve. Assistive listening systems, in which signal is transmitted by infrared light (IR) or modulated ratio waves (FM) are promoted for the special listeners. We compared the effectiveness of assistive systems in a classroom and in a large auditorium with several groups of listeners: normal hearing, hearing impaired, hearing aid users, elderly, and nonnative. In both rooms, the IR and FM systems allowed for better speech perception scores than through loudspeakers. The difference between the two assistive systems was very small. While all groups performed better through the assistive listening systems, the groups who achieved the greatest advantage had the poorest performance in the sound field. Since both systems were equally effective in improving speech intelligibility in less than optimal listening conditions, selection of a system should be based on such factors as cost, maintenance, and expected interferences. [Work supported by NIH.]
Three-dimensional audio image reconstruction
Authors: Romano Anthony J. , Tichy Jiri
Published: October 2005

An imaging technique is developed for the three-dimensional reconstruction of a desired sound field within a playback environment. Through proper coupling of room design and speaker placement, and deduction of the needed inputs to the system to obtain the desired outputs (for a finite length time window of data), a discrete-sliding window algorithm is implemented to obtain the necessary time-domain inputs. As a result of these procedures, the feedback/reverberatory effects of the playback environment are annihilated, leaving any desired sound field, while maintaining the three-dimensionality of the original field.
Two- versus four-tone masking, revisited
Authors: Lutfi Robert A.
Published: October 2005

Canahl [J. Acoust. Soc. Am. 50, 471–474 (1971)] measured thresholds for a 1-kHz sinusoid masked either by two or by four surrounding tones. He reported four-tone masked thresholds that exceeded, by 5–7.5 dB, the energy sum of the masking produced by the individual tone pairs. The present paper reports on a series of experiments investigating the effects of several factors on this 5–7.5 dB “excess” masking. In each experiment, thresholds for a 1-kHz, 250-ms sinusoid were measured as a function of the overall level of two or four equal amplitude sinusoids with frequencies arithmetically centered around 1.0 kHz. For conditions similar to those of the Canahl experiment, 5–6 dB of excess masking were obtained independent of the level of the masking tones. Randomly varying overall level across presentations had no effect on the excess masking, nor did the frequency separation between tones in one condition. The excess masking was reduced or eliminated when the masking tones were generated using an amplitude modulation technique, when they were gated on and off with the signal, or when their waveshapes were fixed across trials. Canahl's result may reflect listeners' ability to detect the signal as a variation in the waveshape of the multitone masker.
Peripheral and central components of comodulation masking release
Authors: Haggard Mark , Harvey Anthony D. G. , Carlyon Robert P.
Published: October 2005

Comodulation masking release (CMR) is the improvement in threshold that occurs when the subaudio amplitude modulations of a masker envelope in the critical band around a target tone are replicated on one or more flanking bands outside the critical band; the phenomenon is of obvious significance for perceptual grouping and attention and may stimulate a literature on parametric variations as large as that for the BMLD. Basic general questions first need to be answered about the nature of the effect, specifically concerning the relative roles of (presumably peripheral) psychophysical suppression and of (presumably central) temporal pattern comparison. In the first experiment the effect was tested monotically, diotically, and dichotically with one monotic control condition implementing 40-dB intensity disparity (as might result from worst-case transcranial conduction in the dichotic condition). Although intensity effects were present, a significant CMR was observed dichotically, suggesting that peripheral explanations such as suppression must be incomplete. The second experiment varied the phase (0°, 90°, 180°, and 270°) of the critical band and flanking band modulations. This experiment also produced results that cannot be completely explained by either a central statistical correction model, or a peripheral suppression model alone; negative CMRs were seen in some subjects.
Spectral locus and spectral spacing as determinants of the perceptual organization of complex tone sequences
Authors: Singh Punita
Published: October 2005

The perceptual organization of complex tone sequences into subsequences or “streams” was studied using spectral differences between tones as a basis for eliciting segregation. The absolute position of the spectra was varied to provide timbral differences between tones, while relative spacing between harmonics was varied to provide differences in pitch. These attributes were put in competition in rapidly occurring sequences of the form: TxPx TyPx TxPy TyPy, with the first pair of tones being assigned the pitch Px but different timbres Tx and Ty, and the second pair pitch Py, and similarly different timbres. Six listeners indicated their percept of such sequences as being groupings based on timbral similarity, pitch proximity, or ambiguous patterns not dominated by either cue. The results demonstrated that the stream segregation phenomenon may be based on relative changes in spectral locus and spectral spacing of the sequential tones, and imply that timbre and pitch can both serve as cues, with tradeoffs between these attributes determining the perceptual organization of a complex auditory input. [Work supported by NINCDS.]
Timbre discrimination by 7-month-old infants
Authors: Clarkson Marsha G. , Clifton Rachel K. , Perris Eve E.
Published: October 2005

While previous research has demonstrated the infant's ability to categorize sounds according to their pitch in the presence of irrelevant spectral variations, little or no research has evaluated discrimination of those spectral variations. In the present study, 7- to 8-month-old infants demonstrated the ability to discriminate harmonic tonal complexes from two timbre categories that differed in their spectral energy distributions. These differences in timbre were identical to the irrelevant variations over which infants categorized sounds in earlier work. In a conditioned head-turning paradigm, infants initially learned to discriminate stimulus tokens that contained the same fundamental frequency, 200 Hz, but different harmonic components (e.g., harmonics 5–10 versus harmonics 3–8 or 7–12). After successfully completing the basic timbre discrimination, the same infants transferred their learning to missing fundamental stimuli that had novel spectral energy distributions (e.g., harmonics 4–9 versus harmonics 2–7 or 6–11). These results suggest that young infants can analyze the spectra of harmonic tonal complexes and discriminate differences in one of the most important cues for adults' timbre perception, the distribution of spectral energy. [Work supported by NSF No. BN583-04419 and NIMH No. MH00332.]
Intracellular recordings from supporting cells in the mammalian cochlea
Authors: Desterle Elizabeth , Dallos Peter
Published: October 2005

The mammalian organ of Corti is constituted of a variety of different cell types. We have utilized horseradish peroxidase marking, along with intracellular recording techniques, to examine whether this heterogeneity is also reflected in the cells' electrical properties. Recordings from third and fourth turn supporting cells, hair cells, and fluid spaces have been made from guinea pig cochleas. A fast Fourier analysis was performed on responses to tone bursts. In all supporting cell types examined, the magnitude of the ac response component recorded from within the supporting cells was less than or equal to that recorded from adjacent fluid spaces. In contrast, differences were observed with respect to the dc response component. Dc components greater than those recorded from adjacent fluid spaces were frequently seen in supporting cell types located in the inner hair cell region. A slowly increasing dc component reminiscent of the slow electrical potential changes that occur in glial cells was also seen. These slow, depolarizing potentials are similar to those recorded from goldfish saccular supporting cells [T. Furukawa, J. Physiol. in press]. [Work supported by NINCDS Grant NS-08635.]
Synchrony and “synchrony suppression” in primary auditory neurons
Authors: Greenwood Donald D.
Published: October 2005

Synchrony of discharge to two-tone stimuli and “synchrony suppression” are analyzed by examining the implications of the definition of vector strength. “Synchrony suppression,” defined as the reduction in the vector strength for one tone when a second is added, occurs by definition when half-wave rectification occurs in an otherwise linear system. The usual shifts of empirical vector strength curves occur, disproving the necessity for compressive nonlinearity. “Synchrony suppression,” has been defined incompatibly as the shift in dB of a vector strength curve—said to be the magnitude of suppression. The identification of half-wave rectification with vector strength reduction disproves this conception, which is a logical fallacy as well. Curve shift is here defined as the shift of the crossover point at which a vector strength growth curve intersects the paired decay curve of the fixed-level tone. Crossover is the point of equality of the output amplitudes, hence vector strengths, in the period histogram. When the vector strength definition is applied to a complex waveform at the output of a compressive nonlinearity that compresses equal inputs equally, the shifts of the crossover points necessarily equal those in the linear case. But differences in unequal inputs will be accentuated in the relative output levels in the histogram, leading to greater differences of the vector strengths, at those relative input levels, than in the linear case. More visible effects of compression result from waveform distortion, which reduces vector strength saturation, and crossover, values and causes them to recede at higher levels. If the auditory periphery includes nonlinearities that compress equal inputs unequally, the basic curve shifts caused by half-wave rectification will change by the amount of differential compression. Since data suggest such nonlinearities may exist, it could be useful in new studies to use the crossover point of the growth and decay curves, as an index of equal output, to seek their confirmation and to infer something about them. [Work supported by NSERC.]
Syllable discrimination in 6–11 year-old normal and handicapped school children
Authors: Elliott Lois L. , Partridge Randy , Rupert Jennifer , DeGraaff Robert
Published: October 2005

We have previously described age-related differences in VOT discrimination [L. L. Elliott et al., J. Acoust. Soc. Am. Suppl. 1 77, S96 (1985)] and discrimination of the place of articulation feature [L. L. Elliott et al., J. Acoust. Soc. Am. 70, 669–677 (1981)] when just-noticeable differences among five-formant synthesized CV syllables are measured using adaptive test procedures. This is the first report of a larger study which also measured vocabulary level, receptive language skill, speech articulation, etc. As previously reported, older children demonstrated smaller just-noticeable differences for syllable discrimination than younger children. Of great interest, however, was the finding that, among normals, there were higher correlations between syllable discrimination and speech production for 6–8 year olds than for 8–11 year-old children. Preliminary factor analyses revealed different factor structures for learning-disabled 8–11 year olds, as compared to their normal age mates. For example, for these learning-disabled Ss, the two measures of VOT discrimination defined one factor while the two measures of the place features, plus the measure of receptive vocabulary, defined another factor. In contrast, the measures of syllable discrimination were distributed across four factors for the older, normal Ss. [Work supported, in part, by NINCDS, NIH.]
Vowel constriction effects on the perception of F 1 offset and vowel duration cues for word final stop consonant voicing
Authors: Fischer Rebecca M. , Ohde Ralph N.
Published: October 2005

The perception of voicing in final velar stop consonants was investigated by systematically varying vowel duration and change in frequency of the final F 1 transition. Several CVC continua were synthesized for each of three vowels, [i, ɪ, æ] which represent a range of tongue constriction values. Subjects used a scaling procedure in responding to these stimuli under both open and closed response set formats. The findings of this study show that both vowel duration and F 1 offset cues influence the perception of final consonant voicing, with the latter more salient in open than constricted vowels and the former more important in constricted than open vowels. In addition, individual preferences for either vowel duration or F 1 offset cues were observed, and these predilections were related to the vowel constriction feature. In summary, the results show that F 1 offset cues were perceived more invariantly as + voice than vowel duration cues across vowel contexts.
Trading relations in the perception of /r/-/l/ by articulation-delayed children
Authors: Manji Shelina , Strange Winifred , Polka Linda , Steffens Michele
Published: October 2005

Articulation-delayed children who did not produce /r/ correctly and who showed slow progress in therapy, were tested on identification of two synthetic series contrasting prevocalic /r/ and /l/ in a trading relations paradigm. The primary spectral cue (F3 and F2 onset and transition) was varied in ten steps in both series. For one series, the secondary temporal cue (F 1 steady-state and transition duration) was /r/-like; in the other, it was /l/-like. Result showed that phoneme boundaries along the spectral dimensions shifted appropriately as a function of the temporal cue value (i.e., a trading relation was demonstrated). However, the shift was somewhat smaller than for adults, and both boundaries were displaced toward the /l/ end of the spectral dimension, relative to the adults. These results suggest that such children have particular difficulty with the perceptual differentiation of spectral information, as might be predicted from their articulatory substitution patterns. [Supported by NIMH, NINCDS.]
Reaction times to categorically perceived speech stimuli
Authors: Schum Donald J. , Hurtig Richard R.
Published: October 2005

The measurement of reaction time for item identification has been suggested as one metric of the equivalence of category members [C. Mervis and E. Rosch, Ann. Rev. Psychol. 32, 89–115 (1981)]. Reaction times were measured while adult subjects (N = 10) categorized computer-generated CV syllables. One set of tokens varied along the VOT continuum (/ga-ka/). A second set of tokens varied along the F2 starting frequency continuum (/ba-da-ga/). The subjects typically demonstrated increased mean reaction times for tokens at or near phoneme boundaries. In certain instances, tokens near a phoneme boundary were consistently identified as belonging to a certain phoneme category; however, significantly longer reaction times were required to make those decisions. This response pattern would suggest that all tokens within a given phoneme category may not be functionally equivalent. The results will be discussed in light of the perceptual demands of listening to ongoing speech.
Perception of sinewave analogs of stop consonant place information II
Authors: Ralston James V. , Sawusch James R.
Published: October 2005

Identification and discrimination procedures were used to further explore a possible auditory basis for the categorical perception of place of articulation. Three 11-element stimulus series were constructed of either two or three pure tone components, with initial 40-ms frequency transitions followed by 200-ms steady-state portions. The initial transitions of the tonal stimuli mirrored the formant structure of synthetic two and three formant stop-place series varying in the onset frequencies of higher formant transitions. Subjects were operantly trained to classify stimuli into either two or three categories. Averaged ABX discrimination functions exhibited local maxima and minima similar to those found previously with analogous synthetic speech stimuli. These results suggest that a common auditory process contributes to discontinuities in the perception of both these tonal series and analogous stop-place series. The effects of stimulus structure on identification performance is also discussed. [Work supported by NINCDS.]
The role of attention in selective adaptation of place
Authors: Mullennix John W.
Published: October 2005

Selective adaptation of speech has been proposed by a number of researchers to be contingent upon a variety of differing mechanisms, including spectral overlap, phonetic featural overlap, and criterion changes. However, the role of attention has long been neglected within the adaptation process. In the present study, focused attention to an adapting syllable was manipulated. Subjects performed a dual-task dichotic monitoring procedure with the adaptor appearing either in an attended or unattended channel (ear). Categorization changes along a /bae/-/dae/ place-of-articulation continuum after adaptation were recorded. Also, the perceptual locus of effects (monaural or central) was assessed by the degree of the interaural transfer of adaptation. In a second and similar experiment, burst-cued adaptors varying in featural similarity to the /dae/ endpoint of the same continuum were utilized. The result of allocation of attentional resources to the adapting stimuli is discussed, along with the general relationship of attention to selective adaptation. [Work supported by NINCDS.]
Mixed excitation for speech III: Effects of manipulation of low-versus high-frequency components on naturalness of voiced fricatives
Authors: Allen George D. , Jamieson Leah H.
Published: October 2005

In our previous presentation to the Society [J. Acoust. Soc. Am. Suppl. 1 77, S11 (1985)], we examined the preceptual effects of a variety of digital manipulations of naturally produced voiced fricatives /z/ and /ȝ/ in V-V contexts. Briefly, local reorderings of pitch periods within the fricative result in a “rough” perceptual quality, while repetitions of periods cause the fricative to sound “buzzy.” In the present study, we examine in greater detail the source of these effects by comparing the relative contributions of low- versus high-frequency components of the pitch period to our perception. Each pitch period of the 24-kHz digitized waveforms was divided (by interactive hand marking) into an initial portion, consisting largely of low-frequency formant energy, and a final portion, consisting largely of high-frequency friction energy. The manipulations used in our previous study were then repeated on these subportions, leaving the order of the other subportions intact. Preliminary results reveal a greater effect of the high-frequency component on the perceived distortions, but with variation from token to token and from talker to talker.
A hybrid domain articulatory speech synthesizer
Authors: Schroeter Juergen , Sondhi Man Mohan
Published: October 2005

Articulatory speech synthesizers model the human vocal tract by means of its geometrical and functional properties. It is believed that this approach can be advantageous in speech coding at bit rates below 4800 b/s. Existing articulatory synthesizers work in the time domain. They either solve a system of differential equations for the vocal tract and the glottis, or synthesize speech using wave digital filters. The first approach is computationally very cumbersome. Both approaches have difficulties in incorporating important acoustic parameters, for example, the radiation impedance at the lips, wall vibration, and other losses. So far a realistic glottis model suitable for the wave digital filter approach does not exist. We have combined a nonlinear time-domain model of the vocal cords [K. Ishizaka and J. L. Flanagan, Bell Syst. Tech. J. 51, 1233–1268 (1972)] with a linear frequency-domain chain-matrix model of the vocal tract [e.g., M. M. Sondhi, paper ♯4.5.1, Proc. Int. Congress on Acoustics, Paris, France, 1983, Vol. 4, pp. 167–170]. The interface between these two models consists of convolving the glottal flow in the time domain with impulse responses of the tract obtained by inverse FFT. Examples of synthesized speech using manually generated and measured tract areas will be given.
Experimental confirmation of horizontal refraction of sound propagation in a wedgelike ocean
Authors: Doolittle R. , Tolstoy A. , Buckingham M.
Published: October 2005

Analysis of experimental data obtained in the region of the East Australian Continental Slope has confirmed theoretical predictions of energetic horizontal refraction due to sound-slope interaction. The observed modal behavior, azimuthal characteristics, and effects of bottom absorption were in accordance with both normal mode and ray theoretical solutions. The experiment, conducted with two ships (one towing a source and the other an array), started at a 400-m depth contour with an initial separation of 33 km and proceeded to deep water on a diverging but constant line of bearing course. Beamformed data-bearing shifts demonstrated the dominance of horizontally refracted arrivals for this geometry. A combination of source-array positions and critical angle effects served to limit both the azimuthal extent of sound received and the total number of modes arriving at any one position.
Measurement of the Biot structural factor δ for sintered bronze spheres
Authors: Baker Steven R. , Rudnick Isadore
Published: October 2005

The Biot structural factor δ of a porous solid [M. A. Biot, J. Acoust. Soc. Am. 28, 168 (1956)] can be extracted from measurements of sound speed and attenuation when the solid is saturated with superfluid helium [S. Baker et al., J. Acoust. Soc. Am. Suppl. 1 74, S59 (1983)]. The results of such measurements made in sintered bronze spheres (nom. diam. 75, 110, and 500 μ) yield values of δ from 3.3 to 4.5. [We take δ to be defined by κ = δ(αωk/vP)1/2, where κ is the argument of Biot's high-frequency correction function F(κ), α is the tortuosity, ω is the angular frequency, k is the permeability, v is the kinematic viscosity, and P is the porosity.] If δvisc is the normal fluid viscous penetration depth and a is a characteristic pore size, then the range of(δvisc/a) covered by these measurements is approximately 10−2 to 1. The values of δ reported here are about 112 to 2 times larger than has been estimated for typical marine sediments [R. Stoll, in Physics of Sound in Marine Sediments (Plenum, New York, 1974)]. [Work supported by ONR.]
An improved formula for wavefront curvature ranging based on measuring the normal curvature of wavefronts
Authors: Wood David H. , Tufts Donald F.
Published: October 2005

The conventional formula for wave-front curvature ranging degrades as the source moves away from broadside. This is because it is known [G. C. Carter, J. Acoust. Soc. Am. 65, 528–530 (1979)] that the conventional formula is equivalent to predicting range by triangulation, and this produces poor predictions in the presence of the refraction that is encountered off broadside. An improved formula that partially corrects for refraction is presented. It calculates the so-called normal curvature of the wave front—the curvature where the wave front intersects the plane containing both the line through the array of receivers and the line perpendicular to the wave front. This normal radius of curvature is an improved approximation for the range when the source moves away from broadside. The improved and the conventional formulas are compared in examples. [Work supported by the Naval Underwater Systems Center, and the Naval Air Development Command.]
Historical account of mutual acoustic radiation interaction between elementary transducers of a large underwater sound propator array
Authors: Hanish S.
Published: October 2005

In the decade 1955–1965 the U. S. Navy through ONR funded the construction of the ARTEMIS projector array for conducting long-range, low-frequency propagation experiments in the ocean. The acoustic size (= ka = 2πaλ) of the elementary transducers was very much less than unity at the transmission frequency. Early trials revealed extensive difficulty in achieving high-power radiation with uniform electrical excitation. It was soon found that the acoustic load varied significantly from element to element, leading to element failure, low transmission efficiency, and operating difficulty with the power amplifier. A historical review is presented for the valuable lessons learned in this experience.
Mutual radiation impedance of multiple mode vibrators
Authors: Benthien George W.
Published: October 2005

Most analyses of mutual radiation coupling to-date have assumed that the velocity distribution on each piston is fixed and known. In some problems, however, the velocity distribution on each radiator is made up of several modes and the modes interact acoustically as well as mechanically. For example, the velocity distribution on each piston might consist of a rigid body mode and one or more flexural modes. In this paper a general framework will be presented for the modal acoustic interaction problem, and several numerical techniques for its solution will be examined.
Mutual radiation impedance of baffled pistons
Authors: Benthien George W.
Published: October 2005

The mutual radiation impedance of pistons on a plane infinite rigid baffle is expressed as a Fourier transform of an impulse response. The technique used is an extension of the work of Lindemann on the self-radiation impedance of baffled pistons [O. A. Lindemann, J. Acoust. Soc. Am. 55, 708–717 (1974)]. As an illustration, the results are applied to the mutual interaction of rectangular pistons on a plane. Numerical results are presented and compared with other published results in order to demonstrate the accuracy of this technique.
A critical look at the coupling between the so-called “Helmholtz air” mode and a beam mode of the whole violin
Authors: Hutchins Carleen M.
Published: October 2005

A study of response curves, made over the last 20 years, of violins of varying musical qualities shows that in most of the instruments owned and considered to be of excellent quality by professional players, there is a close frequency relationship between the “Helmholtz air” mode and that of the beam mode around 270–290 Hz. When these two modes occur at the same frequency, there is enhanced amplitude of the “Helmholtz” resonance as well as increased vibration of the whole instrument, not only at this frequency but also at higher frequencies, as shown by response curves. In addition, this close coupling affects the “feel” of the instrument in the player's hands, particularly through the enhanced vibration of the neck. A quick way to check the relation of these two modes is to hold the violin upside down in thumb and forefinger at about half the distance between the bridge and saddle and alternately tap on the end of the scroll and blow in one f hole. With some practice the pitch of these two modes can be compared and also heard with considerable clarity when their frequencies coincide. Implications for violin making and possible effects on bowing will be discussed and a violin demonstrated.
A dynamic analysis of the violin using finite elements
Authors: Knott George
Published: October 2005

The dynamic response of selected points on a 12 000 degree-of-freedom finite element model of a violin to a simulated bowing force was studied. The model consists of the body (plates, bassbar, soundpost, end and corner blocks, ribs, and rib linings), neck, bridge, tail piece, and strings. All components were modeled as elastic elements in vacuo. Comparisons between the frequency response of a real violin and the model will be discussed.
Group classification of vibration modes in bells
Authors: Rossing Thomas D. , Perrin Robert
Published: October 2005

Several different systems have been used in classifying the vibration modes of church bells, carillon bells, and handbells. No existing classification system, however, has been completely satisfactory in describing modes of vibration in all types of bells. In this paper we describe a new mode classification system which is applicable to small handbells as well as to large church bells and carillon bells. The well-known inextensional modes are arranged into groups (0, I, II, …) according to the number and location of their nodal circles, and a “periodic table” is constructed. Less familiar torsional, breathing, swinging, extensional, and ring axial modes are arranged in groups having similar modal shapes.
Scheduled language for synclavier
Authors: Burnette W. A. , Mathews M. V.
Published: October 2005

A real-time music performance language has been written for the New England Digital Synclavier II. It is based on RTSKED, a scheduled language previously implemented on a Z80 microprocessor. The language allows easy and precise control of the timing of computer controlled events. Events can be controlled by external inputs such as pressing a key, or by an internal clock. Several sequences of events can operate simultaneously to generate complex musical sounds. A schedule is a “pure procedure” that consists of lists of operations, including transfers and conditional transfers. An “active schedule” comprises a set of variables describing the computer's progress through the execution of a schedule. An active schedule may create other active schedules and may generate or wait for “triggers” which are notifications of events from external inputs, internal timeouts, or other active schedules. Sounds may be generated by the combined efforts of several active schedules using schedule parameters and real time sensor inputs to compute a sequence of output channel commands. The language was demonstrated by performing four compositions written by R. Rocco.
Impulse-response measurements of small acoustical systems using shift-register sequences
Authors: Price Rodney D. , Palmer E. Paul
Published: October 2005

A data-processing system, using an IBM PC/XT, was developed to obtain impulse-response measurements of small acoustical systems such as musical instruments or the human vocal tract. The test signal is an m sequence generated using a shift register and converted to an acoustic plane wave by a piezoelectric transducer/tube system. We compensate for the nonideal characteristics of the transducer and microphone by using inverse filters generated with an adaptive digital filter [B. Widrow et al., Proc. IEEE 63, 1692–1716 (1975)]. The impulse response is recovered with a correlation technique using Hadamard matrices [M. Cohn and A. Lempel, IEEE Trans. Inf. Theory 23, 135–137 (1977)]. We present impulse-response measurements, 8191 samples in length at 24-kHz sampling rate, for various tube systems. Extension of the technique to reverberant systems with long impulse responses is straightforward since the data are recorded directly on disk to avoid memory-length limitations. The correlation algorithm, measurement results, and limitations of the transducer-compensation technique are discussed.
Spectral envelopes of orchestral instruments
Authors: Benade A. H. , Kouzoupis S. N.
Published: October 2005

Dependable techniques are available for the measurement of musical spectra. The oscillation and radiation dynamics of wind, bow, hammer, and plectrum excited instruments is well understood. Most of an instrument's dynamical system is shared by the production processes of all its notes, so that similar spectrum envelopes are expected for many notes in its scale. Measured spectra confirm the expectation, and show that these similarities are also shared by most instruments of the same playing range. Thus the notes of violin, piano, trumpet, oboe, and clarinet (odd partials) scales have envelopes caricatured by E0(f) = [1 + (f/fc)2β]−1/2, with β = 3 and fc = 1500 Hz. Corresponding alto instruments, e.g., English horn, alto sax, and alto clarinet, tend to have fc close to 1000 Hz, and β = 3. Several mechanisms, each with its own slope and breakpoint, combine in each instrument to produce its E(f), yet makers and players normally choose combinations that approximate E0(f). This suggests perceptual constraints on desirable tone color, e.g., via the “sharpness” integral of v. Bismarck, refined by Terhardt et al. Numerous measured envelopes will be presented, and discussed via their sharpness functions for treble, soprano, alto, and bass instruments. [Work supported by NSF.]