bluuebunny/crossref_metadata_2025 huggingface dataset:
The perception of voicing in word-final Catalan stops
Authors: Charles-Luce Jan
Published: October 2005

It has been demonstrated previously that Catalan does not always exhibit complete acoustic neutralization of the underlying voice contrast in word-final stops. The present study investigated whether listeners can use the acoustic differences distinguishing underlying voicing to identify the voicing of work-final stops. Native Catalan and English speakers were auditorily presented with natural tokens of three Catalan minimal pairs differing in the underlying voicing of the word-final stop. For these minimal pairs, vowel duration significantly distinguished underlying voicing in two-sentence medial position, but not in a sentence final position. The minimal pairs were excised from the sentences and presented in isolation. The results showed that neither the Catalan nor the English listeners were able to identify significantly above chance the underlying voicing of word-final stops produced in any of the sentential positions. Thus, the acoustic differences differentiating the underlying voice contrast do not appear to be perceptually functional for distinguishing minimal pairs. The apparent dichotomy of production and preception grammars and the implications for phonological neutralization will be discussed. [Work supported by NIH Training Grant NS-07134.]
Forward recognition masking: Another look at the “Echo”
Authors: Tomiak Gail R.
Published: October 2005

Auditory backward recognition masking effects have been interpreted as reflecting the existence of an echoic memory: an unencoded sensory store whose contents decay rapidly and are easily overwritten by succeeding auditory stimulation. However, it is equally likely that masking effects reflect the time course of an early stage of auditory processing. To distinguish these alternative conceptualizations, vowel stimuli were used in a forward recognition masking procedure. The interstimulus interval (ISI) remained constant, while the mask duration (and hence the stimulus-onset asynchrony, SOA) was varied. The simple storage model predicts that the efficacy of a mask is a function of the time between mask offset and target onset (ISI). Since this interval was constant, equal masking effects would be expected at all SOA intervals. The perceptual processing view predicts that target recognition performance would improve with increasing mask duration (increasing SOA). As mask duration/SOA increases, it is likely that processing of the mask would be completed before target onset. The results, and the implications of these findings, will be discussed. [Work supported by NINCDS.]
Evaluation of the constant ratio rule for consonant confusions
Authors: Bell Theodore S. , Dirks Donald D. , Kincaid Gail E.
Published: October 2005

The constant-ratio rule for consonant confusions asserts that the ratio of errors of any consonant to any others remains constant regardless of the size of the matrix from which the cells were chosen. In this study, log-linear models were applied to test the constant-ratio rule. Normal-hearing young adults were presented various closed-set arrangements of syllable tokens, spoken by a male and female speaker, and selected from a set of 14 VCs each paired with the vowel /ɑ/. Stop consonants and fricatives, both voiced and voiceless, were presented at three presentation levels as isolated sets (by feature) and in combination. The relationships among voiceless stop consonants remained constant irrespective of the size and nature of the response foil. Similarly, voiceless fricatives were unaffected by the nature of the closed set. In contrast, patterns of errors among voiced stops were dependent on the set of alternatives, as were voiced fricatives. Speaker differences, individual differences among listeners, and implications relating to the generalizability of confusion data collected in small closed-set arrangements are also discussed. [Work supported by a grant from NINCDS.]
Traceability of acoustical instrument calibration to reciprocity calibrations of laboratory standard microphones at the National Bureau of Standards
Authors: Nedzelnitsky Victor
Published: October 2005

Critical needs of public and private acoustical calibration laboratories for traceability to a reliable source have led to measurement services based upon the pressure and/or free-field calibration of laboratory standard microphones by established reciprocity methods over wide frequency ranges at the NBS. These primary methods and measurement services provided by the NBS at fixed cost to other laboratories are described, and the frequency-dependent uncertainties associated with these methods and services are discussed. Instrument calibration hierarchies comprising direct or implied chains of traceability to the NBS may involve at least four different definitions of traceability, and one or more of a variety of secondary calibration methods. Assessing the adequacy of a particular system for realizing traceability involves judging the capacity of that system to ensure measurements of acceptable accuracy for their intended purpose. Examples of traceability and the frequency-dependent uncertainties in some methods used to achieve it are discussed.
Calibration capabilities—Army Primary Standards Laboratory
Authors: Arrington James R.
Published: October 2005

The Army Primary Standards Laboratory (APSL) generates and maintains measurement standards in acoustics and vibration. Army measurement activities are supported by a three-level system that provides increasing degrees of accuracy starting with field calibration teams and ending with APSL. Traceability, calibration methods, calibration accuracy, transducer densities, and transducer types are discussed.
The calibration of integrating-averaging sound level meters: A manufacturers point of view
Authors: Hedegaard Peter
Published: October 2005

The introduction of the integrating-averaging SLM into common use has given rise to some new calibration and testing problems compared with those already known from the ordinary SLM. Determination of reasonable requirements for the integrating-averaging section and how to verify the performance has been a matter of discussion in connection with the national and international standardization work over the past years. Some aspects concerning the requirements and the test methologies and philosophies are discussed. The new opportunity to carry out spatial averaging in a simple way has given new life to the old discussion about free-field and random-field calibration. The free-field/random-field philosophies are summarized and some ideas about the possibilities for a final closing of “the Atlantic divide” are given.
Acoustic scattering in a range-independent, shallow water waveguide with a penetrable bottom
Authors: Sammelmann Gary S. , Trivett D. H. , Hackman Roger H.
Published: October 2005

At a previous meeting of the Acoustical Society of America, the analysis of the low-frequency, acoustic scattering from an elastic spherical shell in a homogeneous, range-independent waveguide with an impenetrable bottom was presented [G. S. Sammelmann and R. H. Hackman, J. Acoust. Soc. Am. Suppl. 1 79, S76 (1986)]. This previous investigation centered on elucidating the underlying dynamical picture leading to the observed “fine structure” of the scattering resonances of the sphere in the waveguide and to the occurrence of “superresonances.” Here, this analysis was extended to a waveguide with a (liquid) penetrable bottom to more completely explore the dependence of the resonance spectrum of the sphere on the acoustic environment. For a sphere in the sediment layer, the effects of the sediment loading and of the partial “unloading” by the nearby sediment-seawater boundary are discussed. [Work supported, in part, by the Office of Naval Research.]
Analytic solutions to the wave equation in a duct lined with axially nonuniform absorbents
Authors: Vaidya P. G. , Martin D. G.
Published: October 2005

For circular ducts with axially uniform boundary conditions, the acoustic field can be expressed in terms of a set of eigenfunctions. The corresponding eigenfunctions depend on the circumferential and the axial mode number. In this paper, a modification of the analysis, when the axial boundary conditions are nonuniform, is presented. The first part of the paper deals with an exponential variation in a semi-infinite duct. It has been shown that the field can be represented in terms of submodes. The eigenvalues of each of these submodes are distinct from each other, however, if they can be readily calculated. This analysis then is extended to an arbitrary variation of axial boundary conditions for finite length ducts. Numerical calculations are presented for the special case of circumferentially uniform fields (m = 0). It has been noticed that the character of the multiple eigenvalues of the submodes undergoes a qualitative change (akin to nonlinear bifurcation phenomenon) at the third radial mode (n = 3), for the specific case studied.
Analytical solutions to the wave equation in a duct lined with circumferentially nonuniform absorbents
Authors: Martin D. G. , Vaidya P. G.
Published: October 2005

Nonuniform boundary conditions are of fundamental interest from the theoretical point of view. They are also important from a practical point of view because of the potential of substantial enhancement of attenuation, which has been experimentally demonstrated. However, conventional eigenfunction solutions cannot be used to solve such problems. Therefore, much of the work in this area has been of a purely numerical nature. In this paper, the recent work by Vaidya, which was restricted to a single term in the exponential expansion, has been extended to any arbitrary circumferential variation. It has been found in this general case that the concept of submodes can still be used to construct the solutions. However, the eigenvalues of submodes can be altered, as opposed to the observation in the restricted case. A modified procedure to find the eigenvalues and the submodal coefficients has been developed. Specific numerical calculations are presented to show the alteration of the acoustic field and attenuation in hollow circular ducts.
Measurement of coherence with a vector acoustic intensity probe
Authors: Clark Joseph A.
Published: October 2005

Complex coherence is usually defined in the acoustics literature as the (ensemble-averaged) cross spectrum between pressure measurements made simultaneously at two points and normalized with respect to the square root of the product of the autospectra at the same two points. If the two points are sufficiently close together, a considerable simplification occurs which allows the complex coherence to be modeled as a function of one position with two parts. The real part corresponds to the classical definition of plane-wave intensity. However, it is scaler in character (omnidirectional). The imaginary part corresponds to the definition of intensity assumed by acoustic intensity measurement methods of more recent interest. This part exhibits a vector character. Vector acoustic intensity probes consisting of at least four elements arranged in closely spaced pairs along three orthogonal directions, can completely characterize complex coherence fields of the form described above. Ambient noise fields (and fields containing both signals and noise) can, in turn, be characterized by complex coherence measurements. These characterizations are of practical importance in estimating the accuracy and array gain of acoustic intensity measurements.
Review of methods of measuring rooms for speech
Authors: Bradley J. S.
Published: October 2005

Measures of the acoustical quality of rooms for speech should include both the signal-to-noise and the room acoustics aspects of the overall problem. The classical approach was to include these two aspects separately as background noise levels and reverberation times. Newer measures have combined both aspects into a single quantity. These more successful newer measures include an assessment of the beneficial effects of the direct and early reflected sounds and the detrimental effects of later arriving sounds. The prediction accuracies of a number of acoustical measures are compared, and optimum values of each quantity are presented. Optimum values for the classical quantities, background noise levels, and reverberation times are derived as a function of room volume from the newer quantities. The relationships between various measures are discussed.
Speech intelligibility in classrooms using RASTI
Authors: Hojbjerg Klaus
Published: October 2005

Speech intelligibility in classrooms is normally not a problem. However, in certain situations, when teaching hearing-impaired children, the acoustics of the classroom becomes important. Many old classrooms have insufficient acoustic design and therefore must be treated when hearing-impaired children are integrated with children with normal hearing. A quick method of assessing speech intelligibility is the RASTI system. This method, developed by Houtgast and Steeneken in The Netherlands, is based on a measure of the modulation transfer function in two octave bands. These two bands are centered at 500 Hz and 2 kHz, where a total of nine modulation frequencies are measured. This paper will describe the RASTI system, and actual measurements will be shown.
Standardization of speech level measurement
Authors: Pearsons Karl S.
Published: October 2005

The level of speech is measured using a variety of instrumentation and techniques. Most methods serve the needs of the particular user. However, the many different schemes make it difficult to compare one set of speech levels with another. In an attempt to provide a standard method of measurement, a standards working group was formed to develop a simple, meaningful, repeatable method for quantifying speech. This paper discusses the difficulties of providing such a measure and presents the philosophy of the working group's proposed method. The proposed method will be described and relationships with other speech measurement techniques will be presented.
Speech communication in lecture halls and classrooms for hearing-impaired
Authors: Nabelek Anna K.
Published: October 2005

Rooms with good acoustics for normal-hearing listeners might be inadequate for people with hearing impairments. To achieve listening conditions satisfactory for the hearing-impaired listeners, three conditions must be met: (1) Speech should be delivered at higher sound-pressure level than for normal-hearing listeners to compensate for hearing loss; (2) background noise should be at least 15 dB lower than the level of speech; and (3) the reverberation time of the room should be as short as possible. Amplification through public address systems is not practical because hearing-impaired listeners would need levels not tolerated by the general public. Also, because there are limits to reduction in background noise and elimination of sound reflections, listening conditions in lecture halls and in classrooms might be less than ideal for the hearing-impaired. A solution is the use of assistive listening systems. In these systems, sounds which are picked up by microphones located close to the talkers' mouths contain little noise and reverberation. Then the sounds are transmitted on radio or infrared waves throughout a whole room. The sounds are received by individually worn receivers which are equipped with gain controls. The benefits and limitations of assistive listening systems will be discussed. [Work supported by NIH.]
Speech intelligibility measurements sing TEF analysis
Authors: Davis Don , Keele Don , Patronis Eugene
Published: October 2005

Recent investigations into the measurement of speech intelligibility using large listener groups (three groups of 30 each) in a cathedral, a concert hall, and a classic motion picture theater of the early 1930s suggest that a listener's subjective intelligibility score will be best matched in objective testing by dividing the sound considered to be direct sound from sound considered to be reverberant sound at the highest level return within the first 50 ms. This, in a majority of cases, results in using only the first arrival and no integration of early sound. Evidence will be presented that indicates that the use of impulse squared measurements that fail to process the signal as a complex analytic signal often fail to properly record major reflective signals unless extensive spatial averaging is resorted to. The paper will be supported by actual postprocessing of data taken with a Techron TEF analyzer using the Heyser technique for energy time curve measurement. In such measurements, minor movements of the microphone dramatically affect the impulse and doublet responses while the same change barely affects the energy time curve measurement, thus demonstrating the need for the complex analytic signal in such analysis.
Vibrotactile masking as a function of stimulus-onset-asynchrony
Authors: Gescheider G. A. , Verrillo R. T. , Bolanowski S. J.
Published: October 2005

Vibrotactile thresholds for the detection of a 50-ms vibratory stimulus on the thenar eminence of the hand were measured in the presence of and in the absence of a 700-ms suprathreshold vibratory masking stimulus. When thresholds were measured in the presence of the masking stimulus, stimulus-onset-asynchrony was varied so that backward, simultaneous, and forward masking could be measured. The amount of masking, expressed as the difference between thresholds for detecting the test stimulus in the presence of and in the absence of the masking stimulus, was greatest when the test stimulus was presented near the onset or offset of the masking stimulus. For both backward and forward masking, the amount of masking decreased as a function of increasing stimulus-onset-asynchrony with the decay rate being greater for backward than for forward masking. Comparisons were made of the amounts of masking measured when the test and masking stimuli were both sinusoids, when the test and masking stimuli were both noise, and when the test stimulus was a sinusoid and the masking stimulus was noise.
Psychophysical evidence for a four-channel model of vibrotaction
Authors: Bolanowski Stanley J. , Gescheider George A. , Verrillo Ronald T. , Checkosky Christine M.
Published: October 2005

Although previous physiological experiments have identified four sensory channels (Pacinian, RA I, SA I, and SA II) in glabrous skin of the human somatosensory periphery, only three (Pacinian, RA I, and SA I) have been shown to mediate vibrotactile sensation. By using stimuli to selectively mask these various channels, it is demonstrated that a fourth channel does, indeed, participate in the perceptual process. The fourth channel is unaffected by changes in skin-surface temperature (15° to 40 °C) and stimulus area (0.008 to 2.9 cm2) or duration (700 to 2500 ms). Adhering to previous psychophysical nomenclature, the fourth channel is called non-Pacinian III (NP III). The four psychophysically measured channels (P, NP I, NP II, and NP III) combine at threshold to create an operating range for the perception of vibration which extends from 0.4 to > 500 Hz. The four channels mediate specific portions of the overall threshold-frequency characteristic, although their sensitivities partially overlap. Thus suprathreshold stimuli may activate several channels simultaneously, suggesting that the perceptual qualities of touch may be determined by the combined inputs of all four channels. [Work supported by NIH and NSF.]
Noise control case history: Cafeteria environment
Authors: Royster Larry H.
Published: October 2005

Complaints from employees at a local research institution about difficulties in communicating while eating lunch in the Institute's cafeteria led the institution to request recommendations as to possible facility modifications. Premodification sound surveys were conducted with and without the employees present. Recommendations were then made to management as to facility modifications that would be expected to significantly lower the cafeteria's sound level during the peak lunch period. After the recommendations were implemented, sound surveys were conducted to verify the predicted levels of noise reduction. The facility modifications included extensive use of sound-absorbing wall panels, a lowered and improved acoustical ceiling, and the construction of a partial barrier between the food-serving line and the eating area. At maximum usage, a 12 dB(A) (A-weighted sound-pressure level) reduction in the cafeteria's sound level was achieved [from 75 to 63 dB(A)]. No additional complaints have been received from the Institute's employees regarding noise in the cafeteria.
Application of the STSF system to noise source identification in domestic appliances
Authors: Dietschi U. D. , Bolton J. S.
Published: October 2005

The acoustic energy produced by small appliances often originates at one point within the device (e.g., an electric motor) but actually radiates from a number of points on the casing. It is useful to identify these surface sources since they may indicate internal energy transmission paths; however, the task is made difficult by the fact that they are inevitably correlated and are often closely spaced. Recently the newly developed Brüel & Kjaer spatial transformation of sound fields (STSF) system was used to identify sources in this situation. This system is a broadband adaptation of the acoustical holography technique; it makes use of sound-pressure measurements at a number of reference locations and at an array of field points to identify the wavenumber components of the field. These may be used to reconstruct the entire sound field exterior to a plane tangent with the source. By examining intensity maps on orthogonal planes near the source, closely spaced individual surface sources may be identified. The application of this technique to a complete noise control procedure will be illustrated in this paper.
Determination of vehicle noise source levels
Authors: Smart Richard C. , Dunn Stanley E.
Published: October 2005

In 1974 the Florida Department of Transportation participated in a study sponsored by the Federal Highway Administration to determine vehicle source levels. This paper reports the findings of a second study designed to extend the range of the operating speeds of the vehicles in question down to 20 mph (32 kph). A total of 11 sites were included in the study with vehicles operating between 20 to 61 mph. The subsequent data were analyzed to provide distributions of maximum passby sound levels and reference energy mean emission levels as a function of speed for automobiles, medium trucks, and heavy trucks. The data were analyzed over the entire speed range and on the basis of 5 mph speed band to provide empirical relations for use in predictive models. Of particular interest was the extent to which the choice of these formats influenced the noise source relationships and, of more significance, the extent to which the introduction of low-speed data which were not previously available influenced source level relationships. [Work supported by the Florida Department of Transportation.]
A method for presenting dynamic performance characteristics of highly damped vibration isolators
Authors: Plesner Neil G.
Published: October 2005

A method for presenting dynamic performance characteristics of highly damped vibration isolators is discussed. The method utilizes normalized stiffness versus load graphs obtained from laboratory test data and enables determination of dynamic stiffness and natural frequency as a function of isolator load and temperature. Isolators examined were grommets and bushings made from three PVC thermoplastics. Because similar types of isolators show similar isolation characteristics with respect to load and temperature, they can be categorized into families based upon material formulation and isolator geometry. This provides a means for presenting a large amount of performance information in an abbreviated format. Methods of data acquisition and reduction are discussed briefly with major emphasis on the determination of isolator effectiveness for basic noise and vibration control problems.
Ambulance sirens and in-vehicle noise exposure: An investigation into alternative siren speaker placement
Authors: Van Buren Martin W. , Caple Gary
Published: October 2005

Two modular Road Rescue type III ambulances on 1985 Ford E350 XL RV cutaway vans, cab headliners insulated with fiberglass, modules with 112 in. of self-extinguishing polyurethane foam planking, were, respectively, outfitted with Southern D-50-C chrome bell 100-W speakers, forward facing, light-bar mounted above cab at 15° off center, speakers approximately 2.3 m from ground (condition 1) and with Whelen SA-450G speaker strobe 100-W speakers, forward facing, flush-radiator-grill-mounted, 0.94 m apart, speakers approximately 0.9 m from ground (condition 2). Each vehicle was placed on tarmac with sirens set on “wail” mode and SPLs were taken 12 m from ground in front of the vehicles along a 180° arc at points up to 30.5 m and in the vehicle at ear level (referenced to driver, passenger, patient, etc.). SPLs measured outside the vehicle at 30.5 m showed the grill-mounted speakers typically produced 14 dBA more intensity than the conventionally light-bar-mounted speakers at 0° azimuth. SPLs inside the vehicle were 11 to 16 dBA quieter in condition 2 depending on position measured. Efficacy of siren itself for emergency alerting was enhanced and occupant noise exposure was reduced when sirens were grill mounted.
Comparison of hearing acuity of rural, urban, and industrial populations
Authors: Raja S. , Singh K. Harjeet
Published: October 2005

Auditory acuity of 104 rural individuals, 126 urban individuals not exposed to occupational noise, and 118 workers from a textile mill was examined in order to study the influence of noise exposure characteristics on hearing level of the three populations. Pure-tone audiometry in 5-dB steps at selected frequencies was conducted with the help of a portable diagnostic audiometer. Noise level recordings were made by a B&K impulse precision sound level meter and an octave band analyzer. The differences in hearing level at 4 kHz between the populations were found to be statistically significant, with the rural individuals exhibiting superior hearing capacity compared to other groups, and urban individuals showing better ability compared to their industrial counterparts. In the low frequencies (0.5 and 1 kHz), however, significant differences were observed only between rural and industrial workers. The comparison between urban and industrial workers yielded only differences in absolute values which were not significant. Some factors contributing to the poorer hearing capacity of urban individuals have been discussed. The significance of this evaluation in the context of a mass movement of individuals from rural to urban areas to seek employment has been outlined.
Measurement of sound absorption and sound speed in terrigeneous sediments
Authors: Weydert Marco M. P. , Murray Nicholas , D'Alessandro Marco
Published: October 2005

The absorption and speed of sound were measured between two bore holes each 150 m deep and spaced 4 m apart. Sediment profile samples have been obtained from the two wells and the water content, porosity, density, and mineralogy determined. Acoustic measurements were made at selected depths in the water saturated sediments down to a maximum of −83 m. The acoustic frequencies ranged from 7.5–18 kHz; at the lower frequency, the absorption varied from 0.3 dB/m in clay to 0.4 dB/m in sand. At 18 kHz the corresponding values were 2.0 and 4.5 dB/m, respectively. Measurements of sound speeds for clay resulted in a velocity of 1480 + 15 m/s, 1540–1820 + 15 m/s in silt, and 1540 + 15 m/s for sand, at a temperature of 12 °C. The absorption measurements compare well at 12 kHz to those made in late 1986 in deep ocean sediments using penetrators that were able to penetrate up to 55 m into the sediments.
Deep ocean/seabed-satellite relay for long-term quasi-real-time data transmissions
Authors: Murray C. N. , Weydert M. M. P.
Published: October 2005

During the last 2 years, work has been underway at the Joint Research Centre of the Commission of the European Communities on the use of satellite communications for the transmission of data from instrumentation placed in the ocean bed. The setup of the transmission link has required the development of several new systems. These include an underwater vehicle capable of successfully emplacing instrument packages within deep ocean sediment formations, a transmitter-receiver system whose characteristics are such as to be able to send acoustic signals through both 50 m of sediments and a 6-km water column, sensors which will give information on sediment characteristics, and a satellite communications system for the automatic quasi-real-time relay of data from the emplaced instrumentation to the laboratory in Ispra (Europe). The present paper will discuss the acoustic link and the relay/buoy system and its applications to oceanography and meteorology.
Low-frequency acoustic/seismic propagation in a sloping ocean environment: Comparison between measured results and numerical predictions
Authors: Ali Hassan B. , Werby Michael F.
Published: October 2005

The propagation of low-frequency signals and noise in shallow water, and more generally in a bottom-limited environment, is of considerable interest to the Navy. A particular aspect of this problem, namely the relative merits of waterborne versus seismic propagation paths, has been investigated using the results of at-sea measurements and predictions based on geoacoustic and numerical models of the test environment. The experiment was conducted by the Naval Ocean Research and Development Activity in a shallow water region off the southeastern coast of the U. S. A vertical array of hydrophones in the water column and a distribution of tri-axial geophones on the ocean bottom were used to monitor the response to both cw and broadband (explosives) sound sources. Predictions of the propagation were made using the SAFARI version of FFP, the wide-angle version of the Parabolic Equation Method, and the SNAP normal mode model. Preliminary results of the measurements are discussed and comparisons are made with the predictions based on the numerical models.
Horizontally polarized shear waves in shallow marine sediments
Authors: Sutton George H. , Ewing John I. , Barstow Noël , Carter Jerry A.
Published: October 2005

In June 1986 a seismic experiment emphasizing the observation of shear waves in the bottom sediments was conducted off the southern New Jersey coast in water depths between about 10 and 56 m. This paper emphasizes the generation and detection of horizontally polarized shear waves SH produced by a pair of air (mud) guns, mounted to apply horizontal forces, on a sled deployed from the ship and capable of being dragged along the bottom. Signals were digitally recorded from three-component geophones and a hydrophone. Shear waves and boundary waves were recorded in the frequency band 2 to 40–50 Hz having group velocities ranging from about 50 to 300 m/s. Both SH/Love wave and P/SV/Rayleigh-Stoneley-Scholte wave signals were dearly recorded and resolved by combining sums and differences of two properly oriented horizontal components and comparing the resolved horizontal data with vertical and pressure components. Full-waveform synthetic seismograms are being used to aid inversion for shear velocity and attenuation models that are required to be consistent with information available from test borings as well as the shear wave data. [Work supported by ONR.]
Solid particles in sonic field
Authors: Mawardi Osman K.
Published: October 2005

A review is presented for the dynamics of solid particles in suspension in a fluid and subjected to sonic energy. The controlling parameters associated with collective behavior of the particles such as precipitation or coagulation are identified. A brief discussion is then given on a classification of observed phenomena in the processing industry and by means of these parameters.
Acoustic agglomeration of power plant fly ash for environmental and hot gas cleanup
Authors: Reethof Gerhard
Published: October 2005

Emission of small particulates from coal-fired power plants in the range of 0.5 to 5 μ results in particularly serious injury to the human respiratory system. Currently used environmental cleanup devices in power plants such as bag houses, electrostatic precipitators, wet scrubbers, and others are very efficient in removing particulates which are larger than about 2 μ in size, leaving by far the majority of these tiny particulates to be emitted into the air environment. Current regulations for effluent cleanup emphasize the mass removal without any attention to the particulate size. Acoustic agglomeration is a means of causing these tiny particulates to collide with one another, adhere, and thereby result in the formation of larger particles, thus removing the tiny particulates from the aerosol stream. In this manner, the efficiency of particle removal of the conventional removal devices can be substantially enhanced. This paper introduces some of the fundamental aspects of acoustic agglomeration theory and practice and presents some of the results of recent studies.
Removal of particulate contaminants from surfaces
Authors: Montz K. W. , Butler P. B. , Beddow J. K.
Published: October 2005

An experimental investigation was made on the interaction of an acoustic field with particulate matter adhered to a substrate. The particles of interest are in the μ-size range. In the experimental design, a combined acoustic/convective field is used to overcome the adhesive forces of the contaminant materials. A key parameter in this work is the removal efficiency (based on number counts) of contaminant particles from the surface of materials. An image analysis technique is used to determine particle counts before and after the process. The relative adhesive and removal forces are a function of particulate material and substrate properties. Preliminary results using a factorial design illustrate the influence of several acoustic properties on the removal efficiencies.
Acoustic suspension processing
Authors: Feke D. L. , Hazony D.
Published: October 2005

Separation and purification of fine solids suspended in fluids are involved in a variety of technologies. A process utilizing high-energy acoustic planes that are separated by low-energy zones will be described for these purposes. Results of theoretical and experimental studies will be discussed.
A model of the effects of signal delay and masker level on masked measures of frequency selectivity
Authors: Lutfi Robert A.
Published: October 2005

In a previous paper [R. A. Lutfi, J. Acoust. Soc. Am. 76, 1045–1050 (1984)], the following relation was used to predict measures of frequency selectivity obtained in forward masking from measures obtained in simultaneous masking: F(g) = G + H(g) − H(0), where, for a given masker level, F is the amount of forward masking (in dB) as a function of signal-masker frequency separation (g), H is the amount of simultaneous masking, and G is the amount of forward masking for g = 0. In the present study, the relation was tested for a wider range of signal and masker frequencies, masker levels, and signal delays. The relation described thresholds from all conditions well with the inclusion of one free parameter λ corresponding to a constant frequency increment: F(g) = G + H(g + λ) − H(λ). The parameter λ was required to account for observed shifts in the frequency of maximum forward masking. It is suggested that a single tuning mechanism can account for commonly observed differences between simultaneous- and forward-masked measures of frequency selectivity. [Work supported by National Science Foundation Grant BNS 83-08498.]
CMR for complex signals
Authors: Hall Joseph W. , Grose John H. , Haggard Mark P.
Published: October 2005

Previous CMR investigations have examined the effect of across-frequency masker envelope coherence on the detection of a pure-tone signal. The present study extends this investigation to examine the effect of the number of components comprising the signal. The masker was composed of either one, two, or three 30-Hz-wide narrow-band noise components. In multiband conditions, the bands had either correlated or uncorrelated envelopes. For the one-component masker, the signal was a pure tone at the center frequency of the masker. For the two-component masker, the signal was either a one-component or a two-component signal. For the three-component masker, the signal was either a one-, two,- or three-component signal. For the three-component masker, CMR was greatest for the single-component signal. However, CMR was appreciable for two- and three-component signals. CMR generally increased as a function of increasing number of masker bands. Results will be discussed in terms of models of CMR. [Work supported by AFOSR.]
Development and testing of artificial low-frequency speech codes
Authors: Reed C. M. , Power M. H. , Foss K. K. , Durlach N. I. , Braids L. D.
Published: October 2005

Previous work on frequency lowered speech for listeners whose hearing is confined to low frequencies has focused on signal processing of natural speech. In the present research, the use of artificial low-frequency codes for such listeners is explored. This research will help separate limitations imposed by signal processing schemes from limitations imposed by the listener's perceptual system. It will also provide background for the design of new systems based on the use of speech-recognition devices as front ends. Artificial frequency-lowering codes were developed for 24 consonants (C) and 15 vowels (V) for two values of low-pass cutoff frequency F (300 and 500 Hz). Individual phonemes were coded by a unique acoustic signal confined to frequencies less than or equal to F. The ability of normal-hearing listeners to identify coded C's and V's in fixed-context syllables was compared to their performance on single-token sets of natural speech utterances low-pass filtered to equivalent values of F. Performance on coded C's was superior to that on low-pass filtered speech for both values of F (by roughly 20 percentage points). An advantage for coded V's (of roughly 15 percentage points) was observed only at F = 300 Hz. Identification of C and V components in roving-context CV syllables was generally inferior to that obtained in the fixed-context conditions. [Work supported by NIH.]
Vocabulary training via a tactile communication aid with a prelingual deaf child
Authors: Woodard Robert E. , Gavin William J.
Published: October 2005

A wearable, self-contained 24-channel vibrotactile vocoder was used to train receptive vocabulary. The subject was a 6-year-old, prelingual, profoundly hearing-impaired female of average intelligence. Receptive vocabulary training was conducted a minimum of 12 h/day, 4 days/week, with the subject receiving auditory, visual (speechreading), and tactile stimulation during training. Assessments of vocabulary growth were conducted every fifth session via a picture pointing task. Words for training were selected from Benedict [H. E. Benedict, “Language comprehension in 10- to 16-month-old infants,” unpublished doctoral dissertion, Yale University, New Haven, CT (1976)] and clinician suggestions. After approximately 34 h of training over 13 weeks, the subject learned 60 words, a learning rate of 1.79 words/h. This rate is slower than the rate of 2.7 words/h reported for normal-hearing subjects [P. L. Brooks and B. J. Frost, J. Acoust. Soc. Am. 74, 34–39 (1983)] but faster than the rate of 0.53 words/h reported for deaf subjects [S. Englemann and R. Rosov, Except. Child 41, 243–253 (1975)].
Assessment of a tactile communication aid with a post-lingual deaf adult
Authors: Rosov Robert , Woodard Robert E. , Ingrisano Dennis R. , Murrell Darlene
Published: October 2005

A case study was utilized to assess receptive vocabulary training using a wearable, self-contained 24-channel vibrotactile vocoder. The subject was a 58-year-old, post-lingual, adventitiously deafened female of average intelligence and above-average skills in learning word meanings (+ 2 s.d. on vocabulary subtest of the WAIS). It was predicted that assessing rate of vocabulary acquisition with this type of subject would be a more rigorous test of using tactile stimulation as a means of sensory substitution for the deaf than could be achieved using a prelingually deafened subject. The vocabulary training protocol was modified from Brooks and Frost [P. L. Brooks and B. J. Frost, J. Acoust. Soc. Am. 74, 34–39 (1983)]. Training words were presented using only tactile stimulation with feedback. Training was conducted a minimum of 1h/day, 5 days/week. After approximately 88 h of training over 18 weeks, the subject was able to identify 15 words, yielding a word acquisition rate of 0.17 words/h. This differs considerably from acquisition rates previously reported of 2.7 words/h for normal-hearing subjects [P. L. Brooks and B. J. Frost, J. Acoust. Soc. Am. 74, 34–39 (1983)] and 0.53 words/h for hearing-impaired subjects [S. Englemann and R. Rosov, Except. Child. 41, 243–253 (1975)].
Using cued speech to clarify speechreading
Authors: Tonry Kathy L.
Published: October 2005

Speechreading for hearing-impaired individuals is a monumental task due to visual similarities of some speech sounds and the lack of visibility of others. Cued speech is a visual supplement to speechreading used to remediate this problem. It employs a system of 12 visual hand cues. Four hand positions are used to clarify vowels and eight hand configurations are used to differentiate consonants which are visually unclear. Two previous studies by Ling and Clarke [D. Ling and B. R. Clarke, Am. Ann. Deaf 120, 480–488 (1975); and B. R. Clarke and D. Ling, Volta Rev. 78, 23–34 (1976)] evaluated the speechreading capabilities of hearing-impaired subjects using cued speech. Results from the first study (subjects used cues for 1 year) supported the use of cues for identifying words in phrases. The follow-up study (1 year later) supported the idea that cued speech facilitates speechreading for sentences, phrases, and words. The present informal study indicates improved language skills in nine hearing-impaired children using cued speech.
An overview of the functions of the ear
Authors: Sachs Murray B.
Published: October 2005

The function of the peripheral auditory system (external, middle, and inner ears) is to produce a representation of sound across the population of auditory-nerve fibers. This representation is processed by the brain to produce appropriate behaviors (e.g., perception, movement, etc.). This talk will focus on the representation of the spectral features of speech sounds in the auditory nerve. Emphasis will be placed on those aspects of the encoding in the auditory nerve that allow the representation to be robust over a large range of sound levels and in the presence of background noise. The transformations of this representation by populations of cochlear nucleus cells will bc discussed. [Work supported by grants from NINCDS.]
Digital hearing aids
Authors: Levitt Harry
Published: October 2005

Digital hearing aids offer many advantages over conventional hearing aids. These include programmability, communication with and control by a host computer, and the potential for implementing advanced signal processing techniques for reducing background noise and for enhancing speech intelligibility. Three types of experimental digital hearing aids have been developed: (1) a quasidigital system in which the audio signals remain in analog form but are controlled by digital means, (2) a digital filtering system under microprocessor control, and (3) real-time simulation of a prescribed hearing aid on a high-speed, general-purpose computer. Recent experiments using digital hearing aids have evaluated new methods of hearing-aid prescription, techniques for processing speech to improve intelligibility for the hearing impaired, methods of noise reduction, and new approaches to the measurement and specification of hearing-aid characteristics. Existing experimental digital hearing aids are relatively large in size, the smallest being a body-worn unit. Current engineering efforts are directed towards developing a unit small enough to be worn on or in the ear.
The middle ear from the standpoint of the surgeon and the acoustician
Authors: Goode Richard L. , Killion Mead C.
Published: October 2005

Recent data obtained from fresh human temporal bones [Gyo et al., Ann. Otol. Rhinol. Laryngol. (1987)] have provided previously unavailable information regarding the likely effect on the middle ear transmission resulting from surgical modification. The possibility of improving a low-grade ear (from the middle ear transmission standpoint) into a “golden ear” will be discussed. Calculations based on modifications to the analog external-plus-middle-ear model of Killion and Clemis [J. Acoust. Soc. Am. Suppl. 1 69, S44 (1981)] will be presented.
Cochlear implants: What are they and what do they accomplish?
Authors: Eddington Donald K.
Published: October 2005

A large portion of the profoundly deaf are not able to benefit significantly from conventional hearing aids. Even with the tremendous gain provided by high-powered aids, the peripheral auditory system of these impaired individuals is not able to transform the intense sound input to patterns of neural activity that the brain can interpret. Cochlear implants are devices that convert sound to electrical stimuli and use electrodes placed in the cochlea to deliver these stimuli to the auditory nerve. The neural activity elicited by these stimuli is interpreted as sound percepts by the brain. The goal of these devices is to create patterns of neural activity that the brain can learn to interpret and thereby restore some measures of hearing to the deaf. These devices are now commercially available in several forms and can provide significant benefit to the profoundly deaf individual. This presentation will describe the rationale of cochlear implantation, the classes of devices now available, and the benefits provided to users of these devices.
Interpretation of amplitude compression in terms of the modulation transfer function
Authors: Plomp R.
Published: October 2005

The successful application of the speech transmission index for predicting the combined effects of reverberation and noise on the intelligibility of speech [e.g., T. Houtgast and H. J. M. Steeneken, J. Acoust. Soc. Am. 77, 1069–1077 (1985)] demonstrates that the modulation transfer function (MTF) is an appropriate measure of the consequences of temporal modifications of the speech signal on its perception. In view of this, it makes sense to describe also the effect of amplitude compression on speech intelligibility in terms of the MTF. A compression circuit, as applied in hearing aids, reduces the MTF from zero up to a critical modulation frequency of, typically, 10 Hz. This means that a substantial part of the range of modulation frequencies present in the speech signal and relevant for its intelligibility (0.4–20 Hz) is affected by the compression process. This can explain why amplitude compression with short time constants has a negative effect on the intelligibility of speech in noise. The results of other attempts to modify the MTF for better speech understanding will also be discussed.
Piano design factors—Their influence on tone and acoustical performance
Authors: Conklin Harold A.
Published: October 2005

Major design factors that affect significantly the tone and acoustical performance of the modem mechano-acoustic piano are reviewed and the influence of each on the performance of the instrument is briefly described or illustrated. Elements discussed include the string plate, the soundboard, the ribs, the bridges, the “scale,” the strings, the hammers, the frame, the case, and the lid. Some directions are suggested for future work to improve performance. The presentation is from the perspective of the instrument designer.