bluuebunny/crossref_metadata_2025 huggingface dataset:
Critical bandwidth in lateralization: Effect of level
Authors: Scharf Bertram , Meiselman Carol H. , Florentine Mary
Published: October 2005

The onset-time difference ΔT needed to lateralize 30-msec dichotic tone bursts toward the leading ear was measured as a function of the frequency difference ΔF between the burst in one ear and the burst in the other ear. An adaptive procedure (a variation of BUDTIF) was used to concentrate judgments in the vicinity of 75% correct. Signal frequencies were centered at a geometric mean of 2000 or 6000 Hz. The level in the right ear was set near 25, 50, or 80 dB SPL. At each of the three reference SPLs, all tone bursts in the right ear were presented at the same loudness level. The level in the left ear was adjusted to center the image. Threshold ΔT remained approximately constant as ΔF increased up to the critical band, which is 300 Hz at a center frequency of 2000 and 1100 Hz at 6000 Hz. Beyond the critical band, threshold ΔT increased with ΔF. The critical bandwidth did not change with level for either listener, but the rise in threshold beyond the critical band became steeper at lower levels. These results provide further evidence that the dichotically measured critical band has the same width as the monaurally measured critical band, and that critical bandwidth is independent of sound pressure level. [Research supported by NIH Grant No. NS 07270.]
Intensity and frequency discrimination in one and two ears
Authors: Jesteadt Walt , Wier Craig C.
Published: October 2005

Early studies of intensity and frequency discrimination reported that discrimination was generally better with two ears than with one. Unlike binaural summation or MLDs, this binaural phenomenon has received little attention and has not been systematically explored. In the present experiment, intensity and frequency discrimination for pure tones were measured at 250, 1000, and 4000 Hz using a 2IFC-adaptive procedure. Signals were presented at 70 dB SPL in a low-level broad-band noise. Intensity and frequency discrimination were measured for each ear individually (SmNm) and for both ears, with signal and noise in phase (S0N0). In all conditions, discrimination with two ears was better than discrimination with either ear individually. The improvement in performance was approximately equal to the √ increase in d′ that an information model would predict for a system with input from two independent channels.
Magnitude of JNDs for diotic and dichotic perception of spectrally colored noise
Authors: Koenig A. H. , Berkley D. A. , Curtis T. H. , Allen J. B.
Published: October 2005

As part of a series of studies concerned with the binaural perception of reverberant sound, just noticeable differences (JNDs) were determined for spectrally colored noise presented both diotically and dichotically. Test stimuli were prepared by passing white noise samples through simple all-zero filters consisting of a delay, an attenuator, and an adder. By adjusting the delay and attenuation, various amounts of spectral distortion could be obtained. The sensitivity of the listener to changes in the amount of distortion or coloration was measured under two listening modes. For the diotic case, the same spectrally distorted noise was presented to both ears; for the dichotic case, pairs of complimentary filters were made simply by changing the sign of the attenuator in one of the filters. The listeners were less sensitive to the presence of coloration for the dichotic both in terms of absolute threshold and for the number of JNDs. The results will be discussed in terms of existing binaural processing models.
Perception of vowel and syllable duration in VC and CVC syllables
Authors: Raphael Lawrence J. , Dorman M. F. , Liberman A. M.
Published: October 2005

Results of previous experiments [Raphael et al., “Vowel information conveyed by consonant transitions,” J. Acoust. Soc. Am. 57, Suppl. No. 1, S49(A) (1975)] indicate that perceptual cues for initial consonants are, in part, incorporated by listeners into their estimates of vowel/syllable duration when they make voicing judgments of word-final consonants. In the present experiment the inventory of initial consonants tested was expanded to include those containing voiced transitions of varying durations, both contiguous and noncontiguous with those of the following vowel; aperiodic transitions; and transitionless noise. Results indicate that listeners incorporate more of the duration of initial consonant cues when they are conveyed by formants than when they are not; when the formants are periodic rather than aperiodic; and when the formants are contiguous with those of the following vowel than when they are not. The degree of incorporation was also found to vary directly with the duration of the formant transitions.
Adaptation to place-of-articulation cues: How long does it last?
Authors: Porter Robert J. , McCormick Colin L.
Published: October 2005

Others have demonstrated that subjects' perceptions of synthetic speech tokens, taken from a VOT or an F2-transition (place-cue) continuum, are changed by repeated presentations of an adapting stimulus which is acoustically and/or phonetically related to the continuum tokens. The study to be reported further examines such adaptation effects by attempting to determine the length of time over which place-cue adaptation effects persist. The duration of adaptation is of interest since short-term effects would suggest a tempory “fatigue-like” state of the system whereas longer-term effects might suggest changes in the system similar to those involved in “learning” phonetic contrasts or in adjusting to different speakers' pronunciations. Preliminary results suggest that the effects have begun to dissipate by two hours and have approached baseline by eight hours. Similarities and differences between these adaptation effects and those in vision will also be discussed.
Perception of speech style and rate in English
Authors: Semiloff-Zelasko Holly
Published: October 2005

Two speakers of American English recorded sentences in four style-rate combinations: formal-slow, formal-fast, casual-slow, casual-fast. Sixty listeners took a perception test where they were to identify the sentences according to style and rate. One group of 30 made the rate decision first, then the style decision; the second group of 30 made the style decision first, then rate. Results indicated the following: (1) Fast and slow rate differences are easier to perceive than formal and casual style differences. (2) The order in which decisions were made did not affect the overall percent correct. (3) Pauses in formal-fast sentences caused them to be judged as formal-slow, although the pause was not a totally unambiguous cue. (4) Formal-slow and casual-fast combinations received a higher percent of correct responses than formal-fast and casual-slow combinations, indicating that listeners associate a slow rate with a formal style and a fast rate with a casual style. Subsequent research on the interrelation of rate and style should concentrate on more subtle delineations of these parameters and investigate ways in which the notion “speech style” can be experimentally measured.
Cues for stop consonants in natural speech: The role of bursts and formant transitions
Authors: Dorman Michael F. , Raphael Lawrence J. , Studdert-Kennedy Michael
Published: October 2005

A series of experiments explored the role of bursts and formant transitions in the recognition of voiced stop consonants. CVCs were recorded in a sentence frame by two informants. Six types of signal were constructed: (1) each CVC was left in its original form; (2) from each CVC the burst was removed; (3) from each CVC, all of the signal prior to voicing was removed; (4) the bursts removed in (2) were attached to VC syllables; (5) the bursts and aspiration removed in (3) were attached to VC syllables; and (6) bursts from [di] and [gi] were attached to each of eight different VC syllables. The effect of burst removal varied with place of articulation and vocalic environment. Recognition of [b] was little affected; recognition of [d] was poorer in front vowel than back vowel environments; recognition of [g] was poor but varied markedly with vocalic environment. The recognition of syllables without formant transitions was generally poor but varied slightly with place of articulation and considerably with vocalic environment. The [di] and [gi] bursts could not be successfully transposed to all other vocalic environments. In general, neither bursts nor formant transitions are sufficient or invariant cues for voiced stop consonants.
Textile industry weave room acoustical absorptive characteristics
Authors: Cudworth A. L. , Hanson W. J.
Published: October 2005

Due to the complex nature of weave room equipment and materials, absorption coefficients of the various surfaces are difficult to calculate. Knowledge of this information could provide a basis for determining noise reduction due to architectural treatment as well as weave room noise level predictions. A typical weave room consists of masonry walls, wood ceilings, and either wood or concrete floors. Volumes are on the order of 400 000 to 900 000 cubic feet, and surface areas 60 000 to 140 000 square feet. Temperatures of 70° to 80° Fahrenheit, and relative humidities approaching 70% are not uncommon. Because of the unusual room proportions, for acoustic calculation purposes, a weave room is essentially two horizontal parallel planes. Reverberation time (decay rate) measurements were made in several weave rooms, and absorption coefficients calculated. Assuming known absorption coefficients of walls and ceiling, and accounting for high-frequency attenuation due to relative high temperature and humidity, floor-machine absorption coefficients have been calculated. Additional tests were conducted to determine decay with distance effects. A single source (fly shuttle loom) noise level was measured at various distances both along and across the weavers alley. This was done in two weave rooms, one with concrete floors and the other with wood.
Ceiling baffles for the remodeled open plan office
Authors: Harris D. A.
Published: October 2005

Achieving speech privacy in the remodeled or renovated open plan office space is usually complicated by the need to retain ceiling assembly fire resistant properties. The designer has few alternatives from which to choose. He may either install ceiling baffles beneath the existing fire-rated assembly or remove the entire ceiling, lights, air terminals, and board, and install a new one. A sprinkler system may also be required. The expense and disruption of the latter solution poses a monumental obstacle. Acoustical properties of ceiling baffles have been evaluated by both objective and subjective tests. Although the baffles are not as efficient as recently developed open plan office ceiling systems, they can satisfy minimal speech privacy criteria. Design of both the baffle and their array is critical. To support these conclusions, test data and evaluation techniques will be presented.
Management aspects of baggage system noise and vibration problems
Authors: Krzak David W.
Published: October 2005

Historical problems related to noise and vibration generated by baggage handling equipment in airline terminals, are briefly discussed. Case study of UAL management measures and problems related to insuring a problem-free installation at a major new terminal. hospital operating rooms or research laboratories.
Building vibrations
Authors: Stratta James L.
Published: October 2005

The rapidly advancing technology of the electronic industry has gone beyond the state of the art of Building Designer. Smaller and smaller geometries in the semiconductor field have necessitated more and more high-powered magnifications. At present, telescopes with magnifications of 500 to 600 are standard practice. Operators of telescopic equipment working at this magnification are required to align geometries, and vibrations must be kept to a level which will not be noticeable so as to disturb the operators. In as much as it is impossible to remove all vibration that can be caused by such sources as traffic, building mechanical equipment, elevators, footfalls, manufacturing equipment, etc., it has become necessary to attempt to develop maximum amplitudes of structural motion that will still allow good operating conditions.
Vowel length before stops of contrasting series
Authors: Maddieson Ian , Gandour Jack
Published: October 2005

Vowels of different length occur before voiced and voiceless final stops in various languages, including English. An analysis of vowel length in Hindi before the stops of four series (plain voiceless, plain voiced, aspirated voiceless, and murmured or “voiced aspirated”) showed three significant degrees of vowel length occur. Vowels are shortest before plain voiceless stops. Vowels before plain voiced and aspirated voiceless stops have one increment of length. Vowels before murmured stops have two increments of length. This provides support for the treatment of murmured stops as voiced counterparts of the aspirated stops, since both voicing and aspiration add one increment to vowel length. Comparable data from other languages with more than two final stop series, such as Eastern Armenian, are being investigated to develop an understanding of the causes of vowel lengthening.
Pitch detection using slope overload noise in an adaptive delta modulation process
Authors: Pillay S. K. , Canas M. A. , Bradshaw F. T. , Lin W. C.
Published: October 2005

This algorithm uses the slope overload noise or error introduced by the adaptative delta modulation (ADM) coding process. This error, greatest at a pitch pulse, decreases in magnitude during a pitch period and displays distinctive autocorrelation differences between voiced and unvoiced speech segments. In this application, the step adaptation and maximum step size parameters are selected to maximize the slope overload error. Autocorrelation techniques applied to the error signal do not precisely delineate the transition points between voiced and unvoiced segments in the time waveform. Hence several other features of the error signal are incorporated into the algorithm. These are the absolute correlation value, normalized peak correlation value, and percent variation of pitch period in adjacent frames. The exact position of the pitch pulse is determined by searching for the maximum value in the error waveform within a variable size time window. Current experimental efforts are directed towards further validation of the preliminary results. This technique of pitch detection is compared with that of the simplified inverse filtering technique (SIFT), reported by Markel [IEEE Trans. Audio Electroacoust. AU-20(5), 367–377 (Dec. 1972)l. Simplicity of the ADM process over the linear predictor approximation used in the SIFT algorithm makes this new technique more attractive to hardware implementation.
Voicing detection by Kalman filtering
Authors: Arcese A.
Published: October 2005

In this talk, we present an algorithm for the automatic detection of voiced and unvoiced speech segments. The algorithm makes use of the Kalman filter where the state transition matrix is taken as the identity matrix. The attributes used for discrimination are the partial correlation coefficients PARCOR). Initially, the algorithm requires a teacher; thereafter, it becomes self-learning. Audio tapes demonstrate speech processed with this algorithm on an LPC-10 system.
LPC Synthesis from speech inputs containing: quantizing noise and additive white noise
Authors: Sambur M. R. , Jayant N. S.
Published: October 2005

An important problem in some communication systems is the performance of Linear Prediction analysis with speech inputs that have been corrupted by (signal-correlated) quantization distortion or additive white noise. To gain a first insight into this problem, a high-quality speech sample was deliberately degraded by using various degrees of adaptive differential PCM (ADPCM coding (two or more bits of quantization per sample) and by the introduction of additive white noise. The resulting speech samples were then analyzed to obtain the LPC control signals: pitch, gain,and the linear prediction coefficients. A distance metric proposed by Itakura was used to compare the original LP coefficients with the coefficients measured from the degraded speech. Other statistical techniques were used to compare the various measurements of pitch. In addition, the measured control signals were used to synthesize speech for perceptual evaluation, Results suggest that LPC analysis/synthesis is fairly immune to the degragation of ADPCM quantization. The effects of additive white noise are, however, more severe.
Temporal sharpening within response areas of medullary neurons
Authors: Crow G. W. , Rupert A. L. , Moushegian G.
Published: October 2005

It is well known that auditory medullary neurons preserve the temporal characteristics of low-frequency sounds by discharging at integral multiples of the stimulus period. Monaural recordings from kangaroo rat cochlear nuclei and binaural superior olive neurons show that the variability in synchrony of neural discharges decreases as the frequency of the sound increases. Variability in synchrony is apparent from period histogram displays and may be expressed by vector strength and interquartile range. These results provide a possible neural basis for the finding that the difference limen for pitch and the minimum audible angle, in absolute time, decreases as the frequency of the sound increases. Evidence is also presented that relates the shortest intervals of neural discharge to the loss of synchrony as frequency of sound stimulation increases. [Supported by AFOSR and NINDS.]
Representation of intensity and frequency information in auditory cortex of monkeys performing a simple auditory task
Authors: Pfingst Bryan E. , O'Connor Thomas A. , Miller Josef M.
Published: October 2005

Intensity functions were measured for single neurons at the cell's best frequency and at surrounding frequencies within the receptive field. Subjects for these experiments were adult male rhesus monkeys. Single neurons were isolated using a vertical approach to the superior temporal plane. Subsequent electrophysiological and anatomical data indicated that cells were located in cortical areas A1, CM, L, and RL [Merzenich and Brugge Brain Res. 50, 275–296 (1973)]. Data were collected while monkeys were performing a simple auditory reaction-time task. As shown in this and many previous studies, in the absence of such a uniform behavioral control much variation in responses of auditory cortex neurons is encountered. The reaction time task was also used to provide an ongoing measure of the subject's threshold and suprathreshold hearing during recording of neural responses. Data are reported for encoding of stimulus intensity and frequency and discussed relation to behaviorally measured hearing. [Supported by NIH Grants NS-08181, PR-00166, and GM-00260.]
Effects of behavior attention on unit responses in the auditory cortex
Authors: Miller Josef M. , Pfingst Bryan E. , Hienz Robert
Published: October 2005

Evoked activity of single cells from the auditory cortex of monkeys was examined under two behavior conditions: when the monkey was “attending” to the auditory stimulus and when he was not. Monkeys were trained in a reaction-time task, in which lights and tones of a fixed duration were randomly presented to the animal. Under one visual cue, the animal was to perform the reaction-time response only to the tone. Under another visual cue he was to respond only to the light. Thus evoked unit activity to an auditory stimulus was recorded under two conditions: (1) when the animal was responding to that stimulus in a reaction-time task (“attending”) and (2) when the animal was performing in a visual reaction-time task with a free tone stimulus. For excitatory “on” responses, evoked discharges were consistently greater and occurred with a shorter initial latency in the condition under which the animal was behaviorally responding to the auditory stimulus than when he was not. Quantitative information on the changes in the characteristics of excitatory and suppressed responses under these two conditions of light-tone discrimination as well as responses to the same stimuli seen in the simple auditory reaction-time task and under the nonperformance condition are described. [Supported by NIH Grants NS-08181 and RR-00166.]
Effect of click repetition rate on the latency of human auditory brainstem responses
Authors: Don M. , Allen A. , Starr A.
Published: October 2005

Auditory brainstem responses are the far-field reflection of electrical activity originating in the auditory pathway in its course from the cochlea to cortex that can be recorded from scalp electrodes using computer averaging techniques. There are seven components in the initial 10 msec following a click signal which have been shown to have an orderly change in latency as a function of signal intensity. The results of this study show that click repetition rate can also significantly affect the response latency measure. Responses were measured in ten normal hearing subjects at click rates of 10 to 100/sec and at four intensity levels (30, 40, 50, 60 dB SL). The mean latency shift of component V was 0.5 msec when responses at 10 and 100/sec were compared. This is equivalent to a 20-dB decrease of signal intensity at the 10/sec click rate. An analysis of the time of occurrence of this shift using brief click trains at 100/sec showed the shift to be complete by the fifth click. The latency shift was similar at the four signal levels tested. The results are interpreted as an objective measure of adaptation in the human auditory system with implication for the measurement in disorders of hearing.
Underwater acoustic scattering from the random, moving ocean surface
Authors: Middleton David
Published: October 2005

A new approach to scattering from the random, moving surface of the ocean is described, which combines the “classical” or continuum scatter theories with the more recently developed FOM (Faure-Ol'shevskii-Middleton) theory, based on random “specular” point scatterers, obeying Poisson laws. Among the new features of the theory are (1) its ability to deliver exact, statistical solutions to the governing Langevin equations of propagation in their interactions with the scattering interface; (2) the explict inclusion of the dynamic Doppler (omitted in the classical treatments), which produces nonstationary random frequency modulations of the incident signal, in addition to the more familiar, stationary phase modulation arising from the random surface positions; (3) a theoretical and experimental technique for determining when one or the other of the continuum or specular point scattering mechanisms is preemptively predominant (for second-order statistics); (4) a second-order, four-parameter characterization of underwater acoustic channels; and (5) among others, various experiments to obtain point and directional surface wave spectra (acoustic Crombie-Barrick experiments). The general approach is developed and illustrated by recent results of the author [“Characterization of Active Underwater Acoustic Channels,” Rep. ARL-TR-74-61, Applied Research Laboratories, University of Texas at Austin (Dec. 1974)]. [Work supported primarily by the Naval Sea Command.]
National Measurement System for Acoustics
Authors: Pallett David S.
Published: October 2005

Measurement, per se, is one of the fundamental activities upon which the functioning of human cultures depends. Over the past two years, the staff of the National Bureau of Standards has conducted a study of the role of acoustical measurements as an element in the National Measurement System. This system is conceived to be the network of measurement standards, techniques, procedures, personnel, and equipment through which all measurements are conducted. Conduct of a measurement operation, or reliance upon measurement data, thus makes one a participant in the National Measurement System. Most acousticians are participants. The emphasis of the present study has been to review the status of the system in order to determine the adequacy of these important physical measurements and to promote improvements within the measurement system. This paper outlines the significance of measurements in acoustics and indicates the relevant physical quantities. The roles of the acoustical standardization institutions and interactions occurring between participants are specified. Technological, social, and economic impacts are outlined. Finally, the status and trends of the system in adapting to changing technology are discussed.
Analysis of cost effectiveness of noise control by regulation
Authors: Kende L. G.
Published: October 2005

Trends in regulations and guidelines of U.S., Canada, and Europe are summarized. These trends, based on task and sleep interference, as well as annoyance, are presented in terms of normalized A-weighted sound levels. Effectiveness of noise control is measured in relation to the reduction of A-weighted sound levels with respect to the summarized regulatory criteria. Several major complaint sources are identified including residential air conditioning, road transportation, and industrial plant operation. Comparison of retrofitting with planned noise control is made on specific examples on the basis of (1) best identical technological measure, (2) best technological measure; and (3) best preplanning consideration with respect to best retrofitting measure. [Work supported by MOE.]
Average sound level including sonic booms
Authors: Young Robert W.
Published: October 2005

The primary and practical means of monitoring noise in the community or in the factory is to measure the average sound level during a time like an hour, eight hours, or a day. Average sound level is also called equivalent continuous sound level. Sonic booms have been commonly measured and reported in terms of peak wide-band overpressure, and with less than the 60dB dynamic range needed for A weighting. The present measurements were made directly on more than 16 sonic booms that occurred from airplanes flying the medium-altitude supersonic corridor northeast of Edwards AFB. The noise measure of a single event particularly useful for calculation of average sound level is the (time integral of A-weighted squared pressure) sound exposure level. Also measured was the flat (8 Hz–8 kHz) sound exposure level. For aircraft mostly of length on the order of 18 m (59 ft), the peak flat sound pressure level exceeded the flat sound exposure level about 12 dB. The peak flat sound pressure level of 122 dB exceeded the sound exposure level (which possibly included engine noise) by 42 dB, but the difference decreased to 34 dB when the peak pressure level was 135 dB.
The simulation, superposition, and active reduction of higher-order modes in cylindrical ducts
Authors: Oslac M. J.
Published: October 2005

The use of a “Spinning Mode Synthesizer” (SMS) [M.J. Oslac and J.M. Seiner, Proc. Purdue Noise Conf. (1971)] has made possible the study of the generation, propagation, and active reduction of singular and multiple higher-order modes in cylindrical ducts. This paper presents the results of an analytical and experimental investigation into the use of the SMS as an active device for the reduction of higher-order modes. The results of singular mode cancellation as well as multiple mode (superposition of two or more modes) cancellation are presented. Reductions of from 20 to 50 dB have been achieved in the ideal case of using one-half of the SMS as a modal generator and the other half as a “phase-opposite” controllable source. Results will also be presented on the generation and simultaneous propagation of modes at multiple frequencies.
Attenuation of sound in ducts with acoustic treatment—a generalized approximate equation
Authors: Rice E. J.
Published: October 2005

A generalized approximate equation for duct lining sound attenuation is presented. The specification of two parameters, the maximum possible attenuation and the optimum wall acoustic impedance, is shown to completely determine the sound attenuation for any acoustic mode at any selected wall impedance. The equation is based on the nearly circular shape of the constant attenuation contours in the wall acoustic impedance plane. For impedances far from the optimum the equation reduces to Morse's approximate expression. The equation can be easily used for initial acoustic liner design. Not least important is the illustrative nature of the solutions which provide an understanding of the duct propagation problem which is usually obscured in the exact calculations. An additional property of the approximate equation is that empiricism can be injected into the expression to account for phenomena which can not currently be handled by exact analysis. The peak and the bandwidth of the sound attenuation spectrum are presented by quite simple functions of the ratio of actual wall acoustic resistance to optimum resistance.
Psychophysical tuning curves in normal and impaired ears
Authors: Leshowitz B. , Lindstrom R. , Zurek P.
Published: October 2005

Frequency selectivity in ears suffering sensorineural hearing deficits was assessed using a tone-on-tone masking paradigm. Psychophysical tuning curves relating level of a continuous tonal masker required to just mask a probe of constant frequency and intensity to the frequency of the masker were obtained for several normal and impaired ears. In agreement with the recent observation of Vogten (in Facts and Models in Hearing, E. Zwicker and E. Terhardt, Eds.), for high-level probes (≃70 dB SPL) psychophysical tuning curves displayed well-defined minima (corresponding to maximum masking) at masker frequencies below the frequency of the probe. This “negative-maximum-masking frequency” effect was observed at sensation levels of 20 dB or less in both normal ears exposed to wide-band background noise and in ears impaired by acoustic trauma. In one patient having a uniform hearing loss of 60 dB at low frequencies, tuning curves did not contain either negative-maximum-masking frequencies or well-defined minima. Interpretations of the data in terms of the interaction between the masking effects of the tonal background and the effects of sensorineural impairment are discussed.
Intelligibility of low- and high-pass filtered speech set at 1900 Hz for hearing-impaired subjects
Authors: Condon Ellen , Asp Carl
Published: October 2005

Speech intelligibility for Modified Rhyme Test words was measured through a 1900-Hz low-pass and a 1900-Hz high-pass filter condition. The stimuli was presented monuaurally to twelve subjects with a moderately sloping sensori-neural loss at a 20, 30, and 40 dB level above the subject's speech detection threshold. The speech discrimination scores were significantly better through the low-pass filter condition over the high-pass filter condition. No significant difference was observed as a function of sensation levels, or for the interaction of filter conditions and sensation levels.
Effects of the acoustic reflex on speech intelligibility of hearing-impaired listeners
Authors: Martin R. L. , Asp C.
Published: October 2005

Eight hearing-impaired subjects were given Modified Rhyme Tests at three sensation levels, with and without an active reflex in the test ear. The reflex was elicited by filter noise presented in the contralateral ear. The reflex depressed the speech intelligibility scores for the low sensation levels and had no effect upon the highest sensation level. The implications of wearing a hearing aid which elicits bilateral reflexes and the effect of such reflexes upon speech intelligibility were considered.
Quantification of the acoustic reflex and its relation to stimulus loudness
Authors: Block Michael G. , Wightman Frederic L.
Published: October 2005

To develop a rigorous technique for quantifying acoustic reflex magnitude, a computer sampled baseline and perstimulatory conductance and susceptance. For each component, the difference between baseline and perstimulatory means were divided by the pooled standard deviation yielding a statistic similar to d′. Component d′'s were combined by taking the square-root of the sum of d′2's. The resultant d′ represented acoustic reflex magnitude in an objective statistically based format. The level required to elicit criterion magnitudes were estimated using the PEST adaptive procedure. Magnitude-intensity functions were constructed for a 1-kHz tone, 30- and 50-Hz-wide noise bands and white noise. The estimated levels were references in loudness balances among all stimulus pairs, yielding four equal-loudness contours. Equal-loudness and equal-reflex contours had similar shapes suggesting a relationship between loudness and the acoustic reflex. The magnitude-intensity functions were steeply sloped; therefore, small intensity changes produced large magnitude changes. We concluded that the data do not permit rejection of the hypothesis that stimuli eliciting equal reflexes are equally loud. However, the quantification technique has value as an objective alternative to analog tracing interpretation and can replace “threshold” with a mathematically defined magnitude. Experiments to refine this technique are in progress. [Research supported by NIH grant.]
Development of a protocol for fitting a wearable master hearing aid
Authors: Levitt H. , White R. E. C. , Resnick S. B.
Published: October 2005

An experimental protocol for fitting a Wearable Master Hearing Aid (WMHA) has been developed from which an abridged practical protocol will be extracted. There are five stages to the experimental protocol. In the first stage, conventional audiological data are acquired. Second stage data are obtained using a Latin square design, covering a range of operating conditions and parameter values of the WMHA, to determine the relative role of the various factors affecting hearing aid performance for the subject. The third stage consists of obtaining comparative data between the speech discrimination tests used in this study and other tests of speech discrimination. From data acquired in the second stage, an initial estimate is made of the optimum settings of the hearing aid. The fourth stage consists of a search, using standard optimization techniques, for the best estimate of these optimum settings. The fifth, and final, stage consists of a comparison between the subject's original hearing aid and the WMHA, as fitted according to this protocal. [Research supported by NIHCDS.]
Hearing of personnel included in the USAF hearing conservation program
Authors: Gasaway Donald C. , Sutherland Harrell C. , Danford Roy
Published: October 2005

In October 1956 the U.S. Air Force established a formal hearing conservation program that employs threshold monitoring audiometry. A hearing conservation data registry was also established to study and accumulate hearing data of military and civilian personnel who routinely work in noise. Recent revisions of this program have resulted in increased emphasis on hearing conservation studies and evaluations. The authors report the result of one segment of this study—the annual audiometric monitoring of 26 446 personnel (22 817 military and 3629 civilian employees) during January through March 1975. The data derived from this study are reported separately for military and civilian employees and for right and left ears using threshold-averaging techniques. For example, comparisons are reported for average hearing at 500, 1000, and 2000 Hz (speech frequency average) with averages for 1000, 2000, and 3000 Hz (procedure proposed by Working Group 77 of NAS-NRC Committee on Hearing, Bioacoustics, and Biomechnics) and 2000, 3000, and 4000 Hz (procedure proposed by Occupational Safety and Health Act). Results of the study are summarized and discussed.
Comparison of simple criteria used to identify significant amounts of threshold shift in persons who work in noise
Authors: Gasaway Donald C. , Sutherland Harrell C.
Published: October 2005

Since October 1956, the U.S. Air Force (USAF) has monitored the bearing of 3 967 000 military and civilian personnel as part of a hearing conservation program. During this 19-year period, two criteria have been used to identify threshold shifts that require subsequent follow-up. These criteria are somewhat complicated and errors in identification have resulted. The authors initiated a study to develop a simplified technique to significantly reduce errors without seriously compromising early identification of persons who demonstrate shifts in hearing attributable to noise. Values of threshold shift were studied using 32 572 audiometric records where current audiograms (500–6000 Hz) were compared to reference threshold levels. Four proposed criteria are compared with the current USAF criteria: (1) 20 dB TS at any frequency, either ear, (2) 20 dB TS at 2000, 3000, or 4000 Hz, either ear, (3) 15 dB TS at any frequency, either ear, and (4) average of 10 dB at 2000, 3000, or 4000 Hz, either ear. Study results are presented and discussed.
Measurement of cavitation acoustic pressure and bubble dynamics in liquid helium II
Authors: Marston P. L.
Published: October 2005

Acoustic cavitation nucleated in bulk He II at a temperature of 2.10°K was photographed at 6000 frames-per-second. The vapor bubbles were formed at pressure antinodes near the axis of a PZT4 cylindrical transducer operating at 52 KHz with a wavelength of 4.4 mm. The diffraction pattern of light traveling along the cylinder axis was used to adjust the frequency for maximum acoustic intensity and to measure the acoustic pressure. For a pressure of 280:± 70 mbar rms, spherical bubbles originating in the liquid grew to 1 mm diameter in less than 0.3 msec. They fragmented during the next 0.6 msec to form smaller nonspherical cavities which persisted for tens of milliseconds. The smaller nonspherical cavities could also be observed at lower acoustic pressures where they originated on the surface of a probe and not in the bulk fluid. Previous photographs of He II cavitation [A. Mosse, M.L. Chu, and R.D. Finch, J. Acoust. Soc. Am. 47, 1258 (1970)] showed similar nonspherical cavities without revealing their origin.
Ultrasonic relaxation in gas phase mixtures of H2O/N2, H2O/Argon, and H2O/O2 at 500 °K
Authors: Bass H. E. , Keeton Roy G.
Published: October 2005

Ultrasonic absorption in mixtures of H2O and an impurity M=O2, N2, and Argon has been measured at 500 °K for frequency to pressure ratios from 2 to 50 MHz/atm. Measurements were made for mixtures of 85% H2O/15% M, 65% H2O/35% M, and 45 H2O/55% M. From these absorption measurements the vibrational and rotational relaxation times in pure water vapor were deduced as well as the rate at which Argon, O2, and N2 vibrationally de-excite H2O. In addition, by using earlier work at low H2O concentrations, the rate at which H2O de-excites O2 by vibration-translation and vibration-vibration energy transfer have been determined. [This work supported by the U.S. Army Research Office, Durham, NC.]
Parameter-specific voice-difference effects in dichotic CV identification
Authors: Haggard M. P. , Paul D. J.
Published: October 2005

Earlier RT studies showed greater perceptual difficulty in adjusting to voice differences between trials involving vocal tract information (F1F2F3) than larynx information (F0), irrespective of discriminability. Comparable voice differences were used in a dichotic simultaneous masking paradigm where half the subjects heard varying voice on the target ear only and half on the mask ear only. The identification set /bE, dE, b3, d3/ varied on both ears. Sets of trials with the two types of voice difference produced distinct results. For target-varying subjects, intrusions were more frequent overall and F1F2F3 differences depressed both C and V intrusions. This suggests that voice-changes create an attention demand, though this is evident only when attended spatial location and location of the novel voice conflict. F0 differences between ears depressed V intrusions but slightly enhanced C intrusions. The acoustical properties of vowels appear to carry a relatively reliable local sign for ear that can be supplemented, possibly both in analysis and late response retrieval, by differences in F0. For the poorly located consonant features similar enhancement of spatial discriminability may be favouring interruption at an early perceptual stage.
Dichotic listening: reversals in ear advantage
Authors: Speaks Charles , Niccum Nancy , Carney Edward , Marble Kathleen
Published: October 2005

Synthetic CV syllables (stop+/a/) were presented dichotically to two listeners (mean REA = 11%). Although the main findings paralleled those of Studdert-Kennedy and Shankweiler [J. Acoust. Soc. Am. 48, 579–594 (1970)], the result of interest was for the 12 double-contrast pairs. The mean REA of 12% for these pairs obscured the fact that the ear advantage was REA = 62% with voiceless stops in the left ear (V − Left, V + Right), but LEA = 37% with voiceless stops in the right ear (V + Left, V − Right). The same reversal occurred for a third listener (LEA = 4%). This reversal was not obtained for voicing-contrasted, place-shared pairs for two of the three listeners, indicating that it does not reflect simply a V+ feature dominance. The reversal in ear advantage disappeared when the same listeners were tested with natural CVs, suggesting that (1) the synthetic CVs contain artifacts (although monotic scores were nearly 100%); (2) the reversal depends upon overall level of difficulty (scores were higher for natural CVs); or (3) that results with synthetic stimuli differ qualitatively from those with natural stimuli. [Supported by PHS Grant No. NS-12125.]
Investigation of the measurement of transmission loss
Authors: Bies D. A.
Published: October 2005

The measurement of the transmission loss of low noise reduction apertures has been investigated using the two-reverberant-rooms method. It is shown that the standard procedure will be in error if the noise reduction is less than 10 dB, but that a new procedure, which will be described, may be used to obtain good measurements even for test items with noise reductions as low as 3 or 4 dB. The determination of the transmission loss of low-noise-reduction apertures has shown anomalous results for the measurements for a modal overlap of less than unity in one or the other of the two rooms used for the measurements. At higher frequencies, numbers very close to 0 dB for the transmission loss are obtained as would be expected. However, for large apertures, anomolies also begin to appear at high frequencies. The procedures, results and their implications are discussed in the paper.
Characterization of room coloration by moments of room spectral response
Authors: Curtis T. H.
Published: October 2005

Coloration is the perceived effect of changes in the short-time spectrum of speech produced by its transmission in a room, The relationship between moments of the room spectral response and a subjective evaluation of the coloration in speech has been studied. Initial results indicate that the average value of the second moment of the room response correlates with coloration. This method shows promise as a way of characterizing the acoustic quality of rooms for hands-free telephony.
Evaluation of the large anechoic chamber at the National Bureau of Standards
Authors: Hruska G. R. , Koidan W.
Published: October 2005

An estimation of how closely the large anechoic chamber at NBS approximates a true free field was obtained by three different methods. First, a microphone was moved away from a stationary source and the sound pressure recorded on punched paper tape as a function of the distance r from the source, Using an electronic computer, a least-squares fit was obtained to the curve of pressure versus 1/r and deviations from the ideal curve calculated. An integral part of this calculation was the determination of the distance between the acoustic centers of source and receiver. Secondly, the source and receiver were kept a constant distance apart as they were moved across the chamber, Deviations from the mean pressure were recorded. The last method made use of tone bursts to determine the amplitude of sound reflected from the room boundaries compared to the incident sound amplitude. In the first two methods, receiving microphones and sound sources were as omnidirectional as possible to take into account reflections from all surfaces of the chamber. The range of frequencies covered was 40 Hz to 63 kHz.
Effect of time varying boundary on the spatial variance and sound power radiated into the reverberation chamber
Authors: Tichy J.
Published: October 2005

The reverberation chamber with the time varying boundary is represented by a parallelopiped with one wall oscillating at the amplitude equal to 1/4 of the single frequency, radiated by a monopole source at random position inside the chamber or at the chamber wall. Using the theory developed by F. Hwang [J. Acoust. Soc. Am. 52, S46(A) (1975)], the calculations of spatial distributions of p2 and its variance as a function of frequency and source position in the region of low modal overlap were performed, Comparison with a chamber with nonvarying boundary indicates a decrease of p2 variance by a factor of 0.25–0.8. For the source position inside the chamber, at least 1/4 distance from the walls, the sound power averaged over 23 frequencies and five source positions is 3–4 dB lower than the power which would be radiated into the free space. Placement of the source reduces this difference to a fraction of the decibel. This paper also presents data on the qualification according to ANSI 1.21.
Finite-element modeling of piezoelectric ceramic hydrophones
Authors: Winnicki R. T. , Auyer S. E.
Published: October 2005

The finite-element method has been used for several years to predict the response of hydrophones. This paper discusses the application of the technique to the class of hydrophones consisting of right circular air-filled cylinders of piezoelectric ceramic with semirigid nonpiezoelectric end caps. The acoustic sensitivity of this class of hydrophone can be calculated from a solution developed by R.A. Langenvin [R.A. Langenvin, J. Acoust. Soc. Am. 26, 421–427 (1953)]. This solution assumes an ideal response of the hydrophone which does not include the effects of end cap bending or the restriction of movement of the piezoelectric cylinder at the end-cap cylinder boundary. The contribution to the hydrophone's acoustic sensitivity of the radial, axial, and circumferential stresses are calculated by the finite-element method and are used to examine the effects on the hydrophone performance of a nonideal response which includes end-cap bending and the restriction of cylinder movement at the cylinder end-cap boundary. A computer generated film depicts the response of an ideal and nonideal hydrophone.
Optimum segmented acoustic liners for flow ducts
Authors: Sawdy D. T. , Beckemeyer R. J. , Patterson J. D.
Published: October 2005

Mode-matching analytical models for segmented ducts have been developed to take advantage of the liner impedance discontinuities in the design of optimal duct acoustic liners of one, two, and three segments. The analytical studies have been reported previously [R.J. Beckemeyer and D.T. Sawdy, J. Acoust. Soc. Am. 52, Suppl. No. 1 S16(A) (1975)]. In the present paper, the analytical studies have been extended to include flow ducts containing uniform mean flow, and experimental results are reported for Mach numbers from zero to 0.4. Excellent analytical-experimental correlation has been obtained, thus validating the analytical procedures used to design optimum segmented duct liners. [Supported by the NASA Lewis Research Center under Contract NAS 3-18550.]