bluuebunny/crossref_metadata_2025 huggingface dataset:
Psychometric functions and thresholds of audibility obtained with different methods as a function of duration and frequency
Authors: Valerio Michael W. , Wright H. N.
Published: October 2005

Two interval forced choice (2IFC) psychometric functions and thresholds of audibility, measured with the methods of adjustment, Bekesy tracking, and 2IFC tracking [Zwislocki et al., J. Acoust: Soc. Am. 30, 254–262 (1958)], were obtained for 500- and 20-ms tones at 250, 1000, and 4000 Hz in the absence of masking noise. Custom fitted ear inserts maintained constant transducer placement. The signal intensity at a percent correct P(C) of 76% was designated threshold on each psychometric function. Values of P(C) obtained at threshold intensity with each method, designated the relative position each threshold assumed on it's respective psychometric function. Test-retest results indicated that thresholds obtained with different psycho-physical methods have associated with them stable, but nevertheless different P(C) values. Moreover, statistical comparisons among the threshold for tones, as a function of their duration and frequency, revealed a uniform and monotonic decrease in temporal summation as frequency increased with all methods.
Hearing in the jewel cichlid
Authors: Jun Shen , Coombs S. , Fay R. , Popper A. N. , Saidel W. M.
Published: October 2005

An in-depth analysis of hearing sensitivity for the jewel cichlid, Hemichromis bimaculatus was made using neurophysiological techniques to measure sensitivity at various levels of the auditory system and psychophysical techniques to measure whole-system sensitivity. For physiological studies, fish were stimulated by sinusoids generated by a loudspeaker or by direct vibration of the fish's head. Microphonic potentials were recorded from the sacculus of the inner ear and evoked potentials were recorded at several depths from the forebrain, midbrain, cerebellum, and medulla. For psychophysical studies, shock avoidance techniques were used in training fish to respond to tones generated by a loudspeaker. While audiograms based on the various neurophysiological and behavioral measures differed in detail, they showed in general that the auditory system of the jewel cichlid responds to sounds between 70 and 600 Hz, with best sensitivity in the region of 100 to 300 Hz. Evidence exists that within the frequency region of best sensitivity, the auditory system responds to both sound pressure and particle motion. The methodological difficulties in obtaining independent measures of sensitivity to pressure and particle motion will be discussed. [Supported by grants from NSF and NIH to R.F. and A.P.]
Acoustical designing for speech quality in theaters
Authors: Latham Howard G.
Published: October 2005

Speech quality, a subjective measure of overall excellence determined by preference testing, was originally developed for evaluating speech system designs. The application of speech quality measurements to the evaluation of auditorium acoustics design required two models: a real-time multichannel electroacoustic model to simulate physical sound fields, and a linear/scalar products mathematical model to analyze subjective preferences (together with an analysis of variance method for determining statistical significance and scalability levels). In the present experiments, speech quality is shown to be either multidimensional or unidimensional in configuration, and valid at either interval or ordinal levels of measurement. Four consensus dimensions (reflections ratio; excessive reverberation; discrete echo disturbance; naturalness) and three individual differences dimensions (reverberance; spatial coloration; evenness of echogram) were identified. The objective index, signal-to-noise ratio, gave a high correlation with speech quality for simulated sound fields without excessive background noise. [Work supported by the SERC, United Kingdom.]
A behavioral audiogram for the fishing bat, Noctilio leporinus
Authors: Wenstrup Jeffrey J. , Suthers Roderick A.
Published: October 2005

Behavioral audiograms were obtained from the fishing bat Noctilio leporinus which uses both constant frequency (CF) and frequency modulated (FM) pulses for echolocation. Three individuals were tested in a two choice operant conditioning procedure using tone bursts at various frequencies between 500 Hz and 120 kHz. Although there was some individual variation, the maximum auditory sensitivity (−1 to +5 dB SPL re 2.0 × 10−5 N/m2) was always within the range of the fundamental frequencies of the sonar pulse (32 to 58 kHz). Below this range, sensitivity declined at 30–40 dB/octave. Sensitivity below 10 kHz was very poor. There was a sharp reduction in sensitivity just above the CF component (56–58 kHz) of the sonar pulse and a plateau in the auditory sensitivity at frequencies corresponding to the second harmonic. N. leporinus thus lacks the sharply tuned auditory sensitivity characteristic of several other CF bats. However, its audiogram is in good agreement with the spectra of both its orientation pulses and communication sounds. [Supported by NSF grant BNS 79-05606.]
Mechanism for production of ultrasonic sonar pulses by the constant frequency echolocating bat, Pteronotus parnellii: Subglottic pressure and tracheal airflow
Authors: Suthers Roderick A.
Published: October 2005

The echolocating bat, Pteronotus parnellii, emits high-intensity orientation pulses having most of their energy in a nearly constant frequency component at about 60 kHz. Brain stimulation was used to elicit echolocative pulses from animals restrained in front of a microphone. The rate of tracheal airflow and subglottic pressure during phonation were measured directly via a tracheal cannula containing a microbead thermistor heated to a constant temperature and a miniature piezoresistive pressure transducer. The relationship of airflow and pressure to the intensity and frequency of the sonar pulse was investigated. The mean sound pressure level of orientation pulses measured 34 cm in front of the mouth was about 98 dB (re 2.0 × 10−5 N/m2). The mean peak subglottic pressure during pulse emission ranged from 29.7 to 42.5 cm H2O in different animals and the mean rate of tracheal airflow was between 4.0 and 4.3 cc/s. [Supported by NSF research grant BNS 79-05606.]
Detection of moving and stationary targets by Eptesicus fuscus
Authors: McCarty J. Kenneth , Jen Philip H.-S.
Published: October 2005

Sensitivity of a bat's echolocation was studied by training Eptesicus fuscus to echolocate and approach a presented stationary or moving target. The target, a spheric steel ball with a diameter of 2.0 cm, was randomly placed in the direction of one of the two ramps of a Y-shaped platform. The distance between the bat and the target ranged from 30 and 130 cm. By means of an electric motor, the target could be made to oscillate back and forth with an average speed of 4, 6, or 10 cm/s. Our results showed that a bat could detect a moving target with a higher percent of correct responses than a stationary one. This was also true even when one of the bat's ears was plugged, although the overall performance decreased. These data suggest that a moving target contains a higher releasing value than a stationary one. [Work supported by NSF BNS 80 07348 and USPH 1-KO4-NS00433-01.]
Is the equal energy rule applicable to impact noise?
Authors: Henderson Don , Salvi Richard , Hamernik Roger P.
Published: October 2005

The equal energy rule (EER) is a theoretical framework for evaluating the acoustic trauma associated with a wide variety of noise exposures. Most of the data supporting the EER have been large scale demographic studies. Controlled laboratory studies, especially with impulse noise, have brought into question the generality and validity of the EER. The present study assessed the hearing loss resulting from three impact noise exposures having equal energy, but different peak levels (107, 113, 119 dB SPL) and repetition rates (4/s, 1/s, 1/4s). Hearing loss was assessed at 0.5, 2, and 8 kHz in three groups of chinchillas using the auditory evoked response. The animals were exposed for five days and developed an asymptotic threshold shift (ATS) during the exposure. The animals, however, did not develop the same amounts of ATS for each exposure as predicted by the EER; instead the hearing loss increased with peak level at the rate between 0.6 and 1 dB ATS/dB peak SPL. [Work supported by NIOSH.]
A comparison of frequency and intensity discrimination in humans and monkeys
Authors: Sinnott Joan M. , Petersen Michael R.
Published: October 2005

Although Stebbins [Am. J. Phys. Anthropol. 38, 357–364 (1973)] has reported that humans are “slightly” superior to monkeys in frequency and intensity discrimination, these comparisons were made across separate studies using different test procedures. The present experiment compared the discriminative capacities of humans and monkeys (Macaca fuscata) under identical test conditions. A repetitive standard technique was used in which subjects reported changes in the frequency or intensity of an ongoing train of 1-kHz tone pulses, presented monaurally at 50 dB SL. Human and monkey ΔI's for intensity increments were very similar (1.5–2.5 dB). However, human ΔF's for frequency increments (1–2 Hz) were clearly superior to those of monkeys (15–20 Hz). Species differences also occurred when subjects were required to detect decrements in frequency or intensity. Human − ΔF's (2–3 Hz) were slightly larger than those for frequency increments, but monkeys − ΔF's were about two to three times as large (30–50 Hz). Finally, human − ΔI's were similar to those for intensity increments, but we have as yet been unable to train monkeys to reliably report intensity decrements as large as −10 dB. Implications of these results for using monkeys as models of human speech sound perception and for the design of species-specific vocalizations will be discussed. [Work supported by NSF, NIH, and Deafness Research Foundation.]
ROC analysis of neural responses during auditory signal detection in the rabbit
Authors: Kettner Ronald E. , Thompson Richard F.
Published: October 2005

Rabbits were classically conditioned to respond to a white noise stimulus at behavioral threshold (50% correct responding) using movement of the nictitating membrane as the behavioral response. During detection performance unit clusters were recorded from the anteroventral cochlear nucleus, the central nucleus of the inferior colliculus and the ventral division of the medial geniculate body. For each electrode placement an ROC curve was constructed for the neural unit response to stimuli which were behaviorally detected and a similar ROC curve was constructed for nondetected stimuli. Post stimulus time histograms of the average unit response to detected and nondetected stimuli were also created. Both the ROC and the histogram analyses suggest that there are small differences between detected and nondetected stimuli but that these differences are not sufficient to account for signal detection behavior. In all three nuclei studied the neural response was well defined and essentially identical on detection and nondetection trials. Statistical analyses confirmed these results. These data show that the neural response to an auditory stimulus can be well defined even when the auditory stimulus is not behaviorally detected and suggest that auditory signal detection performance is not determined by three primary brain nuclei along the auditory pathway.
Latency loudness functions following COCB section: Recruitment of masked tones
Authors: Moore Maurus J.
Published: October 2005

Latencies to dichotic tones presented either alone or with noise masking were obtained from ten rats before and after surgery. The crossed olivocochlear bundle (COCB) was cut in five animals, but left intact in control subjects. COCB sectioned animals differed from Sham controls only when the tones were masked by noise. The two groups did not not differ (a) in masked or unmasked threshold estimates, or (b) in unmasked loudness functions computed from latency data. Masked loudness functions from COCB sectioned animals exhibited recruitment from unchanged masked thresholds, and saturated within 30 dB. At saturation, masked tones were louder than unmasked tones (latencies were shorter). The lack of effect in the absence of masking agrees with Dewson [J. Neurophysiol. 31, 122–130 (1968)]. These results support the concept of a negative feedback system mediated by COCB neurons, but which may only operate in the presence of broadband input. COCB activity may reduce the gain at the periphery, with an improvement in the dynamic range of the system under masking being the result. [Work supported by USAF.]
A rigorous examination of the real-ear attenuation at threshold method (125 Hz–2 kHz)
Authors: Berger E. H. , Kerivan J. E.
Published: October 2005

The most ubiquitous method of measuring the attenuation of hearing protection devices (HPDs) is that of real-ear attenuation at threshold (REAT). Such tests, which evaluate the hearing thresholds of human subjects both with and without the HPDs in place, have often been criticized for measurement artifact and inappropriate test conditions. We examined two of the more common alleged REAT problems; masking of the occluded thresholds due to amplified physiological noise and the possible errors resulting from extrapolating low sound level performance tests to the high sound levels in which HPDs are typically worn. An ear canal mounted subminiature microphone was used to measure physiological noise in occluded and unoccluded test conditions as well as to measure the insertion loss (IL) of insert, supra-aural and circumaural HPDs. Measurements spanned 1/3 octave bands from 125 Hz to 2 kHz. Attenuation was also measured via REAT procedures and finally the magnitude of the occlusion effect was examined via bone conduction audiometry. The ramifications of these data will be discussed in detail.
Comparison of methods used to determine sound attenuating characteristics of hearing protective devices
Authors: Mozo B. T. , Patterson J. H.
Published: October 2005

Attenuation characteristics of circumaural hearing protective devices measured by two threshold shift methods were compared. The methods used were ANSI Z24.22 (1957) and ASA STDI-1975. In addition, the attenuation characteristics measured by the threshold shift methods were compared to an objective method of measurement which is under development. The objective method measured insertion loss using a miniature microphone placed at the ear canal. Each of the listeners used in the threshold shift method were used in determining the insertion loss of the hearing protectors. The objective method of measurement was performed in the hard walled room in which the ASA STDI-1975 measurements were completed. A sound pressure level of 105 dB was used in the determination of the insertion loss.
A field study of the performance of circumaural hearing protectors by dosimetry
Authors: Chung David Y. , Hardie Richard , Gannon R. Patrick
Published: October 2005

The performance of earmuffs in the actual working environment was studied with two dosimeters which had been modified so that the 1/4 in. microphone connecting to the dosimeter could be placed inside an earmuff with negligible noise leakage. It was found that personal hearing protection, at least with earmuffs, is a potentially adequate method of protection against noise-induced hearing loss in most industrial working situations. Most of the problems involving the use of hearing protectors in a hearing conservation program can be overcome by education and proper fitting.
Motor equivalence in respiratory subsystem control for speech
Authors: Hunker Chauncey J.
Published: October 2005

Motor equivalence coordination in the speech motor systeM has been defined as a neural control process superimposed over several potentially independent speech gestures to ensure their complementary contribution to achieve a common goal [O. Hughes and J. H. Abbs, Phonetica 33, 199–221 (1976)]. The respiratory subsystem has not been specifically examined with regards to motor equivalence. Of particular relevance are the coordinated movements of the rib cage (RC) and abdomen (AB) in achieving lung volume changes. RC and AB movements were transduced in naive subjects via a variable inductance transduction system which provided a measure of relative volume contributions. Intraoral air pressure and speech audio signals were recorded simultaneously as indices of alveolar pressure. These data were analyzed to determine the muscle forces required to achieve a targeted alveolar pressure at various lung volumes as well as the relative volume contributions of the RC and AB. The results of these analyses provided substantial evidence for the operation of motor equivalence in the movement coordination of the respiratory subsystem. These findings will be discussed relative to respiratory subsystem biomechanics and their implication for the complex motor system adjustments required during speech. [Supported by NICDH Grant.]
Multi-channel study of tongue EMG during vowel production
Authors: Alfonso Peter J. , Honda Kiyoshi , Baer Thomas , Harris Katherine S.
Published: October 2005

EMG potentials were simultaneously recorded from extrinsic tongue muscles (anterior and posterior genioglossus, hyoglossus, and styloglossus), from accessory tongue muscles (geniohyoid and mylohyoid), and from the cricothyroid and orbicularis oris superior while a single talker produced ten repetitions of 11 vowels in a /əpVp/ environment. Patterns of activity for the anterior and posterior genioglossus and for styloglossus were consistent with earlier reports. To our knowledge, hyoglossus activity has not been previously reported. This muscle demonstrated qualitatively different patterns of activity for front versus back vowels. For front vowels, an initial peak occurred simultaneously with the peak of orbicularis oris activity associated with the initial stop. This was followed by a partial suppression, the timing and extent of which was related to that of genioglossus activation. Hyoglossus activity then recovered to the level of the initial peak. For back vowels, the EMG patterns were characterized by a single peak associated with tongue positioning for the vowel. Interchannel timing relationships and their implications will be discussed. [Research supported by NINCDS.]
Acoustic and physiologic manifestations of psychological stress
Authors: Brenner Malcolm , Shipp Thomas
Published: October 2005

An effort was made to provide more data on the relationship between vocal fundamental frequency behavior, cricothyroid muscle activity level, and eye pupil diameter, a known measure of psychological stress. Seven normal male subjects underwent a task repeating auditory stimuli comprising strings of various numbers of digits. Recordings were obtained of voice output, pupil size (using a pupilometer), and cricothyroid EMG activity. Results showed that pupil size increased with the difficulty level of the task (implying increased psychological stress), fundamental frequency perturbation decreased with the difficulty level, while cricothyroid muscle activity was highly variable with task difficulty level.
Relationships between vocal jitter, age, sex, and smoking
Authors: Ludlow Christy L. , Coulter David C. , Bassich Celia J.
Published: October 2005

Extended vowel productions were recorded in 25 males and 35 females between 18 and 75 years without laryngeal pathology on indirect laryngoscopy. Variation in the length of the phonatory period was measured from zero crossings of a peak detector output with a system resolution of 2.15 μs at 100 Hz. Fourteen subjects were current smokers while the remaining subjects had no smoking history. Two-way analyses of covariance, with fundamental frequency, intensity, and age as covariates tested whether sex and smoking characteristics affected jitter or jitter ratios (the proportion of the period length perturbed). Fundamental frequency was significantly related to both jitter and jitter ratio while age and intensity were not related to either. Smoking was not significant when jitter and jitter ratios were adjusted for fundamental frequency. The male and female jitter and jitter ration values differed significantly when adjusted for fundamental frequency. Jitter values were greater in the male subjects while jitter ratios were greater in the female subjects. The proportion of the phonatory period involved in perturbation was greater in females at all ages.
Acoustical characteristics of intended syllabic stress in excellent esophageal speakers: Linguistic considerations
Authors: Reich Alan , McHenry Monica , Minifie Fred
Published: October 2005

The ability of excellent esophageal speakers to produce intended syllabic stress was investigated, using hi-syllabic word combinations which could be produced either as compound nouns or adjective/noun combinations depending on which syllable received primary stress. Fundamental frequency (f0), sound pressure level (SPL), and duration were evaluated for each stressed/unstressed vowel nucleus. Difference scores were obtained for each acoustic parameter. Comparisons were made among versions in which the first or second syllable was stressed, and across matched syllable pairs, in which the same syllable was stressed in one sentence and not in the other. Difference scores also were obtained for sentence-embedded syllable pairs, and for sentence-final syllable pairs in which the first or the second syllable was stressed. The magnitude of difference scores for f0 and SPL differed significantly for both groups of subjects as a function of the stressed syllable's position within the bi-syllabic stimulus item and within the sentence. No significant differences were found between the esophageal speakers and their matched normals.
A perceptual study of emotion in esophageal speech
Authors: Ellis Charles C. , Schwartz Ralph J.
Published: October 2005

Two male talkers, one using laryngeal voice, the other using esophageal voice, simulated five emotions for each of five sentences. A randomized audiotape presented three tokens of each utterance to 30 naive listeners. The judges identified the intended emotion of each sentence from a closed set of five labels; and in addition they indicated on a five-point scale the self-confidence with which they made their decision. The objective of the study was to determine whether esophageal voice could adequately convey intended emotion. Results show that given a superior esophageal speaker the emotion can be correctly identified at considerably greater than chance. As expected, ability to make accurate judgments is much better for the sentences from the laryngeal talker. Also, confidence ratings are generally much higher when the laryngeal talker is judged, irrespective of whether the judgment is correct or incorrect.
Iceberg patterns of tongue blade movement in medial /t/ and /d/
Authors: Fujimura O. , Laferriere M.
Published: October 2005

Tongue movement data in the production of medial /t/ and /d/ in American English were used to test the “iceberg” model of the temporal organization of articulatory events. This model [Fujimura, Phonetica 38, 66–83 (1981)] describes speech as comprising compressible articulatory gestures, subject to temporal modulation and other gesture modifications, and relatively constant movement patterns, labeled “icebergs.” Each iceberg represents an inherent dynamic movement pattern of a specific articulator that is crucial for the consonantal element in a given demisyllabic unit. Tongue blade movement for alveolar consonants is one example of the iceberg gestures. Tongue body movements, on the other hand, do not show such invariant patterns in alveolar consonants.
Accounting for errors in /s/ identification by the hearing impaired
Authors: Dorman M. F. , Hannley M. , Raphael L. J.
Published: October 2005

Previous research with hearing impaired listeners has reported a number of substitution errors for /s/, e.g., /∫/, /θ/, /f/, and /t/. The purpose of the research reported here was to determine the conditions under which each of the errors occurs and thus to determine the predictability of errors. Hearing-impaired listeners were first presented /s/-noise in isolation to determine threshold and then were presented a test sequence at several levels above /s/ threshold. The confusion errors for /s/ were found to group into reasonable patterns as a function of /s/ threshold and presentation level.
Development of a portable speech processor for an auditory implant subject: Choice of algorithm and role of learning
Authors: Atlas Les E. , Herndon Matthew K. , White Robert L. , Dent Laurel J.
Published: October 2005

Several real-time portable speech processing algorithms for an ditory implant subject [Simmons et al., Acta Otolaryngol. 87, 170–175 (1979)] have been evaluated. The algorithms, designed in consideration of data collected on loudness and pitch for this implant subject, included various combinations of (1) automatic gain control, (2) instantaneous compression, (3) emphasis of high-frequency energy if unvoiced sounds are detected, (4) differentiation following the instantaneous compression, and (5) encoding zero crossings as pulses with amplitude derived from the speech envelope. Taped speech tests were processed by each of the algorithms and presented to the subject via a single modiolus-implanted electrode. Findings include the observations that conversion of zero crossings to pulses led to improved perception of the question/statement distinction and that initial consonant identifications were facilitated by the differentiation following the compression. Having thus determined the best combination of the above-mentioned processing steps, we incorporated this combination into a portable speech processor. The effect of daily use of the portable processor on speech discrimination and identification will be reported. [Work supported by NIH.]
Comparison of four speech-processing algorithms in one cochlear implant subject
Authors: Herndon Matthew K. , Atlas Les E. , Simmons F. Blair , White Robert L.
Published: October 2005

Four different methods of encoding speech for electrical stimulation were compared in a single cochlear implant subject using taped speech material. Thus the effectiveness of the different speech processing strategies could be compared without subject-to-subject differences. A baseline algorithm presented only information about the amplitude envelope of the acoustic speech signal. A second algorithm added fundamental frequency information; improved comprehension of final consonants in monosyllabic words was seen, apparently due to increased ability to distinguish voiced and unvoiced consonants. A third algorithm consisted of simple log compression of the analog signals; comprehension was somewhat better than with the baseline algorithm, but it is unclear which features of speech were responsible for this performance. The fourth algorithm extracted formant frequencies and mapped them onto the lower usable frequency range [Simmons et al., Acta Otolaryngol. 87, 170–175 (1979)]. This algorithm showed the best overall comprehension, especially for initial and final consonants of monosyllabic words. [Work supported by NIH.]
Effects of modifying the time-varying amplitude patterns on perception of speech by hearing-impaired and normal listeners
Authors: Henrickson Larry K.
Published: October 2005

The acoustical patterns of speech may be analyzed into three primary components: a fine-temporal pattern, a gross-temporal pattern, and a time-varying amplitude pattern. Previous research has supported the importance of both fine- and gross-temporal cues for speech perception. Braida et al. [ASHA Monographs 19 (1979)] reviewed amplitude compression, and indicated that the importance of time-varying amplitude patterns was not well understood. The present research utilized altered versions of the California Consonant Test to examine perception by hearing impaired and normals. The time-varying amplitude patterns had been systematically altered to simulate the effects of several types of single- and multichannel compressors. The effects of frequency response, degree of compression, and number of channels have been statistically analyzed using ANOVA. Results for the hearing impaired showed significant improvements in consonant differentiation for severe rather than mild compression ratios, and for compression preceded by high-frequency emphasis. Multichannel designs did not perform better than optimal single-channel designs. The time-varying amplitude pattern seems of little importance for the speech intelligibility.
Vibrotactile perception of phonetic features of speech: A comparison of single-channel and multi-channel aids
Authors: Carney A. E.
Published: October 2005

In an earlier paper [C. A. Beachler and A. E. Carney, J. Acoust. Soc. Am. Suppl. 1 69, S123 (1981)], two vibrotactile devices, one single-channel and one multi-channel, were compared with regard to their ability to transmit suprasegmental aspects of speech. Subjects trained with a single-channel device were significantly better trained than those trained with a multi-channel device in the identification of number of syllables in a word, syllable stress and intonation contour. As a follow up, two additional groups of subjects were trained with the same two different vibrotactile devices to recognize 20 consonants and eight vowels, all in a CV context. Subjects were trained in three conditions—visual only, vibrotactile only, and visual plus vibrotactile. Results indicated that there was essentially no difference between instruments in the percentage of phonemes transmitted correctly. Some slight differences between instruments were observed in the pattern of errors and the particular phonemes which were transmitted best by each instrument. These data are discussed with regard to the selective applicability of these devices to speech training for the hearing impaired.
Effects of small room reverberation on subjective preference
Authors: Allen J. B.
Published: October 2005

In this talk I shall present some results on quantifying the effects of small room reverberation on diotic subjective preference of speech under conditions of no interfering noise. All reverberation was generated digitally using an image model of a rectangular room [Allen and Berkley, J. Acoust. Soc. Am. 65, 943 (1979)]. Our main result is given by the equation P = Pmax − σT60, where P is the subjective preference in some arbitrary units, Pmax is the maximum possible preference, σ is the room spectral variance [Jetzt, J. Acoust. Soc. Am. 65, 1204–1211 (1979)], and T60 is the reverberation time. The spectral variance is a known function of Δ/Δc, where Δ is the source-receiver distance and Δc is the critical distance (Jetzt). In our experiment, σ went from 0.5 to 5.5, and T60 ranged between 75 and 500 ms. The above result will be explained in some detail.
A unified approach to electronic simulation of auditorium acoustics
Authors: Borish Jeffrey
Published: October 2005

A unified approach to simulating auditorium acoustics electronically is described. The technique begins with a modeling phase in which the geometry and acoustical characteristics of an auditorium are programmed into a computer. The computer calculates the directional impulse response characterizing the transmission of sound from the source to the listener. The impulse response is convolved with an audio signal in real time by using a high-speed transversal filter. The result of the convolution can be presented to a listener with a few speakers and still achieve a result which is natural-sounding and realistic.
“Assisted resonance” 1964–1982
Authors: Jones Anthony J. , Barnett Peter W.
Published: October 2005

The acoustical quality of an auditorium is a subjective impression and even now, after years of research, is difficult to relate to any one simple objective measure. Historically, reverberation time has been adopted as the measure of a hall's acoustic and is still arguably one of the most important single parameters. It is a considerable task to design a concert hall to exhibit the chosen reverberation time and London's Royal Festival Hall is one example where, despite careful design, insufficient low-frequency reverberation resulted. Traditional remedies were discounted on economic grounds and the electronic solution known as “assisted resonance” was conceived. This overcame the lack of warmth by employing multiple channel, frequency selective positive feedback via the normal modes of vibration of the auditorium, thereby increasing the reverberation time. Since this prototype installation, the “assisted resonance” concept has been applied to another ten halls of a variety of types, and the approach can now be considered well proven by example. During this time the elements of the system have been developed considerably, including the recent introduction of sophisticated signal processing equipment. This not only enhances the management of the basic process of reverberation magnification but also brings closer the ability to control and optimize other acoustic parameters which contribute to listening enjoyment.
The detection and diagnosis of faults in sophisticated machinery systems
Authors: Stewart R. M.
Published: October 2005

The detection and diagnosis of faults in sophisticated machinery systems presents the monitoring engineer with some of his most challenging problems. One of these is inter-fault definition, first of all between the various components of the machine and then down at component level. For example, the intermediate casing of an aeroengine gas turbine is literally packed solid with shafts, bearings large and small, gears, splines, seals, etc., each with its own particular set of failure modes. In particular, the gears and bearings can fail in a large variety of ways, some more disastrously than others. For reasons connected with the statistical validity of his result, which is essentially to give the machine a “bill of health,” the engineer must seek ways of identifying the faulty state of each component as though it were an isolated unit running on a test stand. The author will describe what progress is being made in the U.K. to achieve this aim, emphasizing primarily the role of vibration analysis, but touching briefly on other important techniques such as debris, performance, temperature and electrostatic signature analysis.
Vibration monitoring for damage in offshore platforms
Authors: Rubin Sheldon
Published: October 2005

A number of vibration-based techniques have been considered for the detection of underwater structural damage in fixed offshore platforms for oil and gas production. Most platforms are primarily composed of welded steel pipes, and are pile anchored to the sea floor in depths of up to 320 m (1050 ft). The standard inspection technique employs divers to visually survey for severance or cracking of members. This is costly, can be hazardous, and is subject to environment constraints. Various vibration instrumentation approaches are described. All employ accelerometers to measure ambient excited or forced vibration. The crucial issues are (1) the ability to reliably detect damage before there is a significant loss of structural integrity, (2) the avoidance of false alarms due to nondamage changes, and (3) the masking of the desired information by machinery and other noise. A highly promising new technique, soon to be field tested, employs accelerometer placements down special chutes to reach various underwater positions for detection of overall shear stiffness of the individual platform bays. Results are described for a USGS/ONR sponsored blind laboratory test program in which several techniques were evaluated.
Recognition of defects from impact generated surface waves on the rim of a railroad wheel
Authors: Fahmy M.
Published: October 2005

Ultrasonic surface waves have been used to detect thermal cracks on the rim of a railroad wheel. In this paper it is pointed out that surface waves may alternatively be generated by an impact on the rim of the wheel. Several circuits of such waves around the rim can be recognized in the first few milliseconds, after impact, of the signal from a microphone in proximity to the rim or an accelerometer in contact with the wheel. Reflections from cracks can also be recognized. Experimental results were obtained using a ball bearing impact and various types of transducers as receivers. Time domain signals resemble seismograms from which the wave velocity can be calculated. This velocity is in approximate agreement with the theory of surface wave propagation on the edge of a plate. Some developments in this theory will also be presented. The significance of the work lies in the potential for adding surface wave recognition algorithms to an Acoustic Signature Inspection System. [Work supported by Association of American Railroads.]
Sound levels and spectra experienced by drum players
Authors: Hall Donald E. , Dearing James
Published: October 2005

We measured sound levels produced by a typical percussion set (five drums, four cymbals) in a small reverberant room. This acoustic environment would resemble a practicing situation more often than it would performance. Sound levels in dBA for individual components ranged from the low 90s for the ride cymbal up to nearly 110 for the snare drum. For the entire set, readings around 110 have the snare as the strongest contributor, and tend to be 3 or 4 dB higher at the player's ears than in the reverberant field. We will also present time-averaged spectra showing how this sound is distributed over the audible frequency range, and mention some implications for protection of the drummer's hearing.
Equivalent modes and generalized rays for sound fields near critically reflected and bottom-grazing rays
Authors: Niver E. , Kamel A. , Felsen L. B.
Published: October 2005

In a vertically stratified ocean, ray acoustics fails in the vicinity of ray paths that return from the penetrable bottom after critical or grazing incidence. This deficiency can be repaired on deleting the angular spectrum intervals surrounding these rays and filling these intervals with modes. Alternatively, the intervals may be filled with a spectrum of plane waves that generates generalized rays. Both of these options are pursued by applying the previously developed hybrid ray-mode theory [L. B. Felsen, J. Acoust. Soc. Am. 69, 352 (1981)] to a model comprising an exponentially varying refractive index in the water and a homogeneous bottom. The utility of the various formulations is assessed by numerical examples. [Work supported by ONR Ocean Acoustics Branch.]
An inverse method for determining the compressional wave attenuation of the ocean bottom
Authors: Frisk George V. , Rajan Subramaniam D.
Published: October 2005

A perturbation method has been developed for determining small variations in sound speed from normal-incidence measurements of the plane-wave reflection coefficient as a function of frequency [J. K. Cohen and N. Bleistein, SIAM J. Appl. Math. 32, 784–799 (1977)]. We propose an analogous method for determining the compressional wave attenuation profile of an ocean bottom with known velocity profile from measurements of the reflection coefficient as a function of incident angle at a fixed frequency. The technique utilizes the fact that the attenuation can be treated as a small imaginary perturbation in the total wavenumber. Results are presented for a simple example. [Work supported by NORDA.]
Measurement of compressional-wave sound velocity and attenuation in water-saturated sands
Authors: Layton Michael R.
Published: October 2005

The need for additional measurements of compressional-wave sound velocity and attenuation in water-saturated sands has been expressed in the literature [E. L. Hamilton, J. Acoust. Soc. Am. 68, 1327 (1980)]. In the course of selecting a bottom material for a scale-model shallow-water duct, measurements were made of sound velocity and attenuation in water-saturated sands ranging in diameter from 180 to 600 μm, at a frequency of 100 kHz. Results are presented and compared with previously published experimental data as well as with theoretical predictions [E. L. Hamilton, Geophysics 37, 620–646 (1972)]. [Work supported by ONR and DARPA.]
Scattering from a fluid sphere revisited via a transition matrix approach
Authors: Jacobi Nathan
Published: October 2005

The Anderson model for scattering of a sound wave from a fluid sphere, used widely in biological applications, is rederived by using a transition matrix approach. The variant used is MOOT (method of optimized truncation), in which the amount of dissatisfaction of boundary conditions due to a finite basis set is minimized [W. M. Visscher, J. Appl. Phys. 51, 825 (1980)]. The method is applicable to nonspherical scatterers and to layered structures of scatterers, and the computation is greatly simplified for axially symmetric scatterers. The reproductibility of the Anderson model and the properties of the transition matrix render this method very promising for acoustic scattering from nonspherical marine organisms with thin elastic shells and swimbladders. [Work supported by NASA.]
Creeping waves in acoustic scattering and the ringing of resonances
Authors: Gaunaurd G. C. , Überall H.
Published: October 2005

The ringing of resonances in acoustic scattering was experimentally observed by Faran in selected cases. In our resonance scattering theory [G. C. Gaunaurd and H. Überall, J. Appl. Phys. 50, 4642 (1979)], this is obtained simply as the Fourier transform of a pole term in the scattering amplitudes, with poles located in the complex-frequency plane corresponding to the eigenfrequencies of the target system, as described in the radar case by the “Singularity Expansion Method” (SEM). These SEM poles are found to occur in distinct “layers,” and each such layer corresponds to the manifestations of a given circumferential wave, as obtained by us previously from a Watson transformation. We show for impenetrable spheres and cylinders that summing the pole terms over a corresponding layer leads to creeping pulses with arrival times governed by the group velocity, including multiple circumnavigations which constitute the ringing. Phase and group velocity dispersion curves are obtained directly from the SEM poles. [H. Überall is also at Catholic University, Washington, DC, supported by the Office of Naval Research.]
An investigation of approaches to language identification
Authors: Shoup June E. , Cimarusti Deidre
Published: October 2005

This paper will present a discussion of approaches used in the area of language identification. Past efforts have attempted to develop an automatic language identification system based upon the employment of linguistic units, i.e., phones or phonemes, with moderate results. An approach currently under investigation at SCRL is the application of pattern analysis techniques to features extracted from the speech signal, without regard to specific linguistic units. Upon completion of the first phase of the research, it is likely that a very unusual set of features will prove to be the most useful one in identifying languages. It will be of interest to examine the relationships that may exist between those features which are determined to be the greatest contributors to the classification process and basic linguistic units. [Work supported by RADC.]
Interaction of simultaneous and forward masking
Authors: Jesteadt W. , Wilke S.
Published: October 2005

When forward-masking tuning curves are obtained in the presence of a continuous background noise [Green et al., J. Acoust. Soc. Am. 69, 1758–1762 (1981)], the sinusoidal forward masker and the signal are both partially masked by the noise. We have explored the interaction of forward and simultaneous maskers using 1000-Hz sinusoids as forward maskers and as probes, with broadband noise (low-pass filtered at 5000 Hz) as a simultaneous masker. Masked threshold for the probe was determined for various combinations of simultaneous noise maskers (N0 = 0, 10, 20, 30 dB SPL) and sinusoidal forward maskers (30–80 dB SPL). When the noise was present during both the forward masker and the probe, masked threshold increased as a function of both the forward and the simultaneous masker levels over a 30-dB range. The data were very similar when the noise was present only during the probe. The noise had little effect when it was present only during the forward masker. Even when sinusoidal forward maskers are at high levels, background noise plays a role in determining the probe thresholds. The nonadditive effect of the noise cannot be attributed to restriction of off-frequency listening. [Work supported by NIH and NSF.]
Relationship between excitation patterns, loudness, and the auditory filter
Authors: O'Loughlin Bernard J. , Moore Brian C. J.
Published: October 2005

In Zwicker and Scharf's model of loudness summation [Psych. Rev. 72, 3–26 (1965)], excitation patterns and critical bands are treated as separate elements. A more common recent view is that excitation patterns and critical bands (as “auditory filters”) are alternative ways of conceiving the filtering occurring in the peripheral auditory system. If the form typically associated with the auditory filter [e.g., R. D. Patterson. J. Acoust. Soc. Am. 59. 640–654 (1976)] were found to be predictable from a simple excitation-pattern model, the latter position would receive support. Two loudness-matching experiments are reported. The first was similar in design to those often used for determining the form of the filter. A 10-ms, 2-kHz sinusoid (of 45, 60, or 75 dB SPL) was simultaneously presented with a 500-ms broadband noise which was 3 dB below masked-threshold level. A notch, centered on 2 kHz, was introduced into the noise, and the loudness of the sinusoid matched by another 10-ms, 2-kHz sinusoid (using a tracking method) as the notch width varied. It was predicted, using schematic excitation patterns, that the form of the growth of loudness with increasing notch width would be that characteristically associated with the auditory filter (although inverted). In the second experiment, the loudness of three sinusoids of the same levels was measured as the level of the broadband noise (with no notch) was decreased from the masked-threshold level. Data were also collected in both paradigms for a forward-masked sinusoid of 45 dB SPL. The results of both experiments were well predicted by the excitation-pattern model, especially when lateral suppression was taken into account, and suggest that (a) the form of the auditory filter is dependent on the interaction of excitation patterns, and (b) loudness may be simply related to the area under the excitation pattern in dB/log-frequency coordinates.
Auditory adaptation: Some agreement between loudness effects
Authors: Weiler Ernest M. , Sandman David , Goldman Thomas , Davis James M.
Published: October 2005

We have not yet found significant intercorrelations between binaural balances, monaural balances, monaural reaction time, and magnitude estimates of loudness adaptation. Despite this, the intracorrelations for each method have been significant and moderately high. Furthermore, for group averages, the characteristics of adaptation measured by the different techniques share similarities such as the progressive decline of loudness over time, even though individuals are not consistent across methods. Recently, using the modified method of magnitude estimates with short intensity increments, we found effects on loudness estimates consistent with the models proposed for binaural balances by Weiler, Loeb, and Alluisi (1972) and Hood and Weiler (1977). Noting that magnitude judgments of loudness should occur at the highest level of the auditory system, perhaps simple magnitude estimates as evolved by Scharf's group include more of the complexities of auditory function than reaction time or balance measures.