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Journal of the Acoustical Society of America

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Sep 1992

Volume 92, Issue 3, pp. 1203-1802

Page 1 of 6 Pages Next Page | Jump to Page

Method for transducer transient suppression. I: Theory

Jean C. Piquette

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1203-1213 (1992); (11 pages) | Cited 3 times

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The problem of driving a transducer in such a way as to produce a tone burst of steady‐state sound radiation in the surrounding fluid medium is considered. The goal is to determine the driving voltage waveform to apply to a transducer to produce an acoustic pressure waveform in the fluid that is a segment of a steady‐state sine wave, beginning and ending at zero crossings of the sine, i.e., the usual turnon and turnoff transients are suppressed. The theoretical driving voltage waveform for a spherical transducer is shown to consist of a sum of a pedestal voltage, a ramp voltage, and a sinusoidal voltage that is phase shifted with respect to the sinusoid appearing in the fluid. Both theoretical and numerical calculations are given here. The following paper presents results of experimental measurements. The measurements were carried out on several spherical transducers (one of which was selected for presentation) and on an array of piezoelectric tubes. These experiments confirm the validity of the theory.
Show PACS
43.58.Vb Calibration of acoustical devices and systems
43.58.Wc Electrical and mechanical oscillators
43.20.Px Transient radiation and scattering
43.20.Rz Steady-state radiation from sources, impedance, radiation patterns, boundary element methods

Method for transducer transient suppression. II: Experiment

Jean C. Piquette

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1214-1221 (1992); (8 pages) | Cited 2 times

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Results of experiments that were conducted to investigate the validity of the theory presented in the preceding paper are given. The method was applied to a number of spherical piezoelectric sources, one of which was selected for presentation here, and to a source consisting of an array of piezoelectric tubes. A high degree of transient suppression was realized.
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43.58.Vb Calibration of acoustical devices and systems
43.20.Px Transient radiation and scattering
43.20.Rz Steady-state radiation from sources, impedance, radiation patterns, boundary element methods

Improvement of the acoustic and hygroscopic properties of wood by a chemical treatment and application to the violin parts

H. Yano and K. Minato

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1222-1227 (1992); (6 pages) | Cited 1 time

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A chemical treatment was applied to a wood specimen and some parts of a violin. Formaldehyde cross linkings formed between hydroxyl groups of wood constituents decreased the internal friction (Q−1) of a sitka spruce specimen as much as 40% and 45% in the fiber and cross fiber directions, respectively, in the frequency range of 150–500 Hz. The degree of reduction in Q−1 was smaller at high frequencies and larger at low frequencies. The specific dynamic Young’s modulus (E/γ) at 20 °C, 65% RH increased 10% in the cross fiber direction, though it hardly increased in the fiber direction within the frequency range examined. The changes of specific gravity in the air‐dried state were negligibly small. The variations of acoustic properties and dimensions of the wood against humidity change were reduced markedly because of the decreases of hygroscopicity. By a sensory evaluation test, a violin having a treated bridge was judged to be more bright, sonorous, and less poor, dull, and comprehensively better than the same violin having an untreated bridge. The effect of the treatment was more pronounced when the top plate, bass bar, and sound post were treated in addition to the bridge. The variation in sound spectra and the decrease in tuned frequency of violin against humidity change diminished as a result of the treatment.
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43.75.De Bowed stringed instruments

The influence of talker differences on vowel identification by normal‐hearing and hearing‐impaired listeners

Anna K. Nábělek, Zbigniew Czyzewski, and Lata A. Krishnan

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1228-1246 (1992); (19 pages) | Cited 9 times

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Vowel identification was tested in quiet, noise, and reverberation with 20 normal‐hearing subjects and 20 hearing‐impaired subjects. Stimuli were 15 English vowels spoken in a /b–t/ context by six male talkers. Each talker produced five tokens of each vowel. In quiet, all stimuli were identified by two judges as the intended targets. The stimuli were degraded by reverberation or speech‐spectrum noise. Vowel identification scores depended upon talker, listening condition, and subject type. The relationship between identification errors and spectral details of the vowels is discussed.
Show PACS
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.71.Ky Speech perception by the hearing impaired

Temporal cues for consonant recognition: Training, talker generalization, and use in evaluation of cochlear implants

Dianne J. Van Tasell, Donna G. Greenfield, Joelle J. Logemann, and David A. Nelson

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1247-1257 (1992); (11 pages) | Cited 19 times

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Limited consonant phonemic information can be conveyed by the temporal characteristics of speech. In the two experiments reported here, the effects of practice and of multiple talkers on identification of temporal consonant information were evaluated. Naturally produced /aCa/ disyllables were used to create ‘‘temporal‐only’’ stimuli having instantaneous amplitudes identical to the natural speech stimuli, but flat spectra. Practice improved normal‐hearing subjects’ identification of temporal‐only stimuli from a single talker over that reported earlier for a different group of unpracticed subjects [J. Acoust. Soc. Am. 82, 1152–1161 (1987)]. When the number of talkers was increased to six, however, performance was poorer than that observed for one talker, demonstrating that subjects had been able to learn the individual stimulus items derived from the speech of the single talker. Even after practice, subjects varied greatly in their abilities to extract temporal information related to consonant voicing and manner. Identification of consonant place was uniformly poor in the multiple‐talker situation, indicating that for these stimuli consonant place is cued via spectral information. Comparison of consonant identification by users of multi‐channel cochear implants showed that the implant users’ identification of temporal consonant information was largely within the range predicted from the normal data. In the instances where the implant users were performing especially well, they were identifying consonant place information at levels well beyond those predicted by the normal‐subject data. Comparison of implant‐user performance with the temporal‐only data reported here can help determine whether the speech information available to the implant user consists of entirely temporal cues, or is augmented by spectral cues.
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43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.66.Ts Auditory prostheses, hearing aids
43.71.Ky Speech perception by the hearing impaired

The time course and magnitude of perceptual acclimatization to frequency responses: Evidence from monaural fitting of hearing aids

Stuart Gatehouse

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1258-1268 (1992); (11 pages) | Cited 7 times

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At high presentation levels, normally aided ears yield better performance for speech identification than normally unaided ears, while at low presentation levels the converse is true [S. Gatehouse, J. Acoust. Soc. Am. 86, 2103–2106 (1989)]. To explain this process further, the speech identification abilities of four subjects with bilateral symmetric sensorineural hearing impairment were investigated following provision of a single hearing aid. Results showed significant increases in the benefit from amplifying speech in the aided ear, but not in the control ear. In addition, a headphone simulation of the unaided condition for the fitted ear shows a decrease in speech identification. The benefits from providing a particular frequency spectrum do not emerge immediately, but over a time course of at least 6–12 weeks. The findings support the existence of perceptual acclimatization effects, and call into question short‐term methods of hearing aid evaluation and selection by comparative speech identification tests.
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43.71.Ky Speech perception by the hearing impaired
43.66.Yw Instruments and methods related to hearing and its measurement
43.66.Ts Auditory prostheses, hearing aids

Vowel perception strategies of normal‐hearing subjects and patients using Nucleus multichannel and 3M/House cochlear implants

Vivien C. Tartter, Sharon A. Hellman, and Patricia M. Chute

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1269-1283 (1992); (15 pages)

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Vowel perception strategies were assessed for two ‘‘average’’ and one ‘‘star’’ single‐channel 3M/House and three ‘‘average’’ and one ‘‘star’’ Nucleus 22‐channel cochlear implant patients and six normal‐hearing control subjects. All subjects were tested by computer with real and synthetic speech versions of [i, i, u, u, q, 1, c, v, a, ϵ], presented randomly. Duration, fundamental frequency, and first, second, and third formant frequency cues to the vowels were systematically manipulated. Results showed high accuracy for the normal‐hearing subjects in all conditions but that of the first formant alone. ‘‘Average’’ single‐channel patients classified only real speech [hVd] syllables differently from synthetic steady state syllables. The ‘‘star’’ single‐channel patient identified the vowels at much better than chance levels, with a results pattern suggesting effective use of first formant and duration information. Both ‘‘star’’ and ‘‘average’’ Nucleus users showed similar response patterns, performing better than chance in most conditions, and identifying the vowels using duration and some frequency information from all three formants.
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43.71.Ky Speech perception by the hearing impaired
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.66.Ts Auditory prostheses, hearing aids

Effects of short‐term auditory deprivation on speech production in adult cochlear implant users

Mario A. Svirsky, Harlan Lane, Joseph S. Perkell, and Jane Wozniak

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1284-1300 (1992); (17 pages) | Cited 12 times

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Speech production parameters of three postlingually deafened adults who use cochlear implants were measured: after 24 h of auditory deprivation (which was achieved by turning the subject’s speech processor off); after turning the speech processor back on; and after turning the speech processor off again. The measured parameters included vowel acoustics [F1, F2, F0, sound‐pressure level (SPL), duration and H1–H2, the amplitude difference between the first two spectral harmonics, a correlate of breathiness] while reading word lists, and average airflow during the reading of passages. Changes in speech processor state (on‐to‐off or vice versa) were accompanied by numerous changes in speech production parameters. Many changes were in the direction of normalcy, and most were consistent with long‐term speech production changes in the same subjects following activation of the processors of their cochlear implants [Perkell et al., J. Acoust. Soc. Am. 91, 2961–2978 (1992)]. Changes in mean airflow were always accompanied by H1–H2 (breathiness) changes in the same direction, probably due to underlying changes in laryngeal posture. Some parameters (different combinations of SPL, F0, H1–H2 and formants for different subjects) showed very rapid changes when turning the speech processor on or off. Parameter changes were faster and more pronounced, however, when the speech processor was turned on than when it was turned off. The picture that emerges from the present study is consistent with a dual role for auditory feedback in speech production: long‐term calibration of articulatory parameters as well as feedback mechanisms with relatively short time constants.
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43.70.Aj Anatomy and physiology of the vocal tract, speech aerodynamics, auditory kinetics
43.70.Dn Disordered speech
43.70.Bk Models and theories of speech production
43.66.Ts Auditory prostheses, hearing aids

Acoustic and perceptual effects of changes in vocal tract constrictions for vowels

T. Gay, L‐J. Boé, and P. Perrier

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1301-1309 (1992); (9 pages) | Cited 1 time

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The purpose of this study was to use vocal tract simulation and synthesis as means to determine the acoustic and perceptual effects of changing both the cross‐sectional area and location of vocal tract constrictions for six different vowels: Area functions at and near vocal tract constrictions are considered critical to the acoustic output and are also the central point of hypotheses concerning speech targets. Area functions for the six vowels, [i], [a], [u], [q], [1], and [c] were perturbed by changing the cross‐sectional area of the constriction (Ac) and the location of the constriction (Xc). Perturbations for Ac were performed for different values of Xc, producing several series of acoustic continua for the different vowels. Acoustic simulations for the different area functions were made using a frequency domain model of the vocal tract. Each simulated vowel was then synthesized as a 1‐s duration steady‐state segment. The phoneme boundaries of the perturbed synthesized vowels were determined by formal perception tests. Results of the perturbation analyses showed that formants for each of the vowels were more sensitive to changes in constriction cross‐sectional area than changes in constriction location. Vowel perception, however, was highly resistant to both types of changes. Results are discussed in terms of articulatory precision and constriction‐related speech production strategies.
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43.70.Bk Models and theories of speech production
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech

Speech changes following reimplantation from a single‐channel to a multichannel cochlear implant

Alexandra Economou, Vivien C. Tartter, Patricia M. Chute, and Sharon A. Hellman

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1310-1323 (1992); (14 pages) | Cited 7 times

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The speech of a postlingually deafened preadolescent was recorded and analyzed while a single‐electrode cochlear implant (3M/House) was in operation, on two occasions after it failed (1 day and 18 days) and on three occasions after stimulation of a multichannel cochlear implant (Nucleus 22) (1 day, 6 months, and 1 year). Listeners judged 3M/House tokens to be the most normal until the subject had one year’s experience with the Nucleus device. Spectrograms showed less aspiration, better formant definition and longer final frication and closure duration post‐Nucleus stimulation (6 MO. NUCLEUS and 1 YEAR NUCLEUS) relative to the 3M/House and no auditory feedback conditions. Acoustic measurements after loss of auditory feedback (1 DAY FAIL and 18 DAYS FAIL) indicated a constriction of vowel space. Appropriately higher fundamental frequency for stressed than unstressed syllables, an expansion of vowel space and improvement in some aspects of production of voicing, manner and place of articulation were noted one year post‐Nucleus stimulation. Loss of auditory feedback results are related to the literature on the effects of postlingual deafness on speech. Nucleus and 3M/House effects on speech are discussed in terms of speech production studies of single‐electrode and multichannel patients.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation
43.66.Ts Auditory prostheses, hearing aids

Target detection, shape discrimination, and signal characteristics of an echolocating false killer whale (Pseudorca crassidens)

Randall L. Brill, Jeffrey L. Pawloski, David A. Helweg, Whitlow W. Au, and Patrick W. B. Moore

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1324-1330 (1992); (7 pages) | Cited 3 times

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This study demonstrated the ability of a false killer whale (Pseudorca crassidens) to discriminate between two targets and investigated the parameters of the whale’s emitted signals for changes related to test conditions. Target detection performance comparable to the bottlenose dolphin’s (Tursiops truncatus) has previously been reported for echolocating false killer whales. No other echolocation capabilities have been reported. A false killer whale, naive to conditioned echolocation tasks, was initially trained to detect a cylinder in a ‘‘go/no‐go’’ procedure over ranges of 3 to 8 m. The transition from a detection task to a discrimination task was readily achieved by introducing a spherical comparison target. Finally, the cylinder was successfully compared to spheres of two different sizes and target strengths. Multivariate analyses were used to evaluate the parameters of emitted signals. Duncan’s multiple range tests showed significant decreases (df=185, p<0.05) in both source level and bandwidth in the transition from detection to discrimination. Analysis of variance revealed a significant decrease in the number of clicks over test conditions [F(5,26)=5.23, p<0.0001]. These data suggest that the whale relied on cues relevant to target shape as well as target strength, that changes in source level and bandwidth were task‐related, that the decrease in clicks was associated with learning experience, and that Pseudorca’s ability to discriminate shapes using echolocation may be comparable to that of Tursiops truncatus.
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43.66.Gf Detection and discrimination of sound by animals
43.80.Ka Sound production by animals: mechanisms, characteristics, populations, biosonar
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Avoiding conflicts between the natural behavior of the animal and the demands of discrimination experiments

J. M. Harrison

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1331-1345 (1992); (15 pages)

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Auditory discrimination experiments are traditionally designed without regard for ethological or ecological concerns, yet land dwelling mammals may have specialized behavior with respect to sound sources. Auditory discriminations occur under field conditions, and there is some fit or matching of the animal’s behavior to the acoustic environment. Understanding this fit requires a knowledge of specializations. Understanding the specializations may also guide the design of discrimination experiments. This paper reviews a number of auditory discrimination experiments that were designed to reveal some of the specialized behaviors. These experiments showed the following: (i) The position of a sound source is the dominant sensory dimension, over riding the quality of the sound; (ii) the effect of reinforcing a response in the presence of a sound is to strengthen the response of approaching the source. This effect is ubiquitous in discrimination tasks; (iii) sounds that are novel at the start of discrimination training more rapidly gain control of responding than sounds to which the animal has been pre‐exposed; (iv) novel low‐intensity sounds elicit approach and exploration of the source. These behaviors rapidly adapt. These four behavioral attributes are considered in terms of their impact upon behavior in the field, and of the requirements they impose on the design of experimental discriminations.
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43.66.Gf Detection and discrimination of sound by animals
43.66.Ba Models and theories of auditory processes
43.66.Qp Localization of sound sources
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Spectral and temporal weights in spectral‐shape discrimination

Huanping Dai and Bruce G. Berg

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1346-1355 (1992); (10 pages) | Cited 9 times

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The COSS analysis [B. G. Berg, J. Acoust. Soc. Am. 86, 1743–1746 (1989)] was used to estimate spectral and temporal weights of a three‐component, amplitude‐modulated stimulus in a spectral‐shape discrimination task. In all experiments, the task of the observer was to detect an increment in the level of the center component. A spectral–temporal weight quantifies the relative influence of a spectral component on the decisions of an observer during a specified segment of the total stimulus duration. In the first two experiments, the signal was added to all three temporal segments of the center component. The ideal weights for each component should be the same across temporal segments. Spectral–temporal weights were obtained for four conditions with different stimulus durations. In general, the estimated weights for each component were not equal at different temporal segments. In the third experiment, the signal was added to only one of three segments of the center component. Ideally, weight patterns should have changed when the temporal position of the signal segment was altered. Two stimulus durations, 300 and 15 ms, were used. For the 300‐ms condition, the signal was added to only the end segment, and for all three observers the weight patterns are different from that obtained in experiment 1 with the signal added to all segments. For the 15‐ms conditions, altering the signal position changed the estimated weights for only one observer.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Yw Instruments and methods related to hearing and its measurement
43.66.Fe Discrimination: intensity and frequency

Middle‐ear phenomenology: The view from the three windows

Christopher A. Shera and George Zweig

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1356-1370 (1992); (15 pages) | Cited 13 times

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To provide a common ground for the comparison between theory and experiment, this paper presents a framework for the phenomenological description of middle‐ear mechanics. The framework defines those measurements sufficient to characterize the transduction properties of the middle ear and its components. Phenomenological equations are represented in the form of an equivalent electrical circuit that can be used to deduce testable relations among measurable quantities. Two applications are then discussed. First, the classical concept of the middle‐ear transformer ratio is generalized to include any effects of eardrum flexion or nonrotational ossicular motion. Middle‐ear models predict that the resulting transformer ratios vary considerably with frequency. Second, the conditions under which the topology of existing circuit analogs satisfactorily approximates middle‐ear mechanics are given. Most middle‐ear models cannot be used to correctly predict the absolute pressures in the cochlea.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Bt Models and theories of the auditory system

Analyzing reverse middle‐ear transmission: Noninvasive Gedankenexperiments

Christopher A. Shera and George Zweig

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1371-1381 (1992); (11 pages) | Cited 15 times

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The phenomenological framework outlined in the companion paper [Shera and Zweig, J. Acoust. Soc. Am. 92, ■■–■■ (1992)] characterizes both forward and reverse transmission through the middle ear. This paper illustrates its use in the analysis of noninvasive measurements of middle‐ear and cochlear mechanics. A cochlear scattering framework is developed for the analysis of combination‐tone and other experiments in which acoustic distortion products are used to drive the middle ear ‘‘in reverse.’’ The framework is illustrated with a simple psychophysical Gedankenexperiment analogous to the neurophysiological experiments of Fahey and Allen [J. Acoust. Soc. Am. 77, 599–612 (1985)].
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Bt Models and theories of the auditory system
43.64.Jb Otoacoustic emissions

An empirical bound on the compressibility of the cochlea

Christopher A. Shera and George Zweig

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1382-1388 (1992); (7 pages) | Cited 4 times

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Effects of a possible inner‐ear compressibility on middle‐ear transfer functions are explored and a small upper bound on the magnitude of that compressibility established. Consequently, the traditional two‐port representation of middle‐ear mechanics remains valid to within a few percent. If the compressibility of the cochlea is small but finite, a simple phenomenological model of that compressibility correctly predicts hearing thresholds in the ‘‘middleless’’ ear at low frequencies. Experiments to establish the value of cochlear compressibility and to explore further its possible contributions to residual hearing in patients with missing or disarticulated middle‐ear ossicles are suggested.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system

Temporal structure model of binaural masking level difference

Yehuda Albeck, Isaiah Nebenzahl, and Aaron Lewis

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1389-1393 (1992); (5 pages)

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It is shown that a simple cross‐correlation model is not adequate to explain both binaural masking level difference (MLD) and spatial selective attention. The reason is that for a low‐intensity signal in NoSπ condition the maximal activity in the binaural analyzer as a function of interaural delay in single spectral channel is independent of signal intensity. On the other hand, if detection ability is associated with the isolation of tonically firing units, MLD is simply explained as the increase in firing synchronization as a function of the signal’s interaural phase difference (IPD). Quantitatively results are presented based on numerical solutions of the model.
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43.64.Qh Electrophysiology of the auditory central nervous system
43.64.Bt Models and theories of the auditory system
43.66.Pn Binaural hearing
43.66.Gf Detection and discrimination of sound by animals

Musical fundamental frequency tracking using a pattern recognition method

Judith C. Brown

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1394-1402 (1992); (9 pages) | Cited 5 times

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In a recent article [J. C. Brown, ‘‘Calculation of a Constant Q Spectral Transform,’’ J. Acoust. Soc. Am. 89, 425–434 (1991)], the calculation of a constant Q spectral transform that gives a constant pattern in the log frequency domain for sounds with harmonic frequency components has been described. This property has been utilized in calculating the cross‐correlation function of spectra of sounds produced by musical instruments with the ideal pattern, which consists of one’s at the positions of harmonic frequency components. Therefore, the position of the best approximation to the ‘‘ideal’’ pattern for the spectra produced by these instruments has been determined, and in so doing the fundamental frequency for that sound has been obtained. Results are presented for scales produced by the piano, flute, and violin as well as for arpeggios played by a wide variety of instruments.
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43.60.Gk Space-time signal processing, other than matched field processing
43.75.Yy Instrumentation and measurement methods for musical acoustics

A ground‐based narrow‐band passive acoustic technique for estimating the altitude and speed of a propeller‐driven aircraft

Brian G. Ferguson

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1403-1407 (1992); (5 pages) | Cited 4 times

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The Doppler effect is the change in the observed frequency of an acoustic or electromagnetic wave due to the relative motion of source and observer. The acoustical Doppler effect is utilized here to estimate the altitude and speed of a propeller‐driven aircraft. The acoustic energy emitted by the aircraft is received by a microphone located just above ground level. The acoustic spectrum of the aircraft is dominated by a spectral line corresponding to the propeller blade rate which is equal to the product of the propeller shaft rotation rate and the number of blades on the propeller. For the present experiment, the aircraft is flown at a constant altitude and speed over the microphone. A frequency estimation technique is applied to the acoustic data from the microphone so that the Doppler shift in the propeller blade rate can be observed at short time intervals during the aircraft’s transit overhead. Using the altitude and speed of the acoustic source as the variable parameters, a simple model is fitted to the observed variation of the blade rate with time; estimates of the aircraft’s altitude and speed correspond to a least‐mean‐squares curve fit of the model to the observations. These estimates are then compared with the actual altitude and speed recorded by the aircraft itself as it flew over the microphone at each of the nominated altitudes: 250, 500, 750, 1000, 1250, and 1500 ft, and at each of the nominated speeds: 150, 200, and 250 kn.
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43.60.Gk Space-time signal processing, other than matched field processing
43.28.Bj Mechanisms affecting sound propagation in air, sound speed in the air
43.50.Lj Transportation noise sources: air, road, rail, and marine vehicles

Matched‐field minimum variance beamforming in a random ocean channel

Jeffrey L. Krolik

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1408-1419 (1992); (12 pages) | Cited 25 times

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Matched‐field source localization methods that employ deterministic full‐wave acoustic propagation models can be seriously degraded due to the presence of random inhomogeneities in the ocean channel. In this paper, a minimum variance (MV) matched‐field beamformer is presented that achieves greater robustness to random inhomogeneities in the sound‐speed profile between the source and receiver. The proposed modification of the MV beamformer consists of employing multiple linear constraints derived from predicted pressure fields obtained using a set of perturbed sound‐speed profiles. In order to investigate the nature of wave‐front variations due to random sound‐speed perturbations, a normal mode model based on adiabatic and first‐order perturbation approximations is examined. The signal wave‐front spatial correlation implied by this model suggests that the coherence among modes can remain high even in a fluctuating ocean environment. This in turn implies that the dimension of the signal perturbation constraint space for the MV beamformer can be small for typical sound‐speed variations at moderate source ranges. Given the signal constraint space, design of the MV beamformer with sound‐speed perturbation constraints is achieved by selecting its quiescent response to maximize the average signal‐to‐noise ratio gain against spatially uncorrelated noise. This leads to a computationally efficient realization of the beamformer that avoids the need to repeatedly compute perturbed pressure fields. Simulation experiments using a realistic deep‐water Pacific Ocean environment are presented, which suggest that robust unambiguous low‐frequency source location estimates can be achieved in the presence of mesoscale inhomogeneities given only knowledge of the second‐order statistics of the random range‐dependent sound‐speed profile plus a single environmental measurement at the receiving array.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Wi Passive sonar systems and algorithms, matched field processing in underwater acoustics

Signal detectors for random ocean media

Isabel M. G. Lourtie and G. Clifford Carter

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1420-1427 (1992); (8 pages) | Cited 3 times

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This paper reports on acoustic signal detection in a stationary random multipath environment. The multipath transmission channel is modeled considering that both multipath time delay and attenuation coefficients characterizing the emitter/receiver transfer function are random variables with an a priori given distribution. Under the above condition, and assuming the signal‐to‐noise ratio (SNR) is either low or high, the likelihood ratio (LR) detector structure is developed, analyzed, and interpreted. A Monte Carlo study is also carried out. For low SNR conditions, the statistical behavior of the achieved processor is evaluated and compared to that of two classical detectors: (i) the standard detector derived based on a presumed known multipath channel structure, and (ii) the ad hoc detector developed for inaccurate multipath time delay modeling assumptions.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Cg Statistical properties of signals and noise
43.30.Ft Volume scattering

Principal component calibration models in the acoustic evaluation of zooplankton size spectra

Carlos M. Martínez and Pascal M. David

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1428-1439 (1992); (12 pages)

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The theoretical basis of application of PCR (principal components regression) methods to the estimation of zooplankton size spectra by multifrequency acoustical methods is presented. Preliminary performance studies on simulated data are preformed to test the capacities of this technique as a function of the acoustic signal‐to‐noise ratio (SNR) and the type of acoustic scattering model. The results are compared with those obtained with a non‐negative least squares (NNLS) algorithm, and finally, an application to a real data matrix is shown. The main advantage over the NNLS method is that it avoids the knowledge of the scattering responses for each class of particles. The results obtained on simulated data indicate that PCR calibration models are good alternatives for the estimation of size spectra, in the range tested. For all the conditions analyzed, the error estimators and correlation coefficients show a better fit than NNLS models. The real data matrix tested is particularly complex, avoiding the use of the NNLS method. The PCR method, however, gives acceptable results for most classes, despite the undetermination of the acoustic data matrix (five frequencies for 26 groups). The addition of nonacoustical variables to the acoustic data set, correlated with the spatial distribution of organisms, increases the precision of the estimates. Further analysis will be made to test the capacities of this technique in the presence of a more complex noise structure and nonlinear phenomena.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Qv Signal processing instrumentation, integrated systems, smart transducers, devices and architectures, displays and interfaces for acoustic systems
43.30.Xm Underwater measurement and calibration instrumentation and procedures
43.30.Ft Volume scattering

Range estimation using backpropagation

Tat‐Jin Teo and John M. Reid

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1440-1442 (1992); (3 pages)

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The method of backpropagation is used to estimate the range between two planar surfaces. This method is shown to be more tolerant in a noisy environment than another method that uses the spatial derivative of the measured wave field to estimate the range.
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43.60.Gk Space-time signal processing, other than matched field processing

Lateral low‐frequency components of reflected sound from a canopy complex comprising triangular plates in concert halls

Tatsumi Nakajima, Yoichi Ando, and Keisuke Fujita

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1443-1451 (1992); (9 pages)

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Applying the diffraction integral of Rubinowicz, the transfer functions (TFs) for reflection by canopy panels of various shapes were calculated. Numerical calculation of TF shows that the optimum interior angle of an isosceles triangle giving a flat response lies between 90° to 120°. And, the use of a number of canopy panels above both sides of a concert hall helps to control lateral low‐frequency components for the audience area, and produces a small value of interaural cross correlation (IACC). It is interesting to observe that the interior angle of the triangular panels composing the canopy in the Tanglewood Music Shed falls in the optimum range. The TFs of large triangular panels suspended in front of the top of the proscenium arch in Fujita Hall, a concert hall designed by us, indicate the existence of relatively strong low‐frequency components of reflection from lateral directions to a listener at the front section of the hall. Further, higher frequency components of the reflections tend to arrive near to the median plane of each listener.
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43.55.Fw Auditorium and enclosure design
43.55.Gx Studies of existing auditoria and enclosures

Sound transmission through two kinds of porous concrete blocks with attached drywall

A. C. C. Warnock

J. Acoust. Soc. Am. Volume 92, Issue 3, pp. 1452-1460 (1992); (9 pages)

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The airflow resistance of concrete blocks plays a significant role in determining sound transmission losses for block walls with drywall added on furring. Two block types were tested. One was so porous that sound passed easily through it and there was no sign of a mass–air–mass resonance behind the added drywall. Such blocks must be plastered or else their contribution to the attenuation of sound is very small. Once plastered, the low porosity increased the effective depth of the air cavity behind the added drywall on the unplastered side with attendant increases in sound transmission loss. The mass–air–mass resonance occurred at a lower frequency corresponding to the distance from the drywall through the block to the plaster. The other block type tested had a higher airflow resistance that added a loss mechanism to the wall systems; the mass–air–mass resonance was damped and transmission loss was not reduced by the resonance. The benefits of having drywall added on furring were still obtained with these blocks. Their sound transmission losses were increased when plaster was applied on one face but the increase was not so great as for the very porous blocks. Airflow resistance is an important parameter that should be considered when designing block walls.
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43.55.Rg Sound transmission through walls and through ducts: theory and measurement
43.55.Ti Sound-isolating structures, values of transmission coefficients
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