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Sep 1983

Volume 74, Issue 3, pp. 677-1100

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A Lamb wave voltage sensor

Kohji Toda and Koichi Mizutani

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 677-679 (1983); (3 pages)

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A voltage sensor using a Lamb wave delay line oscillator is described. The device consists of two pairs of interdigital transducers and one plate electrode at the central part of the device. All of these transducers have the counter electrodes on the bottom surface of the substrate. The oscillation frequency of the device changes significantly with the voltage applied to the central plate electrode. Performances of the voltage sensor are given, including sensitivity and its frequency dependence.
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43.38.Fx Piezoelectric and ferroelectric transducers
43.38.Hz Transducer arrays, acoustic interaction effects in arrays
43.35.Pt Surface waves in solids and liquids

Foil electret transducer for blood pressure monitoring

J. E. West, I. J. Busch‐Vishniac, G. A. Harshfield, and T. G. Pickering

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 680-686 (1983); (7 pages)

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A foil electret microphone for use under the cuff of an automatic blood pressure monitoring system is described. The transducer is designed to operate with relatively flat sensitivity over a static pressure range of 40 to 250 mm Hg (5.33×104 to 3.33×105 dyn/cm2). The new electret microphone differs from conventional microphones used for airborne sound reception in two ways: (1) the diaphragm thickness is 50  μm rather than the typical 12.5 or 25  μm, and (2) the backplate contains a set of annular ridges spaced at 4 mm rather than the typical 7–10 mm. This microphone offers three advantages over the piezoelectric microphone now in use: (1) greater tolerance in positioning the microphone over the brachial artery, (2) nearly 20‐dB higher sensitivity and signal‐to‐noise ratio, and (3) the ability to obtain measurements with the microphone placed midway between the elbow and shoulder. Tests of the new foil electret microphone in conjunction with the automatic blood pressure monitoring system indicate that the automatic and conventional measurements of systolic and diastolic blood pressure agree to within 5 mm Hg at least 90% of the time. In addition, the electret microphone is able to obtain automatic measurements on subjects with a wider range of ages and sizes.
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43.38.Kb Microphones and their calibration
43.38.Fx Piezoelectric and ferroelectric transducers
87.57.-s Medical imaging
87.63.-d Non-ionizing radiation equipment and techniques
87.85.Pq Biomedical imaging
43.80.Vj Acoustical medical instrumentation and measurement techniques

Effect of half‐wavelength membranes on the axial resolution of real‐time ultrasonic scanners

Jonathan Ophir and P. A. Narayana

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 687-690 (1983); (4 pages)

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The effect of interposing a half‐wavelength membrane between the transducer and the load in mechanically sectored ultrasonic scanners was investigated. In particular, the effects of impedance mismatch and detuning on broadband spectra were theoretically and experimentally studied. These studies indicate that for certain common cases, a bandwidth reduction of 20%–30% can be anticipated, resulting in a proportional decrease in the axial resolution.
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43.80.Vj Acoustical medical instrumentation and measurement techniques
87.57.-s Medical imaging
87.63.-d Non-ionizing radiation equipment and techniques
87.85.Pq Biomedical imaging
43.35.Yb Ultrasonic instrumentation and measurement techniques
43.60.Gk Space-time signal processing, other than matched field processing

Characteristics of the glottal turbulent noise source

Robert E. Hillman, Elizabeth Oesterle, and Lawrence L. Feth

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 691-694 (1983); (4 pages) | Cited 3 times

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This investigation utilized a reflectionless tube technique to obtain direct estimates of turbulent noise produced at the glottis during whispered vowels. In the past, the glottal turbulent noise source has been described theoretically as a series pressure source having a spectrum that is relatively flat for 2 or 3 oct around a center frequency [K. N. Stevens, J. Acoust. Soc Am. 50, 1180–1192 (1971)]. Center frequency is determined primarily by the area of the constriction at which turbulence is produced with the volume velocity of air flowing through the constriction. The present results were shown to substantiate this theoretically based description of the glottal turbulent noise source. In addition, there was no significant difference between the glottal turbulent noise spectra of male and female speakers. The application of these findings to the synthesis of whispered vowels is discussed.
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43.70.Bk Models and theories of speech production
43.72.Ar Speech analysis and analysis techniques; parametric representation of speech

Dynamic specification of coarticulated vowels

Winifred Strange, James J. Jenkins, and Thomas L. Johnson

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 695-705 (1983); (11 pages) | Cited 40 times

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An adequate theory of vowel perception must account for perceptual constancy over variations in the acoustic structure of coarticulated vowels contributed by speakers, speaking rate, and consonantal context. We modified recorded consonant–vowel–consonant syllables electronically to investigate the perceptual efficacy of three types of acoustic information for vowel identification: (1) static spectral ‘‘targets,’’ (2) duration of syllabic nuclei, and (3) formant transitions into and out of the vowel nucleus. Vowels in /b/–vowel–/b/ syllables spoken by one adult male (experiment 1) and by two females and two males (experiment 2) served as the corpus, and seven modified syllable conditions were generated in which different parts of the digitized waveforms of the syllables were deleted and the temporal relationships of the remaining parts were manipulated. Results of identification tests by untrained listeners indicated that dynamic spectral information, contained in initial and final transitions taken together, was sufficient for accurate identification of vowels even when vowel nuclei were attenuated to silence. Furthermore, the dynamic spectral information appeared to be efficacious even when durational parameters specifying intrinsic vowel length were eliminated.
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43.70.Dn Disordered speech

Effect of burst amplitude on the perception of stop consonant place of articulation

Ralph N. Ohde and Kenneth N. Stevens

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 706-714 (1983); (9 pages) | Cited 8 times

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We have examined the effects of the relative amplitude of the release burst on perception of the place of articulation of utterance‐initial voiceless and voiced stop consonants. The amplitude of the burst, which occurs within the first 10–15 ms following consonant release, was systematically varied in 5‐dB steps from −10 to +10 dB relative to a ‘‘normal’’ burst amplitude for two labial‐to‐alveolar synthetic speech continua—one comprising voiceless stops and the other, voiced stops. The distribution of spectral energy in the bursts for the labial and alveolar stops at the ends of the continuum was consistent with the spectrum shapes observed in natural utterances, and intermediate shapes were used for intermediate stimuli on the continuum. The results of identification tests with these stimuli showed that the relative amplitude of the burst significantly affected the perception of the place of articulation of both voiceless and voiced stops, but the effect was greater for the former than the latter. The results are consistent with a view that two basic properties contribute to the labial–alveolar distinction in English. One of these is determined by the time course of the change in amplitude in the high‐frequency range (above 2500 Hz) in the few tens of ms following consonantal release, and the other is determined by the frequencies of spectral peaks associated with the second and third formants in relation to the first formant.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Perception of intervocalic stop consonants: The contributions of closure duration and formant transitions

Vivien C. Tartter, Donna Kat, Arthur G. Samuel, and Bruno H. Repp

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 715-725 (1983); (11 pages) | Cited 2 times

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Acoustic analyses of vowel–consonant–vowel (VCV) utterances indicate that they generally include formant transitions from the first vowel into a period of closure (VC transitions), and transitions out of the closure into the second vowel (CV transitions). Three experiments investigated the perceptual importance of the VC transitions, the CV transitions, and the closure period in identification of medial stop consonants varying in place of articulation. Experiment 1 compared identification of members of synthetic VC and CV continua with those from VCV series made by concatenating corresponding VC and CV stimuli using various closure durations. Experiment 2 examined identification of VCV stimuli constructed with only VC, only CV, or both VC and CV transitions; again closure duration was systematically varied. Experiment 3 correlated CV and VC identification with identification of VCV stimuli. Neither closure duration nor formant transition structure (i.e., only VC, only CV, or both) had an independent effect on identification. Instead, the formant structure and closure duration together strongly affected stop identification. When both VC and CV transitions were present, the CV transitions contributed somewhat more to identification of medial stops with short closures, than the VC transitions did. With longer closure durations, neither set of transitions appeared to determine perceived place of articulation in any simple way. Overall, the data indicate that the perception of a medial consonant is more than simply a (weighted) sum of its parts.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Intelligibility of VCV segments excised from connected speech

A. Schmidt‐Nielsen

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 726-738 (1983); (13 pages)

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Confusions between pairs of intervocalic consonants excised from connected speech were investigated under several conditions of speech degradation: digital voice processing, noise, and bandpass limiting. The distribution of the types of errors that were made, e.g., voicing, nasality, place of articulation, differed from those made on citation form syllable‐initial consonants. Intervocalic consonants that were taken from word‐initial position scored higher with no degradations than those taken from word‐medial or word‐final position, but the medial and final segments suffered less under mild degradations than did the initial ones.
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43.70.Dn Disordered speech
43.71.Gv Measures of speech perception (intelligibility and quality)
43.72.-p Speech processing and communication systems
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Effect of a single interfering noise or speech source upon the binaural sentence intelligibility of aged persons

A. J. Duquesnoy

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 739-743 (1983); (5 pages) | Cited 38 times

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The free‐field speech‐reception threshold (SRT) for sentences was investigated in quiet and under nine conditions involving noise or competing speech for a group of 20 elderly subjects (ten male, age 75–85; ten female, age 76–88) and a reference group of ten young normal‐hearing subjects. The noise source had the same long‐term average spectrum as the competing speech. The interfering signals were presented at a constant level of 55 dBA. All elderly subjects had moderate, nearly symmetrical pure‐tone hearing losses with an average loss at 500, 1000, and 2000 Hz of between 9 and 40 dB re: ISO‐389. The main results are (1) the SRT values in noise and competing speech are about equal, whereas the normal‐hearing subjects showed a lower SRT (7 dB lower for the condition that both sound sources are in front) in competing speech than in noise; apparently, the elderly subjects do not benefit from the relatively silent periods in competing speech; (2) the gain obtained by moving the interfering noise source from the front to the lateral position is only 2.5 dB, in contrast to a gain of 9.6 dB for the young subjects; apparently, the elderly are unable to make full use of the spatial divergence between primary speaker and noise source.
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43.71.Gv Measures of speech perception (intelligibility and quality)
43.50.Qp Effects of noise on man and society

Experimental manipulation of speaking rate for studying temporal variability in children’s speech

Bruce L. Smith, Michael D. Sugarman, and Steven H. Long

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 744-749 (1983); (6 pages) | Cited 6 times

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Children’s speech timing is often more variable than adults’. In the present study, two hypotheses that have been proposed to account for this observation are considered. One claims that children do not have neuromotor control capabilities comparable to adults. The other suggests that the greater variability is a statistical consequence of children’s longer segment durations. These two hypotheses were examined by having children and adults speak at both faster and slower rates than normal. Within‐group comparisons across different rates and between‐group comparisons for similar durational values were made from spectrographic measurements. Results indicate that both statistical and neuromotor factors seem to contribute to the greater variability commonly observed in children’s speech.
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43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation
43.70.Bk Models and theories of speech production
43.72.Ar Speech analysis and analysis techniques; parametric representation of speech

Suggested formulae for calculating auditory‐filter bandwidths and excitation patterns

Brian C. J. Moore and Brian R. Glasberg

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 750-753 (1983); (4 pages) | Cited 105 times

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Recent estimates of auditory‐filter shape are used to derive a simple formula relating the equivalent rectangular bandwidth (ERB) of the auditory filter to center frequency. The value of the auditory‐filter bandwidth continues to decrease as center frequency decreases below 500 Hz. A formula is also given relating ERB‐rate to frequency. Finally, a method is described for calculating excitation patterns from filter shapes.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking

Suppression in simultaneous masking

Hugo Fastl and Matthias Bechly

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 754-757 (1983); (4 pages) | Cited 2 times

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Suppression, i.e., the decrease of masked threshold caused by the addition of a second masker M2 to a first masker M1, is measured for the case of simultaneous masking. The magnitude of suppression decreases with increasing test tone duration; pulsed maskers elicit somewhat more suppression than continuous maskers. In comparison to suppression effects obtained in nonsimultaneous masking (post‐masking, pulsation threshold) suppression in simultaneous masking is considerably smaller and was found only at the lower slopes of the two maskers. Suppression in simultaneous masking would not be predicted by those models of suppression which require nonsimultaneous presentation of maskers and test sound.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Cb Loudness, absolute threshold

Broadband masking noise and behavioral pure tone thresholds in cats

John A. Costalupes

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 758-764 (1983); (7 pages) | Cited 2 times

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The threshold for detection of pure tones in broadband noise was determined for three cats using an auditory reaction time procedure. Critical ratio is defined as the ratio of the signal power at the masked threshold for detection to the spectrum level of the noise. Critical ratios were obtained for 250‐, 500‐Hz, and 1‐, 2‐, 4‐, 8‐, and 16‐kHz tones over a wide range of noise intensities. Results indicate that critical ratios increase with the frequency of the tone stimulus. At frequencies below 4 kHz, critical ratios remain constant at moderate and high noise intensities. For frequencies above 4 kHz, critical ratios increase as the level of the masking noise is raised from moderate to high levels. The difference between low‐ and high‐frequency behavior of the level dependence of critical ratios is considered in terms of two possible mechanisms: (1) different mechanisms may be involved in the encoding of low‐ and high‐frequency information by the nervous system or (2) the difference in level dependence may be due to attenuation by the action of the middle ear muscles at high sound levels.
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43.66.Gf Detection and discrimination of sound by animals
43.66.Dc Masking
43.66.Fe Discrimination: intensity and frequency
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Time course of adaptation and recovery of channels selectively sensitive to frequency and amplitude modulation

B. W. Tansley and J. B. Suffield

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 765-775 (1983); (11 pages) | Cited 16 times

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In a series of experiments we investigated the time course of adaptation and recovery of channels in the human auditory system selectively sensitive to frequency and amplitude modulation (FM and AM). We determined the rate of loss of sensitivity to modulation using sinusoidal frequency or amplitude modulation (SFM or SAM) of a 50 dB SL, 500‐Hz pure tone carrier over a 30‐min period. Adaptation stimuli were modulated at ten times the preadaptation modulation detection threshold, as determined immediately before the 30‐min adaptation session. Modulation rates investigated were 2, 4, 8, 16, and 32 Hz. Long exposure to SFM always elevated thresholds for detection of SFM more than this exposure elevated thresholds for detection of SAM. Similarly, adapting to SAM always elevated SAM detection thresholds more than SFM thresholds. Loss of sensitivity during adaptation was relatively slow; asymptotic loss of modulation sensitivity took 20 to 30 min. The recovery of modulation sensitivity after cessation of the modulation component of the adapting stimulus was determined in a second experiment. Recovery was found to be rapid; most of the recovery occurred within the first 60 sec. Our evidence suggests that there exist two types of modulation‐sensitive channels in the human auditory system—one selectively sensitive to amplitude modulation and the other to frequency modulation. They appear to have similar time courses for adaptation and for recovery.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Ba Models and theories of auditory processes

Performance of hearing‐impaired listeners under various types of amplitude compression

Igor V. Nábělek

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 776-791 (1983); (16 pages) | Cited 2 times

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Speech perception by subjects with sensorineural hearing impairment was studied using various types of short‐term (syllabic) amplitude compression. Average speech level was approximately constant. In quiet, a single‐channel wideband compression (WBC) with compression ratio equal to 10, attack time 10 ms and release time 90 ms produced significantly higher scores than a three‐channel multiband compression (MBC) or no compression when a nonsense syllable test (City University of New York) was used. The scores under MBC, WBC, or no compression were not significantly different when the modified rhyme test (MRT) was used. But when overshoots caused by compression were clipped, the MRT scores improved significantly. The influence of MBC on reverberant speech and of WBC on noisy speech were tested with the MRT. Reverberation reduced the scores, and this reduction was the same with compression as without. Noise added to speech before compression also reduced the scores, but the reduction was larger with compression than without. When noise was added after compression, an improvement was observed when WBC had a compression ratio of about 5, attack time 1 ms, and release time 30 ms. Other compression modes (e.g., with high‐frequency pre‐emphasis) did not show an improvement. The results indicate that WBC with a compression ratio around 5, attack time shorter than 3 ms, and release time between 30 and 90 ms can be beneficial if signal‐to‐noise ratio is large, or, if in a noisy or reverberant environment, the effects of noise or reverberation are eliminated by using listening systems.
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43.66.Sr Deafness, audiometry, aging effects
43.66.Ts Auditory prostheses, hearing aids
43.72.-p Speech processing and communication systems
43.70.Dn Disordered speech

Acoustically evoked radial current densities in scala tympani

W. E. Brownell, P. B. Manis, M. Zidanic, and G. A. Spirou

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 792-800 (1983); (9 pages)

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We have developed a method for measuring current density within the fluid spaces of the cochlea and report the existence of stimulus evoked radial currents in scala tympani of the guinea pig cochlea. The spatial distribution of electrical potentials in scala tympani was measured along a radial path parallel to the basilar membrane. Click evoked potentials were recorded at successive points separated by a fixed increment as the electrode was either advanced from the spiral ligament or withdrawn from a position near the modiolus. Potential differences were found to exist between recording points and gradients were calculated from the evoked potential measurements. Evoked potential gradients are observed at the same position along the path of the electrode both on advancing and on withdrawing the electrode. The largest potential gradients are located beneath the organ of Corti. Condensation and rarefaction clicks produce radial currents in opposite directions at a given location along the electrode’s path. The magnitude and spatial distribution of radial currents is a function of stimulus intensity. Potential gradients of small magnitude are observed at locations other than below the organ of Corti in some penetrations. Control experiments suggest the smaller gradients are artifactual and may result from displacement of the spiral ligament by the recording electrode. The locations, magnitude, and direction of intracochlear ionic flow relate directly to the mechano‐electrical transduction process in the organ of Corti.
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43.64.Nf Cochlear electrophysiology
43.64.Ri Evoked responses to sounds

Suppression of auditory nerve responses. II. Suppression threshold and growth, iso‐suppression contours

Eric Javel, JoAnn McGee, Edward J. Walsh, Glenn R. Farley, and Michael P. Gorga

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 801-813 (1983); (13 pages) | Cited 15 times

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Two‐tone ‘‘synchrony suppression’’ was studied in responses of single auditory nerve fibers recorded from anesthetized cats. Suppression thresholds for suppressor tones set to a fiber’s characteristic frequency (CF) were approximately equal to discharge rate thresholds for CF tones. Suppression thresholds above and below CF were usually lower than the corresponding discharge rate thresholds. However, at all frequencies studied (including CF), suppression thresholds were higher than the corresponding thresholds for discharge synchronization. Across fibers, rates of suppression growth for suppressors at CF were greatest in low‐CF fibers and least in high‐CF fibers, and there was a systematic decrease in suppression growth rate at CF as CF increased. Within fibers, rates of suppression growth above CF were typically less than at CF, and slopes were monotonically decreasing functions of frequency. Within‐fiber rates of suppression growth below CF were variable, but they usually were greater than rates of growth at CF. Iso‐suppression contours (frequencies and intensities producing criterion amounts of suppression) indicated that tones near CF are the most potent suppressors at near‐threshold intensities, and that the frequency producing the most suppression usually shifts downward as the amount of suppression increases. These data support the notion that synchrony suppression arises primarily as a passive consequence of hair cell activation.
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43.64.Pg Electrophysiology of the auditory nerve

Equivalent bandwidth of a general class of polynomial smoothers

Lawrence C. Ng and Robert A. LaTourette

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 814-826 (1983); (13 pages)

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This paper presents a detailed investigation of the properties of a general class of least‐mean‐square‐fit (LMSF) smoothers in the presence of a white or colored input sequence. The results of the investigation show that the LMSF can be described as a low‐pass filter whose frequency response characteristics can be calculated exactly. A particularly useful result derived from the frequency response characteristics is the LMSF equivalent bandwidth. It was shown that knowledge of LMSF bandwidth, plus knowledge of the input bandwidth, provides the second‐order statistical description of the LMSF output noise process. Results of the analysis are verified by extensive computer simulation.
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43.60.Cg Statistical properties of signals and noise
43.60.Gk Space-time signal processing, other than matched field processing

Nonstationarity in acoustic fields

Y. H. Tsao and J. K. Hammond

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 827-839 (1983); (13 pages)

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Acoustic pressure fields measured by an observer when the source or observer or both are moving is a nonstationary random process even if the source generates a random process which is stationary in the reference frame of the source. The causes of nonstationarity are classified as being due to wave expansion, directivity, and Doppler shift. This paper is concerned with developing two‐dimensional (frequency–time) spectral descriptions for the processes by constraining the processes to fit within the framework of the ‘‘evolutionary spectral density.’’ Earlier literature has described how evolutionary spectra may be estimated from single sample realizations. Spectral representation forms for free‐field acoustic processes produced by moving monopole and dipole excitations are derived from the fundamental wave equations.
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43.60.Cg Statistical properties of signals and noise
43.50.Cb Noise spectra, determination of sound power

Side‐lobe reduction in the ring array pattern for synthetic aperture imaging of coherent sources

Ajay K. Luthra and Saleem A. Kassam

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 840-846 (1983); (7 pages)

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A new technique is described for image formation using a circular array of transducer elements in which each element acts as both a transmitter and receiver. By processing the amplitude and phase measurements acquired in such a system in a particular way, it is shown that it is possible to approach the performance of a transmit–receive filled circular aperture of the same dimension. The technique combines a synthetic aperture scheme with an earlier approach for array pattern synthesis ( J2 synthesis). The proposed technique does not require spatial incoherence of reflections from the object plane, which was a severe limitation in earlier use of J2 synthesis. The effect of array sampling with a finite number of transducer elements is also examined.
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43.60.Gk Space-time signal processing, other than matched field processing
43.35.Sx Acoustooptical effects, optoacoustics, acoustical visualization, acoustical microscopy, and acoustical holography
43.20.Bi Mathematical theory of wave propagation

Some simple expressions for the beamforming properties of focused high‐resolution circular arrays, with applications to refocusing systems

James F. Lynch

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 847-850 (1983); (4 pages)

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Some simple analytical expressions for two important optical characteristics of focused high‐resolution circular arrays are derived, specifically those for beam patterns and depths of field. The expressions generated, in addition to providing simple estimational tools for array design purposes, also lend themselves to straightforward interpretation in terms of the Fourier (farfield) and Fresnel (nearfield) kernels. Applications to refocusing arrays are discussed, as well as similarities and differences between such (sonar oriented) systems and radar and optical systems.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems
43.28.Tc Sound-in-air measurements, methods and instrumentation for location, navigation, altimetry, and sound ranging

Optimum SNR enhancement of narrow‐band signals in surface reverberation

Frank W. Symons

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 851-860 (1983); (10 pages)

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Theoretical performance capabilities of the optimum filter for maximization of signal‐to‐noise ratio (SNR) and a suboptimum approximation, easy to realize in practice, are compared to the conventional matched filter (designed for white noise only) in the case of low Doppler signals masked by surface reverberation. Spectrum models for surface backscattering as a function of local environmental parameters are based on recent results describing scattering from a subsurface bubble layer. Significant improvements in SNR in the low Doppler region as well as a dramatic dependence of performance on windspeed and direction, and the ratio of reverberation to uncorrelated noise power are predicted. Results serve as average performance limits to be compared to those obtained by algorithmic implementations of optimum and adaptive filters.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems
43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries
43.58.Kr Spectrum and frequency analyzers and filters; acoustical and electrical oscillographs; photoacoustic spectrometers; acoustical delay lines and resonators

On the application of coherence techniques for source identification in a multiple noise source environment

M. E. Wang and Malcolm J. Crocker

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 861-872 (1983); (12 pages) | Cited 2 times

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Two approaches for noise source identification based on theory for multiple‐input systems have been investigated. The concepts of the frequency response function and the coherent residual spectral density function were used to estimate the spectra of the noise sources that were accounted for in the multiple‐input model. The key factors which determine the applicability of the techniques for noise source identification are found to be the degrees of coherence between the measured inputs. The experimental results showed that, for a noise system with multiple sources which were slightly coherent to each other, the techniques presented might be able to estimate the spectra of the noise sources, provided there was only modest measurement contamination.
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43.60.Gk Space-time signal processing, other than matched field processing
43.50.Yw Instrumentation and techniques for noise measurement and analysis

Calculation of subjective preference at each seat in a concert hall

Yoichi Ando

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 873-887 (1983); (15 pages) | Cited 6 times

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This paper represents a method of calculating the subjective preference of sound fields in concert halls before construction. Subjective preference judgments (paired comparison tests) were systematically performed using fully independent objective parameters of acoustic information which describe the signals to the two ears. The sound fields with various combinations of listening level, delay of early multiple reflections, subsequent reverberation time, and magnitude of the interaural cross correlation were simulated with the aid of a digital computer. The optimal conditions maximizing the subjective preference could be found for each objective parameter, because the parameters had an almost independent effect on the subjective preference judgments. Based on the linear scale value, which is obtained by applying the law of comparative judgment, we can calculate a total preference value according to the ‘‘principle of superposition.’’ Examples of calculating the preference values by use of the plan and the cross section of a concert hall are described.
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43.55.Fw Auditorium and enclosure design
43.55.Br Room acoustics: theory and experiment; reverberation, normal modes, diffusion, transient and steady-state response

Community response to blasting

Sanford Fidell, Richard Horonjeff, Theodore Schultz, and Sherri Teffeteller

J. Acoust. Soc. Am. Volume 74, Issue 3, pp. 888-893 (1983); (6 pages) | Cited 1 time

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Annoyance due to chronic exposure to blast noise and vibration was assessed in residential areas near two surface mines and a quarry. It was found possible to base useful prediction of the prevalence of high annoyance on a metric of outdoor ground vibration related to high centiles of the long term distribution of exposure levels.
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43.50.Ba Noisiness: rating methods and criteria
43.50.Qp Effects of noise on man and society
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