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Journal of the Acoustical Society of America

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Dec 1993

Volume 94, Issue 6, pp. 3051-3544

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EDITORIAL: Voluntary page charges

Daniel W. Martin

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3051-3051 (1993); (1 page)

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Abstract Unavailable
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43.05.Gv Publications, ARLO, Echoes, ASA Web page, electronic archives and references
43.10.Gi Editorials, Forum

Development of (3,1) drive low‐frequency piezofilm hydrophones with improved sensitivity

J. Jagannath Bhat, P. Philip Thomson, and P. R. Saseendran Pillai

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3053-3056 (1993); (4 pages)

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Piezopolymers are becoming popular as the active material for the design of probes for sensing ultrasonic fields and quantitative determination of acoustic field parameters in water and biological media. A new innovative transducer design proposed here utilizes a poled piezofilm which is made to vibrate in (3,1) drive by a modified structural assembly. The voltage generated in this design is found to be greater compared to that in the conventional design, due to the concentration of acoustic pressure to a very small cross‐sectional area. The prototype design consists of a prestretched piezofilm fixed to a phosphor bronze diaphragm through a driver pin. The proposed design yielded sensitivities to the extent of −170 dB re: 1 V/μPa in water at around 1.5 kHz.
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43.38.Ar Transducing principles, materials, and structures: general
43.38.Fx Piezoelectric and ferroelectric transducers
43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration

Distributed acoustic echo cancellation system with double‐talk detector

Sen M. Kuo and Zhibing Pan

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3057-3060 (1993); (4 pages)

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The acoustic echo cancellation system using the least‐mean‐square algorithm is an effective technique to reduce acoustic echoes in a hands‐free telephone system. However, very high‐order adaptive filters and multiple echo paths have computational and convergence problems. Recently, a new acoustic echo cancellation microphone (AECM) system [S. M. Kuo and Z. Pan, J. Acoust. Soc. Am. 93, 1629–1636 (1993)], which uses two closely spaced directional microphones and a low‐order adaptive filter, was proposed to solve these problems. In order to make the AECM system work automatically, a double‐talk detector, which is presented in this paper, is required. This detector utilizes the power difference between the system output signal and the microphone output signal. Computer simulations were conducted to demonstrate the effectiveness of the detectors. The complete AECM system was implemented on the TMS320C25, and real‐time experiments were conducted in the conference room.
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43.38.Si Telephones, earphones, sound power telephones, and intercommunication systems
43.72.Kb Speech communication systems and dialogue systems
43.60.Gk Space-time signal processing, other than matched field processing

Broadband determination of ultrasonic attenuation and phase velocity in insulating materials

Thierry Ditchi, Claude Alquié, and Jacques Lewiner

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3061-3066 (1993); (6 pages) | Cited 7 times

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Ultrasonic attenuation and phase velocity in a solid insulating material are determined in a broad frequency band by studying the propagation of a pressure pulse in a sample of this material submitted to an electric field of known value. During the propagation of the pulse in the sample, which is placed between short‐circuited electrodes, a current is generated in the external circuit. This signal gives directly the time dependence of the pressure pulses entering and exiting the sample, from which the frequency‐dependent attenuation and phase velocity are deduced by Fourier analysis. The pressure pulse is generated by the impact of a laser pulse on an absorbing surface adjacent to one face of the sample. New results obtained in polyethylene and silicone samples are presented. The proposed method presents the following advantages: It avoids the use of a transducer, reduces the impedance matching requirements, and provides information in a broad frequency band by a single measurement.
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43.58.Dj Sound velocity
43.35.Cg Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in solids; elastic constants
43.20.Jr Velocity and attenuation of elastic and poroelastic waves

The sounds of bicolor damselfish (Pomacentrus partitus): Predictors of body size and a spectral basis for individual recognition and assessment

Arthur A. Myrberg, Jr., Samuel J. Ha, and Michael J. Shamblott

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3067-3070 (1993); (4 pages) | Cited 5 times

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Evidence is provided that the ‘‘chirp,’’ a sound commonly produced by males of the bicolor damselfish (family: Pomacentridae) possesses an anatomical constraint: The peak frequency within its power spectrum reflects a clear inverse relationship to body size. For every 1‐mm change in the standard length of a male (range: 50–69 mm), the peak frequency of its sounds shifts by approximately 20 Hz. The ultimate constraint appears to be the volume of an individual’s gas bladder. This provides an individualistic feature to the sounds of different sized colony members, all of whose sounds possess an otherwise extremely stereotyped temporal pattern of their included pulses. This finding may aid in clarifying the mechanism that provides the clue for the already established acoustical recognition of individuals within colonies of the species.
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43.80.Jz Use of acoustic energy (with or without other forms) in studies of structure and function of biological systems
43.80.Nd Effects of noise on animals and associated behavior, protective mechanisms
43.30.Re Signal coherence or fluctuation due to sound propagation/scattering in the ocean

Behavioral measures of auditory thresholds in developing tree shrews (Tupaia belangeri)

Elke Zimmermann

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3071-3075 (1993); (5 pages)

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The development of hearing in tree shrews (Tupaia belangeri) was tested through behavioral measures in the range of 0.3–60 kHz from the first postnatal thresholds detectable through to those of weanlings. The onset of hearing was defined using unconditioned startle responses, while, later on, pinna reflexes and head movements were used as behavioral indicators of sound perception. Startle responses could be evoked from day 16 after birth (DAB) onwards, indicating that the hearing system is capable of transferring airborne sound, although no reactions to tones were recorded at this age. The first unconditioned reactions to tones were found at DAB 18 in the range of 1–5 kHz (range of mother–infant call). Responses were restricted to the range of greatest sensitivity in adults (1–10 kHz). An increase in sensitivity was detected from DAB 18–38. The lowest tested frequency (0.3 kHz) reached adult levels earlier than higher frequencies. No further improvement of auditory thresholds could be observed from DAB 38 (end of nutritional weaning) to DAB 42. Results were related to recent findings on the development of acoustical behavior and of the peripheral and central auditory system of tree shrews and other mammals.
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43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing
43.80.Jz Use of acoustic energy (with or without other forms) in studies of structure and function of biological systems

Vocal affect in three‐year‐olds: A quantitative acoustic analysis of child laughter

Evangeline E. Nwokah, Patricia Davies, Asad Islam, Hui‐Chin Hsu, and Alan Fogel

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3076-3090 (1993); (15 pages) | Cited 3 times

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Recordings were obtained of the laughter vocalizations of four 3‐year‐old children during three sessions of spontaneous free‐play between mother and child in a laboratory playroom. Acoustic analysis was used to determine laughter durations, laughter events, F0, and harmonic characteristics, and to suggest a taxonomy of laughter types. Melodic contours were assessed from patterns of F0 change during laughter. Mean duration of laughs ranged from 200 ms to 2.0 s, but events within a laugh were usually about 200‐ms duration. Laughs were intuitively classified into four major types, and, following the acoustic analyses, were further defined and classified into types and subtypes of exclamatory and dull comment; chuckle; basic, variable, and classical rhythmical; and squeal. Melodic contours included more rising contours than previously reported for cry, but there was great variability in the types of contours produced especially for rhythmical laughs. The results of the acoustic analyses are discussed in relation to (a) the development of a taxonomy of laughter and (b) different features of the vocal affect characteristics of high‐intensity emotion.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.71.Gv Measures of speech perception (intelligibility and quality)

Speaker‐independent consonant classification in continuous speech with distinctive features and neural networks

Renato De Mori and Giovanni Flammia

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3091-3103 (1993); (13 pages)

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This paper provides experimental evidence to the assertion that the design of appropriate neural networks (NN) for speech recognition should be inspired by acoustic and phonetic knowledge, and not only by knowledge in pattern recognition. Rather than investigating the NN learning paradigm, the paper is focused on the influence of the input parameters, of the internal structure, and of the desired output representation on the classification performance of recurrent multilayer perceptrons. As an instructive example, the paper analyzes the problem of classifying ten stop and nasal consonants in continuous speech independently of the speaker. Experiments are reported for the TIMIT database, using 343 speakers in the training set and 77 different speakers in the test set. Comparative experiments show that good performance is obtained when many input acoustic parameters are used, including a time/frequency gradient operator related to transitions of the second formant, and when the desired outputs represent context‐dependent articulatory features. Classification is performed by principal component analysis of the NN outputs. Refinement of the design parameters yield increasingly better performance on the test set, ranging from 45% errors for a perceptron without hidden nodes to 23.3% errors for the best NN.
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43.72.Ne Automatic speech recognition systems
43.72.Bs Neural networks for speech recognition

Dynamics of the two‐mass model of the vocal folds: Equilibria, bifurcations, and oscillation region

Jorge C. Lucero

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3104-3111 (1993); (8 pages) | Cited 13 times

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The dynamics of the large‐amplitude oscillation of the vocal folds is analyzed using the two‐mass model. First, the equilibrium positions are determined in the case of a rectangular prephonatory glottis, and the existence of two equilibrium positions besides the rest position is shown. Their stability is examined and a bifurcation diagram is derived with a normalized subglottal pressure and a coupling coefficient as control parameters. Phase plane plots are shown to illustrate the results. The cases of convergent and divergent prephonatory glottis are then briefly considered. The main results are finally discussed relative to previous analytical works; it is shown that they disprove the previous oscillation theory based on the existence of a glottal negative differential resistance.
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43.70.Aj Anatomy and physiology of the vocal tract, speech aerodynamics, auditory kinetics
43.70.Bk Models and theories of speech production

Informational masking for multicomponent maskers with spectral gapsa)

Donna L. Neff, Theresa M. Dethlefs, and Walt Jesteadt

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3112-3126 (1993); (15 pages) | Cited 17 times

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Simultaneous maskers comprised of a few random‐frequency sinusoids can produce considerable informational (uncertainty‐based) masking if the component frequencies are drawn from a wide range and changed with each stimulus presentation. The present experiments examined the effect on informational masking of removing masker energy from large frequency regions around the signal. Threshold for a 1000‐Hz signal was measured in the presence of maskers comprised of 2, 4, 6, 10, 50, or 100 random‐frequency sinusoids, notched‐noise, or two fixed‐frequency sinusoids. The multicomponent maskers had a maximum frequency range of 300–3000 Hz, typically excluding a 160‐Hz band around the signal. In comparison conditions, masker frequencies were limited to the high or low side of the signal, or the gap around the signal was progressively widened. Four listeners showed substantial informational masking which was not eliminated even by extreme spectral gaps in the maskers. Four other listeners showed much smaller effects of masker uncertainty across all conditions. Notched‐noise measures of auditory‐filter width did not distinguish the two subject groups, but indices of processing efficiency were typically poorer for the high‐threshold listeners, as were measures of both the width and processing efficiency of presumed ‘‘attentional filters’’ under conditions of masker‐frequency uncertainty.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Lj Perceptual effects of sound

Auditory filter shapes of normal‐hearing and hearing‐impaired listeners in continuous broadband noisea)

Marjorie R. Leek and Van Summers

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3127-3137 (1993); (11 pages) | Cited 15 times

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Listeners with sensorineural hearing impairment typically exhibit auditory processing deficits such as reduced frequency and/or temporal resolution. Such deficits may represent separate sequela of auditory pathology or may result directly from the sensitivity loss and the requirement to listen at high levels. To assess the impact of increased thresholds on frequency resolution, auditory filter characteristics were determined for hearing‐impaired and normal‐hearing listeners at 500 and 2000 Hz in the presence of continuous broadband noise meant as a rough simulation of hearing loss. In the fitting procedure, the low‐frequency skirt of the derived auditory filter was allowed to vary as a function of signal level, permitting different filter shapes to be estimated at high versus low signal levels. Listeners with moderate hearing losses at 2000 Hz demonstrated near‐normal auditory filter shapes for lower signal levels, but increasingly broad and asymmetric filters as signal level was raised. At 500 Hz, where hearing losses were mild, filter bandwidths increased little at the higher signal levels. The presence of broadband noise had essentially no effect on filter shapes of either listener group. The filter shape abnormalities demonstrated by listeners with moderate hearing loss, which were not observed in normal‐hearing listeners at the same signal levels, indicate that poor frequency resolution in these patients for high‐intensity stimuli does not follow directly from decreased sensitivity, but instead reflects an independent pathology.
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43.66.Dc Masking
43.66.Sr Deafness, audiometry, aging effects

Effects of modulation rate and rate of envelope change on modulation discrimination interference

Michael J. Shailer and Brian C. J. Moore

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3138-3143 (1993); (6 pages) | Cited 3 times

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Modulation discrimination interference (MDI) refers to a reduced ability to detect a change in modulation depth of a target carrier frequency in the presence of other modulated carriers with frequencies remote from that of the target. These experiments examined whether the tuning of MDI in the modulation domain depends primarily on the similarity in modulation periodicity of the target and masking sounds or on the similarity in rate of change of envelope of the target and masking sounds. This question was addressed by manipulating the ac component of the modulation waveform of the target and masker along part of a continuum that ranged from a sinusoid at one end to a square wave at the other; the transition time (rise/fall time) of the modulator was 6.25, 12.5, 25, or 50 ms. When the target and masker carriers were modulated at the same rate [10 periods per second (pps)], MDI decreased with decreasing rise/fall time of the target and with increasing rise/fall time of the masker. Maximum MDI did not occur when the rise/fall times of the target and masker were equal, as might be expected if MDI were caused by perceptual grouping of the target and masker. This pattern of results is consistent with the idea that, for a given carrier, shorter rise/fall times lead to greater salience. When the target and masker had the same rise/fall times, MDI was strongly affected by a difference in modulation rate of the target and masker and slightly affected by the common rise/fall time of the target and masker. Taken together, these results indicate that both modulation periodicity and rate of change of envelope influence MDI.
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43.66.Dc Masking
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Reaction times to unidirectional frequency changes: A test of Zwicker’s excitation‐pattern model

Wolfgang Ellermeier and David S. Emmerich

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3144-3152 (1993); (9 pages)

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A reaction‐time methodology was used to evaluate excitation‐pattern models of frequency discrimination. These models assume that in order to detect frequency changes, the auditory system actually monitors intensity changes in the filter array along which the excitation pattern is shifted. According to Zwicker’s model, it is along the steep low‐frequency slope of the excitation pattern that small frequency changes will be detected. Consequently, small downward shifts in frequency should produce an increment in excitation in the filter monitored, while upward shifts should be associated with a decrement in activity in that filter. Since simple auditory reaction time (RT) is differentially sensitive to increments and decrements in intensity, it seems particularly suitable to studying this conceptualization. In experiment 1, it was demonstrated that increments in intensity elicit faster responses than decrements of equal magnitude. Experiment 2 found that when subjects had to respond to small changes in frequency on the order of 4–20 Hz, reaction time to upward shifts in frequency was faster than reaction time to downward shifts by about 20 ms. This asymmetry, however, rapidly reduced to zero when the magnitude of the frequency shift was made larger, as demonstrated in experiment 3. While these results are inconsistent with the idea that increments and decrements in excitation level in a single channel tuned to the low‐frequency slope of the excitation pattern determine frequency‐discrimination performance, they can be accounted for when it is assumed that listeners monitor more than one channel to detect a frequency change. Further support for this notion was found in experiment 4 in which parts of the excitation pattern were selectively masked.
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes

Spectral pattern and the perceptual fusion of harmonics. I. The role of temporal factors

Brian Roberts and Peter J. Bailey

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3153-3164 (1993); (12 pages) | Cited 6 times

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A single even harmonic can be segregated from an odd‐harmonic complex more easily than its odd neighbors for a fundamental frequency of 100 or 200 Hz, but not 400 Hz [Roberts and Bregman, J. Acoust. Soc. Am. 90, 3050–3060 (1991)]. It has been proposed that these findings reflect a cross‐channel comparison with limited temporal resolution of amplitude‐modulation (AM) rates at the outputs of the auditory filterbank. If this ‘‘temporal‐pattern’’ hypothesis is correct, then the even–odd difference should be reduced or abolished by a manipulation that degrades information about AM rate, either by minimizing the depth or changing the pattern of AM. This was achieved by changing the phase relations between the components of the complex (experiment 1) and by adding noise to the complex (experiment 2). Subjects were presented with complex tones consisting exclusively of odd harmonics or containing one additional even harmonic. They were required to hear out one of the components and to rate its clarity. The results did not provide support for the temporal‐pattern hypothesis. First, the phase manipulation predicted to minimize AM depth did not reduce the even–odd difference. Second, the reduction in the even–odd difference resulting from the addition of noise did not appear to result from a degradation of information about AM rate.
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43.66.Lj Perceptual effects of sound
43.66.Jh Timbre, timbre in musical acoustics

Spectral pattern and the perceptual fusion of harmonics. II. A special status for added components?

Brian Roberts and Peter J. Bailey

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3165-3177 (1993); (13 pages) | Cited 8 times

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A single even‐numbered harmonic can be segregated from an odd‐harmonic complex more easily than its odd‐numbered neighbors, for low fundamentals [Roberts and Bregman, J. Acoust. Soc. Am. 90, 3050–3060 (1991)]. It has been proposed that this effect occurs because the even harmonic is inconsistent with the regular spectral pattern formed by the odd harmonics. However, in this study we evaluate some alternative accounts of the effect. The stimuli were harmonic complex tones for which one of the components was cued by a preceding pure tone. Subjects were required to listen for the cued component and either to rate its clarity (experiments 1 and 3) or to judge its pitch in relation to the preceding tone (experiment 2). It is argued that the weaker fusion of an added even harmonic depends on its status as an added component rather than as an even harmonic (experiment 1), and on its immediate perceptual salience rather than on auditory learning. An even–odd difference was also found when even and odd harmonics were tested in an identical spectral context (experiment 2), and when one of the neighboring odd harmonics was removed (experiment 3). These results are consistent with the proposal that it is easier to segregate a harmonic from a periodic complex tone when it does not form part of the regular pattern of spectral spacing defined by the other harmonics.
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43.66.Lj Perceptual effects of sound
43.66.Jh Timbre, timbre in musical acoustics

Psychophysical and speech perception studies: A case report on a binaural cochlear implant subject

R. J. M. van Hoesel, Y. C. Tong, R. D. Hollow, and G. M. Clark

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3178-3189 (1993); (12 pages) | Cited 12 times

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Further improvements in speech perception for cochlear implant patients in quiet and in noise should be possible with speech processing strategies using binaural implants. For this reason, presented here is a series of initial psychophysical and speech perception studies on the authors’ first binaural cochlear implant patient. For an approximate matching of the places of stimulation on the two sides, the patient usually reported a single percept when the two sides were simultaneously stimulated. Lateralization was strongly influenced by amplitude differences between the electrical stimuli on the two sides, but only weakly by interaural time delays. Speech testing, comparing monaural with binaural electrical stimulation, showed a binaural advantage particularly in noise.
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43.66.Ts Auditory prostheses, hearing aids
43.66.Pn Binaural hearing
43.66.Qp Localization of sound sources
43.71.Ky Speech perception by the hearing impaired

Effects of frequency on the detection of decrements and increments in sinusoids

Brian C. J. Moore, Robert W. Peters, and Brian R. Glasberg

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3190-3198 (1993); (9 pages) | Cited 9 times

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Thresholds for the detection of decrements in level of sinusoidal signals were measured as a function of decrement duration and of frequency (250, 1000, and 4000 Hz) in nine normally hearing subjects. Thresholds for detecting a brief increment in level were also measured. The sinusoids were presented in a background noise intended to mask spectral splatter associated with the decrement or increment. Performance tended to improve with increasing frequency, for all decrement durations and for increment detection. The results were analyzed using a model consisting of an array of bandpass filters (the auditory filters), each followed by a nonlinearity, a sliding temporal integrator and a decision device. The analysis indicated that the worsening in performance with decreasing frequency can be attributed mainly to changes in the efficiency of the decision device following the temporal integrator; at lower frequencies a larger change is required at the output of the integrator for threshold to be reached. At 250 Hz, the effect of the auditory filter in ‘‘smoothing’’ the internal representation of the stimuli appears to be comparable to the effect of the temporal integrator, making it difficult to determine the characteristics of the integrator.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Dc Masking
43.66.Ba Models and theories of auditory processes

Self‐suppression in a locally active nonlinear model of the cochlea: A quasilinear approach

Luc J. Kanis and Egbert de Boer

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3199-3206 (1993); (8 pages) | Cited 20 times

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Mechanical input–output functions of the cochlea for pure‐tone stimuli are nonlinear for frequencies around the characteristic frequency. To simulate these functions, a long‐wave model of the cochlea containing a saturating pressure generator (located at the site of the outer hair cells) is solved in the frequency domain with a quasilinear method. In this method distortion products in the basilar‐membrane (BM) response are treated as perturbations and the nonlinear pressure waveform is approximated by the first‐order Fourier component. Because the saturating pressure generator forms part of a feedback loop the solution of the model is achieved in a number of iteration steps. Model results show flattening of the BM response at higher input pressures; this property, called self‐suppression, is due to saturation of the pressure generator. The resulting input–output functions display the main features of experimental curves. The third‐order distortion product in the BM velocity is always more than 25 dB below the primary BM velocity and does not influence the results of the computation; this justifies the use of the quasilinear method.
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43.64.Bt Models and theories of the auditory system
43.64.Kc Cochlear mechanics

Auditory brain‐stem response correlates of resistance to noise‐induced hearing loss in Mongolian gerbils

Flint A. Boettcher

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3207-3214 (1993); (8 pages) | Cited 3 times

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The auditory brain‐stem response (ABR) was recorded from young adult Mongolian gerbils exposed to noise (octave band of noise centered at 4 kHz, 80 dB SPL, 6 h on, 18 h off) for 12 days. Temporary threshold shift (TTS) of 20–50 dB was measured at 4–8 kHz and TTS of 10 dB or less was measured at 1–2 and 16 kHz immediately after the initial exposure. Immediately following the final (12th) exposure, each animal had 10 dB or less threshold shift at all frequencies, demonstrating as much as 40‐dB resistance to TTS. Because significant TTS was limited to the high frequencies, the apical portion of the cochlea was left relatively unaffected by the exposure. Amplitudes of waves ii–iii and iv of the ABR were unaffected at low frequencies and reduced at all stimulus levels for 8 kHz on the first day of exposure; the amplitudes recovered to near‐baseline levels by the 12th day of exposure. ABR latencies of waves ii and iv were prolonged at low stimulus levels on days one and six of exposure, but recovered to baseline levels by the 12th day of exposure. Because resistance to noise exposure was observed in all subjects and resistance was limited in spectrum, the results suggest that the gerbil is an excellent model for examining mechanisms of resistance to noise‐induced hearing loss.
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43.64.Qh Electrophysiology of the auditory central nervous system
43.64.Ri Evoked responses to sounds
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

The envelope following response (EFR) in the Mongolian gerbil to sinusoidally amplitude‐modulated signals in the presence of simultaneously gated pure tones

William Ford Dolphin and David C. Mountain

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3215-3226 (1993); (12 pages) | Cited 4 times

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The envelope following response (EFR) is an auditory‐evoked potential recorded from the scalp which is elicited by long duration, amplitude‐modulated stimuli. In this paper, the results of a series of experiments exploring the behavior of the EFR elicited with sinusoidally amplitude modulated (SAM) tones in the presence of simultaneously gated, continuous, pure‐tone interfering signals of varying intensity are reported. Probe stimuli consisted of SAM tones with carriers ranging in frequency from 800 Hz–4 kHz, modulated at frequencies between 30–150 Hz. Probe signals were presented at intensities between 50 and 75 dB pSPL. Pure‐tone interfering signals consisted of frequencies between 100 Hz and 10 kHz and ranged in intensity from −10 to +20 dB re: the probe. In these experiments a maximum reduction in the response to the probe tone, measured at the probe modulation frequency, appeared as a sharp peak within a narrow frequency band above the frequency of the probe carrier and a broader region of reduced response extending to higher frequencies. This reduction in response was asymmetrical, spreading more to high than to low frequencies. With an increase in the intensity of the interfering signal the maximum reduction of the response increased in a saturating, monotonic fashion with a concomitant broadening of the frequency region affected. The obtained interference response pattern may be attributable to both ‘‘synchrony capture’’ (i.e., capture of the EFR of the system by envelope components arising due to the interaction of probe and interfering signals) and ‘‘synchrony suppression’’ (i.e., a reduction in the synchronized response from neurons excited by the probe in the presence of the added interfering tone). It appears that the EFR to SAM stimuli of low to moderate intensity arose primarily from neuronal populations tuned to frequencies at or above the probe fc. The results of the present study suggest that at low intensity levels SAM signals are indeed relatively frequency specific and warrant further study for audiometric applications.
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43.64.Qh Electrophysiology of the auditory central nervous system
43.64.Ri Evoked responses to sounds

Discharge suppression in the silent interval preceding the tone burst in pause‐build units of the dorsal cochlear nucleus of the unanesthetized decerebrate cat

K. Parham and D. O. Kim

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3227-3231 (1993); (5 pages)

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A recent intracellular study of dorsal cochlear nucleus (DCN) neurons in vitro by Manis [P. B. Manis, J. Neurosci. 10, 2338–2351 (1990)] suggests that the expression of the pause‐build discharge pattern is in large part dependent on hyperpolarization of their membrane potentials in a period just preceding a depolarizing stimulus (‘‘hyperpolarization conditioning’’ hypothesis). Our examination of the activity of a sample of pause‐build units (n=72) revealed suppression of discharge activity during a time window of the silent interstimulus interval (SII) just preceding the tone burst relative to the spontaneous rate in 74% of all units. The discharge suppression of a subset of DCN pause‐build units in the SII satisfies a requirement of the ‘‘hyperpolarization conditioning’’ hypothesis, and thus supports the hypothesis.
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43.64.Qh Electrophysiology of the auditory central nervous system

Measures of auditory brain‐stem responses, distortion product otoacoustic emissions, hair cell loss, and forward masked tuning curves in the waltzing guinea pig

Barbara Canlon, Kasper Marklund, and Erik Borg

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3232-3243 (1993); (12 pages)

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Measures of the auditory brain‐stem response (ABR), distortion product otoacoustic emission (2f1−f2), hair cell loss, and forward masked tuning curves were obtained from waltzing guinea pigs and their age‐matched controls at postnatal day 2, 9, 15, and 30. A mild ABR threshold shift (10–15 dB) is seen by 2 days postnatal and gradually increases to a more severe threshold shift (40–50 dB) by postnatal day 15. Already by 30 days the auditory brain‐stem response, being beyond the output of the instrumentation, could not be elicited. The mean distortion product otoacoustic emission (DPOE) amplitude as a function of f1 amplitude for the postnatal day 2, 9, and 15 waltzing guinea pigs were only between 3 and 8 dB below the control values for stimulus levels below 65 dB SPL. The DPOE audiogram constructed for the waltzing guinea pigs shows no more than an 8‐dB mean difference from the control values when the intensity of f1 was 50 dB SPL, and no more than 10 dB when f1 was 60 dB SPL. Analysis of the individual cases revealed that the DPOE amplitude could be greater than control values. On the contrary, when f1 stimulus levels were below 65 dB SPL, DPOEs could not be detected for the postnatal day 30 waltzing guinea pigs. At stimulus levels above 65 dB SPL, DPOEs could be recorded yet these responses were depressed from control values by 10 to 25 dB. Analysis of surface preparations of the organ of Corti from the day 15 waltzing guinea pig reveals that the prominent alteration occurs on the third row outer hair cells.
To a lesser extent, the second row outer hair cells, and then the first row outer hair cells are affected while the inner hair cells appear normal. In contrast, the organ of Corti from the postnatal day 30 waltzing guinea pig showed a more extensive outer hair cell loss among all three rows as well as a considerable degree of inner hair cell loss. Forward masked auditory brain‐stem response tuning curves were generated at 8, 4, and 1 kHz for control and waltzing guinea pigs between 2 and 12 days of age. All tuning curves obtained from waltzing guinea pigs showed progressive decreases in sensitivity with increasing age. The Q 10 dB values of the 1‐ and 4‐kHz tuning curves were not different between the controls and the waltzing guinea pigs at any age. However, compared to the controls, the Q 10‐dB values of the 8‐kHz tuning curves were reduced by 50% for the waltzing guinea pigs.
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43.64.Wn Effects of noise and trauma on the auditory system
43.64.Ri Evoked responses to sounds
43.64.Jb Otoacoustic emissions
43.64.Pg Electrophysiology of the auditory nerve

Doppler effect for sound emitted by a moving airborne source and received by acoustic sensors located above and below the sea surface

Brian G. Ferguson

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3244-3247 (1993); (4 pages) | Cited 3 times

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The acoustic emissions from a propeller‐driven aircraft are received by a microphone mounted just above ground level and then by a hydrophone located below the sea surface. The dominant feature in the output spectrum of each acoustic sensor is the spectral line corresponding to the propeller blade rate. A frequency estimation technique is applied to the acoustic data from each sensor so that the Doppler shift in the blade rate can be observed at short time intervals during the aircraft’s transit overhead. For each acoustic sensor, the observed variation with time of the Doppler‐shifted blade rate is compared with the variation predicted by a simple ray‐theory model that assumes the atmosphere and the sea are distinct isospeed sound propagation media separated by a plane boundary. The results of the comparison are shown for an aircraft flying with a speed of about 250 kn at altitudes of 500, 700, and 1000 ft.
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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
43.30.Es Velocity, attenuation, refraction, and diffraction in water, Doppler effect

Adaptive feedback cancellation with frequency compression for hearing aids

Harry Alfonso L. Joson, Futoshi Asano, Yōiti Suzuki, and Toshio Sone

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3248-3254 (1993); (7 pages) | Cited 4 times

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The use of an adaptive feedback canceler (AFC) for howling suppression in hearing aids seems very attractive since it is not only unaffected by the changes in the operating environment, but it also limits signal degradation due to the feedback signal. This, however, requires a reference signal which is correlated with the feedback signal but not with the input signal. In hearing aids, such a signal is hard to obtain. The output signal could be used as reference if its correlation with the input signal could sufficiently be removed. If the reference signal is correlated with the input signal, the input signal will also be canceled by the AFC. Here, the use of a frequency compressor as a decorrelator is proposed. The performance of this system is then investigated via digital simulation. Results indicated that with the use of the proposed system and the proper choice of system parameters, an increase of about 18 dB in the howling margin could be achieved with minimal deterioration in output signal quality.
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43.60.Lq Acoustic imaging, displays, pattern recognition, feature extraction
43.38.Lc Amplifiers, attenuators, and audio controls
43.66.Ts Auditory prostheses, hearing aids

Binaural simulation of concert halls: A new approach for the binaural reverberation process

J. Martin, D. Van Maercke, and J‐P. Vian

J. Acoust. Soc. Am. Volume 94, Issue 6, pp. 3255-3264 (1993); (10 pages)

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A geometrical cone‐tracing method associated with the signal processing technique is used to calculate the binaural impulse response of a concert hall. Some inaccuracy and the computation time of the geometrical algorithm tend to limit the method for the high‐reflection orders which are necessary to provide a good listening effect. In order to extend the response, a new approach is presented based on different statistical processes that depend on both the acoustical and geometrical characteristics of the hall. After a theoretical presentation (new statistical results are proposed to describe the sound field behavior in a concert hall), some simulations are given in order to illustrate the different statistical processes. This simulation technique seems to be a very convenient tool both for the design of a new concert hall and for the study of the important parameters in auditory spaciousness.
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43.55.Fw Auditorium and enclosure design
43.55.Hy Subjective effects in room acoustics, speech in rooms
43.55.Lb Electrical simulation of reverberation
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