Some remarks on Pastore (1988)

Pastore [J. Acoust. Soc. Am. 84, 2262-2266 (1988)] has written a lengthy response to Kewley-Port, Watson, and Foyle [J. Acoust. Soc. Am. 83, 1133-1145 (1988)]. In this reply to Pastore's letter, several of his arguments are addressed, and new data are reported which support the conclusion of the original article. That conclusion is, basically, that the temporal acuity of the auditory system does not appear to be the origin of categorical perception of speech or nonspeech sounds differing in temporal onsets.     

Pitch and rhythm paradoxes: comments on "Auditory paradox based on fractal waveform"

Schroeder [J. Acoust. Soc. Am. 79, 186-189 (1986)] describes the paradox of acoustic waveforms that sound lower when reproduced at higher speeds. The author has also demonstrated this paradox [J.C. Risset, J. Acoust. Soc. Am. 46, 88 (A) (1969); see also Seventh ICA, Budapest, S10, 613-616 (1971)]; in addition he has recently synthesized a rhythmic analog of the paradox, namely rhythmical sequences that can sound slower when reproduced at higher speeds.     

Is pitch a learned attribute of sounds? Two points in support of Terhardt's pitch theory

Terhardt [J. Acoust. Soc. Am. 55, 1061-1069 (1974)] postulated a pitch perception model wherein a learning stage constitutes an integral part: it is only repeated exposure to patterns of spectral pitch that will generate the percept of virtual pitch (i.e., the residue). Two examples, one clinical and one musical, are cited to support the idea that perception of the pitch of complex tones represents a case of pattern perception which is acquired with experience.     

Unification and extension of monolithic state space and iterative cochlear models

Time domain cochlear models have primarily followed a method introduced by Allen and Sondhi [J. Acoust. Soc. Am. 66, 123-132 (1979)]. Recently the "state space formalism" proposed by Elliott et al. [J. Acoust. Soc. Am. 122, 2759-2771 (2007)] has been used to simulate a wide range of nonlinear cochlear models. It used a one-dimensional approach that is extended to two dimensions in this paper, using the finite element method. The recently developed "state space formalism" in fact shares a close relationship to the earlier approach. Working from Diependaal et al. [J. Acoust. Soc. Am. 82, 1655-1666 (1987)] the two approaches are compared and the relationship formalized. Understanding this relationship allows models to be converted from one to the other in order to utilize each of their strengths. A second method to derive the state space matrices required for the "state space formalism" is also presented. This method offers improved numerical properties because it uses the information available about the model more effectively. Numerical results support the claims regarding fluid dimension and the underlying similarity of the two approaches. Finally, the recent advances in the state space formalism [Bertaccini and Sisto, J. Comp. Phys. 230, 2575-2587 (2011)] are discussed in terms of this relationship.     

Parabolic equation solution of seismo-acoustics problems involving variations in bathymetry and sediment thickness

Recent improvements in the parabolic equation method are combined to extend this approach to a larger class of seismo-acoustics problems. The variable rotated parabolic equation [J. Acoust. Soc. Am. 120, 3534-3538 (2006)] handles a sloping fluid-solid interface at the ocean bottom. The single-scattering solution [J. Acoust. Soc. Am. 121, 808-813 (2007)] handles range dependence within elastic sediment layers. When these methods are implemented together, the parabolic equation method can be applied to problems involving variations in bathymetry and the thickness of sediment layers. The accuracy of the approach is demonstrated by comparing with finite-element solutions. The approach is applied to a complex scenario in a realistic environment.     

Examining the use of a time-varying loudness algorithm for quantifying characteristics of nonlinearly propagated noise (L)

A previous letter by Gee et al. [J. Acoust. Soc. Am. 121, EL1-EL7 (2007)] revealed likely shortcomings in using common, stationary (long-term) spectrum-based measures to quantify the perception of nonlinearly propagated noise. Here, the Glasberg and Moore [J. Audio Eng. Soc. 50, 331-342 (2002)] algorithm for time-varying loudness is investigated. Their short-term loudness, when applied to a shock-containing broadband signal and a phase-randomized signal with equivalent long-term spectrum, does not show a significant difference in loudness between the signals. Further analysis and discussion focus on the possible utility of the instantaneous loudness and the need for additional investigation in this area.     

Multiresolution spectrotemporal analysis of complex sounds

A computational model of auditory analysis is described that is inspired by psychoacoustical and neurophysiological findings in early and central stages of the auditory system. The model provides a unified multiresolution representation of the spectral and temporal features likely critical in the perception of sound. Simplified, more specifically tailored versions of this model have already been validated by successful application in the assessment of speech intelligibility [Elhilali et al., Speech Commun. 41(2-3), 331-348 (2003); Chi et al., J. Acoust. Soc. Am. 106, 2719-2732 (1999)] and in explaining the perception of monaural phase sensitivity [R. Carlyon and S. Shamma, J. Acoust. Soc. Am. 114, 333-348 (2003)]. Here we provide a more complete mathematical formulation of the model, illustrating how complex signals are transformed through various stages of the model, and relating it to comparable existing models of auditory processing. Furthermore, we outline several reconstruction algorithms to resynthesize the sound from the model output so as to evaluate the fidelity of the representation and contribution of different features and cues to the sound percept.     

The influence of musical training on the perception of sequentially presented mistuned harmonics

The question of whether musical scales have developed from a processing advantage for frequency ratios based on small integers, i.e., ratios derived from relationships among harmonically related tones, is widely debated in musicology and music perception. In the extreme position, this processing advantage for these so-called "natural intervals" is assumed to be inherent, and to apply to sequentially presented tones. If this is the case, evidence for this processing advantage should show up in psychoacoustic experiments using listeners from the general population. This paper reports on replications and extensions of two studies from the literature. One [Lee and Green, J. Acoust. Soc. Am. 96, 716-725 (1994)] suggests that listeners from the general population can in fact determine whether sequentially presented tones are harmonically related. The other study [Houtgast, J. Acoust. Soc. Am. 60, 405-409 (1976)] is interpreted in different terms, but could be confounded by such an ability. The results of the replications and extensions, using listeners of known relative pitch proficiency, are consistent with the idea that only trained musicians can reliably determine whether sequentially presented tones are harmonically related.     

Spectral weighting strategies for sentences measured by a correlational method

Listeners' ability to understand speech in adverse listening conditions is partially due to the redundant nature of speech. Natural redundancies are often lost or altered when speech is filtered, such as done in AI/SII experiments. It is important to study how listeners recognize speech when the speech signal is unfiltered and the entire broadband spectrum is present. A correlational method [R. A. Lutfi, J. Acoust. Soc. Am. 97, 1333-1334 (1995); V. M. Richards and S. Zhu, J. Acoust. Soc. Am. 95, 423-424 (1994)] has been used to determine how listeners use spectral cues to perceive nonsense syllables when the full speech spectrum is present [K. A. Doherty and C. W. Turner, J. Acoust. Soc. Am. 100, 3769-3773 (1996); C. W. Turner et al., J. Acoust. Soc. Am. 104, 1580-1585 (1998)]. The experiments in this study measured spectral-weighting strategies for more naturally occurring speech stimuli, specifically sentences, using a correlational method for normal-hearing listeners. Results indicate that listeners placed the greatest weight on spectral information within bands 2 and 5 (562-1113 and 2807-11,000 Hz), respectively. Spectral-weighting strategies for sentences were also compared to weighting strategies for nonsense syllables measured in a previous study (C. W. Turner et al., 1998). Spectral-weighting strategies for sentences were different from those reported for nonsense syllables.     

Analyzing reverse middle-ear transmission: noninvasive Gedankenexperiments.

The phenomenological framework outlined in the companion paper [C. A. Shera and G. Zweig, J. Acoust. Soc. Am. 92, 1356-1370 (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 P. F. Fahey and J. B. Allen [J. Acoust. Soc. Am. 77, 599-612 (1985)].     

A comparison of broadband models for sand sediments

Chotiros and Isakson [J. Acoust. Soc. Am. 116(4), 2011-2022 (2004)] recently proposed an extension of the Biot-Stoll model for poroelastic sediments that makes predictions for compressional wave speed and attenuation, which are in much better accord with the experimental measurements of these quantities extant in the literature than either those of the conventional Biot-Stoll model or the rival model of Buckingham [J. Acoust. Soc. Am. 108(6), 2796-2815 (2000)]. Using a local minimizer, the Nelder-Mead simplex method, it is shown that there are generally at least two choices of the Chotiros-Isakson parameters which produce good agreement with experimental measurements. Since one postulate of the Chotiros-Isakson model is that, due to the presence of air bubbles in the pore space, the pore fluid compressibility is greater than that of water, an alternative model based on a conjecture by Biot [J. Acoust. Soc. Am. 34(5), 1254-1264 (1962)], air bubble resonance, is considered. While this model does as well or better than the Chotiros-Isakson model in predicting measured values of wave speed and attenuation, the Rayleigh-Plesset theory of bubble oscillation casts doubt on its plausibility as a general explanation of large dispersion of velocity with respect to frequency.     

The acoustic reflex threshold in aging ears

This study investigates the controversy regarding the influence of age on the acoustic reflex threshold for broadband noise, 500-, 1000-, 2000-, and 4000-Hz activators between Jerger et al. [Mono. Contemp. Audiol. 1 (1978)] and Jerger [J. Acoust. Soc. Am. 66 (1979)] on the one hand and Silman [J. Acoust. Soc. Am. 66 (1979)] and others on the other. The acoustic reflex thresholds for broadband noise, 500-, 1000-, 2000-, and 4000-Hz activators were evaluated under two measurement conditions. Seventy-two normal-hearing ears were drawn from 72 subjects ranging in age from 20-69 years. The results revealed that age was correlated with the acoustic reflex threshold for BBN activator but not for any of the tonal activators; the correlation was stronger under the 1-dB than under the 5-dB measurement condition. Also, the mean acoustic reflex thresholds for broadband noise activator were essentially similar to those reported by Jerger et al. (1978) but differed from those obtained in this study under the 1-dB measurement condition.     

Acoustic radiation torque and the conservation of angular momentum (L)

This note concerns the evaluation of the static acoustic radiation torque exerted by an acoustic field on a scatterer immersed in a nonviscous fluid based on far-field scattering. The radiation torque is expressed as the integral of the time-averaged flux of angular momentum over a spherical surface far removed from the scattering object with its center at the centroid of the object. That result was given previously [G. Maidanik, J. Acoust. Soc. Am. 30, 620-623 (1956)]. Another expression given recently [Z. W. Fan et al., J. Acoust. Soc. Am. 124, 2727-2732 (2008)] is simplified to this formula. Comments are made on obtaining it directly from the general theorem of angular momentum conservation in the integral form.     

On boundary conditions for the diffusion equation in room-acoustic prediction: Theory, simulations, and experiments

This paper proposes a modified boundary condition to improve the room-acoustic prediction accuracy of a diffusion equation model. Previous boundary conditions for the diffusion equation model have certain limitations which restrict its application to a certain number of room types. The boundary condition employing the Sabine absorption coefficient [V. Valeau et al., J. Acoust. Soc. Am. 119, 1504-1513 (2006)] cannot predict the sound field well when the absorption coefficient is high, while the boundary condition employing the Eyring absorption coefficient [Y. Jing and N. Xiang, J. Acoust. Soc. Am. 121, 3284-3287 (2007); A. Billon et al., Appl. Acoust. 69, (2008)] has a singularity whenever any surface material has an absorption coefficient of 1.0. The modified boundary condition is derived based on an analogy between sound propagation and light propagation. Simulated and experimental data are compared to verify the modified boundary condition in terms of room-acoustic parameter prediction. The results of this comparison suggest that the modified boundary condition is valid for a range of absorption coefficient values and successfully eliminates the singularity problem.     

On the perception of voicing in syllable-initial plosives in noise

Previous studies [Lisker, J. Acoust. Soc. Am. 57, 1547-1551 (1975); Summerfield and Haggard, J. Acoust. Soc. Am. 62, 435-448 (1977)] have shown that voice onset time (VOT) and the onset frequency of the first formant are important perceptual cues of voicing in syllable-initial plosives. Most prior work, however, has focused on speech perception in quiet environments. The present study seeks to determine which cues are important for the perception of voicing in syllable-initial plosives in the presence of noise. Perceptual experiments were conducted using stimuli consisting of naturally spoken consonant-vowel syllables by four talkers in various levels of additive white Gaussian noise. Plosives sharing the same place of articulation and vowel context (e.g., /pa,ba/) were presented to subjects in two alternate forced choice identification tasks, and a threshold signal-to-noise-ratio (SNR) value (corresponding to the 79% correct classification score) was estimated for each voiced/voiceless pair. The threshold SNR values were then correlated with several acoustic measurements of the speech tokens. Results indicate that the onset frequency of the first formant is critical in perceiving voicing in syllable-initial plosives in additive white Gaussian noise, while the VOT duration is not.     

The normalized interaural correlation: accounting for NoS pi thresholds obtained with Gaussian and "low-noise" masking noise.

Recently, Eddins and Barber [J. Acoust. Soc. Am. 103, 2578-2589 (1998)] and Hall et al. [J. Acoust. Soc. Am. 103, 2573-2577 (1998)] independently reported that greater masking of interaurally phase-reversed (S pi) tones was produced by diotic low-noise noise than by diotic Gaussian noise. Based on quantitative analyses, Eddins and Barber suggested that their results could not be accounted for by assuming that listeners' judgments were based on constant-criterion changes in the normalized interaural correlation produced by adding the S pi signal to the diotic masker. In particular, they showed that a model like the one previously employed by Bernstein and Trahiotis [J. Acoust. Soc. Am. 100, 3774-3784 (1996)] predicted an ordering of thresholds between the conditions of interest that was opposite to that observed. Bernstein and Trahiotis computed the normalized interaural correlation subsequent to half-wave, square-law rectification and low-pass filtering, the parameters of which were chosen to mimic peripheral auditory processing. In this report, it is demonstrated that augmenting the model by adding a physiologically valid stage of "envelope compression" prior to rectification and low-pass filtering provides a remedy. The new model not only accounts for the data obtained by Eddins and Barber (and the similar data obtained by Hall et al.), but also does not diminish the highly successful account of the comprehensive set of data that gave rise to the original form of the model. Therefore, models based on the computation of the normalized interaural correlation appear to remain valid because they can account, both quantitatively and qualitatively, for a wide variety of binaural detection and discrimination data.     

Effects of perilymph viscosity on low-frequency intracochlear pressures and the cochlear input impedance of the cat

Cochlear model calculations are shown to be in reasonable agreement with recent low-frequency measurements of intracochlear pressures and the cochlear input impedance of the cat [V. Nedzelnitsky, J. Acoust. Soc. Am. 68, 1676-1689 (1980); T. J. Lynch, III, V. Nedzelnitsky, and W. T. Peake, J. Acoust. Soc. Am. 72, 108-130 (1982)]. Included in the cochlear model are perilymph viscosity, the measured variation of the area of the scala vestibuli with distance from the stapes [P. Dallos, J. Acoust. Soc. Am. 48, 489-499 (1970)], and finite impedance of the round window membrane. The WKB approximation and its extension to the low-frequency region is used in order to exhibit explicitly the dependence of the model results on the cochlear parameters.     

Models of auditory masking: a molecular psychophysical approach

Gilkey et al. [J. Acoust. Soc. Am. 78, 1207-1219 (1985)] measured hit proportions and false alarm proportions for detecting a 500-Hz tone at each of four starting phase angles in each of 25 reproducible noise samples. In the present paper, their results are modeled by fitting the general form of the electrical analog model of Jeffress [J. Acoust. Soc. Am. 48, 480-488 (1967)] to the diotic data. The best-fitting configurations of this model do not correspond to energy detectors or to envelope detectors. A detector composed of a 50-Hz-wide single-tuned filter, followed by a half-wave rectifier and an integrator with an integration time of 100 to 200 ms fits the data of all four subjects relatively well. Linear combinations of the outputs of several detectors that differ in center frequency or integration window provide even better fits to the data. These linear combinations assign negative weights to some frequencies or to some time intervals, suggesting that a subject's decision is based on a comparison of information in different spectral or temporal portions of the stimulus.