bluuebunny/crossref_metadata_2025 huggingface dataset:
Relation of mean loudness matches to intensity JND's
Authors: Lim J. S. , Rabinowitz W. M. , Braida L. D. , Durlach N. I.
Published: October 2005

The preliminary theory of intensity resolution [N. I. Durlach and L. D. Braida, J. Acoust. Soc. Am. 46, 372–383 (1969)] is extended to encompass mean loudness matches between different types of stimuli. According to this extension, two stimuli S1 and S2 are matched in loudness when their intensities I1 and I2 divide the dynamic ranges proportionately in terms of number of JND's. This theoretical relation between loudness matching and resolution (originally proposed by Riesz) is compared to experimental data on equal-loudness contours for tones, loudness matching of a tone in silence versus the same tone in noise, and loudness matching of a tone presented to a normal ear versus the same tone presented to a recruiting ear. [Research supported by NIH.]
Resolution and bias-edge effects in the perception of intensity
Authors: Berliner J. E. , Braida L. D. , Durlach N. I.
Published: October 2005

There is a tendency in both one-interval identification and two-interval roving-level discrimination for intensity resolution to be better near the extremes of the intensity range than in the middle (the resolution edge effect). Also, there is a tendency in the latter paradigm for the signal in the second interval to sound louder than the signal in the first interval near the top of the range and to sound softer near the bottom of the range (the bias edge effect). This paper summarizes a variety of experimental results concerning the dependence of these effects on the parameters of the paradigm. In identification, the resolution edge effect increases with the intensity range. In roving-level discrimination, the resolution edge effect increases with range and with interstimulus interval, and the bias edge effect increases with range for small ranges, decreases with range for large ranges, and increases with interstimulus interval. [Research supported by NIH.]
Monaural beat-tone existence region
Authors: Soderquist D. R. , Hall J. W.
Published: October 2005

An experiment was performed which yielded the upper and lower thresholds for beats. A beat-tone existence region, the difference in decibels between the lower and upper beat-tone thresholds, was determined for several frequencies. A primary tone was presented at various sensation levels (−15 to 65 dB SL) and then a beat-tone was added 4 Hz below the primary. Subjects adjusted the beat-tone intensity to obtain the upper and lower beat thresholds. Beats were shown to occur when the intensity of the beat-tone and primary-tone differed by as much as 30 dB. The existence region initially increased as a function of the primary-tone intensity (between −15 and 20 dB SL) and then remained fairly constant (about 45 dB difference) for values of the primary between 20 and 65 dB. The data suggest that experiments which depend on the monaural perception of beats to indicate when a combination tone has been cancelled (i.e., the probe-tone, cancellation-tone method of measuring the intensity of CT) may error in their estimation of the CT intensity if the CT is actually above about 35 dB or below threshold (but still physically present). When the CT is assumed to be between 15 and 35 dB SL the probe-cancellation-tone method of estimating the intensity of a CT was judged as adequate.
Discrimination of spectral shape by blinded veterans
Authors: De l'Aune W. R. , Bauer S. L. , Bathrick P. H. , Dolan M. T. , Finn W. A. , Grimmelsman T. A. , Healey J. D. , Kevorkian G. A. , Milbier D. I. , Webber M. C.
Published: October 2005

The analysis of the acoustic environment for the purpose of gleaning spatial information is a very important sensory skill potentially available to an individual lacking sight. The purpose of the experimental procedure outlined in this paper was twofold: (1) to provide the means for predicting the level of auditory skills of individual clients and (2) to enhance the auditory skills of blinded clients in a manner more efficient than techniques presently available. The task involved the detection of a notch filtered out of a broad-band noise background. A standard (low pass-900 Hz, high pass-1300 Hz) and three different notch widths (850–1450, 800–1600, 750–1750) filtered by means of a General Radio 1952 Universal filter were used, providing the experimenters with three levels of difficulty in the comparison task. It was found that the test results correlated with performance in real-world acoustic assessment situations to a much greater degree than previously possible with pure tone audiometric results. Significant learning effects generalized to the real world situation will also be discussed.
Use of laryngeal measurements as control parameters in a dynamic speech synthesizer
Authors: Lofgren K. M. J.
Published: October 2005

Six simultaneous physiological measurements were recorded during speech: subglottal pressure; transillumination of the vocal-cord opening; electromyographic activity in the crico-thyroid, posterior crico-arytenoid, and inter-arytenoid muscles; and the sound pressure speech waveform. To first order, the crico-thyroid muscle is related to voice pitch, posterior crico-arytenoid to vocal-cord abduction, and inter-arytenoid to cord adduction. A dynamic vocal-cord/vocal-tract synthesizer produces speech from control parameters which are functions of time and which represent subglottal pressure, vocal-cord separation at rest, vocal-cord tension, and the cross-sectional area of the vocal tract at 20 equispaced points along its length. The area functions were obtained from a program which generates parameters for the speech synthesizer from printed text. These functions were modified in time scale to conform to that of the laryngeal parameters. The six physiological measurements were digitally processed and substituted for the laryngeal control parameters in the synthesizer. As a result, speech syntheses have been tape-recorded using crico-thyroid activity for pitch control and posterior crico-arytenoid and inter-arytenoid activity for control of glottal areas at rest.
Linguapalatal contact characteristics during /s/ production
Authors: Wolf M. B. , McCutcheon M. J. , Hasegawa A. , Fletcher S. G.
Published: October 2005

Measurements of linguapalatal contact and sound spectographic data were obtained in five male subjects during production of CV, VC, and symmetric VCV utterances where the consonant was /s/ and the vowels /i, ɑ, u/. Tongue-palate contact was determined at 10-msec intervals using the PAGIS instrumental system [S. G. Fletcher, Final Report, NINDS Research Grant ♯1 R01 NS10540-01, Department of Biocommunication, University of Alabama at Birmingham, Birmingham, AL 35294]. In this study 48 electrodes were imbedded in the subjects pseudopalate spaced at 2-mm and concentrated in the alveolar-postalveolar region. The minimum width of the central groove (CG) at the alveolar region was selected as the parameter to characterize the linguapalatal contact pattern. The data for all subjects showed a stable CG for 100–150 msec corresponding to the time of /s/ frication sound generation. Preliminary analysis indicated that the CG was (1) essentially independent of vowel context and (2) negatively correlated with speaking rate. These data indicate that a critical feature of /s/ production is constancy of the minimum CB at a given speaking rate, i.e., this parameter is apparently resistant to coarticulatory influences. The data also suggest that the tongue tip may overshoot its CB target as speaking rate increases. [Supported by NIH Grant NS-11852-01A from NINCDS.]
Oscillatory movements of the velum: A factor of intraoral air pressure
Authors: Williams W. N. , Brown W. S. , Hildebrand B.
Published: October 2005

It is known that velar elevation in closure for speech correlates with specific sound productions. In addition, the velum has been observed to move up and down in an oscillatory manner during repeated syllable or word utterances. The purpose of this study was to determine whether this observed oscillatory movement of the soft palate is related to the build up and release of intraoral air pressure and/or to the opening and closing patterns of the mandible. Simultaneous cinefluorographic films and intraoral-air-pressure recordings were taken for three subjects during the repeated production of /ɑpɑ/, /ɑbɑ/, /ɑsɑ/ and /ɑzɑ/. From the cinefluorographic films, variations in velar and mandibular height were determined and correlated with variations in intraoral air pressure. Initial analyses indicate that the oscillatory motions of the velum associated with consonant productions are highly correlated with variations in intra-oral air pressure. These results will be discussed in terms of a theory/model of articulatory behavior for oral consonant production.
Consonant response analyses of speech with simulated single reflection for normal and hearing-impaired subjects
Authors: Slawinska Elizabeth , Nabelek Anna K.
Published: October 2005

Intelligibility scores of speech (Modified Rhyme Test) with simulated single reflections, reported at the 90th ASA Meeting [J. Acoust. Soc. Am. 58, S114(A) (1975)] were analyzed for consonant errors. The perception of the initial consonants was about 6% better than for the final consonants for binaural and monaural listening by normal-hearing subjects and for binaural listening by hearing-impaired subjects. There was no initial position advantage for monaural listening by impaired subjects. For all listening conditions for both groups of subjects, the voicing feature was perceived better than the manner feature, which in turn was perceived better than the place feature. The place feature had an advantage in the initial position, and the manner feature had an advantage in the final position for binaural and monaural listening for both groups. The voicing feature had a small advantage for the initial position for all conditions. The prolongation of the delay between the primary and delayed signal had the greatest influence on the place errors in the initial position for both groups of subjects. [Supported by NIH NISND grant.]
Sound radiation from an accelerating sphere
Authors: Akay A. , Hodgson T. H.
Published: October 2005

The sound radiated by an accelerating sphere has been studied. A careful critique of the existing literature has led to a more complete theoretical analysis of the problem. The acoustic field of an accelerating sphere has also been calculated for different fluid media. The radiated sound pressure from an impulsively accelerated sphere has been compared with a finite acceleration case to emphasize the effect of rate of change of velocity on the pressure waveform. The acoustic energy calculations have been made in both the time and frequency domains in order to further demonstrate that the energy lost in the acceleration is dissipated as sound.
Depth scaling of shallow explosive source levels
Authors: Gaspin Joel B.
Published: October 2005

A new technique for estimating spectral energy levels, for burst depths at which data is not available, has been developed. By use of this technique, the digital pressure-time history from an explosion at a particular depth may be modified so as to yield, after transformation, an estimate of the spectral energy levels at another burst depth. The method involves modification of the negative phases of the pressure-time history, so as to model the bubble-pulse periods, without unduly distorting the positive pulse shapes. The method is shown to be internally consistent to better than 1 dB for depth changes of up to an order of magnitude for small charges. The technique can be applied to digital waveforms with a minimal input of time and effort. A previously used practice, scaling the time series data by a linear scale factor depending on the 5/6 power of the hydrostatic pressure, is shown to be inaccurate, for even small depth changes. [Work supported by ONR.]
A Helmholtz-Kirchhoff theorem for spatially extended radiators in uniform rectilinear motion
Authors: Rudgers Anthony J.
Published: October 2005

This paper discusses an integral representation of the radiated field of an acoustic source region that moves at uniform subsonic speed along a straight trajectory. The source region considered must be of finite size. It may radiate an arbitrarily complex, noise-like signal so long as this signal is of finite duration, however long. Individual sources within the source region need not undergo rigid body translations so long as they can be enclosed within a uniformly translating surface whose shape remains constant. The acoustic field of such a source region is synthesized from the radiated fields of an ensemble of moving point sources by using the principle of superposition in both space and frequency. The integral representation of the moving source's radiated field is analogous to the familiar Helmholtz-Kirchoff theorem, or Helmholtz-integral formulation, that expresses the radiated field of a nonmoving spatially extended acoustic source. In the representation for the field of the moving source, however, the freespace Green's function is replaced by a moving-point-source solution to the inhomogeneous wave equation.
Expansion of a dense gas bubble underwater
Authors: Epstein D.
Published: October 2005

In a recent paper [J. Acoust. Soc. Am. 57, 1427–1435 (1975)] the effect of a dense-gas equation of state on underwater gas-bubble expansion was investigated. A high-pressure region in the immediate vicinity of the bubble minimum radius, where an asymptotic form of the Kirkwood-Bethe differential equation is assumed to hold, is defined. A power series expansion in at/c of this equation is developed. The method of Esipov and Naughol'nykh [Sov. Phys. Acoust. 18, 194–197 (1972)] reduces the problem of integration of this power series to a single quadrature. This procedure has been applied to the dense-gas-bubble case. Because of analytic complexity, only the lowest order term is usually retained. For certain applications, however, more terms may be required. These points and others are critically examined. Also, some numerical errors in Esipov and Naughol'nykh have been corrected.
Purification of cochlear carbonic anhydrase
Authors: Drescher D. G.
Published: October 2005

Carbonic anhydrases appear to function in the excretion of CO2 and secretion of K*. To characterize cochlear carbonic anhydrase at the molecular level, I have purified this enzyme from membranous lateral wall (stria vascularis, spiral ligament, spiral prominence, and part of the outer sulcus) of the inner ear of guinea pig. To my knowledge, this represents the first purification of a cochlear enzyme. Membranous lateral walls from 60 cochleae were dissected at 2°C in 100 mM Na2SO4 + 10 mM TRIS sulfate pH 7.4, homogenized, centrifuged, and the supernatant was applied to a column of Bio-Rad P-100 resin. Since the cochlear enzyme appears to be inhibited by chloride, sulfate was used throughout. The initial gel filtration showed that the enzyme had a molecular weight of about 30 000 and that at least 98% was not associated with larger macro-molecules. Absorbance at 410 nm indicated that contamination by blood accounted for less than 4% of the enzyme activity. Further purification by DEAE agarose chromatography and gel filtration with Bio-Rad P-60 resin resulted in one active peak, suggesting one major charged species. This study indicated that 2% (2–3 μg/dissected cochlea) of the protein of the membranous lateral wall is carbonic anhydrase. [Supported by Intramural Research Program, NIH]
Energy balances in dissipative flow regions
Authors: Peter Abraham C.
Published: October 2005

The formation of dissipative mechanical fluctuations is due to the utilization of the frictional heat, generated by dissipative motion of the fluid, for the performance of mechanical work. Such dissipative energy conversions take place by means of cyclic thermodynamic work processes. [A. C. Peter, J. Acoust. Soc. Am. 58, S36(A), S108(A) (1975)]. These fluctuations consist of two distinct kinds: namely, those whose physical nature is purely “hydrodynamic” and those transmitted by the fluid in the form of acoustic disturbances. The current investigation highlights the energy balances of hydrodynamic fluctuations for the case of perfect gases. It is shown that for a given amount of heat input into the fluid due to the dissipation of its kinetic energy, the following energy balances apply when the specific heat ratio γ = 1.4: 89.3% of the dissipative heat input is used to build up the internal energy of the fluid, the remaining 10.7% being utilized in the performance of expansion work upon the surroundings. The succeeding collapse of the disturbance uses 17.9% of the frictional heat input for the performance of isentropic expansion work. The isentropic collapse is followed by a succession of fluctuations around the ambient pressure of the surroundings whose average work contribution may be considered isobaric. These fluctuations, which return the fluid to its initial equilibrium state, consist of a total work output utilizing 28.6% of the frictional heat (this work is implented on the fluid by the surroundings). In the same process the internal energy of the fluid loses 71.4% of the original frictional heat to the surroundings, thus completing the dissipative work cycle.
Absorption of sound in moist air
Authors: Bass H. E. , Shields F. D. , Bolen L. N.
Published: October 2005

Absorption of sound in air has been measured for temperatures from 0° to 100°F, relative humidities from 0% to 100%, at frequencies from 4 to 100 kHz. Measurements were made by propagating pulses of plane waves down a large cylindrical tube. The results have been compared to the value predicted using a new standard for sound absorption in still air proposed by the S1-57 Working Group. [Work supported by NASA-Lewis Research Center.]
Spectral broadening of sound by strong turbulence
Authors: Clifford S. F. , Brown E. H.
Published: October 2005

A previous study of the authors' found that the spectral broadening of sound propagating through weak turbulence varied as the ratio w/L0 of the speed of the transverse wind w to the outer scale of turbulence L0. However, the condition of “weak” turbulence restricts the usefulness of the result to relatively calm atmospheric conditions or short acoustical paths. The present work extends the analysis to spectral broadening of sound by arbitrarily “strong” turbulence. In contrast, the spectral broadening in strong turbulence varies as the ratio w/ρ0, with ρ0 the transverse coherence length of the sound waves. Here we adapt recent optical theory to sound propagation and to the spectral broadening of sound.
Speaker verification—an operational example
Authors: Hydrick Barbara M. , Doddington George R.
Published: October 2005

An entry control system using speaker verification is now operational at Texas Instruments. This system has performed over 200 000 verifications during the past year and a half. Verification is based upon a 4-word utterance randomly constructed from a set of 16 reference words, each represented by filter-bank spectra. Voice prompting helps to stabilize the user's pronunciation. A sequential decision strategy provides good verification performance with an average of 1.3 utterances. Average verification time is less than 6 sec. Impostor acceptance is established at 1%. User rejection (0.2%) usually indicates anomalous user data. Speech reference data is adaptively updated at each successful entry attempt. Ten thousand bits of reference data storage are required for each user. Thorough user orientation is required to ensure successful enrollment. Enrollment lasts less than 2 min. [Development sponsored in part by the Air Force Systems Command, Rome Air Development Center and Electronics Systems Division.]
Statistical modeling and automatic parameter estimation in speech recognition
Authors: Bahl L. R. , Baker J. K. , Jelinek F. , Mercer R. L.
Published: October 2005

Statistical models for characterizing various aspects of speech production and processing are discussed. Such models are being used effectively in speech recognition systems. The basic philosophy and rationale behind the use of such models is presented. One major advantage of these models is that they can be trained automatically by the use of a parameter estimation algorithm. This makes training the system for new speakers a fairly simple task. This training algorithm and other algorithms that can be used in conjunction with statistical models are discussed. In particular, results on the effect of training the phonological variation and acoustic processor models used in a speech recognition system developed at the IBM Research Center are presented. Some ideas relating to statistical modeling of natural language are also discussed.
Recursive approximations to speech signals
Authors: Bauer W. L.
Published: October 2005

An algorithm is presented which, given a sequence, x = x1, ⋯, xT, and a degree, n, produces a nearly optimal approximation to x, among those sequences, y = y1, ⋯, yT, which satisfy linear recursions of degree n. The recursion associated with the best approximation is not, in general, the optimum predictor in the sense of linear predictive coding. The advantages of this algorithm over linear prediction are: more accurate representation of the signal, hence of formant frequencies and bandwidths, and the derivation of zeros as well as poles. The disadvantages of the algorithm are greater computational work (i.e., several linear predictive analyses must be done), and the necessity of doing pitch synchronous analysis.
Harpy, a connected speech recognition system
Authors: Lowerre Bruce P. , Reddy B. Raj
Published: October 2005

The Harpy connected speech recognition system is the result of our attempt to understand the relative importance of various design choices of two earlier speech recognition systems developed at Carnegie-Mellon University, The Hearsay-I system (D. R. Reddy, et al., IEEE Trans. AU, 229–238 (1973); L. D. Erman, Technical Report, Computer Science Department, Carnegie-Mellon University, (1974) and the Dragon system [J. K. Baker, Ph.D. Thesis (Carnegie-Mellon University)]. Knowledge is represented as procedures in Hearsay-I and as a Markov network with a-priori transition probabilities between states in Dragon. Hearsay-I uses a best first search while Dragon searches all the possible (acoustic syntactic) paths through the network to determine the optimal path. Hearsay-I uses segmentation and labeling and Dragon is a segmentation-free system. Systematic performance analysis of various design choices resulted in the Harpy system which represents knowledge as a finite state transition network but without the a-priori transition probabilities, searches only a few “best” paths, and uses segmentation to reduce the number of state probability updates that must be done. The system achieves between 70% and 100% sentence accuracy, depending upon the task, and runs between 1.5 and 10 times real time. Complete details of design, implementation, and experimental results are given in B. P. Lowerre, Ph.D. Thesis (Computer Science Department, Carnegie-Mellon University, 1976).
Use of an impedance tube as a standard sound-power source
Authors: Proctor T. M. , Cook R. K.
Published: October 2005

In 1966 William A. Jack proposed the use of an Impedance tube as a standard sound-power source for measurements in reverberation rooms. He proposed the use of the pressure standing-wave ratio to determine power radiated from the tube's open end. We measured the time-averaged intensity (Umov) vector in the tube by multiplication of sound pressure and particle velocity. Two small electret microphones separated by a known distance and a small analog computer form the basis for this measurement. The pure-tone results for the limited range of the tube (300–1000 Hz) are compared with sound power measurements done in the free field. Such comparisons are made with the aid of the theoretical radiation pattern of an open tube as determined by Levine and Schwinger. Edge effects at the mouth of the tube were examined. The impedance-tube source is also compared to the results of sound-power measurements made in the reverberation chamber.
Sound-intensity patterns for vibrating plates
Authors: Brito J. Daniel , Lyon R. H. , Manning Jerome E.
Published: October 2005

The spatial patterns of sound intensity from a baffled, simply supported plate are determined theoretically and experimentally. The first is obtained by a modification of Rayleigh formula and the second by scanning with an accelerometer-microphone pair and digitally processing the signals. Contours of equal intensity are developed from computer simulation, that show zones of energy radiation and zones of energy absorption. The space-average radiation efficiency is found to agree with previous results obtained from farfield equations and with experimentally determined values obtained from reverberant field measurements, for single modes and for 1/1- and 1/3-octave band excitation.
Bubble resonance in a water-cooled reactor
Authors: Burton T. E. , Whittemore W. L.
Published: October 2005

General Atomic's TRIGA research reactor is essentially an array of cylindrical fuel rods of conventional geometry at the bottom of a large pool of water, cooled by natural convection and operated at power levels up to 2.0 MW. There can be some local boiling at higher reactor powers. Of many TRIGA reactors of similar design now in operation, only one makes an objectionable noise. Frequency spectral analysis of microphone data suggests that the source of the noise is volume oscillation of gas bubbles near the core, resonating with a standing acoustic wave in the pool. The bubble-oscillation natural frequency varies inversely with aggregate-bubble volume. Resonance has been almost eliminated by improving the coolant flow and reducing the aggregate-bubble volume. Investigations into the problem and its solution are continuing.
Schlieren imaging of unipolar acoustic pulses
Authors: Bucaro J. A. , Flax L. , Dardy H. D. , Carome E. F.
Published: October 2005

The schlieren observation of broadband acoustic pulses is considered. The optical intensity distributions in both the image and the Fourier transform plane (stop plane) are discussed. Experimental observations are presented for Gaussian-like pressure transients generated by narrow voltage pulse (∼0.1 μsec) excitation of thick piezoelectric plates. The intensity profiles obtained from these images are compared to calculations we have formulated for Gaussian pressure pulses. These calculations apply in the phase-grating limit and include optical diffraction effects due to a finite optical beam.
Experimental studies of the propagation of wide-band acoustic impulses
Authors: Carome E. F. , Layton M. R. , Bucaro J. A.
Published: October 2005

Schlieren and acoustic probe techniques are being used to study quantitatively the propagation of wide-band acoustic impulses in water. Gaussian-shaped pressure impulses (∼0.1 μsec wide) are generated using thick ceramic transducers. Acoustic probes capable of accurately representing the time profiles of the pressure impulse are being used in conjunction with fast Fourier-transform techniques to obtain information that may be correlated with the results of high-resolution schlieren measurements. These two methods complement one another and their simultaneous use yields interesting new information on acoustic diffraction, scattering, and other propagation phenomena. [Supported in part by the Office of Naval Research.]
Acoustic nearfield of a baffled piston using Fourier integral techniques
Authors: Stepanishen P. R.
Published: October 2005

A method is presented to evaluate the acoustic nearfield of a baffled piston of arbitrary shape. The method is based on first transforming the well-known Kirchhoff surface integral representation of the field into a Fourier integral. The Fourier integral is then either numerically evaluated via the use of FFT algorithms or analytically evaluated at large frequencies via the use of applicable asymptotic techniques. The relative advantages and disadvantages of both techniques are discussed. It is noted that the FFT technique is more useful in the low-to-mid-normalized frequency range; however, the asymptotic technique is most useful in the high frequency range. In order to illustrate the method, the nearfield of a circular and square piston is investigated the method. The characteristics of the fields are readily obtained and shown to be in agreement with earlier results. [This work was supported by a grant from NIH.]
Vector sound
Authors: White Richard W.
Published: October 2005

Vorticity in linearly viscous flow obeys a diffusionlike equation containing the infinite propagation speed of its parabolic form. Under incompressible conditions, where sound speed is effectively infinite, there is probably no contradiction. For compressible flow, with finite sound speed, it is suggested that the vorticity equation be modified so as to also reflect a finite propagation speed. An attempt to accomplish this leads to an assult on one assumption entrenched in the lore of acoustics, that of strictly longitudinal sonic waves. The consequences are examined and implications for aeroacoustics and turbulence theories are considered.
Vortex simulation of the pressure field for a jet
Authors: Fung Y. T. , Liu C. H.
Published: October 2005

Variations of the pressure field of a jet are simulated numerically by axisymmetric vortex rings with viscous cores submerged in an inviscid uniform stream. The results are compared with the experimental data measured near a jet. The intervals of shedding time between successive vortex rings are first matched with time intervals between successive pressure peaks of the experiments. and then generated by a numerical program to simulate the randomness of the pressure distribution in the acoustic field. This numerical program randomly generates shedding time intervals with the same probability distribution as that given by those measured in the experiments. Statistical comparisons between numerical and experimental results including probability distributions of shedding periods between successive peaks and variations of peak time period of vortex shedding along the cone of measurement are made up to six diameters downstream of the jet. The role played by the axisymmetric model in the pressure field is also discussed.
Diffraction of impulse noise by head shadow
Authors: Keim R. J.
Published: October 2005

The frequency dependent aspects of sound shadow effect on steady state noise have been well established. Little is known of this relationship for impulse noise because of inaccessibility of controlled subjects. Pre and post exposure pure tone thresholds were studied in 23 selected subjects who completed an M-16 rifle qualification course. All had identical noise exposure conditions. Significant differences between right and left ear thresholds were demonstrated and agreed with previous reports on sound shadow. [K. D. Kryter and G. Garinther, Acta Oto-Laryngol. Suppl. 211 (1965); R. J. Keim, Arch Otolaryngol. 90, 65–68 (1969).] Attenuation characteristics of the subjects exposed to impulse noise were compared with those determined in the laboratory [F. M. Weiner and D. A. Ross, J. Acoust. Soc. Am. 18, 401–408 (1946)] for steady-state noise. Statistically significant correlation was determined only when external canal resonance features were also considered in the calculations of attenuation. The results indicate that sound shadow affects impulse noise in the same way as observed for steady state noise and that the “acoustic notch” is primarily attributable to canal resonance features.
Eardrum pressure at absolute threshold: Where is the “missing 6 dB”?
Authors: Killion Mead C.
Published: October 2005

Eardrum pressures at hearing threshold have been calculated from both earphone data (ISO R389-1964 and ANSI S3.6-1969) and free-field data (ISO R226-1961). When head- and ear-diffraction effects and an apparent error in ISO R226 are accounted for in the free-field data, and real-ear versus coupler differences and physiological noise are accounted for in the earphone data, the agreement between the two derivations is excellent. At the audiometric frequencies of 125, 250. 500, 1000, 2000, 4000, and 8000 Hz. the estimated eardrum pressures at absolute threshold are 30, 19, 12, 9, 15, 13, and 17 dB SPL, respectively. At the middle frequencies—where the comparisons should be on a solid footing—no evidence of the “missing 6 dB” is seen. an observation consistent with the experimental results of several recent studies.
Binaural frequency response function
Authors: Ahumada A.
Published: October 2005

The relative sensitivity of the binaural lateralization process to neighboring frequencies was measured using 50-Hz bandwidth noise bursts. The 300-msec bursts had 16 pure tone components spaced 3.33 Hz apart and in random phase. Bursts had center frequencies of 100, 200, 300, or 400 Hz. Amplitudes of the components in decibels were varied linearly in log frequency with slopes that ranged from −36 to 60 dB octave in 12-dB steps. The noise burst presented to one ear differed from that simultaneously presented to the other ear only in that the eight higher frequency components were phase shifted corresponding to a 1 msec delay in one ear and the eight lower components were phase shifted to give a 1-msec delay in the other ear. Observers listened to a binaural burst, 300 msec of silence, and then the same stimulus sent to opposite ears. They judged the direction of movement to be right, left, or none. The lateralization judgments are consistent with an over-all frequency response which peaks between 200 and 300 Hz. Results are compared with neural modulation frequency response functions. [Work supported by Stanford—Ames Joint Institute for Aeronautics and and Acoustics.]
Anomalous lateralization in the precedence effect
Authors: Gaskell H.
Published: October 2005

When two 20-μsec click pairs are presented binaurally in rapid succession, observers hear a single sound from a location which depends mainly on the properties of the first pair (precedence or Haas effect). The effect is usually demonstrated with interaural delays but has also been obtained with interaural intensity differences. In particular, if a click pair is presented simultaneously to the two ears followed approximately 400 μsec later by a second simultaneous binaural pair, the clicks having amplitudes L1 = R1 = L2 and R2 = 0.5 × L2, one might expect to hear the resulting single sound either centrally or on the left. However, the apparent location (measured in a standard two-alternative forced-choice lateralization experiment) in these and other nontrivial conditions is clearly to the right of the midline. It is suggested that these unexpected results occur because the relative amplitude and phase spectra of the stimuli have frequency regions in which both amplitude and phase cues indicate that the lateralization is opposite to that suggested by the temporal waveform. Further experiments, (1) filtering out appropriate frequency bands and (2) making low-energy regions of the spectra, support this hypothesis.
Model for the localization of sound in the azimuthal plane
Authors: Kuhn George F.
Published: October 2005

An objective study of the interaural time difference (ITD) was performed on a manikin comprised of a head and a torso. Data were taken for both a bare and a closed torso. The measured ITD's correspond reasonably accurately at the low and the high frequencies to the computed theoretical values for a rigid sphere of an effective radius a. The theoretical ratio of the low-frequency (<500 Hz) ITD to the high frequency (>2000 Hz) ITD is 32. The measured ITD is a minimum between 1.4 and 1.6 kHz for angles of incidence, θinc, of sound between 15° and 60°. At both the low and the high frequencies the data can be expressed by universal curves when the ITD is normalized by (a/c0) sinθinc, where c0 is the speed of sound in air and θinc is the angle of incidence. Both the ITD and the interaural sound pressure level difference (ILD) show differences between measurements made with the bare torso and those with a clothed torso. These objective results support the subjective measurements of past experiments which showed that in a man there was no localization improvement below approximately 500 Hz, poor localization between 1000 Hz and 2000 Hz, and a change in the localization cue around 1400 Hz from ITD to ILD.
Effect of perturbations in loudspeaker directivity patterns on localization
Authors: Queen D.
Published: October 2005

Because localization cues probably occur during the first millisecond after the radiation of a sound toward a listener, the presence of reflections within this period, having higher intensity than the direct sound, can confuse localization. While such a phenomenon is rare in nature, it may be common in multichannel sound reproduction systems where transducers have frequency-variant nonuniformities in their directivity patterns. In this paper, numerical ratings of localization error, based on directivity equations of the forms p(θ) = 2J1F(θ)/F(θ)2 for a single source and p(θ) = sin[nπmf sinθ/c)]/[n sin(πmf sinθ/c)] for an array; and on empirical relationships of arrival time and intensity, are computed for various combinations of listening positions, room configurations, and transducer configurations. Data suggest that small nonuniformities can produce significant localization error.
Temporal masking for listeners with high-frequency sensorineural hearing loss
Authors: Cudahy Edward A.
Published: October 2005

One off-frequency and two on-frequency temporal masking conditions were employed to examine temporal processing as a function of frequency in hearing-impaired and normal listeners. Forward, backward, and gap masking were measured with a 2AFC procedure for a 5-msec tonal signal in the presence of a narrowband noise masker. In agreement with previously reported data employing signals at a frequency above the masker, normals and hearing-impaired listeners displayed similar masking patterns in forward sad 200-msec gap masking conditions for either a 1500-Hz signal with an 1175–1725-Hz noise masker or a 2500-Hz signal with a 2400–2800 noise masker. Moreover, in the backward and 50-msec gap masking conditions the masking patterns for the hearing-impaired listeners were less steep than normal. In contrast to previous data for the 20-msec gap conditions, both normal and hearing-impaired subjects showed flat masking patterns. Interestingly, when the signal was below the masker in frequency (1500-Hz tonal signal and 2400–2800-Hz noise masker), the masking patterns for normal and hearing-impaired listeners were similar for all conditions. The implications of these results for speech processing in noise by listeners with high-frequency sensorineural hearing loss will be discussed.
Relations between suppression and masking in normal hearing-impaired listeners
Authors: McGee T. M. , Wightman F. L. , Kramer M. B.
Published: October 2005

Tone-on-tone masking patterns from simultaneous and forward masking paradigms are dramatically different. It has been suggested that some of this difference is an effect of suppression, presumably present only in the simultaneous-masking data. To study this. suggestion psychophysical tuning curves were measured using simultaneous and forward-masking techniques and measures of suppression were obtained from the same subjects using Houtgast's two-component forward-masking paradigm. Both normal and hearing-impaired listeners were tested. With normal listeners, simultaneous and forward-masked tuning curves were different in the frequency regions where suppression was most evident. Similar results were obtained from hearing-impaired listeners in regions of normal sensitivity. However, in regions of decreased sensitivity, simultaneous and forward-masking tuning curves were the same and no suppression was evident. These results suggest that much of the difference between simultaneous and forward-masking data is a result of suppression. [Work supported in part by NIH Grant NS12045.]
Ear protection on ships
Authors: Schwartz A. , Munch P. , Naim L.
Published: October 2005

With the intention of incorporate it in a revised version of the “Regulations for the Prevention of Harmful Noise on Ships (1972)” a special “Notice to Shipowners and Shipbuilders” (SHN 4/76) was prepared by a technical committee and issued by the Ministry of Transport, Dept. of Shipping and Ports. Ear protection devices are required in machinery spaces where the noise level exceeds NR 85. The minimum required noise attenuation characteristics of the ear protection devices are: 19 dB at 250 Hz, 30 dB at 500 Hz, 36 dB at 1000 Hz and 38 dB at 2000 Hz and 4000 Hz. Special high quality ear protection devices are required where personnel are exposed to noise levels exceeding NR 110. [The above recommendations were published to show the way to a higher level of crew safety and health, as envisaged in Recommendation No. 141 adopted at the 55th (maritime) session of ILO and IMCO Assembly Resolution A.343 (IX) adopted in 1975.]
Comparative coupler and ear simulator measurements of earphones
Authors: Silbiger Herman R.
Published: October 2005

A series of measurements were performed to compare the performance of handsets and supra-aural and ear-insert-type headsets on standard and modified couplers, and on the KEMAR ear simulator. These measurements have yielded data enabling comparisons to be made of the frequency response of earphones referred to the same location in the ear, at the eardrum, The effects of ear canal resonances on the upper frequencies, as well as the effect of the imperfect seal that the earphone makes with the ear on the receiver's low-frequency response, are thus evidenced. One of the measured effects of the imperfect seal is a reduced low frequency response. The other measure of effect is leakage of room noise into the ear covered by the headphone. These effects are related to a lowering of speech transmission quality ratings as the room noise is increased, where headset types are differentially affected. Finally, data will be presented on the equivalent free-field SPL's to SPL's measured at the eardrum for the various headset types.
Effect of earmold modifications measured by coupler, threshold and probe techniques
Authors: Studebaker G. A. , Cox R. M. , Wark D. M.
Published: October 2005

The validity of probe-tube microphone measurements in providing data indicative of the magnitude of the change in the psychoacoustic experiences of subjects resulting from changes in earmold configuration was investigated. The relationship between these measures in the real ear canal and those changes observed in 2-cc and Zwislocki couplers under similar circumstances of earmold modification was also measured. Standard, vented and “open” earmold conditions were utilized. Threshold and probe-tube measurements were made sequentially without moving the earmold at 21–29 discrete frequencies. Statistical evaluation revealed that these two techniques did not produce significantly different results except at 125–165 Hz where noise masking may have been a factor influencing the threshold data. Neither coupler as used gave an accurate quantitative estimate of the effects of vented earmolds or the open earmold configuration but the Zwislocki coupler gave a somewhat better approximation than the 2-cc coupler. [Work supported by NIH.]
Certain “perceptually consistent” spectral distance measures and their properties when applied to single formant spectra
Authors: Visvanathan R. , Makhoul J. , Russell W.
Published: October 2005

Recently, we proposed a class of spectral distance measures to compute the amount of deviation between two smoothed short-time speech spectra [R. Viswanathan et al., “Towards Perceptually Consistent Measures of Spectral Distance,” Proc. IEEE International Conference on Acoust. Speech Signal Process, Philadelphia, pp. 485–488, April 1976]. These measures produce results consistent with published subjective perceptual results on formant frequency difference limen (DL). More recently, we have adjusted these measures, by a process we call “scaling,” such that spectral distances due to formant frequency shifts equal to the DL at each frequency are approximately the same for different formant frequency values. In this paper, we discuss these “scaled” distance measures, and present the results of using the measures with single formant spectra. The results are in the form of plots of spectral distance obtained under different conditions of either formant frequency shift or formant bandwidth shift. Comparisons are made of these data with published subjective DL data. [Work supported by ARPA-IPTO.]
Semiautomatic segmentation of speech for obtaining synthesis data
Authors: Olive Joseph P.
Published: October 2005

Recent work by Olive and Spiekenagel [J.P. Olive and N. Spickengal, “Speech Resynthesis from Phoneme Related Parameters,” J. Acoust. Soc. Am. 59, 993–996 (1976)] has demonstrated that LPC derived area functions of continuous speech can be represented by straight lines connecting the segment boundaries of the speech. Speech segments were defined as the steady state and transition portions of the phonemes in the continuous speech. The boundaries of the segments in the original experiment were determined by a trial and error procedure, and the criterion for establishing the boundaries was listening to speech synthesized from the original and smoothed parameters. In this paper we will describe a new semiautomatic approach to determine the boundaries of the segments. The method consists of fitting the original area functions by a least-square algorithm; the original boundary points used by the algorithm need be specified only approximately. Speech synthesized from area parameters obtained from the semiautomatic smoothing algorithm sounds as good as the original speech. This method will enable a faster, more accurate segmentation of speech for our work in rule synthesis of speech from dyadic units.
Perceptual evaluation of several pitch detection algorithms
Authors: McGonegal C. A. , Rabiner L. R. , Rosenberg A. E.
Published: October 2005

Recently an objective performance evaluation of seven pitch detection algorithms was made by Rabiner et al. [IEEE Trans. Acoust. Speech Signal Process., Oct. (1976)]. Each of the seven algorithms was compared on the basis of its performance with respect to five different types of errors. The standard of comparison was a semiautomatically determined pitch contour of each utterance. To complement this investigation we describe the results of a subjective evaluation of the synthetic speech quality of LPC analyzed and synthesized speech. The pitch contour used in the synthesis was either one of the seven pitch contours or the result of the semiautomatic analysis. Using a computer controlled sort board, an experiment was run in which each of eight listeners was asked to rank the nine versions of each utterance (the natural version was included to provide a stable anchor point). Results are presented on the over-all preference for each pitch detector. In addition, subject preference as a function of the pitch range of the speaker and the transmission environment used in the recording is discussed. The present results are compared to those obtained in the earlier objective performance study.
Dual input noise cancellation for robust speech processing
Authors: Boll Steven F.
Published: October 2005

A digital preprocessing technique has been developed for removing additive noise from speech prior to Linear Prediction analysis. The algorithm requires two signals: a primary input consisting of speech plus noise, and a second reference input consisting only of correlated noise. (The reference signal could be obtained from a second microphone placed in the noisy environment but isolated from the speaker.) Noise present in the primary signal which is correlated with that in the reference noise from the primary input. The impulse response for the reference signal filter is adaptively computed at the analysis frame rate to minimize the energy of the difference between the primary and filtered reference signals. In addition to improving the estimate of the smoothed spectral envelope, the method can be used to improve pitch detection where inverse filtering by Linear Prediction is employed for spectral flattening.