bluuebunny/crossref_metadata_2025 huggingface dataset:
Frequency specific stimuli for use in sound field audiometry
Authors: Walker Gary , Dillon Harvey
Published: October 2005

Sound field audiometry is used to estimate the hearing thresholds of children and in hearing aid evaluation and selection. The determination of hearing thresholds in the sound field is complicated by the presence of room modes in the test enclosure. These have the effect of selectively amplifying particular frequencies relative to others. The frequency response depends heavily on the position of the speaker and subject in the room. Slight imprecision in the stimulus frequency or the position of the subject can therefore result in substantial errors in threshold determination. This has led to the suggestion that stimuli other than sine waves be used in sound field audiometry. The requirement for frequency specificity imposes an upper limit on the bandwidth of the stimulus that can be used. Suggested stimuli include, frequency modulated tones, amplitude modulated tones, pulsed tones and narrow band noise. Comprehensive data are presented on the behavior of pure tones and the above stimuli in an audiometric enclosure. On the basis of this research specific suggestions are made regarding stimuli and calibration procedures for use in sound field testing.
Pure tone sensitivity and ear canal pressure at threshold in children and adults
Authors: Yoneshige Yvonne S. , Elliott Lois L.
Published: October 2005

The purpose of this study was to relate pure tone sensitivity in six-year-old children and adults to ear canal sound pressure level at threshold. Pure tone sensitivity was measured using a three-alternative forced choice adaptive test procedure, with trial-by-trial feedback. Pure tone thresholds of 6-year-old children were significantly higher than those of adults at both 500 and 2000 Hz. For the condition where control input levels were applied, sound pressure levels under the earphone were essentially the same for children and adults. Results are interpreted as demonstrating true differences in auditory sensitivity between children and young adults—a finding that has also been reported in other studies. [Supported in part by NSF and Sigma Xi.]
Effects of speech competition on word recognition for hearing-impaired listeners
Authors: Morgan Donald E. , Dirks Donald D. , Velde Therese M.
Published: October 2005

Word recognition performance was measured for 15 young normal-hearing listeners and 20 persons with mild/moderate sensorineural hearing loss. Recorded spondaic words and monosyllables (NU ♯6) were delivered monaurally, via earphone in a background of multi-talker babble. The primary signal was presented at four fixed levels (60, 72, 84, and 96 dB SPL). The intensity of the babble was adjusted to achieve a performance criterion of 50% recognition, using a best two-out-of-three adaptive procedure [H. Levitt, J. Acoust. Soc. Am. 49, 467–477 (1971)]. Three separate estimates of “threshold” were obtained at each primary-signal presentation level. For normal hearing subjects, mean 50% recognition was observed at S/N's ranging from -9 (60 dB SPL) to -5 dB (96 dB SPL) for spondaic words and from -4 (60 dB SPL) to + 1.5 dB (96 dB SPL) for monosyllabic words. Performance varied widely among the hearing-impaired. Magnitude of hearing loss, configuration of hearing loss and word-recognition in quiet did not provide a straightforward prediction of performance under difficult communicative conditions. Measurement methods such as those utilized in this experiment may ultimately provide a more accurate prediction of those individuals whose hearing loss severely effects their ability to accurately recognize speech signals under conditions of competition. [Work supported by NINCDS.]
Comparison of interaural time and amplitude jnds in noise
Authors: Cohen Marion F. , Koehnke Janet
Published: October 2005

Interaural time jnds and interaural amplitude jnds for a 500-Hz sinusoid were measured in the presence of either perfectly correlated noise (M0), negatively correlated noise (Mn), or no additional noise using a 21FC adaptive procedure. The correlated noise was presented at a level which would produce a 30-dB threshold shift and this level was maintained for the negatively correlated noise condition. Signal levels in the noise were 10, 20, 30, and 40 dB above detection threshold in the correlated noise (M0S0), and signal levels in quiet were 10, 20, 30, and 40 dB SL. The data indicate that at equal sensation levels, both interaural time and amplitude jnds are smaller when measured against a background of correlated noise than when measured in quiet, and that the largest jnds occur for the negatively correlated noise. Although the differences are greater at low signal levels, they continue to exist at the higher levels for both types of interaural discrimination.
Interaural time and amplitude discrimination in noise
Authors: Slocum Janet E. , Stern Richard M.
Published: October 2005

Just-noticeable differences (jnds) of interaural time delay (ITD) and interaural amplitude differences (IAD) were measured for 50-dB SPL 500-Hz binaural tones in the presence of 100–1000 Hz broadband markers. Data were collected using maskers with various combinations of ITD and IAD. The time and amplitude jnds exhibit similar dependencies on target-to-masker ratio, and they vary with masker type in a fashion previously described by Cohen [J. Acoust. Soc. Am. Suppl. 1 64, S35 (1978)] and Ito et al. [J. Acoust. Soc. Am. Suppl. 1 65, S121 (1979)]. Many of these data trends are consistent with the predictions of simple models based on the instantaneous interaural time and amplitude differences of the stimuli and their variability. Our model that generates lateral position estimates from operations on auditory nerve activity [Stern and Colburn, J. Acoust. Soc. Am. 64, 127–140 (1978)] does not accurately describe these results, for reasons at least partly related to inadequacies in its description of the auditory-nerve response to these stimuli. [Supported by NIH.]
Modern multiple scattering theory
Authors: Lax Melvin
Published: October 2005

A survey of multiple-scattering techniques will be presented (M. Lax, SIAM-AMS Proceedings Vol. VI, 1973, American Mathematical Society) covering the mean-field approximation of Foldy, the quasi-crystalline approximation of Lax (QCA), the average T matrix approximation, modified medium procedures such as the coherent potential approximation (CPA) and a combination of the QCA and CPA to include correlation effects. Severe tests of the adequacy of these approximations are obtained by calculating the density of states for highly random systems and comparing with analytic and Monte Carlo results for one-dimensional systems. We also make a detailed comparison between the CPA, other approximate methods, and a new exact solution [T. Odagaki and M. Lax, Phys. Rev. Lett. (submitted)] of the one-dimensional band percolation problem.
Scattering of acoustic waves by finite elastic cylinders immersed in water
Authors: Numrich S. K. , Frank L. J. , Mason J. P.
Published: October 2005

Backscattering measurements on finite elastic cylinders first reported by Numrich and Varadan [J. Acoust. Soc. Am. Suppl. 1 67, S3 (1980)] have been extended. Comparisons with T-matrix computations are shown at 0°, 45°, and 90° with respect to the major axis of the cylinder and for values of (12)kl ranging from 3–10. For these measurements l is the length of the major axis of the cylinder. Experimental results at higher values of (12)kl are compared with resonance theory.
Resonance theory of acoustic wave propagation through bubble screens
Authors: Gaunaurd G. , Uberall H.
Published: October 2005

Liquid media containing clouds of gas bubbles may be characterized by a set of “effective (and frequency-dependent) material parameters” which differ from those of the liquid and/or gas. A method first proposed by Ament, is used here to determine the effective density and the effective (complex) propagation constant (i.e., the effective sound speed and attenuation) of liquids containing screens or clouds of spherical gas bubbles, by comparing low-frequency sound scattered from the bubble cloud with scattering from an effective sphere of bubbly liquid. The novelty of our resonance treatment is that we have retained some high-order terms in the required low-frequency expansions in order to include the previously ignored dominating effect of the giant monopole resonance. Many numerical results for air bubbles in water are obtained and displayed in graphs. The effects of surface tension and shear viscosity of the liquid and of thermal conduction in the gas, all modify the radiation damping results. We have incorporated all these corrections [in the manner given by R. Nishi, Acustica 33, 65–74 (1975)] into our resonance formulation in order to assess their relative importance in various situations. Comparisons to available experimental results shows very good agreement. [H.Ü. is also at Catholic University, Washington, D.C., and was additionally supported by Code 421 of ONR.]
Immunohistochemical visualization of transmitter related substances in the cochlea
Authors: Fex Jorgen
Published: October 2005

Immunofluorescence and horseradish peroxidase indirect markers for antibodies have been used to study neurotransmitter-related substances in the cochlea of the guinea pig and cat. General aspects of the methods will be discussed and specific examples presented.
Nonlinear behavior of neural responses from lateral line hair cells
Authors: Strelioff David , Sokolich W. Gary , Honrubia Vicente
Published: October 2005

The characteristics of neural responses to combinations of sinusoidal electrical (E) and mechanical (M) stimuli were studied in the lateral line hair cell system of the African clawed toad, Xenopus laevis. The effect of a 4-Hz E or M stimulus on the neural response to a 40-Hz E or M stimulus was a 4-Hz modulation of the mean firing rate and of the response to the 40-Hz stimulus. Discrete Fourier transforms of PST histograms of the neural responses to various combinations of E and M stimuli yielded components at the fundamental and second harmonic frequencies of the applied stimuli and at their sum and difference frequencies. When both stimuli are electric the amplitudes and phases of the observed components are similar to those produced by a quadratic nonlinearity. The most important finding in this study is that the responses to electrical stimuli are nonlinear. This finding implies that nonlinearities observed in the neural responses from hair cell systems are due to nonlinearities in the electrical properties of the hair cells in addition to those in the mechano-electrical transduction process.
Differences in the CF dependence of response phase of gerbil and guinea pig auditory nerve fibers
Authors: Sokolich W. G. , Ohima W. , Strelioff D.
Published: October 2005

Time-locked responses of auditory nerve fibers to acoustic sinusoids having frequencies between 100 and 400 Hz were studied in gerbils and guinea pigs. With the exception of the responses of gerbil fibers with CFs above 10 kHz, which produce double-peaked histograms, the responses of most fibers in both species produce single-peaked histograms when the modulation of spontaneous activity is less than 100%. Within this restricted modulation range, the preferred phase of response is relatively independent of stimulus amplitude. Nonetheless, the variation of the response phase with CF is fundamentally different in the two species. Whereas gerbil fibers show an abrupt 180° phase transition in the CF range between 2 and 4 kHz, the response phase of guinea pig fibers is CF-independent for stimulus frequencies that are at least 12 octave below fiber CF. The finding of different phase functions in the two species indicates that caution should be exercised in basing models of the neural excitation process in the cochlea solely on data obtained from one mammalian species.
Directivity of a long line array with periodically missing sensors
Authors: Thompson William
Published: October 2005

A proposed design of a very long line array of 80 sensors, spaced one-half wavelength apart, envisioned the array as being comprised of four equal subarrays that would be connected together mechanically and electrically. However, mechanical considerations dictated that every twentieth sensor would be eliminated and replaced by mounting hardware. Computations of the directional response pattern of this perturbed array indicated the annoying presence of numerous grating-type lobes that could not be eliminated or even substantially reduced by changing the amplitude weightings. By a simple superposition consideration, i.e., that the array in question is the sum of an intact 80 sensor array and a second array of four negative amplitude sensors, each spaced ten wavelengths apart, the number, bearing, and amplitude of these grating-type lobes can be easily predicted. [Work sponsored by NAVSEA 63 R1.]
A surface pressure transducer using PVDF film
Authors: Brown Richard N. , Brown Neal A.
Published: October 2005

A large area transducer was desired for the measurement of wall pressures beneath turbulent boundary layers. Characteristics desired were small thickness so that it could be applied to a surface without disrupting the flow, and cheapness so that it could be considered expendable and could be applied in mosaic patterns as wave-vector filter elements. A degree of flexibility was also desired for application to curved surfaces. A transducer design using PVDF film was developed toward these ends. The principle problem to be overcome was the sensitivity of the device to vibratory motion of the surface to which it is applied. The particular design includes a sandwich of films on a substrate with a pattern of line supports. The assembly is arranged to suppress outputs due to vibratory bending motion of the substrate while enhancing outputs due to normal stresses (pressure). The preliminary design was 0.020 in. thick and 1 in. square. It had a calculated sensitivity of -220 dB re V/μPa. The actual sensitivity to pressure and vibration was also measured.
The sonic Doppler flowmeter
Authors: Karplus H. B. , Raptis A. C. , Canfield Dennis
Published: October 2005

Industrial applications of the Doppler principle to flow measurement have recently increased in acceptance. Reasonably accurate measurements have been limited to turbulent flow conditions. In the Argonne Doppler flowmeter, as in most commercial units, ultrasonic energy (either in cw or tone burst form) is transmitted at an angle through the pipe wall, and sound backscattered from particles or bubbles, is detected by a second transducer. The Doppler frequency difference between received and transmitted signal is obtained by subtraction. This is a noise of finite bandwidth. For turbulent flow the bandwidth is small enough to permit simple techniques such as counting or frequency to voltage conversion to serve as the Doppler shift indicator with a scale factor close to unity. For laminar flow, a band limited fiat Doppler spectrum is predicted and observed, with the cutoff frequency at twice the predicted Doppler shift. Measurement of such a spectrum is automated with servo controlled adjustable filters. This technique opens up the possibility of using the Doppler method on high viscosity slurries.
State-of-the-art electronics and computerized modeling techniques yield significant improvements in air condenser microphone specifications
Authors: Larson Brian G. , Davis Larry J.
Published: October 2005

Improved design techniques in the evolution of a high-performance microphone system are discussed. Techniques leading to improvements in microphone frequency response, temperature tracking, high-temperature operation, ruggedness, and corrosion characteristics are described. The new design methods enhance microphone sensitivities by more than 10 dB when compared with similar diameter microphones.
Paired comparison judgements of relative speech intelligibility
Authors: Studebaker Gerald , Bisset Jan D. , Van Ort Debra M.
Published: October 2005

The reliability, validity, and sensitivity of paired comparison judgements of speech intelligibility, organized into round robin and elimination tournaments, were studied. Normal hearing and hearing impaired subjects listened to paired samples of speech that were processed with two levels of background noise by 23 hearing aids in two replications. Monosyllabic word intelligibility scores were obtained under each condition in both replications. The results indicated that this paired comparison procedure was a valid, reliable and reasonably sensitive means of evaluating the relative intelligibility of hearing aid processed signals. For example, subjects chose the aid with the best, or second best average discrimination score in 94% of the elimination tournaments. Regarding reliability, the aid that was chosen first on the first trial was the first or second choice on the second trial more than 90% of the time at the O dB S/N ratio. The sensitivity of the procedure was such that at the O dB S/N ratio, the better aid was chosen in 75% of the matches when the difference between the “true” discrimination scores was in the 9%–12% range. [Work supported by NINCDS.]
F
Declination is situation dependent
Authors: Umeda N.
Published: October 2005

One explanation for the current controversy over pitch declination is that different observers have studied different kinds of material. In order to explore the conjecture that voice fundamental frequency (F0) falls consistently in certain types of sentences and not others, we measured F0 contours in three distinctively different modes of speaking—a list of 43 short unrelated sentences, an essay, and a free conversation between two people. We found evidence of declination in only half of the short isolated sentences; and in very few sentences from essay readings and conversation. We conclude that declination observed in isolated sentences comes from two sources: (1) a signal for a new idea marked by high F0 at the beginning of the sentence; and (2) lack of any particular emphasis in a random sentence, and hence a dry and mechanical reading.
Formant estimation with linear prediction
Authors: Gish Walter C.
Published: October 2005

Although the vocal tract transfer function for vowels is approximately all-pole, the resulting vowel waveform is not. We have investigated the effects of using an all-pole analysis technique (linear prediction) on real and synthetic vowels. Formant estimation accuracy and pole location stability were studied as a function of the length and placement of the speech segment analyzed, its pitch, and the analysis technique used. The synthetic speech consisted of a glottal waveform model [G. Fant, STL-QPSR 1 (1979)] with time varying formant damping. Covariance and autocorrelation linear prediction was performed over closed glottis phases, entire pitch periods, and randomly placed 20–25 ms frames. Closed glottis phases for real speech were identified from simultaneous laryngograph measurements and verified by inverse filtering. Covariance linear prediction over the closed glottis phase gave the most accurate formant estimates, stable pole locations, and little or no degradation with increasing pitch. Other analysis intervals or techniques gave significantly poorer results. [Work supported by NSF.]
Management of noise-dominated acoustic data bases to support undersea surveillance processes
Authors: Anderson Nancy D.
Published: October 2005

The properties of noise-dominated tau-Doppler acoustic data bases were examined in relation to data base management requirements. Volume, rate of arrival, rate of aging, and distribution over different fields were among the characteristics studied. Distributions of relatively constant shape due to noise dominance, such as those according to tau; Doppler, and frequency bin, were compared with less stable distributions such as that of coherence. Data organization requirements of algorithms designed to eliminate noise were also examined. Additional areas of study included the requirements of algorithms to associate data from different types of sensors and the effects on data base distribution of sensor tuning and cueing in response to detected target activity. Implementation of data management software designed according to the findings of this study resulted in an order of magnitude increase in data throughout rates for undersea surveillance software.
An adaptive scheme to synthesize broadband signals for a parametric array
Authors: Singhal Sharad , Zornig John G.
Published: October 2005

An adaptive signal synthesis technique is presented that computes the waveform needed as input to a parametric array source so that the farfield received signal has a prescribed waveform for a particular source receiver geometry. The technique uses the Fourier transform of the digitized received signal waveform as a basis for modifying the source signal recursively until the received waveform is close to a desired shape. The method avoids completely the numerical integrations normally associated with parametric source signal design and converges to a satisfactory result within a few iterations. Thus, it is fast enough to be used on line. The theory underlying the method is outlined and several examples of signals designed using the instrumentation of the Yale Hydroacoustic Communications Facility are shown. [Work supported by NSF.]
An analysis of seismic arrivals from large explosive charges received on hydrophones and seismographs
Authors: Safford W. , Cobb A. D. , Browning D. G.
Published: October 2005

In a previous paper [A.D. Cobb et al., J. Acoust. Soc. Am. 67, S30 (1980)] a comparison was made between water-borne and seismic arrivals from large explosive charges during Project LADLE. This paper presents a detailed analysis of the seismic arrivals and a comparison with data obtained from the geophysical investigations. In the 0–10 Hz filter band two types of principal seismic paths were found: an oceanic crust travel path (average sound speed 5.66 km/s) was obtained from three shots; an upper mantel path (average sound speed 7.82 km/s) was obtained from four shots. The seismic arrivals received on the bottomed hydrophones were approximately 100 dB lower in level than the water-borne acoustic arrivals.
Low-frequency ocean ambient noise depth dependence
Authors: Anderson Aubrey L. , Martin Robert L.
Published: October 2005

The depth dependence of low-frequency ocean ambient noise, as observed by omnidirectional sensors, has received considerable attention recently. Similarly, questions of coherence of noise from some surface sources have been raised. In this paper, a near surface ambient noise data set is used to demonstrate the influence of surface decoupling on the depth dependence of noise in this frequency/depth regime. Deviations of 10 dB or more are shown from nearby deep sensor data. Comparisons with theory are used to explain the observations and to illustrate techniques for extrapolation of the results.
Synoptic oceanic wind fields
Authors: Kaye G. T. , Wright T. L.
Published: October 2005

Assessment of the contribution of long-distance storms to the ambient noise background requires a synoptic sampling of the surface wind and wave fields over large oceanic areas. Wind fields based upon satellite observations of cloud cover motions over the Atlantic and Pacific Oceans are operationally reported thrice daily by the National Environmental Satellite Service of NOAA. In the Pacific the Geostationary Operational Environmental Satellite (GOES) covers latitudes 50°N–50°S and longitudes 170°E–110°W with sampled points every 212° in latitude and longitude. Although these are data taken at cloud altitudes, a combination of cloud cover motions for wind direction and satellite scatterometer data for wind speed appears promising. Conformal mapping of wind speed to acoustic noise source level is via Wilson [J. Acoust. Soc. Am. 66, 1499–1507 (1979)]. [Work supported by NAVELEX-320.]
Applications for signal aliasing in gas turbine engine data processing
Authors: Guerette A. R.
Published: October 2005

During the development of gas turbine engines, a wide variety of engine parameter measurements are necessary using many different measuring transducers. Signals from these transducers are subjected to an array of sophisticated signal processing techniques, among them being Signal Spectrum Analysis. Of great concern to the person using this technique is protection from spectrum aliasing. This phenomenon occurs during the FFT analysis of sampled data unless adequate precautions are taken. Under the proper conditions, however, aliasing can be used to produce results which are otherwise difficult or impossible to obtain. This paper presents two situations in which intentional aliasing can be beneficial in the processing of sampled data. The first of these allows the analyst to achieve any desired degree of spectral resolution without the use of unusual or possibly unavailable signal processing equipment. The second shows a method by which the difficult problem of relating phenomena which are sensed at different frequencies by rotating and stationary transducers can be handled.
The measurement and analysis of active system acoustic impedance
Authors: Lagnese T. J. , Sneckenberger J. E.
Published: October 2005

A considerable part of the structure borne vibration in small aircraft is caused by both mechanically and acoustically induced vibration from the aircraft engine. The measurement method suggested within this work obtains the acoustic impedance of an active acoustic source; particular emphasis will be placed on the measurement procedure and apparatus for acquiring the source impedance characteristics of an internal combustion engine. The acoustic impedance is directly used to identify a mismatching impedance and when physically implemented with the acoustic source eliminates much of the acoustic energy output. This project utilized a two sensor sampling technique for obtaining the acoustic impedance of an active acoustic system. The two sensor method of analysis incorporates a Digital Signal Analyzer system to calculate incident-reflected acoustic properties. The laboratory method of experimentation includes exposure of the engine to a broadband random noise while performing sensor signal analysis using a transfer function technique. [Work supported by Nelson Industries, Inc.]
Ray modeling in shallow water
Authors: Tindle C. T. , Bold G. E. J.
Published: October 2005

The lateral displacement of a beam which is totally reflected from an interface is well known. This beam displacement is used to modify the ray paths in a ray model of shallow water propagation in the simple Pekeris channel. Inclusion of the beam displacement for total reflection at the bottom greatly alters the eigenray distribution. Coherent summation of pulses travelling along each eigenray yields a series of arrivals with the depth dependence characteristic of normal modes. The calculations are in good agreement with experimental data obtained at 5.1 km range in water 50 m deep for 4-cycle pulse transmissions in the range 60–140 Hz. There is also good agreement with the results of normal mode calculations. It is concluded that coherent ray tracing is possible even in water a few wavelengths deep.
Intraconvergence zone acoustic propagation: Fluctuations
Authors: Worcester Peter F. , Williams Gordon O.
Published: October 2005

The fluctuations in acoustic transmission between a broadband source and an autonomous receiver moored somewhat below the sound channel axis at 23 km range have been examined. The transmitted signal was centered at 2272.73 Hz. A variety of measures of the observed fluctuations (pulse and cw) for three well-resolved paths computed from 72 h of data are compared with theoretical predictions based only on sound-speed perturbations due to internal wave vertical displacements.
Path identification studies in the Atlantic
Authors: Weinberg Norma L. , Clark J. G.
Published: October 2005

For two long-range paths in the Atlantic, extensive data sets are available both for fixed source to fixed receiver multipath structures, and for sound velocity profiles along the paths. The sound velocity data have been used in range-dependent ray computations in an attempt to identify the observed multipaths. Data and computations are presented, along with the results of a special study of the sensitivity of computed multipath structure to range dependence in measured profiles.
Investigation of low-frequency reflectivity from the ice-water interface
Authors: Yang T. C. , Votaw Charles
Published: October 2005

Preliminary data of low-frequency (⩽1 kHz) reflectivity from the ice-water interface at shallow angles will be presented. Distinct frequency dependence is observed in the data and possible theoretical interpretations suggested. [Work supported by ONR.]
The monitoring and mixdown environment
Authors: Augspurger George L.
Published: October 2005

The selection of monitor loudspeakers is only one factor that contributes to the evaluation of a commercial recording. Judgments of tonal balance, definition and stereo imaging are also affected by the physical geometry of the monitor installation and the acoustical idiosyncrasies of the monitoring room. Even the electronic audio chain can add its own discernible signature to an extent that affects esthetic decisions. Regardless of philosophies embraced by individual producers and recording engineers, the monitoring and mixdown environment is usually designed not to produce a typical listening experience but as a tool to identify that product which will retain its integrity when heard in a wide variety of real-life situations.
From recording console through the home loudspeaker; the promise of state-of-the-art technology
Authors: Locanthi Bart N.
Published: October 2005

If the listening environment and the loudspeakers in the home and the studio are identical, and if the communications links between are perfect, then the record producer can deliver products to the consumer which are exact copies of his originals. We will discuss some communications links between producer and consumer which can deliver copies having errors of less than 0.01% when compared with the original.
Ultrasonic imaging in diagnostic medicine
Authors: Fry Francis J.
Published: October 2005

Imaging of soft tissues of the human body for medical diagnostic purposes using pulse echo ultrasound has become a major method in the field of medicine. The characteristic of tissue which has made this achievement possible is differences in bulk modulus, density, acoustic velocity, and attenuation. Some or all of these differences are responsible for the particular type of two-dimensional cross section image displayed on a monitor for physician observation. From such images and experience a body of relevant diagnostic information has evolved. A number of images of human tissues will be presented. These images were obtained from eight different commerical and experimental devices. Both real time and nonreal time images will be shown. Although the present state of ultrasonic medical diagnosis is mainly of a qualitative nature it is anticipated that a more quantitative approach will evolve along lines to be discussed.
The physical basis for tissue identification by ultrasound scattering
Authors: Barger James E.
Published: October 2005

The original objective of diagnostic medical ultrasound was to reveal by noninvasive means the boundaries of soft tissues, thereby providing images of organs that were analogous to x-ray images of hard tissues. The mechanical properties of soft tissues that determine their interaction with sound vary over dimensions comparable to or larger than a wavelength. Therefore such physical processes as propagation, scattering, attentuation, reflection, diffraction, and refraction can be observed to determine some parameters of the tissue properties. When these parameters are uniquely. associated with a tissue, then that tissue can be identified by acoustical measurements made in sound that has interacted with it. This paper outlines procedures of tissue identification based on scattering of sound. The frequency of interrogating ultrasound spans the range extending from about 0.5 MHz for diagnosis in the brain to about 10 MHz for diagnosis in the eye. In each case, the spatial distribution of the tissue's mechanical properties are estimated from analysis of scattered acoustical signals. [Work supported by NIH and NSF.]
Subjective and objective considerations of the loudspeaker interface
Authors: Heyser Richard C.
Published: October 2005

The loudspeaker and the listening environment constitute a final interface in the music recording and reproducing chain. Eventual success or failure of the music experience often depends upon subjective considerations which are not adequately handled by traditional loudspeaker technical measurements. A discussion is presented of practical experience gained during evaluation of loudspeakers for review in a leading consumer publication, and some examples given of the degree to which some loudspeakers met or failed the requirements for music reproduction.
Glory contribution to backscatter from large elastic spheres
Authors: Marston P. L. , Flax L.
Published: October 2005

Partial wave theory for backscatter by large aluminum spheres and cylinders in water give a form function for spheres several times greater than that for cylinders when each is normalized to the geometric scattering of a rigid reflector of the same size and shape [L. Flax, J. Acoust. Soc. Am. 62, 1502 (1977)]. The physical-optics approximation is used in the present work to show that twice-reflected, twice-refracted glory rays are a significant contribution to the largeness of the backscatter by elastic spheres. These rays lead to a back-facing toroidal wavefront which appears to originate at a virtual ring-like source of radius b < the sphere radius a. The modulus of the glory contribution to the normalized form function obtained by computing the farfield diffraction of this wavefront is |fa| = (ka)1/2G|J0 (kb sin γ)|, where k is the wavenumber in water, G is a function of material properties, γ = 180° − θ, and θ is the scattering angle. When all glory rays in aluminum are S waves, b/a ≃ 0.27 and |fa ≃ 2.8 for backscatter with ka = 938. The sum of 2.8 and the geometric form function of 0.8 due to reflection from the first surface is similar to the prediction of partial wave theory; however, a complete physical-optics calculation should include the phase of these terms and other reflections. The (ka)1/2 enhancement is not present in cylinders. [Work supported by ONR. P. Marston is an Alfred P. Sloan Research Fellow.]
Weak multiple scattering by a linear array of randomly sized spheres
Authors: Young J. W. , Ma K. S.
Published: October 2005

Scattering of an incident plane wave by a linear array of randomly sized spheres exhibits both deterministic and stochastic features. For the case of noninteracting spheres, moments of the scattering cross section are determined by moments of the sphere size distribution function multiplied by the geometrical pattern function of the linear array. When interaction between spheres (multiple scattering) is included, this simple product relationship is modified. In this paper, we evaluate the effect of weak multiple scattering on the moments of the scattering cross section of a linear array of small spheres. Both hard and soft boundary conditions are considered in the limit of kti kā ≪ I and d̄ where ā is the mean radius and d̄ is the mean separation.
Two-tone suppression as a function of masker level, suppressor level, and signal delay
Authors: Neff D. L. , Jesteadt W.
Published: October 2005

Forward masking was measured for two-tone maskers constructed from all combinations of a 1000-Hz sinusoid at 40. 50, 60, or 70 dB SPL and an 1136-Hz sinusoid at 60, 70, 80, or 90 dB SPL. A 1000-Hz, 20-ms sinusoidal signal followed the offset of the masker by 4, 8, 16, or 32 ms. The forward-masking effect of each masker component alone was also assessed for all levels and signal delays. In addition, the simultaneous-masking effect of each component of the two-tone masker on the other was measured. The reduction in forward masking associated with the addition of the 1136-Hz masker component provides a commonly used, albeit indirect, measure of suppression. More direct measures of suppression, based on estimates of the reduction in effective level of the 1000-Hz component, will be described. Differences in suppression across listeners can be accounted for in part by large individual differences in the amount of forward masking of the signal by the 1136-Hz component. There is no simple relation that allows the amount of suppression to be described in terms of amount of simultaneous masking and signal delay. [Work supported by NIH.]
Psychophysical tuning curves and off-frequency listening
Authors: Green David M. , Shelton Brian R. , Picardi Maria C. , Hafter Ervin R.
Published: October 2005

Psychophysical tuning curves were obtained for three subjects in a forward masking paradigm, with signal levels of 21, 30, and 50 dB at 1000 Hz and 30, 40, and 50 dB with a 3000-Hz signal. All tuning curves were measured in quiet and with a background noise adjusted to produce a constant signal-to-noise ratio of E/N0 = 15. The results indicated that the tuning curves obtained in quiet change shape with increasing signal level, whereas the shape of tuning curves measured in noise are the same at all signal levels. A control condition demonstrates that only signal energy in a narrow band near the signal frequency is useful in detecting the signal. This control along with the invariance of tuning curve shape with signal level support the interpretation that the broadening of tuning curves with increased signal level, observed when the signals are presented in quiet, is caused by the presence of off-frequency components.
Effect of a transitional layer on the transmission of vibrational disturbances between two elastic media
Authors: Pierucci Mauro
Published: October 2005

Classically, the transmission of disturbances between adjacent media is obtained by equating velocities and/or stresses on both sides of the interface. If a transition layer is placed between two media, then the discontinuity in the elastic properties may be smoothed out over a small but finite distance. In this paper the vibration transmitted across the two elastic media is obtained from the continuous transition layer approach and the results are compared with the classical approach. The effect of the thickness of the transition layer for different frequencies and different elastic materials is presented. Results indicate that the effect of the thickness of the transitional layer is felt not only within the layer itself but it is also very dominant at large distances from the interface.
Asymptotic grazing field on a locally excited, homogeneous fluid-loaded plate before critical frequency
Authors: Berkman E. F. , Barger J. E. , Smith P. W.
Published: October 2005

The farfield solution for the pressure field on the surface of a locally driven plate below critical frequency is developed. The solution to this problem relates to the propagation of sound along ship hulls. Only the grazing field propagating principally in the fluid at sonic speed is considered here. The flexural wave carried principally by the plate is excluded. As farfield radiation is directive with a null at grazing, a higher order asymptotic solution is required to describe the pressure on the plate surface.
Edge diffraction interference in baffle performance measurements
Authors: Sachs D. A.
Published: October 2005

Edge diffraction has usually been a limiting interference mechanism in baffle performance measurements. In this paper, Kendig's theory of diffraction by an impedance-covered half-plane [R. P. Kendig, Ph.D. thesis, Pennsylvania State University (1977)] is used to generate predictions of edge diffraction interference levels for baffles with general surface impedances. The effectiveness of an array receiver for reducing the interference is discussed.
Spectral normalization of vowels based on frequency warping
Authors: Ohyama Ghen , Wakita Hisashi , Matsumoto Hiroshi
Published: October 2005

In a previous paper [J. Acoust. Soc. Am. Suppl. 1 64, S180 (1978)] speaker normalization based on a frequency warped spectral matching was discussed. A frequency normalized distance between test and reference spectra is defined on the basis of the minimum mean square difference over all possible choices of frequency. The slopes of the warping functions are restricted by an assumed range of the vocal tract length, so that they mostly take account of linear scaling for the difference in vocal tract length and of slight nonlinear difference. In this paper a modification of the technique will be proposed. Test and reference spectra are converted to a log scale frequency axis so as to give a heavy weighting in the region of the low frequencies and light weighting in the region of high frequencies. Then the distance between test and reference spectra is computed by means of dynamic programming, after eliminating the individual clottat characteristics. The experimental results show a better recognition score than previous ones.
On the effects of varying analysis parameters of an LPC based, isolated word recognizer
Authors: Rabiner L. R. , Wilpon J. G. , Ackenhusen J. G.
Published: October 2005

Speech recognition systems based on LPC features sets have been applied successfully to a number of speech recognition tasks including an airlines information system, a directory assistance system, a voice repertory dialer, and a connected digit recognizer. All these systems have been based on a recognition model, originally proposed by Itakura, with a fixed analysis parameter set. To get an appreciation for the robustness of this feature set, an experimental investigation was undertaken to vary each of the parameters of the analysis system, and to measure the effect of recognition accuracy. The parameters chosen for study included—N, the size of the analysis frame; L, the shift between frames; p the number of LPC poles; α, the preemphasis factor, and LP, the analog filter cutoff. Four talkers were used in this study. Each talker recited the vocabulary (the digits, the letters of the alphabet and three command words) seven times for training, and ten times for subsequent testing. The speech was stored in digital form at a sampling rate of 20 000 Hz. Speaker trained templates were obtained from the training data. The results of this study indicate that the recognition accuracy remained high over a wide range of variation of analysis parameters.
A new training method for isolated word recognition
Authors: Wilpon J. G. , Rabiner L. R.
Published: October 2005

One of the most important operations in isolated word speech recognition systems is the method used to obtain the word reference templates. For speaker-trained systems, the techniques that have been used include casual training, averaging, and use of statistical pattern recognition clustering methods. In a recent study Rabiner and Wilpon showed that the statistical techniques, when combined with the technique of averaging the autocorrelation coefficients of all tokens within the cluster, provided a reliable, robust set of reference templates. The only drawback to this method was the extensive, burdensome training required for the statistical analysis. Since the statistical training method could not be used in most practical situations, techniques were investigated for obtaining a simplified, robust training procedure which would incorporate many of the ideas of the statistical approach. Such a training method is described in this talk. The advantages of this new training procedure, over (previously used) casual training include: ease of use, robustness of the templates, minimized computation in the recognition algorithm, normalized durations of all references, and recognition accuracy comparable to previous systems with double the number of templates.
Speaker recognition in conjunction with speaker independent word recognition
Authors: Rosenberg A. E. , Shipley K. L.
Published: October 2005

Our method for accomplishing speaker independent word recognition is to select a small set of word templates that typify and span individual speaker reference templates obtained from a large population of speakers. An unknown utterance is processed and compared with a set of such templates for each word in the vocabulary. Word recognition decision functions are based on combinations of template distances obtained in the comparison. In this study it is hypothesized that distributions of template distance scores are reasonably consistent for individual speakers and vary characteristically from speaker to speaker. This property is exploited to provide a speaker recognition capability in conjunction with speaker independent word recognition. It is shown that good speaker identification performance is dependent on the input of a sequence of distinct words. For a 20 speaker experimental population a mean speaker identification error less than 1% is obtained over a sequence of seven distinct words. Techniques are also described for grouping individual speaker references into class references.
ADPCM voice storage in a microprocessor
Authors: Flanagan J. L. , Johnston J. D. , Upton J. W.
Published: October 2005

Adaptive differential pulse-code-modulation (ADPCM) is a simple, moderately efficient technique for digitally encoding and storing speech signals. Signals typically can be represented with acceptable quality by 4 bits of adaptive quantization per Nyquist sample. The ADPCM code format is therefore convenient for use with currently-evolving microprocessors. In this paper we describe an experimental implementation in which an MC68000 microprocessor is programmed for real-time ADPCM encoding at 20K bits/s, and for storage of input speech signals in on-board memory. A simple packing program puts two 4-bit code words into each memory byte, providing the capacity for over 10 s of stored speech per 32K bytes of memory. An independent read program unpacks the words, decodes them to linear PCM and outputs them for analog conversion. An alternative arrangement utilizes the Digital Signal Processor (DSP) integrated circuit for external encoding and decoding. Dual DSPs communicate with data ports of the MC68000 and the latter edits, packs, stores and retrieves the ADPCM code words.
Longitudinal study of children's hearing: Its relationship to noise and other factors
Authors: Roche Alex F. , Siervogel Roger M.
Published: October 2005

Six-month auditory threshold levels (ATL) have been obtained from children aged 6 to 18 years over a 4-year period. The means decrease with age especially at lower frequencies. Past otological problems are associated with poorer hearing. Stature and thresholds are not associated in boys, but taller girls tend to have worse hearing at younger ages and better at older ages. After the effects of stature are removed, rapid maturation is associated with lower thresholds in boys from 10 to 18 years and in girls at 8 and 9 years. Mean thresholds are significantly higher in those reporting higher noise exposure, especially from loud TV and power tools. High thresholds are associated with elevated systolic pressures in boys but with low pressures in girls. There are no associations with diastolic pressure or between noise exposure and blood pressure. Leq(24) levels have been obtained using General Radio and Metrosonics dosimeters. The mean Leq(24) for these children is between 77 and 84 dB with no significant age or sex effects; Leq(24) was correlated with ATL in girls but not boys. [Work supported by USAF and EPA.]