bluuebunny/crossref_metadata_2025 huggingface dataset:
Neural networks for estimating articulatory positions from speech
Authors: Atal Bishnu S. , Rioul Olivier
Published: October 2005

This talk describes an application of neural networks for estimating positions of various articulators (such as the tongue, the lips, etc.) in the vocal tract from the speech signal. In general, a neural network consists of a large number of interconnected computational elements. The networks that will be discussed in this paper include an input layer of nodes connected directly or through an intermediate layer of hidden nodes to an output layer. Iterative gradient search procedures are often used for determining the unknown parameters of the neural network, but these procedures are very slow for training neural networks with a large number of hidden nodes. For estimating articulator positions, it was found that the weights in the first layer could be set to fixed random values during the training procedure without degrading the performance of the network. The random fixed weights in the first layer permit the use of a fast noniterative procedure for determining the unknown parameters of the second layer. Tests on a vocal tract model with ten articulatory variables show that the articulator positions can be determined accurately, using a network with 500 hidden nodes in the intermediate layer, from ten LPC parameters derived from the speech output of the model.
Diffraction of a point source about a disk
Authors: Dlubac J. J. , Jiang C. W. , Yim B.
Published: October 2005

Computations of diffraction of an acoustic point source about a hard and soft disk are presented. The solution, given as a series of oblate spheroidal wavefunctions, is based on the classical analysis [J. J. Bowman, T. B. A. Senior, and P. L. E. Uslenghi, Electromagnetic and Acoustic Scattering by Simple Shapes (North-Holland, Amsterdam, The Netherlands, 1969), Chap. 14]. In the limit as the source goes to infinity, the plane-wave solution is recovered. Computations with the field point on the disk verify the boundary conditions. Parameter studies are presented, and convergence criteria are discussed.
Detection of intensity decrements by the chinchilla
Authors: Graf Christine J. , Saunders Samuel S. , Salvi Richard J.
Published: October 2005

Three monaural chinchillas were trained to detect intensity decrements in broadband noise (20 kHz) using a shock-avoidance conditioning procedure. The intensity decrements were presented at one of nine different durations between 2 and 35 ms at noise levels of 25, 45, and 65 dB SPL. At each intensity-duration combination, the level of the decrement was varied to obtain a decrement threshold. The minimal detectable decrement decreased from approximately 20 dB at the shortest duration to an asymptote of roughly 4 dB at approximately 30 ms. The data were modeled by a low-pass filter with an 11-ms time constant. The decrement detection function of the chinchilla is similar to that of humans. However, long duration decrement thresholds are larger in the chinchilla, as would be predicted from the large intensity difference limen of the chinchilla. In general, there was little change in the decrement function across background intensities except that 2-ms decrements were not detected at the 25 dB SPL background intensity. [Research supported by grants RO1NS16761 and RO1NS23894.]
Ultrasonic studies of collective modes in superfluid He3
Authors: Ketterson John B.
Published: October 2005

Liquid He3 undergoes a transition at a temperature of 2×10−3 K into a superfluid state. The state is related to the paired-electron state of the BCS theory of superconductivity but differs in that the He3 atoms are paired in an L = 1 (p wave), S = 1 (triplet) state [rather than the L = 0 (s wave), S = 0 (singlet) state of conventional superconductivity]. Various excited states of the superfluid exist, which are referred to as excitonic or collective modes, and, loosely, they correspond to various values of the total angular momentum J = L + S. Some of these modes couple strongly with ultrasound. A brief review is presented of these collective modes and of the ultrasonic experiments performed to date, which probe their behavior as a function of temperature and magnetic field.
A learning word-spotting method for speaker-independent word recognition in noisy environments
Authors: Kanazawa Hiroshi , Takebayashi Yoichi
Published: October 2005

A learning word-spotting method has been proposed for the purpose of robust speaker-independent word recognition in noisy environments. In order to avoid word boundary detection errors at the recognition stage, the method employs word spotting based on the multiple similarity method, which was shown to be effective for noisy speech data. The learning process uses synthesized noisy speech data, a mixture of pure speech data and noise data, to design reliable word reference vectors for the word spotting. Word feature vectors with maximum multiple similarity values are automatically extracted by the word spotting. During the learning process, the signal-to-noise ratio (SNR) of the synthesized data is gradually decreased to perform word spotting accurately. Experiments were carried out for 13 words, including ten Japanese digits, spoken by 50 males. Under the condition of 10 dB, SNR contaminated by concourse noise, the recognition scores of 85.5 % and 94.1% were obtained by word spotting, without learning and with learning, respectively. The results have shown the effectiveness of the proposed method in noisy environments.
Goal-directed speech in a spoken language system
Authors: Rudnicky Alexander I.
Published: October 2005

The advent of reliable speaker-independent continuous speech recognition systems has made it possible to design systems that use speech as a replacement for keyboard input. To understand the nature of a system that accepts spontaneous goal-directed speech (as opposed to the current standard of read speech), a spoken-language spreadsheet was implemented and users performing a series of tasks using this system were studied. The system was instrumented to allow the collection of detailed timing information about the components of the interaction cycle. The (HMM-based) recognition system incorporates a lexicon of 273 words and a language of perplexity 51. Four users performed a series of 40 tasks (involving the entry of personal financial information) alternating voice and keyboard input. Users completed 30 tasks in one block of sessions, then returned a month later to complete the remainder. The utterances spoken into the system (over 7500) were stored for later analysis. The data collected provide a comprehensive picture of how users learn to use a spoken-language system and provide an understanding of the factors that govern performance in spoken-language systems.
The intelligibility of native and non-native speakers of American English using spelling alphabet test materials
Authors: Schmidt-Nielsen A.
Published: October 2005

Male and female speakers who were native speakers of French and Dutch, as well as native American Engish speakers, were recorded reading randomized lists of the International Civil Aviation Organization (ICAO) spelling alphabet words (ALFA, BRAVO, CHARLIE, etc.) and the digits 0 to 9. The resulting tapes were subjected to a variety of degradations, including Gaussian noise, interruption by silence and by noise, and LPC coding. Non-native speakers were less intelligible than the native speakers, and the decrement was greater for some of the degraded speech conditions. Results are compared with selected DRT scores for the same speakers. [Work supported by NAVSPAWARSYSCOM.]
Speaker-independent recognition of the DARPA Naval Resource Management Task
Authors: Lee Chin-Hui , Rabiner Lawrence R. , Wilpon Jay G. , Pieraccini Roberto
Published: October 2005

In this talk, the issue of acoustic modeling of the fundamental units for recognition is examined. Modeling approaches for both context independent (CI) and context dependent (CD) units are studied. The acoustic modeling approaches were tested on speaker independent recognition of the DARPA Naval Resource Management Task. The set of context independent units in this study is a fixed set of 47 phonelike units (PLUs), in which each PLU is associated with a linguistically defined phoneme symbol. Each CI/PLU is modeled using a continuous density hidden Markov model (CDHMM) with Gaussian mixture state observation density. The set of context dependent units includes PLUs defined by left, right, and both left and right context. Only those CD/PLUs with enough occurrences in the training data are selected for modeling. Two approaches are presented to model the CD/PLUs. Both CI/PLUs and CD/PLUs are obtained using the segmental k-means training procedure. In the case of context independent acoustic modeling, the maximum number of mixtures was varied in each state from 1 to 256 and it was found that the word accuracy increased from 56% to 89% which indicates that sufficient acoustic resolution is essential for improved performance. The 89% word accuracy is the highest performance reported based on context independent units. When context dependency modeling is incorporated, both modeling approaches achieved better than 92% word accuracy.
Automated training for speech recognition
Authors: Smith L. A. , Scott B. L. , Lin L. S. , Newell J. M.
Published: October 2005

A method is described for automated training of a speaker-independent isolated word recognizer. The training process generates vocabulary templates from a database of collected training utterances. These templates are then modified through adaptive training, an iterative process of testing and modifying templates in order to optimize recognition. Robustness of the templates is enhanced by varying the presentation of the collected utterances during adaptation; varying the utterance sampling rate, for example, has the effect of presenting the same utterance at differing pitches and time scales. Adaptive training continues until the error rate falls to an acceptable level. Results will bc presented for similar vocabularies developed with and without adaptation and under varying adaptation conditions.
Attentional allocation during phoneme monitoring: An investigation into the unit of perceptual analysis and selective attention during speech perception
Authors: Pitt Mark A. , Samuel Arthur G.
Published: October 2005

A variant of the phoneme monitoring technique (in which the probability with which the target phoneme occurred at a particular consonant location in a word was varied) was employed to investigate whether the phoneme or the syllable is the smallest linguistic unit of perceptual analysis, and to examine selective attention during speech perception. The results suggest that the phoneme may be a unit of perception and that subjects were able to attend selectively to a single consonant location in a word. [Research supported by AFOSR.]
On the combined use of “full” and “reduced” diphones in speech synthesis
Authors: Drullman Rob , Collier René
Published: October 2005

In the generation of synthetic speech by means of diphones, the latter are usually excised from accented syllables. Therefore, all the speech sounds that result from the concatenation of diphones have phonetic properties that are typical of this accented context only. In particular, their temporal and spectral characteristics are close to the target values that one finds in the production of carefully spoken, isolated, accented syllables. Speech generated by means of such diphones often sounds over-articulated. In unaccented syllables (and perhaps elsewhere, too), one might prefer to use “reduced” diphones, which may lead to shorter and spectrally less extreme speech sounds. Perception experiments were run in order to assess the improvement that can be achieved by also using “reduced” diphones, which are extracted from unaccented syllables. In these tests, the use of “reduced” diphones was determined by the following variables: word stress, sentence accent, word and phrase boundaries, metrical structure. [Research sponsored by the Dutch government as part of the national SPIN program “Speech analysis and synthesis.”]
Three-dimensional matched-field processing with an eigenvalue method
Authors: Collins Michael D. , Kuperman W. A. , Perkins John S. , Tinker Laurie E. , Glattetre John
Published: October 2005

An eigenvalue method for three-dimensional matched-field processing with a vertical array has been developed for localization of multiple sources in range and depth (bearing can also be determined with the method if the environment is asymmetric). The method involves applying the Bartlett beamformer to the eigenvectors of the cross-spectral density matrix. This approach appears to be more effective than the standard eigenvalue approach involving beamforming on a linear combination of several eigenvectors. Eigenvalue beamforming methods are based on the orthogonal targets approximation [R. F. Gragg, NRL Rep. 9143, Naval Research Laboratory, Washington, DC (1989)]. The spatial correlation over a horizontal array of two sources is small provided the sources are sufficiently separated in bearing. On the other hand, the spatial correlation over a vertical array of two point sources may be relatively large for some source locations (i.e., the Bartlett matched-field processor for a vertical array has relatively large sidelobes). However, simulations suggest that this effect does not seriously degrade the eigenvalue processing method for vertical arrays. Source motion will further enhance this processing method.
A perturbation method for normal mode theory with variable velocity profile
Authors: Werby M. F. , Ellis Paul
Published: October 2005

Normal mode theory is a very well-developed technique for predicting the propagation loss in the ocean for most stratified environments and velocity profiles, as well as some range-dependent environments. So why another method? The reason is that sometimes it is desirable to retain the simple trigonometric functions obtained from constant velocity profiles. This is usually done by employing the approximate scheme of using multiple layers in each of which the velocity profile is assumed constant. In this work, it is shown that this can be carried out nicely by representing the velocity profile in terms of a mini-max fit using Tschebyscheff polynomials and a first-order perturbation technique derived from Sturm-Liouville theory. The derivation leads to a compact closed expression that is easily programmable. The results from the formulation are compared with other methods.
Perception of vowels by budgerigars (Melopsittacus undulatus)
Authors: Dooling Robert J. , Brown Susan D. , Nespor Amy , Hawks John W.
Published: October 2005

Budgerigars (parakeets) were trained using operant conditioning techniques to respond to differences between speech stimuli. Response latencies were used to construct similarity matrices and multidimensional scaling procedures were then used to produce perceptual (spatial) maps of these stimuli. For natural vowel tokens, budgerigars showed evidence of perceiving phonetic categories in spite of variation in talkers including talker gender and talker age. Experiments with synthetic vowel tokens generally confirmed and extended these findings. Multiple regression techniques revealed that the perceptual dimensions obtained from MDS were highly correlated with the frequencies of the first and second formants. These results suggest that both natural and synthetic vowels are probably perceived in similar ways by budgerigars and humans. These results have relevance for theories of speech perception and language development in humans. [Work supported by NIH.]
Psychophysical studies on cochlear implant patients deafened prior to 4 years of age
Authors: Busby P. A. , Tong Y. C. , Clark G. M. , Alcantara J. I.
Published: October 2005

Psychophysical studies were conducted on eight prelingually deaf cochlear implant patients, between 5 and 23 years of age at the time of surgery, who were deafened prior to 4 years of age. Electrical stimulation of the auditory nerve was by means of the Nucleus multiple-electrode prosthesis. The average difference limens for the prelingual patients in electric current level (7% of the dynamic range between threshold and comfort level), duration (17 ms for a standard of 30 ms), and the detection of a gap in a 1-s signal (9 ms) were in the same range as those reported for postlingually deafened adult implant patients. The average difference limens in repetition rate for the prelingual patients (19% for a standard of 100 rep/s and 28% for a standard of 200 rep/s) were poorer than those found for the postlingual patients (5% for 100 rep/s and 8% for 200 rep/s). Individual differences in performance were observed for the prelingual patients in all studies.
A pitch perception study for a cochlear implant user with residual hearing in the contralateral ear
Authors: van Hoesel R. , Tong Y. C.
Published: October 2005

In order to combine electrical and acoustic signals for multichannel cochlear implant users with residual hearing in the unimplanted ear, it is important to determine the degree of overlap in pitch sensations on the two sides. A pitch scaling experiment was performed maintaining constant and identical repetition rates on both the implanted (left) and normal (right) ear. The acoustic signals consisted of bandpass-filtered pulses {with center frequencies ranging from 350–850 Hz). These were transmitted in synchrony with electrical pulses to one of five selected bipolar electrode pairs spanning 15.75 mm of the scala tympani (from approximately 2.25 to 18 mm beyond the base). The spacing between the two electrodes of each bipolar pair was 1.5 min. The experiment was repeated for repetition rates of 112.5 and 205 Hz. Pitch sensations on the left (electrical) side were reported to be as low as, and at times even lower than, on the right (acoustic) side. All signals were sealed with respect to an acoustic reference stimulus that had a bandpass filter center frequency at 450 Hz. Synthetic vowels were derived from the pitch scaling results and acoustic strategies only as well as combined electrical and acoustic strategies were tested. It was found that the best results were obtained from a combined electrical and acoustic strategy where the electrode information is chosen such that pitch perception overlap with the acoustic side is minimized. [Work supported by NIH Grant NS21027.]
Test materials for evaluating speech perception ability of cochlear implant patients
Authors: Fukuda Yumiko
Published: October 2005

A system for assembling the test materials that are used in evaluating speech perception ability by combined use of cochlear implant and lip-reading has been developed. In order to construct balanced lists, a database of Japanese basic vocabulary furnished with specifications regarding their auditory, visual, temporal, and linguistic components is utilized in this system. Factors whose distribution are balanced between each other list are the kind of vowel and the manner/place of articulation of the consonant, special Japanese syllables (elongated vowel, choked sound, and syllabic nasal), phones having the same mouth shape, number of syllables in a word and phrases in a sentence, class of familiarity, and grammatical data such as part of speech. All these data and specification of category of meaning of words in the database are used in searching suitable words for the lists, as well as in analyzing characteristics of perception ability of the patients. Problems in requirements for the evaluation methods, construction of balanced lists from the database, and video recordings of speech and mouth shape of utterances of the lists are discussed in this paper.
The effect of varying the amplitude-frequency response on the masked speech-reception threshold for sentences in conditions with low-frequency noise
Authors: van Dijkhuizen Janette , Festen Joost M. , Plomp Reinier
Published: October 2005

Within a study on the merits of a frequency-dependent automatic gain control in hearing aids, the beneficial effect of adapting the amplitude-frequency response to situations of seriously interfering low-frequency noise is studied. For 12 normal-hearing and 12 hearing-impaired listeners, the masked speech-reception threshold (SRT) for sentences is measured. The noise has a spectrum identical to the long-term average spectrum of the sentences, but with its level in one octave band raised by a constant 20 dB in the first experiment, and slowly increasing by 20 dB during sentence presentation in the second experiment. Results indicate that, in conditions with steady-state noise, selective attenuation of the signal in the band containing the extra noise gives an improvement of speech intelligibility in terms of speech-to-noise ratio of up to about 4 dB for both the normal-hearing and hearing-impaired group. In conditions with time-varying noise, a similar improvement was observed for both groups.
Evaluation of a 32-channel electrotactile vocoder
Authors: Rakowski Kathleen , Brenner Christine , Weisenberger Janet M.
Published: October 2005

Previous evaluations of laboratory prototype multichannel tactile aids have shown them to provide useful speech information. However, few of these prototypes have been successfully converted into wearable devices. The present study evaluated a device that has recently become commercially available, the Audiotact (Sevrain-Tech, Inc.), a 32-channel electrotactile vocoder worn on the abdomen. Four subjects pacticipated in the study, two of whom had previous experience with other tactile aids. In addition, a visual display designed as a training aid for new users of the Audiotact was evaluated. This display provides a visual representation of the activity on the tactile array. Two of the subjects were instructed to use this visual information while wearing the Audiotact. Results of testing with minimal pairs and small set phoneme identification tasks showed that manner features were transmitted quite well. Subjects were also able to integrate information available through lipreading with information from the Audiotact. However, preliminary results for connected discourse tracking with the Audiotact showed only a small improvement over lipreading alone. The results of this study are discussed in terms of comparability to other multichannel tactile devices.
Nonlinear effects in a driven vibrating wire
Authors: Hanson Roger J. , Macomber Hilliard K.
Published: October 2005

A horizontal wire is driven sinusoidally with a vertical magnetic force. Horizontal and vertical optical detectors follow the motion of a point on the wire. With a moderately large driving force, the usual predicted nonlinear effects are observed as the driving frequency is increased from below resonance. These include the onset of a horizontal component of motion, phase shifted 90° with respect to the vertical component, and ultimately resulting in circular motion of the wire. However, with a much weaker driving force, the phase relation between the horizontal and vertical components is no longer fixed at 90° and becomes very frequency sensitive in a certain range. It appears to vary from 180°, producing inclined straight-line motion, to 90°, to 0°, again producing straight-line motion but with a positive slope. Further increase of the driving frequency causes the vertical excursion of the straight-line motion to increase while the horizontal excursion changes little. A second harmonic in the horizontal component becomes pronounced and leads to figure-eight motion. These effects appear not to have been predicted and will be illustrated on videotape.
Separation of a one-dimensional bending wave field into propagating and standing parts based on the reflection coefficient estimation
Authors: Linjama Jukka , Lahti Tapio
Published: October 2005

The structural intensity methods offer a way of determining the absolute flow of mechanical power anywhere on the surfaces of some basic structures. In one-dimensional cases such as beams, these methods can be extended to the determination of mechanical impedance and reflection coefficient. In this paper, a separation of the bending vibration of the beam is suggested, which divides the field quantities into perfectly propagating and standing parts, based on the estimation of the complex reflection coefficient. Such a division is hoped to open an alternative view in quantifying the flow and distribution of power in beams. The technique is restricted to farfield regions of the beam. It is also directly applicable in other one-dimensional (vibrational, acoustical, etc.) fields. Properties of the suggested separation are illustrated with an experiment, and they are compared to an alternative approach that is based on the impedance. The relationship of the separated parts to the active and reactive intensity is discussed. It is shown that each of the field quantities involved in power transmission can be separated, and the propagating parts correspond to the net (active) intensity. The standing parts show properties that may be usable in modal analysis sense.
Design evolution of miniature electroacoustic transducers
Authors: Egolf David P. , Carlson Elmer V. , Madaffari Peter L. , Mostardo August F.
Published: October 2005

This presentation traces the evolution of the design of miniature electroacoustic transducers during the last 40 years. Of particular emphasis are those contributions to the research, design, and development of such transducers made by Hugh Knowles and his colleagues. Pioneering efforts in the miniaturization of receivers (i.e., small loudspeakers or earphones) and microphones made possible the transition of hearing aids from the bulky body-worn aids of the past to the tiny all-in-the-canal aids of today. Included in this presentation is an overview of his innovations in piezoelectric, condenser (including electret) and electromagnetic transduction technology. Mr. Knowles' last contribution was the research and development of the miniature transducer—a hearing-aid receiver—incorporating a special electronic circuit to reduce current consumption and extend battery life of a hearing aid.
Evaluation of a real-time adaptive-beamforming hearing aid
Authors: Greenberg Julie E. , Zurek Patrick M.
Published: October 2005

A real-time two-microphone monaural hearing aid, based on the constrained adaptive beamformer proposed by Griffiths and Jim [IEEE Trans. Antennas Propag. AP-30, 27–34 (1982)], has been implemented and evaluated. The beamformer adapts to preserve the target signal (assumed to be straight ahead) and to minimize the power of jammer signals arriving from all other directions. The basic Griffiths-Jim algorithm is augmented with a method to inhibit adaptation in the presence of strong target signals [Greenberg et al., J. Acoust. Soc. Am. Suppl. 1 85, S26 (1989)]. The real-time system employs the Motorola DSP56001 and was evaluated using a speech target and a single jammer for a variety of free-space and head-mounted microphone configurations in anechoic and reverberant environments. The system demonstrates very good performance (30- to 40-dB gain from input to output in an intelligibility-weighted spectral average of target-to-jammer ratio) for several broadside microphone configurations in the anechoic environment. Adaptation inhibition provides a robust insensitivity to target misalignment at high-input target-to-jammer ratios. As expected, performance degrades with increasing reverberation; at the critical distance, the system provides 5- to 10-dB gain. In extreme reverberation, the system performs at least as well as a delay-and-sum beamformer. [Work supported by NIH.]
The scattering of sound by a cylindrical bubble cloud
Authors: Korman Murray S. , Roy Ronald A. , Crum Lawrence A.
Published: October 2005

Preliminary experiments are preformed on the scattering of acoustic pulses by a cylindrical bubble cloud. The goal is to investigate the role played by experimental length scales in establishing the angular dependence of the scattered intensity. These length scales include bubble diameter, interbubble distance, acoustic wavelength, and cloud diameter. The bubbles (which are larger than resonance size) are much smaller than the wavelength in water, which, in turn, is smaller than the diameter of the cloud. Experiments are carried out in a 1-m3 tank using 250-kHz acoustic pulses generated by a 5-cm-diam circular plane array. An identical receiving array detected the scattered energy as a function of angle. The cylindrical bubble cloud is generated by a constant volume velocity airflow through a fritted ceramic disk located at the bottom of the tank. Results are compared with theories involving spherical clouds as well as homogeneous fluid cylinders with an “effective” compressibility and density. [Work supported by ONR.]
Acoustic scattering from a “Pierson-Moskowitz” sea surface
Authors: Thorsos Eric I.
Published: October 2005

An exact integral equation method has been used to study low-frequency (200 Hz) acoustic coherent and incoherent scattering from surfaces that vary in only one horizontal direction. The surface roughness spectrum is obtained from the Pierson-Moskowitz frequency spectrum for a fully developed sea. Comparisons are made with standard theoretical predictions using the Kirchhoff approximation, perturbation theory, and the composite-roughness model. The Kirchhoff approximation is found to be reasonably accurate for near-specular scattering and for the coherent reflection loss at grazing angles as low as 10°. Perturbation theory, when carried to order (kh)4 for the scattering cross section, is in essentially perfect agreement with exact results for bistatic backscattering. This is true for kh values as great as 1.8, where k is the acoustic wavenumber and h is the rms waveheight. The standard composite-roughness model is found to be accurate for backscattering, except at the lowest grazing angles, where it over-predicts the scattering. [Work supported by ONR.]
Nonlinear bias of travel time in modal ocean acoustic tomography
Authors: Shang E. C.
Published: October 2005

In a previous paper [E. C. Shang, J. Acoust. Soc. Am. 85, 1531–1537 (1989)], ocean acoustic tomography based on the adiabatic normal mode theory has been developed. Mesoscale structure can be inverted by using a linear relationship connecting the modal travel time perturbation and the sound-speed perturbation. This linear relationship is only an approximation due to the following two facts: (1) neglecting the difference between perturbed modal eigenfunction and the background model eigenfunction; (2) taking the first-order expansion of the “potential function” ΔK2 (z,r) to calculate the modal wavenumber perturbation. Comparing with ray tomography, item (1) is the term corresponding to the ray path perturbation. It has been found that item (1) is not significant and the problem caused by item (2) can be recovered by a “post-correction” procedure proposed in this paper. [Work supported by NOAA and ONR.]
Experimental results of sound propagation in a penetrable wedge with a fluid bottom
Authors: Glegg Stewart A. L. , House Ian
Published: October 2005

This paper will describe the results of a model scale experiment on sound propagation in a penetrable three-dimensional wedge with a fluid bottom. The wedge model is 1.2×2.4 m and has a Mylar-kevlar membrane that separates two fluids (water and ethylene gylcol) with sound speeds of 1500 m/s and 1700 m/s. The model is suspended in a large water tank so that reflections from the edges of the model are not significant. The experiment has concentrated on across slope propagation at several different downslope ranges, using a high-frequency pulse. The pulse spectra have been compared with Buckingham's theoretical model for sound propagation in a penetrable wedge. [Work supported by ONR.]
Numerical analysis of sound propagation with acoustic shock waves in varying area ducts
Authors: Zhu Z. C. , Zhang Y.
Published: October 2005

In recent results, several second-order finite difference schemes were adapted to obtain shocked solutions for sound propagation in varying area hard-wall ducts with transonic flow [N. J. Walkington and W. Eversman, J. Sound Vib. 104, 81–107 (1986)]. The second-order MacCormack scheme with a fourth viscous term was then recommended by the authors. But its numerical error still seemed to be a bit large. In this paper, the fourth-order MacCormack scheme with a fourth viscous term is used to improve the solution. It is shown from three sample calculations with acoustic shock waves that the new method is much better than the second-order MacCormack method. Its numerical error, on an average, is only one fifth of that in the latter method. Here the new method is depicted in detail, considering especially the treatment of boundary conditions, the selection of viscous coefficient, the stability condition, and the computation correctness criterion. This method needs less grid points and is more stable. It is also time-saving and storage-saving. Maybe it is a good method for calculating shocked solutions of sound propagation in ducts.
Focusing of ultrasound in solids with applications to QNDE
Authors: Harris John G.
Published: October 2005

Ultrasound can be focused by the curved fluid-solid interface of a solid part being interrogated for cracks or by using a lens as part of an imaging system such as the acoustic microscope. In the latter case it is usually difficult to form a diffraction-limited spot when the focus lies in the solid because the focused beam field must cross a fluid-solid interface. This paper summarizes recent investigations of these two situations. A calculation of the wave fields near the caustics formed when a two-dimensional (unfocused) beam scatters from a concave fluid-solid interface is discussed. Also discussed is a calculation of the wave fields within the focal region of a three-dimensional, wide aperture source when the focused beam is directed across a plane fluid-solid interface and the focal region lies in the solid. In both cases attention is paid to the interplay between the diffraction effects of the source and of the focal region, and also to the excitation of leaky surface waves. The implications of these considerations for the quantitative interpretation of ultrasonic measurements are explored.
Radiated fields determined from moments of surface velocity
Authors: Bennink David D. , Pate Anna L.
Published: October 2005

A quick and efficient method is derived for calculating the field radiated by a finite but otherwise arbitrary velocity profile. The method calculates the pressure field radiated into the half-space (z > 0) from the normal component of velocity on the boundary plane (z = 0). The derivation starts from the Rayleigh integral and utilizes a power series expansion of the kernel in the radial coordinate of the source point. As a result, a formulation is obtained that contains separate terms for the source and for propagation. The source terms represent the moments of the velocity profile. The propagation terms can be obtained from a recurrence relation starting from certain special cases for which analytical expressions are known. Convergence and stability of the method will be discussed, and examples of field calculations for a few simple profiles will be given. [Work supported by Center for NDE at Iowa State University.]
Behavioral thresholds for tone pips in 3- and 6-month-old infants and adults
Authors: Olsho Lynne Werner , Feeney M. Patrick , Folsom Richard C.
Published: October 2005

Infant behavioral thresholds for tone-pip stimuli at 1, 4, and 8 kHz (3-cycle rise-fall and 1-cycle plateau) were estimated using the observer-based psychoacoustic procedure (OPP). Behavioral thresholds for these stimuli were also obtained on a group of normal-hearing adults. Across frequency comparisons revealed that 6-month-old infant threshold curves paralleled adult curves, but infant thresholds were elevated 20 to 30 dB. For 3-month-old infants, OPP threshold curves were not parallel to adult curves due to relatively poorer infant thresholds at 8 kHz: Furthermore, in comparison with wave V auditory brain-stem response (ABR) thresholds for the same stimuli, 3-month-old infant behavioral thresholds are the same as ABR thresholds at 1 and 4 kHz, whereas behavioral thresholds are poorer at 8 kHz. A similar elevation in threshold at 8 kHz for 3-month olds has been reported for longer duration stimuli [L. W. Olsho et al., J. Acoust. Soc. Am. 84, 1316–1324 (1988)]. The results suggest a frequency-dependent maturation of behavioral thresholds between 3 and 6 months for these brief stimuli. [Work supported by NIH.]
Central processing in patients with cochlear implants
Authors: Daly D. M. , Owens F. D. , Stephenson P.
Published: October 2005

Because of locations and orientations of auditory cortex, studies of central processing in man typically exploit psychophysical phenomena or disease altered structure/functioning. Cochlear implants afford a view of central processing through sparse, specifiable peripheral transfer functions. Six patients were examined who have unilateral implants (cochlear 22 channel; 14–21 active electrodes in BP + 1; 4 AS, 2 AD implants; 2–100 weeks post-op) using prerecorded sets of BW, GY, BDG, and iɪe delivered directly through the processor [Daly et al., J Neurophysiol. 140 (2), 141-62 (1980)]. The representation of stimuli was sufficient for three patients to classify stop/glide sets consistently (p<0.001); although in a set which included both BW and GY none distinguished [be] from [gel or [we] from [ye]. Performance reverted to chance when the processor was configured as a single channel device and all stimuli had the same total duration. None distinguished among BDG. One patient classified vowels iɪe consistently; boundaries changed significantly when a central electrode malfunctioned and was inactivated in configuration map. Results suggest, with these stimuli, patients resolved brief temporal changes more accurately than brief frequency changes alone.
Electrotactile encoding and recognition of a 16-obstruent system
Authors: Piroth H. G. , Gfroerer S. , Tillmann H. G.
Published: October 2005

In continuation of a long-term training experiment [H. G. Tillmann and H. G. Piroth, J. Acoust. Soc. Am. Suppl. 1 84, S45 (1988)] the recognition of 16 obstruents in Cə syllables was tested using a modified quasiarticulatory coding method. Tactile syllable patterns were delivered by the 16-channel electrocutaneous system “SEHR-2” to 16 electrode pairs that were arranged to build four equidistant rings around the left forearm. Consonants were assigned to rings or pairs of rings in order to yield more than four places of articulation. Fricatives were coded as subsequent stimulations of four or eight electrodes (forming one or two rings, respectively) while plosives were represented by oscillating movements between the neighboring electrodes of a ring. The fortis/lenis distinction was coded by different pattern velocities. Since recognition with the initial coding of the two-ring fricative patterns as parallel and synchronous movements was poor, an alternative coding method was applied with contrarotating movements in order to avoid local masking or inhibition resulting from the stimulation of adjacent electrodes. Using this alternative coding method, one highly trained subject reached 85.6% correct identifications of the 16 obstruents.
Coding of phase structure in complex sounds by bullfrog auditory-nerve fibers
Authors: Simmons Andrea Megela , Ferragamo Michael
Published: October 2005

This experiment explored how the peripheral coding of the spectral and temporal structure of a complex communication signal is affected by the phase relations between individual frequency components in the sound. Complex sounds with 21 components, all of which were harmonics of a common, low-frequency fundamental (100 Hz), were digitally synthesized. The frequencies and amplitudes of the individual components matched those of the bullfrog's species-specific advertisement call. For one stimulus, the 21 components were generated in cosine phase, and for the other stimulus, the 21 components were in random phase. Most amphibian papilla (AP) and basilar papilla (BP) fibers synchronized to the envelope only of the phase-coherent stimulus, and the complex spectral structure of the stimulus was not accurately encoded by population firing patterns. Envelope synchronization to the random-phase stimulus was not seen in the responses of some mid-frequency-sensitive AP and BP fibers, although low-frequency-sensitive AP fibers responded similarly to both stimuli. Behavioral experiments suggest that bullfrogs can discriminate between in-phase and random-phase versions of their advertisement calls.
Neurophysiological observations on the auditory cortex in the FM bat, Eptesicus fuscus
Authors: Moss Cynthia F. , Fritz Jonathan , Ferragamo Michael , Simmons James A.
Published: October 2005

The echolocating FM bat, Eptesicus fuscus, integrates target range and shape (range profile) into unified images along a synthetic target-range axis. The response properties contained in single- and multi-unit recordings from the auditory cortex of Eptesicus in relation to the process of image formation revealed in psychophysical data were examined. A tonotopic map was found in primary auditory cortex similar to what has been described previously, but also a much larger auditory cortical representation than hitherto reported. Echo delay-tuned neurons are interspersed with frequency-tuned neurons, showing no apparent topographic organization. The response profiles of some echo-delay-tuned and delay-tracking neurons are dependent on the harmonic structure of the FM stimuli, as required for representing the unusual dual time- and frequency-domain images that the bat perceives. Strong suppression effects may sharpen perceived target range and be the basis for encoding the depth structure of sonar targets in Eptesicus.
ABER latencies are prolonged in rat pups fed a biotin deficient diet
Authors: Rybak L. P. , Whitworth C. , Scott V. , Bhardwaj B.
Published: October 2005

The effects of biotin deprivation on ABER were studied in rat pups fed a diet that lacked this compound. Pregnant rats were either fed a normal rat diet (controls) or a special diet without biotin (Teklad, Harlan Sprague Dawley, Madison, WI). After weaning, the respective diets were continued in pups from each group. ABER was tested under chloral hydrate anesthesia (380 mg/kg) using clicks (100 μs, 5/s) and tone bursts (3-ms duration, 1.5-ms rise-fall time at 2, 8, 16, and 32 kHz) presented free field with a driver suspended 8 cm above the vertex in the median plane. Responses were obtained using an active lead from the vertex with a reference electrode in the nose. Blood levels of biotin were measured using a microbiological assay. Animals on the biotin-free diet showed alopecia and dry, scaly skin. There was no significant difference in ABER threshold in control versus experimental animals. However, there was a significantly longer brain-stem transmission time (interpeak latencies from waves I to IV) in biotin-deficient animals. The lowering of serum biotin in experimentals was confirmed by microbiologic assay. These findings suggest that biotin deficiency in the developing rat adversely affects auditory-nerve and/or brain-stem development. These effects could be mediated by a delayed myelination of these structures. [Work supported by Deafness Research Foundation.]
Auditory-nerve discharge characteristics that distinguish between basilar papilla neurons and amphibian papilla neurons in the leopard frog, Rana pipiens
Authors: Ronken Don A.
Published: October 2005

The frog's inner ear has two sensory organs for auditory reception. One organ, the amphibian papilla (ap), has many of the complexities of mammalian ears. The other organ, the basilar papilla (bp), is much simpler and offers special opportunities for comparison to the mammalian inner hair cell configuration. These anatomical differences make it important to identify the site of the receptor cell when recording from individual afferent neurons in the main trunk of the auditory nerve. In routine experiments, this can be a difficult problem, since the characteristic frequency (cf) of the highest-frequency (ap) neurons overlaps the cf of bp neurons in many species. To approach this problem, a statistical data base was created using responses to tone bursts and clicks, tuning curve characteristics, and spontaneous rates. A standard statistical method, discriminant analysis, produced reliable classification of neurons without using cf estimates and offers the possibility of a fully automated decision procedure. [Work supported by NIH grants NS21592 and RR01379.]
Prior auditory stimulation influences the precedence effect
Authors: Litovsky Ruth Y. , Freyman Richard L. , Balakrishnan Uma , Clifton Rachel K.
Published: October 2005

Previous studies have demonstrated that echo threshold for clicks is elevated by a preceding click train. The present study examined the influence of a preceding click train for three variables presented in 16 combinations: click rate (1–16/s), number of clicks in the train (3–17), and duration of train (0.5–8.0 s). Subjects were tested in an anechoic chamber, with two speakers positioned at 45 deg left and right of midline. Leading clicks were always presented from the left speaker, while lagging clicks were presented from the right speaker with delays ranging between 2–14 ms. Each trial consisted of a click train followed by the test click. Subjects pressed a button when they heard the lagging test click on the right. Subjects' echo threshold was defined as the click delay at which the echo was reported on 50% of trials. Relative to an isolated test click, thresholds were elevated approximately 3 to 6 ms when preceded by a click train. The most critical variable influencing magnitude of shift in echo threshold was the number of clicks in the preceding train. For a constant number of clicks, duration of the train had little effect, whereas rate sometimes interacted with number. [Work supported by NSF.]
Is a power function description of intensity-jnd data compatible with the loudness function?
Authors: Hellman William S. , Hellman Rhona P.
Published: October 2005

A sequence of approximation for calculating loudness functions from the intensity-jnd function for a 1-kHz tone is performed. It is shown that a single power function representation of pure tone intensity-jnd data can generate a loudness function from a McGill-Goldberg type model in better agreement with loudness measurements than has heretofore been demonstrated. This improvement arises from the presence of an integration constant that had been previously ignored. The inclusion of the integration constant markedly improves the overall shape of the derived loudness function, somewhat more at low then at high intensity levels. Using the same near miss power function, the consequences of setting the integration constant equal to zero are indicated. These effects are compared to those obtained from the unaltered intensity-jnd data [W. S. Hellman and R. P. Hellman, J. Acoust. Soc. Am. Suppl. 1 82, S25 (1987)]. Corrections resulting from going beyond first order in the calculational procedure are discussed. [Partially supported by the Rehabilitation R&D Service of the VA.]