bluuebunny/crossref_metadata_2025 huggingface dataset:
A cognitive perspective on audio delay tolerance in telecommunications
Authors: Marturano Larry
Published: October 2005

A set of tests are described in which the effects of echo-free audio delay on PCM-encoded speech are evaluated. In these tests, a conversation-type subjective speech assessment technique is used, which is augmented with several objective measures of conversational performance. At the same time, a measure of mental workload is studied to assess the cognitive effects of audio delay on the human end user. The workload measure that is used is a secondary task involving short term, or working, memory. In describing these experiments, relevant background information is provided on audio delay, which illustrates the difficulties in measuring user acceptability of audio delay. Information on human memory models and key experimental findings are reviewed, which motivate the choice of workload measure used in this study. The test methodology and results are then reviewed in detail. User tolerance of audio delay is found to decrease with increasing audio delay, in accordance with other researchers findings. In addition, added cognitive load is demonstrated in the presence of long delays via decreased working memory performance.
Perception of multiple echoes
Authors: McCall Daniel D. , Costantino Brian L. , Clifton Rachel K. , Freyman Richard L.
Published: October 2005

Since Ebata etal. [ 537–541 (1968)], little experimental or theoretical work has been done on the perception of multiple echoes. In the current experiment, listeners seated in an anechoic chamber were presented with 4-ms noise bursts in a lead-lag-lag configuration. The lead burst was presented from a loudspeaker at 45° left of midline and one lag burst (lag A) was presented at midline. Listeners discriminated the location of a second lag burst (lag B) presented from a loudspeaker at either 35° or 55° right of midline. Keeping the delay between lead and lag B constant, a range of delays between lead and lag A was presented to determine whether changing the delay of lag A had an effect upon the perception of lag B. Presentation of lag A interfered with the perception of lag B over a wide range of lag A delays, such that subjects’ discrimination of the location of lag B was poorer when compared to the discrimination of lag B when lag A was absent. The data suggest that the introduction of an earlier echo serves to suppress directional information from a subsequent echo. [Work supported by NIH grant DC01625.]
Smart foam design and applications in active structural acoustic control
Authors: Gentry Cassandra A. , Guigou Cathy , Fuller Chris R.
Published: October 2005

Acoustic foams possess inherent high-frequency sound absorbing capabilities for radiation and reflection control of structures. Comparatively, active noise control systems generally do not perform well at high frequencies due to the increasing complexity of the physical model and the necessary high speed digital signal processing. The ‘‘smart foam’’ concept and design originates from the combination of these two control strategies formulating a passive/active noise suppression device that can efficiently operate over a broad range of frequencies. In this research, the experimental device is comprised of a layer of piezoelectric material embedded in a partially reticulated polyeurethane foam. Depending on the specific application, the distributed actuator alters the surface impedance of the passive absorber such that an incident acoustic wave is totally reflected or absorbed. Error sensor configurations include microphones or piezoelectric sensors integrated directly into the noise suppression device creating a compact passive/active system. Experimental results are presented for piston radiation control. An analytical model of smart foam is described and simulations are presented to illustrate the potential of the device and the interaction between the passive and active components.
Musical oscillator simulations: From curiosity-driven science to commercial product
Authors: Schumacher Robert T.
Published: October 2005

The first computer simulations of bowed string oscillations were done by McIntyre and Woodhouse in 1977. Generalizations to wind instruments followed within five years. The first commercial product using the principles of these simulations appeared in 1994. This talk will give a brief history of these developments, and will then concentrate on recent work on computer simulation of bowed string motion, much of which is presented in detail by other papers in this and an associated session. the relative success of bowed string simulations compared to simulations of wind instrument oscillations requires some explanation, which will be provided. The talk will conclude with a summary of the recent work by the author with James Woodhouse using fast computation to explore the enormous parameter space of the string instrumentalists’ world.
Evaluation of transients through observations and simulations of bowed strings
Authors: Boutillon Xavier
Published: October 2005

Despite their well-established importance on the perception of sound and their crucial role in music, transients are far less studied than steady-state regimes in musical instruments. How long a transient lasts is certainly a major question, among others: Is pitch variation noticeable? How different are transients from note to note, from one instrument to another? In order to investigate these questions, direct and reciprocal observations will be presented as well as various partial simulations: fully numerical simulations, tests with a computer-controlled bowing machine, simulations based on transfer functions. This variety of approaches is aimed at sorting out what could be attributed to the player, the bow, the string and the body of the instrument. The question of adequate signal processing or representation will be addressed and the relationship between the transient length and the quality of the instrument will be outlined. [Work supported in part by the French Ministry of Culture.]
Some aspects of bow resonances—Conditions for spectral influence on the bowed string
Authors: Guettler Knut
Published: October 2005

The resonant bow seems to some extent capable of modifying the power spectrum of the bowed string, even during a steady-state Helmholtz motion, where only one slipping interval occurs during a fundamental period. Longitudinal resonances in the bow hair, strongly coupled to the transverse vibrations of the stick, are being excited by the changes of the frictional force which occur during each individual period. In general, these velocity fluctuations—superimposed on the steady bowing velocity—show small amplitudes compared to those of the string under the bow. In certain frequency regions however, dependent on the bow/bridge distance, impedances, etc., their energy content is sometimes great enough to cause noticeable modification of the string velocity. Such spectral changes would at any rate be small, but may still bear some acoustic significance. The present analysis is based on measurements on a violin bowed by a computer-controlled bowing machine, and supported by computer simulations. [Thanks are extended to Dr. Anders Askenfelt for sharing his experience in measuring bows, and to The Swedish Royal Institute of Technology for the use of their bowing machine and other laboratory facilities.]
Mode frequency and damping changes due to chemical treatments of the violin bridge
Authors: Bailey Machele , Bissinger George
Published: October 2005

Hammer-impact modal analysis using a microphone as a no-load vibration transducer was performed on violin bridge blanks chemically treated two different ways to extract free–free vibrational mode frequencies, dampings and shapes for comparison with untreated bridges. Data were taken for both in-plane (IP) and out-of-plane (OP) vibrations over a 0–20 kHz range. Average effects of these chemical treatments on mode frequency and damping were determined from IP and OP single-point excitation on six Formalization (FA), six Resorcin/formaldehyde (RE) bridges, and eight untreated (UN) maple violin bridge blanks. All the bridges exhibited three well-defined IP modes and six OP modes. Few significant changes were observed between the frequency and damping parameters of the IP and OP mode shapes, although the RE treated bridge exhibited a possibly unique doublet at ∼9 kHz. Since no mode frequency variations ≳1.5% were found, even though the RE treatment increased bridge masses 9.1%, it was concluded that the combined mass/stiffness changes of the wood were related to the acoustic effects observed with these bridges [H. Yano and K. Minato, 1222–1227 (1992)].
Informational masking in the identification of simple auditory patterns
Authors: Kidd Gerald , Rohtla Tanya L. , Mason Christine R.
Published: October 2005

Listeners identified six auditory patterns comprised of eight sequential tone bursts [cf. Kidd etal., 2916(A) (1994)]. Several factors affecting performance were examined including center frequency, level and the size of the frequency steps forming the patterns. In masked conditions, a multitone masker was played synchronously with each pattern element. The components of the multitone masker were chosen randomly on every burst within a presentation interval. Masker components were excluded from a region of 20% of the pattern center frequency to minimize ‘‘energetic masking.’’ The results in quiet indicated that identification performance was nearly perfect for pattern frequency steps as small as 1% and at levels as low as 20 dB SL. Considerable ‘‘informational masking’’ was produced by the random multitone masker bursts. The masked identification-level functions were shallow, rising on average about 2% per dB. Several signal and masker presentation schemes were examined that were intended to reduce the observed masking. To date, the most effective means for improving signal pattern identification has been to present the signal to one ear and identical maskers to both ears with the masker leading in time (8 ms) in the ear contralateral to the signal. [Work supported by NIH.]
The influence of pattern-based frequency expectations on resolving power for components of multitone sequences
Authors: Kidd Gary R.
Published: October 2005

The effect of deviations from frequency expectations on resolving power for components of tonal sequences was studied with a method similar to that used earlier to determine the effect of temporal deviations [G. R. Kidd, 2966(A) (1994)]. In a same–different task, listeners were asked to compare the last two of four twelve-tone patterns presented on each trial. The first two patterns were identical and served to establish expectations from which the third and fourth patterns could deviate. In the third pattern, the frequency of a single ‘‘target’’ tone was lowered by either 3 or 7 semitones, or left unchanged. The frequency change to be detected was added to the target component in the fourth pattern on half the trials. Thresholds for this frequency increment were estimated for each deviation condition (as defined by the target-tone frequency used in the third pattern). Just as with deviations from temporal expectations, deviations from frequency expectations produced threshold increments that increased with the size of the deviation. These findings suggest that the superior discriminability typically observed with familiar patterns is due to the appropriate targeting of attention in frequency and in time during pattern presentation. [Work supported by NIDCD.]
Identification of the incremented component in profile stimuli
Authors: Gockel Hedwig , Colonius Hans
Published: October 2005

The ability to identify the frequency of the incremented component in a multitone complex was assessed in two conditions, both using the same 11-tone complexes (fixed frequency, equilog spaced, random phases). All components had equal amplitude, except one of the inner 9, which was incremented. The overall level was randomly varied on each presentation. In condition A the tone complex, with the increment position randomized, was followed by a sinusoid either at the increment frequency or at that of an adjacent component (below or above). In condition B the same complex with flat power spectrum was additionally presented on each trial. The sequence of flat and incremented profile stimuli was randomized over the first two presentation intervals, followed by the sinusoid. In both conditions subjects had to indicate whether the frequency of the incremented component was same or different from that of the sinusoid. Thresholds for these two conditions were estimated by the use of an adaptive procedure and compared to the thresholds of the same subjects in discriminating between the flat and an incremented profile (with signal uncertainty). Results indicate that identification ability is at least as good as discrimination. [Work supported by DFG.]
Effects of interaction of low- and high-frequency tone complexes on periodicity pitch perception
Authors: Krishnamurti Sridhar , Hawks John W.
Published: October 2005

The effects of interaction of low-frequency tone complexes and high-frequency tone complexes on the thresholds of discrimination (DLs) for a change in periodicity pitch in one of the two frequency regions were studied. Stimuli were digitally constructed by summing sine waves spaced at 100- or 200-Hz intervals. The amplitude of components of one of the interacting complexes was fixed while the periodicity pitch of the other complex was obtained by decrementing amplitude of components in 1-dB steps. DLs were obtained for (1) low-frequency complexes only, (2) high-frequency complexes only, (3) low-frequency complexes in the presence of high-frequency complexes, and (4) high-frequency complexes in the presence of low-frequency complexes. Preliminary results indicate elevated DLs for high-frequency complexes in condition 2 and elevated DLs (significant interference effects) in conditions 3 and 4. Interference effects were greater for condition 4 relative to condition 3. Implications in terms of temporal fine structure and envelope cues are discussed.
Temporal resolution and CMR can depend upon frequency
Authors: Bacon Sid P. , Lee Jungmee
Published: October 2005

An estimate of temporal resolution can be obtained by subtracting the signal threshold obtained in the presence of a modulated masker from that obtained in the presence of an unmodulated masker (yielding the modulated-unmodulated difference, or MUD). An estimate of comodulation masking release (CMR) can be obtained by measuring the MUD with both a broadband and critical-band masker. The present study obtained MUD and CMR for signal frequencies from 250 to 4000 Hz. The masker was unmodulated or sinusoidally amplitude modulated at rates from 2 to 16 Hz. For all rates and both masker bandwidths, the MUD increased monotonically with signal frequency; it was larger for the broadband masker, indicating an across-channel CMR. The CMR increased from a few dB at 250 Hz to 14 dB at 4000 Hz. The frequency effect (for MUD and CMR) was considerably reduced when the modulation depth (m) of the masker was decreased from 1.0 to 0.9, and was essentially eliminated at a depth of 0.5. The results of a forward-masking experiment using the unmodulated masker at both masker bandwidths suggest that the frequency effect is due to differences in the recovery of forward masking, and that suppression can influence CMR (and MUD) at very large modulation depths. [Work supported by NIDCD.]
Modeling modulation perception: Modulation low-pass filter or modulation filter bank?
Authors: Dau T. , Kollmeier B. , Kohlrausch A.
Published: October 2005

In current models of modulation perception, the stimuli are first filtered and nonlinearly transformed (mostly half-wave rectified). In order to model the low-pass characteristic of measured modulation transfer functions, the next stage in the models is a first-order low-pass filter with a typical cutoff frequency of 50 to 60 Hz. From physiological studies in mammals it is known that many neurons in, e.g., the inferior colliculus, show a bandpass characteristic in their sensitivity to amplitude modulation. Results from psychophysical studies of modulation masking also suggest some kind of bandpass analysis of modulation frequencies. Results of two experiments on modulation detection that allow discrimination between models incorporating a low-pass filter and those using a modulation filterbank are presented. In the first experiment, modulation detection thresholds were measured for noise carriers of bandwidths between 3 and 6000 Hz. In the second experiment, modulation detection for a sinusoidal carrier was measured in the presence of interfering modulation components with a bandpass characteristic in the modulation spectrum. The results from these experiments could not be simulated by a model including a modulation low-pass filter, but were successfully simulated by a model using a modulation filterbank.
Multiband detection of energy fluctuations
Authors: Grose J. H. , Hall J. W.
Published: October 2005

The filtering properties of the ear permit the auditory periphery output to be viewed as a bank of parallel quasi-independent channels. The purpose of this study was to examine how information contained among the independent channels is combined. Four related experiments were undertaken in which the detection of an energy change within a channel was measured as a function of the number of channels carrying that change. Each experiment examined the detection of a different signal type: (1) The detection of multitone signals masked by narrow bands of noise; (2) the detection of intensity increments (intensity discrimination) for multiple noisebands; (3) the detection of intensity decrements for the same stimuli as (2); and (4) the detection of temporal gaps in multiple noisebands. Eight normal-hearing listeners participated and performance was measured for a 1-, 2-, 4-, and 8-channel stimulus. The center frequencies for the 8 channels were 356, 494, 663, 870, 1125, 1442, 1838, and 2338 Hz. For each signal type, performance improved with increasing number of channels carrying the signal. The comparative patterns of improvement will be discussed in terms of multiband detection models. [Work supported by the NIDCD R01-DC01507.]
Synthesis of common-envelope signal pairs
Authors: Anantharaman Jayanth , Krishnamurthy Ashok K. , Feth Lawrence L.
Published: October 2005

A large class of sounds can be described in terms of rapid pressure fluctuations known as the ‘‘fine structure’’ and relatively slower amplitude changes called the ‘‘envelope.’’ Signals having the same envelope and occupying the same range of frequencies, but differing in their fine structure, are important to the study of fine structure perception. Voelcker’s [Proc. IEEE 54, 340–353, 735–755 (1966)] complementary two-tone complex pair is one of the simplest signals in this class, and has been used in a number of pitch related experiments. As an extension of these ideas, a computational algorithm is described to generate a common-envelope pair occupying the same range of frequencies of any bandlimited signal. Several properties of common-envelope signals are presented and methods of combining these signals to create other common-envelope signals are given. A few examples of multicomponent common-envelope signals are spectrally reversed linear-spaced complexes, spectral ramps, and multiple Voelcker two-tone pairs. Applications of these signals to psychoacoustic experiments are discussed. [Work supported by AFOSR.]
Identification of flat-spectrum vowels on the basis of amplitude modulation
Authors: Moore Brian C. J. , Alcántara Joseph I.
Published: October 2005

These experiments investigate whether amplitude modulation of ‘‘formants’’ allows vowel sounds to be identified. The periodic complex sounds had a fundamental frequency of 100 Hz and a level of 85 dB SPL. Harmonics were added either in cosine or random phase. Harmonics with frequencies below 3000 Hz were equal in mean amplitude. Three pairs of successive harmonics, located at the first, second, and third formant frequency values of six possible vowels, were sinusoidally amplitude modulated. For a 10-Hz modulation rate, vowel identification of cosine-phase stimuli improved with increasing modulation index, m, up to m=0.4, after which it reached an asymptote of about 80%. For the random-phase stimuli, performance was essentially at chance for all modulation depths, despite the fact that the amplitudes of the ‘‘formant’’ harmonics were greater than those of ‘‘background’’ harmonics at certain points of the modulation cycle. For a 2-Hz modulation rate, performance improved for the random-phase stimuli, but remained below that for the cosine-phase stimuli. Modulating the ‘‘formants’’ at different rates or modulator phases made performance worse for random-phase stimuli but had little effect for cosine-phase stimuli. The results are interpreted in terms of the cochlear-filtered waveforms and in terms of perceptual grouping mechanisms.
Seasonal acoustic propagation effect
Authors: Lundberg Wayne R.
Published: October 2005

Seasonal variations in wind and temperature gradients are known to influence the propagation of aircraft noise and sonic booms. This paper examines predictive techniques in current use for potential refinement and application to seasonal environmental impact statements. The effects of seasonally averaged refractive atmospheres were studied using sonic-boom propagation models which include refractive effects. The atmospheric environment was chosen to be that near Edwards Air Force Base where sonic booms are known to affect the public. Sonic-boom carpet predictions for a representative flight of an F-111 at Mach 1.3 and 10 000 ft above mean sea level were documented. The effect of such atmospheres on predicted noise exposure contours due to a real-world mix of supersonic flights was calculated using appropriate adjustments to the CORBOOM model. Two distinct atmospheric averaging techniques were applied for seasonal and annual comparisons. The resulting single-event and sonic-boom exposure predictions were compared to each other and to predictions obtained using the U.S. Standard Day Atmosphere and the Annual Reference Atmosphere for Edwards AFB. The results show the potential benefit of conducting seasonal acoustic predictions in forecasting the effects which low-altitude aircraft flight operations have on the public.
Application of model experiment data to test sonic-boom propagation models
Authors: Lipkens Bart
Published: October 2005

Model experiments are successful in simulating the propagation of sonic booms through the atmospheric turbulent boundary layer. In the model experiment spark-produced N waves propagate across the turbulent flow field of a plane jet. The N waves are measured by a wide band condenser microphone and the turbulence is measured by hot-wire anemometry. The results from the model experiment, pressure waveforms and turbulence characteristics, are used to test various sonic-boom models for propagation through turbulence. Both waveform distortion models and rise time prediction models are tested. Only Pierce’s model [A. D. Pierce, 906–924 (1971)] based on the wavefront folding mechanism at a caustic yields a fairly accurate prediction for the rise time of the mean waveform after propagation through the turbulence. Results from other models are not confirmed by the model experiment data. [Work supported by NASA Langley.]
Comparison of computer codes for propagation of sonic booms through the atmosphere
Authors: Chambers James P. , Bass Henry E. , Raspet Richard , Cleveland Robin O. , Hamilton Mark F. , Blackstock David T.
Published: October 2005

NASA initiated an exercise to compare sonic-boom propagation codes. Three different codes developed at our two universities are compared here: SHOCKN (Mississippi), THOR (Texas), and ZEPHYRUS (Texas; developed by L. D. Robinson). Each code includes nonlinear steepening, absorption and dispersion (relaxational and thermoviscous), geometric spreading, and atmospheric stratification. All three codes calculate steepening in the time domain. The main difference is in the calculation of absorption and dispersion. SHOCKN and ZEPHYRUS use the frequency domain; the FFT is used to transfer back and forth between the time and frequency domains. ZEPHYRUS employs weak shock theory to reduce the number of FFT operations. All of THOR’s calculations are done in the time domain. Code accuracy and effectiveness is judged by comparing ground level predicted waveforms (input waveforms are specified by NASA). The codes are capable of dealing with arbitrary atmospheric data. Results are presented here for uniform and isothermal atmospheres, with and without the effects of molecular relaxation. Early comparisons revealed discrepancies, which led to improvement of codes. Current comparisons show excellent agreement. Ground waveforms show surprising sensitivity to input stratification data. Some comparisons of code running time have been made. The FFT operations apparently make SHOCKN and ZEPHYRUS run more slowly than THOR. [Work supported by NASA.]
Psychophysics of speech and speech-like stimuli
Authors: Houtgast Tammo
Published: October 2005

The starting point is the representation of speech in the spectro-temporal domain. From this representation, a number of interesting physical characteristics can be derived on the dynamic properties of speech. These include the temporal and spectral envelopes, correlation patterns, and synchrony. The perceptual relevance of these characteristics will be illustrated by reviewing a number of psychophysical data from experiments including speech-like sounds of specific laboratory-made dynamic stimuli. Within the context of low bit-rate coding and automatic recognition, the ultimate goal of this comparison between speech characteristics and psychophysics is to help identify the cues in the speech signal which are perceptually significant.
Speech perception, memory, and cognition: Implications for automatic speech recognition systems
Authors: Kuhl Patricia
Published: October 2005

Recent experiments suggest that people form perceptual representations for speech as a result of experience with a specific language. These representations are argued to take the form of perceptual maps stored in memory that specify distances between speech stimuli. This point will be illustrated with data from cross-language experiments on American and Japanese adults using the segments /r/ and /l/. The results show that the underlying perceptual space encompassing /r/ and /l/ varies greatly in American and Japanese listeners—their perceptual maps are dramatically different. Cross-cultural developmental studies show that language-specific speech representations are present early in life and differ in infants reared in different linguistic environments. Further work suggests that the perceptual maps resulting from speech experience not only influence auditory speech perception, but influence auditory-visual speech perception in adults and speech production in infants. Implications for automatic speech recognition systems will be discussed. [Work supported by NIH.]
Automatic speech recognition using signal processing based on auditory physiology and perception
Authors: Stern Richard
Published: October 2005

Signal processing for automatic speech recognition has traditionally been inspired more by models of speech production than by models of auditory perception. While some aspects of human auditory processing have been implicit in traditional signal analysis for speech recognition, there is growing interest in the development of more computationally demanding signal processing strategies that are directly motivated by knowledge of auditory physiology and perception. The use of physiologically motivated signal processing has been shown to improve the accuracy of some automatic speech recognition systems, particularly in difficult acoustical environments. This talk will review and discuss the role that knowledge of human hearing has played in the design of speech recognition systems. The common features of auditory models that have emerged from various laboratories as well as their differences will be discussed. The recognition accuracy obtained using auditory models will be compared with the accuracy obtained using conventional signal analysis techniques, as well as the accuracy obtained using other approaches to robust recognition that are not physiologically based. Finally, an attempt will be made to identify aspects of monaural and binaural auditory processing that appear to be most helpful for robust recognition. [Work supported by ARPA.]
Meeting the challenges of the Americans with Disabilities Act: Providing access for people with communication disabilities
Authors: Williams Jo
Published: October 2005

The American with Disabilities Act (ADA) is significant not only as landmark civil rights legislation, but also because it is a turning point in public policy regarding the importance of communication and communication access. In the past, accessibility policy has focused on physical/mobility concerns and largely ignored needs related to sensory/cognitive impairments. More than 34 million Americans of all ages—approximately one of every ten people—have hearing, speech, and/or cognitive language impairments that affect their ability to work, learn in school, and participate in everyday activities. ADA regulations address communication access in employment, community programs and businesses, transportation, and telecommunications. Strategies for communication access and employment accommodations vary with the individual/disability and the type of activity/business. Most individuals need strategies that optimize auditory and visual information exchange, including improvements in environmental conditions (such as reduced ambient noise levels). New approaches and rethinking are needed regarding (1) assessing communication (dis)ability and needs, (2) the concept of accessibility, and (3) the role of professionals, including interdisciplinary collaborative efforts, in ADA implementation.
Room acoustics needs of people with hearing impairment
Authors: Nabelek Anna K. , Nabelek Igor V.
Published: October 2005

Speech perception of people with hearing impairment is negatively affected to a greater degree by background noise and reverberation than the speech perception of people with normal hearing. As a consequence, rooms which have adequate acoustics for normal-hearing people might be too noisy or too reverberant for hearing-impaired users. Causes of perceptual difficulties of hearing-impaired listeners and data comparing effects of noise and reverberation on speech perception of normal-hearing and hearing-impaired listeners will be presented. The number of people with hearing loss grows with age. Two major causes are noise exposure and aging processes. People with hearing loss can obtain improvement in speech communication with hearing aids but poor room acoustics can limit hearing benefits and cause hearing aid rejection. Advantages of acoustical modifications, preferential sitting, and use of amplification in small rooms will be discussed. Listening systems such as induction loops, FM, and infrared will be considered as alternatives or additions to acoustical modifications of large rooms. The listening systems can be utilized by hearing aid users and by people who do not use hearing aids, even those with normal hearing.
Evaluation of accessibility needs in a home for the aged
Authors: Pichora-Fuller M. Kathleen
Published: October 2005

Audiological rehabilitation has traditionally focused on the hard-of-hearing individual. In contrast, the needs of groups who perform specific activities in specific locations must be evaluated in planning facility-based accessibility programs. Such evaluations are necessary when planners design programs where solutions may be achieved by a combination of strategies including modification of the environment, provision of personal or institutional assistive technology, or communication training. The present study demonstrates how accessibility needs were evaluated in a home for the aged. For a group of 30 residents, scope of participation in activities and quality of communication during activities were measured four times over two years, twice before and twice after program implementation. In a pilot study, groups of residents and staff identified situations at the home where they felt that hearing was important. The evaluation of communication function was keyed to these situations. The patterns of scope and quality of communication were used to formulate an accessibility plane. Baseline and outcome data regarding the extent to which the program accomplished accessibility will be presented. For example, the program rendered meetings and chapel services accessible to most residents whereas situations such as dining in the dining hall remained inaccessible. [Work supported by NHRDP.]
Acoustical design of the New York railway stations with improved accessibility for hearing and visually impaired persons
Authors: Nichols Robert , Henning Gerald
Published: October 2005

An acoustical design of the New York railway stations is proposed to improve the acoustical accessibility for persons with hearing and visual impairments. The acoustical design includes methods and materials for reducing reverberation times in the stations. Reducing reverberation times increases the capability for auditory communication by: (1) lowering interfering reverberant noise levels generated by people and trains, and (2) increasing the clarity of speech from people and audio systems providing railway information.
Influence of impulse noise on speech transmission index
Authors: Mijić Miomir , Popović Miodrag
Published: October 2005

This paper analyzes the influence of impulse noise impairments in a telecommunication channel on the speech transmission index (STI) value, which was measured using RASTI method. So far STI has not been applied in telecommunications as a speech transmission quality measure. Impulse noise components were formed as IEEE standard impulse streams. Impulse noise amplitude level relative to the level of reference (RASTI) signal as well as the duration between successive impulse events were varied in a channel. Consequently, STI value was determined as a function of these two parameters. In the same channel PB-Word Score evaluation was also performed. As a result, the correlation between PB-Word Score and STI value for a telephone channel with impulse noise impairments present was established in this paper.
The internal tide of the western North Atlantic observed using long-range reciprocal acoustic transmissions
Authors: Dushaw Brian D. , Worcester Peter F. , Cornuelle Bruce D. , Howe Bruce M.
Published: October 2005

Time series of ray travel times were obtained at 350-, 410-, and 670-km ranges in the western North Atlantic during the 1991–2 Acoustic Mid-Ocean Dynamics Experiment (AMODE). Transmissions were recorded for approximately 300 days between six transceivers in a pentagonal array. Sound-speed variability is separated from current variability by calculating the sum of reciprocal travel times. Both phase-locked and narrow-band internal-tide sound-speed variability, caused by the internal-tide isotach displacement, are observed by the sum travel times. The description of the acoustic array as a high-directivity antenna for the incident internal tide will be given. Beamforming the internal tide M2 and K1 harmonic constants determined on each of the 15 acoustic paths suggests two origins of significant internal tide generation are the Blake Escarpment and the continental shelf north of Puerto Rico. The nontidal, high-frequency variability (≳1 cpd) is due to internal-wave sound-speed and current variability.
Scattering of acoustic tomographic signals in shallow water by internal waves
Authors: Lynch James F. , Jin Guoliang , Traykovski Peter , Chiu Ching-sang , Miller James H.
Published: October 2005

This talk examines how internal waves in shallow water cause travel time fluctuations in acoustic tomographic transmissions. Ray and mode theory expressions are developed and calculations are compared to travel time fluctuation data taken from the 1992 Barents Sea Polar Front experiment. While there is basic agreement, there is still much room for improvement of the basic theory and experiments. Directions of further research in both theory and experiment are discussed. [Work supported by ONR.]
Numerical simulation of the sensitivity for the temperature remote acoustics control in Fram Strait environmental conditions
Authors: Naugolnykh Konstantin A. , Johannessen Ola M. , Esipov Igor B. , Ovchinnikov Oleg B. , Tujilkin Yury I. , Zosimov Viktor V.
Published: October 2005

An important role of the general problem of global climate changes is the heat inflow from the Atlantic Ocean to the Arctic basin control. The effective tool to perform it is the acoustic methods application. Some aspects of this problem connected with the average water temperature measurements in some cross section of the Fram Strait are considered. It is supposed that the temperature change takes place in the upper 200-m layer of the ocean which is typical for the winter Fram Strait environmental conditions. Variations of the amplitude and the arrival time of the acoustic signals during their propagation in the Fram Strait cross section with variable upper layer mean temperature were considered on the base of the ray theory approximation. The results of computer simulation of the process are presented.
A full field approach to acoustic tomography of oceanic currents
Authors: Godin Oleg A. , Mikhin Dmitry Yu.
Published: October 2005

Traditionally it is nonreciprocity in travel times along identified pairs of spatially close eigenrays that serves as input data for a current’s velocity inversion. In shallow water many of the rays are bottom-interacting and cannot be resolved or identified. Hence the traditional approach results in a very poor or nonexistent resolution in the vertical plane [D. S. Ko etal., J. Geophys. Res. 94, 6197–6211 (1989)]. To avoid these problems it is suggested that one match nonreciprocity of acoustic continuous waves measured at a set of points, e.g., at a vertical array. Analytic studies indicate that nonreciprocity of the phase of a cw signal is sensitive to the flow velocity but does not depend, to first order, on fluctuations in sound speed or on uncertainties in transceiver location and, therefore, can be used as input data for currents inversion. Respective matched-phase nonreciprocity tomography (MPNT) is simulated numerically, including effects of acoustic SNR, systematic and random environmental mismatches, number of transceivers in the array, uncertainty in their positions, etc. and compared to other possible inversion schemes. It is concluded that MPNT could be a practical tool for monitoring ocean dynamics in shallow water. [Work supported by NRC and RBRF.]
Acoustic footprints of an interthermocline eddy: Field experiment and numerical simulations
Authors: Mikhin Dmitry Yu. , Aleynik Dmitry L. , Burenkov Sergey V. , Chepurin Yury A. , Selivanov Viktor G. , Molchanov Sergey Ya. , Godin Oleg A.
Published: October 2005

A moving ship tomography experiment (MOST) was carried out in the western Mediterranean basin in 1994. Six moored transceivers deployed in the area by IfM (Kiel, Germany), IFREMER (Brest, France), and WHOI (Woods Hole, USA), in the framework of the THETIS-2 project, were utilized as wide-band sound sources for the MOST. An interthermocline eddy with cold core was found during the MOST. Detailed CTD surveys were performed along several sections of the eddy. In the horizontal, it was close to an ellipse with axes of 25 and 40 nm. The sound-speed difference at given depth inside and outside the eddy was up to 2 m/s. Within the eddy core the sound-speed gradient was an order of magnitude less than in surrounding waters. Acoustic signals from the transceivers were recorded at 16 points within and around the eddy in order to study feasibility of reconstruction of mesoscale inhomogeneities of this type by acoustic tomography means. A technique of coherent signal processing used made it possible to resolve multiple eigenray arrivals from four transceivers at each point of observations. Results of mathematical modeling of the acoustic effects of the eddy and their comparison with the experimental data will be reported. [Work supported by ISF and INTAS.] a)On leave from P. P. Shirshov Oceanography Institute, Moscow.
Active control of noise and vibration: Overview and current applications
Authors: Fuller Chris R.
Published: October 2005

In this introductory presentation the principles and components of active control systems will be briefly overviewed. Current research topics and areas of application will be discussed including duct noise, machinery isolation, wave control in beams, interior noise in aircraft, and structurally radiated sound. The use of adaptive materials in active noise and vibration control will also be touched upon. The presentation will finish with a discussion on the present limitations of active control and future promising directions.
Active control: New paradigms for structural acoustics and dynamics
Authors: Tucker Albert J.
Published: October 2005

Originally the concept of active control conveyed the simple notion of creating ‘‘antinoise’’ or ‘‘antivibration;’’ these ideas implied that simple antiphase motions could be propagated to cancel unwanted noise. However, recent advances in signal processing, coupled with gigaflop digital signal processors, augur new ways to understand and control complex dynamic systems. Among these are shifting wave numbers from radiating to nonradiating components and modifying structural impedances to create large dynamic masses, stiffnesses, and/or damping. Hence one electronically modifies the dynamics of complex systems to achieve acoustic radiation reduction, or to have a physically lightweight structure emulate the dynamics of a heavier, stiffer structure. This fundamental scientific understanding offers new opportunities for development of new technology for control algorithms, sensors, and actuators. As a result, active control is emerging as viable addition to the engineering toolkit for noise and vibration reduction.
Simple intensity type fiber optic vibration sensors and sensor systems
Authors: Carome Edward F. , Kubulins Vilnis E.
Published: October 2005

Research is continuing on the development of simple intensity type fiber optic vibration sensors suitable for use in hostile environments, such as automotive and aircraft engine cavities. The transduction element consists of a single or a pair of optical fibers mounted on a cantilever reed. The intensity of the optical output signal is directly proportional to the reed displacement and, therefore, to acceleration at frequencies well below the reed fundamental. Using readily available materials an operating temperature range of −40 ° to +225 °C has been achieved. In addition, by employing simple multiplexing techniques, small groups of these sensors may be addressed with a single optical source and/or detector. The basic design of the sensors and the results of extensive laboratory tests are presented in detail. [Work supported in part by NASA.]
An innovative probe microphone design for use in high-frequency sound field measurements
Authors: Connelly Sean M. , Franzoni Linda P.
Published: October 2005

There is a need for nonintrusive measurement devices. When the wavelength of the sound wave is physically the same scale as that of the microphone, its sound pressure cannot be measured accurately. Acousticians have worked around this difficulty by using smaller microphones and, when necessary, using probe tubes with even smaller diameters attached to small microphones. This need for very small diameter devices is especially critical at high frequency. The use of probe tubes introduces another level of complexity to acoustic measurements. The resonant dynamics of the tube itself must be taken into account. The tube supports standing waves that cause the microphone frequency response to be very irregular. Typically, damping material, such as foam or steel wool is introduced into the probe tube to attenuate standing waves. However, even when damping is successfully applied, the response remains irregular and the damping material results in a significant reduction of signal-to-noise ratio. The new concept is designed to eliminate standing waves without introducing damping in the signal path between the microphone and the end of the tube. Therefore, the signal-to-noise ratio will not suffer significant degradation, and behavior will be consistent from probe to probe.
Use of a background noise correction
Authors: Staiano Michael A.
Published: October 2005

A correction scheme has been proposed which consists of repetitive measurements of the source-signal-with-background and background noise alone than the computation of a signal estimate and prediction interval. The procedure assumes that both the source of interest and background noise are: uncorrelated, normally distributed, random processes which are stationary over the duration of the measurements. For useful results, the numbers of measurement must be selected to provide for a calculated confidence interval which acceptably contains the prediction errors. These requirements are strongly influenced by the variability of the measured parameters. For relatively low background noise situations, the technique is useful primarily for quantifying expected measurement confidence bounds. Application of the procedure to a field measurement situation with mean source band sound levels ranging approximately from 2 to 15 dBA re: mean background band sound levels gave a 99.5%-confidence upper-bound margin of 0.3–1.3 dB with 17–19 measurements per band.
Precision of reference sound source hemianechoic sound power calibration using a fixed microphone array: Relation to ISO 6926 revision
Authors: Campanella Angelo J.
Published: October 2005

Reference sound source calibration by the hemianechoic ISO 3745 method allowed a course array of 10 measurement microphone positions on a hemisphere. Draft revision of ISO 6296 provides for a difficult meridional arc constant vertical velocity microphone scan. An improved array of multiple fixed microphone positions is proposed as an alternate to the meridional scan method. Bias of the improved array was simulated with Mathcad 4.0. Agreement was predicted to be within 0.2 dB up to 4 kHz for 10 points and up to 8 kHz for 20 points. Experimental results for three microphone position arrays were compared to reverberation room results over the limited frequency range of 125 to 8000 Hz. The standard deviation of the difference was 0.84 to 0.89 dB (0.81 dB predicted) for the old ISO 3745 array. For an array having 10 fixed height difference positions from 0.15R to 0.95R, the standard deviation was 0.62 dB (0.61 dB predicted). For 20 fixed heights from 0.025R through 0.975R the standard deviation was found to be 0.63 (0.18 predicted). Calculated results and experimental test results, compared with reverberation room results, will be presented.
Measurement of acoustic properties of materials at low frequencies in a water-filled tube
Authors: Kenney Debra M. , Rogers Peter H.
Published: October 2005

Two of the traditional methods for measuring the complex reflection coefficient of a material, the standing-wave tube method and the pulse tube method, each have shortcomings when the measurements are to be made in a water-filled tube at low frequencies. An acoustic impedance tube has been designed to overcome these shortcomings. The design uses a transient incident signal as opposed to a cw signal in order to minimize the problem of vibrations in the tube wall. It also uses a dual sensor system [D. Guicking and K. Karcher, Trans. ASME: J. Vib. Acoust. Stress Reliability Design 106, 393–396 (1984)] to separate the incident signal from the reflected signal without them being temporally separated. Active impedance control at the end of the tube opposite from the samples reduces the reflection from that end. The tube has been designed so that ambient pressure and temperature can be controlled during measurements to simulate an ocean environment. The analytic design of the tube and preliminary experimental results will be presented. [Work supported by ONR and NSF.]