bluuebunny/crossref_metadata_2025 huggingface dataset:
Residential sound attenuation—A quest for quiet
Authors: Murphy Deborah W.
Published: October 2005

While aircraft noise annoyance does not cause a negative impact on public health, it does affect public welfare. Airports around the country are constantly looking for innovative ways to reduce the impact of airport operations on their neighbors. The Airport’s Residential Sound Attenuation Pilot Project is one of several ways the airport has played an active role in the community, to help improve their neighbors’ quality of life. The Pilot Project was an initial demonstration which introduced the concept of sound attenuation to residents of homes located between the DNL 65- and 75-dB noise contours. Implemented as part of the Airport’s FAR Part 150 Noise Compatibility Program, the Project was designed to provide costing information, design guidelines, and construction methodology for a possible continuing program in which all eligible homes could be sound attenuated. The overall results of the program were extremely successful, providing four homes with a substantial reduction of interior noise levels due to aircraft overflights. The Pilot Project was successful not only as a demonstration of sound attenuation effectiveness, but also as a public relations program in that it showed evidence of the Airport’s commitment to address noise as an issue of importance to the community.
Noise distribution and reduction in industrial enclosures
Authors: Rybak S. A.
Published: October 2005

Noise is one of the main emission factors in industry, and a tendency of production progress is such that further increase in noise emission should be expected. In practice, various methods for noise reduction at its propagation path are used such as reduction of the reflected energy fractions, and proofing of a certain zone with a screen. Both acoustic designing of rooms and noise reduction, require the data on acoustic characteristics of materials and constructions, and sound power of sound sources. Methods of determination of these characteristics are based on the measurements carried out in special rooms with specified acoustic properties—reverberation, ‘‘dead’’ and ‘‘half-dead’’ rooms. Such rooms are also used for other purposes, for example, for generating sound fields of high levels with special uniformity of frequency and spatial spectra needs when testing instrumentation and response constructions for acoustic strength. At present, several methods of representation of the sound field in rooms have been developed: An energy method, a method of acoustic mirages, and a wave theory. In practice, the most widely used method is the energy method which has been considerably modified in this report.
Investigation of the ocean acoustic signatures from strong explosions at a long distance in the ocean sound channel by computer simulation
Authors: Kamegai Minao , White John W. , Clarke Douglas B.
Published: October 2005

The identification and location of ocean acoustic signatures are the principal objectives of a program to discourage clandestine testing of nuclear explosives. Difficulties arise primarily from variations in the water column. In turn, these variations affect acoustic propagation in the SOFAR channel. In this study, the path effects on the signals generated by strong explosions (1 and 10 kn) are investigated. The goal is to make a quantitative correlation between the initial source description and the final acoustical signatures received at a great distance under various conditions. The study is performed entirely by computer simulations applying two computer programs in succession. First, the explosions are simulated by a 2-D hydrodynamic computer program, CALE, which was originally developed to calculate astrophysical problems. The computed signals have reached more than 700 m deep approaching the SOFAR channel. At this point, the CALE output is linked to a hydro-acoustic computer program, the NPE code, by which wave propagation in the SOFAR channel is modeled. The NPE code was developed at the Naval Research Laboratory to study ocean acoustics. [Work supported by the U. S. Department of Energy under Contract No. W-7405-ENG-48.]
Jovian acoustic tomography
Authors: Collins Michael D. , McDonald B. Edward , Kuperman W. A. , Siegmann William L.
Published: October 2005

Recent work in ocean acoustic tomography indicates that acoustic waves propagating global distances contain planetary-scale information [A. B. Baggeroer and W. Munk, Phys. Today 45(9), 22–30 (1992)]. The expected collision of the fragments of comet Shoemaker–Levy 9 with Jupiter in July [J. K. Beatty and D. H. Levy, Sky & Telescope 87(1), 40–44 (1994)] provides an opportunity to explore the use of techniques developed in ocean acoustics on a different planet. The nonlinear progressive wave equation [B. E. McDonald and W. A. Kuperman, J. Acoust. Soc. Am. 81, 1406–1417 (1987)] can be used to model the propagation of shock waves near the impact sites. Adiabatic modes can be used to determine horizontally refracted propagation paths on an oblate spheroid [Heaney etal., J. Acoust. Soc. Am. 90, 2586–2494 (1991)]. Normal-mode and parabolic equation (PE) models can be used to obtain two-dimensional (range and depth) solutions, which illustrate geometric dispersion and cycle distances. The adiabatic mode PE [M. D. Collins, J. Acoust. Soc. Am. 94, 2269–2278 (1993)], generalized to include horizontal wind fields, can be used to determine the horizontal distribution of energy and the locations of caustics. Available input data (from the Voyager missions and other sources) for making pre-impact predictions include wind velocity as a function of latitude, the locations of the impacts, and information for estimating sound speed as a function of depth.
Directional hearing in the goldfish (Carassius
Auratus)
Authors: Lewis Thomas N. , Rogers Peter H.
Published: October 2005

Directional hearing in fish is a complex, ambiguous, and poorly understood phenomenon. Some nonostariophysian fish, particularly the cod, have shown the ability to unambiguously determine the direction of a monopole sound source and that their angular resolution is perhaps as good as 10°. Although considered ‘‘hearing specialists,’’ there is little evidence that ostariophysian fish are capable of directional hearing. An attempt to classically condition the ostariophysian goldfish to discriminate sound sources differing only in location failed. The sound signal was pulsed bandpass filtered noise at least 20 dB above previously determined sound pressure masked thresholds. Ostariophysian fish are characterized by the presence of Weberian ossicles, coupling the air filled swim bladder to the inner ears. This may enhance pressure sensitivity at the expense of directionality. This apparent lack of directional hearing capability will be examined in light of the known ability of goldfish to show directionality in their Mauthner mediated acoustic startle response and the physiological data demonstrating directionality in the individual auditory end organs. [Research supported by ONR.]
Diffraction and chaos for sound propagation through ocean internal waves
Authors: Simmen Jeffrey , Wang Guang-yu , Flatté Stanley
Published: October 2005

The effect of the dynamical ocean on the propagation of sound impulses over long ranges (exceeding several hundred kilometers) is an area of renewed interest, since there is a growing appeal for the use of acoustical tomography as a means of measuring or monitoring the ocean. This presentation will emphasize recent numerical results describing the effects of ocean internal waves on long-range pulse propagation. Elements of the presentation will include the nature of wave front folding as a function of propagation range, the chaotic behavior of rays, as well as its sensitive dependence on average sound-speed profile, and the diffraction due to internal waves. (Comparisons between solutions to the parabolic wave equation and its semiclassical ray counterpart will illustrate the nature of the diffraction.)
High-frequency acoustic propagation measurements during solitary wave events on the eastern continental shelf edge
Authors: Levine Edward R. , Shell Richard R. , Medeiros Michael R.
Published: October 2005

High-frequency environmental acoustics studies were conducted during July 1993, on the continental shelf edge east of New Jersey. Internal solitons previously observed in this region near the shelf/slope front propagate in packets, usually in the summer seasonal thermocline, and have been associated with anomalous low frequency sound propagation. Acoustic pings were collected using a towed sled instrumented with sonar arrays. Synoptic measurements to characterize the solitons including sound velocity profiles sampled every 10 min over a tidal cycle, and moored data including current, temperature, and conductivity. Acoustic measurements were taken during sled tows parallel to the bottom bathymetry, normal to the propagation direction, over a region determined from bottom cores to be nearly homogeneous fine sand. Measurements were taken using the sled as a source for backscatter measurements, and also using moored acoustic sources and the sled based transducers as receivers. The observed solitons had amplitudes of approximately 10 m and periods of several minutes. The backscatter variability during soliton events was observed to approximately 10–20 dB, and will be compared to modeled predictions based on environmental data.
Acoustic transmission of Taiwan’s Northeast Sea
Authors: Chen Chi-Fang , Lee Tai
Published: October 2005

The Northeast Sea of Taiwan is characterized by the turning of the Kuroshio current and mixing of water masses from the same current, East China Sea, and Taiwan strait. Its bottom topography consists of the shelf and the shelf slope. Kuroshio export exchange program (KEEP) was conducted by oceanographers from Taiwan to study the complexities of this region. An acoustic study was also included. An acoustic transmission experiment was conducted via fixed receivers and a towed source emitting cw tone of 170 dB re; 1 μPa, 1 m at 400 Hz along three predetermined tracks in the region. The tracks are of flat bottom and slope bottom. The ambient noise in this region is high; its level reaches 80 dB or higher at 400 Hz. The region is basically a bottom-limited acoustic environment. A PE model was used with an acoustic model constructed by real data of water column and bottom material. The numerical results agree well with the experimental data. The acoustic effect of a cold eddy in the region was studied numerically using a PE model with KEEP oceanography data. A cold eddy of 50 km in diameter along 200-km track was studied. The eddy causes 5 dB more losses in acoustic transmission.
Temporal modulations of pulse response amplitude in shallow water resulting from a combination of sound-speed variability and bottom loss
Authors: Monjo Charles L. , Deferrari Harry A.
Published: October 2005

Reciprocal transmission tomography experiments in the Florida Straits have yielded a month long time history of reciprocal pulse responses over three ranges approximately 22 km in length and 250 m in depth. A transmitted pulse of 8.75 ms in duration is spread out approximately 100 ms. In a previous paper a strong correlation was reported between the sound-speed gradient at middepth and the total pulse width. A plausible explanation, that the modulation of the sound-speed profile was causing a temporal modulation of the ray arrivals was set forth as a hypothesis for this study to be examined with range-dependent ray and PE models. The models predict a different and unexpected relation. Variations in the sound-speed profile shape from linear to convex increase the bottom grazing angle to the early arrivals, while decreasing the bottom grazing angles of the slower arrivals. This results in an attenuation of the early arriving portion of the pulse to below ambient noise and therefore an apparent narrowing of the pulse. Likewise, a change in profile shape from linear to concave, produces smaller bottom grazing angles for the fastest arrivals, less loss, and an overall wider appearing pulse. [Work supported by ONR.]
Computer simulation of drop behavior in an acoustic levitation chamber
Authors: Bhargava Neil A.
Published: October 2005

This simulation was written in order to describe the behavior in two dimensions of a liquid droplet inside the drop physics module (DPM). The DPM is a chamber which uses standing sound waves in order to levitate and position a liquid droplet placed within it. The translational motion of the liquid droplet in this acoustic field can be described by a second-order differential equation which is integrated in order to obtain droplet position information. Stokes’ viscous drag on the droplet’s translational motion is incorporated. Due to the reliance on the use of standing waves, any fluctuations in the resonance frequency are monitored and evaluated. The slow temperature increase in the DPM will change the resonance frequency of the chamber (temperature detuning) and is also taken into account. The resonance frequency is also affected by the existence of the inclusion and by its position within the chamber (volumetric detuning.) The simulation includes an analytic expression which describes this phenomenon. Since the DPM is designed for use in the microgravity environment of an orbiting space shuttle, calculations are also adjusted to account for behavior specific to the shuttle motion. [Work supported by Jet Propulsion Laboratory, Contract No. 958722.]
Computer graphic simulations of microscopic particle collisions for determining macroscopic thermodynamics along a path
Authors: Espindle Devin P. , Korman Murray S.
Published: October 2005

The heat absorbed by an ideal monotomic gas during an isothermal expansion—when a piston slowly moves from volume point V1 to V2 in a cylinder—is a standard problem in thermodynamics. Students taking the SP226 Heat, Light and Sound course learn that the process involves (a) an understanding of the kinetic energy loss that a gas molecule makes with the slowly moving piston ‘‘called the bunt,’’ (b) the drop in internal energy of the gas as a collection, which leads to a slight drop in temperature of the gas, and (c) the transfer of heat from the reservoir to the gas until an equilibrium temperature is reached. This problem is modeled in two dimensions on a computer. The collisions involve N elastic scatterers that each have a cross section of radius r0. The heat resevoir is modeled by molecules having a random distribution of velocities at the wall boundary which is in contact with the gas. The piston is modeled as a wall which moves at the speed Vwall. Thermodynamic calculations involving work, heat and entropy are attempted for the model system. Adiabatic and free expansions are also simulated and compared with the well-known theoretical results. Work is extended to cover collisions involving intermolecular potentials.
Behavior of an air jet interacting with a Helmholtz resonator
Authors: Kottke Jason A. , Cottingham James P.
Published: October 2005

The internal spectrum of a Helmholtz resonator excited by an air jet was studied as a function of jet speed for a variety of jet parameters. Special attention was given to the relation between jet speed and amplitude of the Helmholtz mode, which shows a pattern of characteristic ‘‘domains’’ first reported by Khosropour and Millet [J. Acoust. Soc. Am. 88, 1211–1221 (1990)]. The effects of variation of air-jet length on major features of these domains were studied. The deduced behavior of the air jet was compared with previous models, in particular that of Fletcher and Thwaites [Acustica 42, 323–334 (1979)]. Good agreement with the predictions of this model was discovered except at high jet flow rates.
Speaking rate and associated kinematic patterns of tongue movements
Authors: Lu Henry Y. , Perkell Joseph S. , Matthies Melanie L.
Published: October 2005

Kinematically based measures of effort for tongue movements were examined for a speaker with a cochlear implant. Turning the speech processor of the implant on- and off-induced changes in the subject’s speaking rate. An electromagnetic midsagittal articulometer (EMMA) was used to track movements of the tongue body toward and away from the steady-state target for the /u/ vowel in the utterance ‘‘...who hid...’’. Although the tongue body movements toward the /u/ were quite small—on the order of a millimeter—kinematic measurements for those movements still exhibited relationships that were strikingly similar to results of past findings, which have shown strong relations between movement distance and duration. Similarly, values for velocity profile shape (the ratio of peak velocity to average velocity) appeared to follow increases in movement duration. According to the measures of peak velocity and energy, the most ‘‘effort’’ for these small movements was exerted in the condition when the cochlear implant had been off for the longest time. Examination of the number of velocity peaks and of tongue body trajectories for each utterance revealed that ‘‘smoother’’ and less effortful movements appeared with greater frequency when the cochlear implant was on. [Work supported by N.I.D.C.D.]
Phonetic structures of an endangered language: Montana Salish
Authors: Flemming Edward
Published: October 2005

Montana Salish, or Flathead, is an Interior Salishan language spoken by about 70 people on the Flathead reservation in Northwest Montana. This study utilized acoustic and aerodynamic data to examine three aspects of this language: (1) Like many other Salishan languages, Montana Salish permits extremely complex consonant clusters: initial sequences of five or more consonants are possible. (2) In addition to contrasts between plain and ejective stops, there are contrasts between glottalized and nonglottalized sonorants (lateral, nasals, glides, and pharyngeals). In Montana Salish, glottalized sonorants typically involve a glottal constriction early in the sonorant, i.e., they are pre-glottalized, rather than being produced with creaky voice throughout. (3) Montana Salish pharyngeals provide an interesting contrast to Semitic pharyngeals, in that they are very vocalic in character, sounding much like low back vowels. However, three distinguishing characteristics of these sounds have been identified: a pharyngeal constriction, resulting in F1 raising and F2 lowering, lowered fundamental frequency, and a change in voice quality. However there is considerable variability and not all of these properties are observable in all instances of these sounds.
Speech recognition with minimal spectral cues
Authors: Shannon Robert V. , Zeng Fan-Gang , Wygonski John , Kamath Vivek , Ekelid Micheal
Published: October 2005

Speech recognition was measured in conditions that systematically reduced the amount of spectral information while preserving temporal envelope information. Speech stimuli were spectrally separated into several frequency bands. The temporal envelope in each band was extracted by half-wave rectification followed by low-pass filtering. Each resulting envelope was then used to modulate a noise band with the same bandwidth and cut-off frequencies as the original analysis band. Identification of consonants (16 consonants in aCa context), vowels (8 vowels in hVd context), and words in simple sentences (CUNY sentences) was measured as a function of the number and frequency distribution of analysis bands, the envelope filter cut-off frequency, and overall spectral shaping. Results as a function of the number of channels show that consonant recognition improves from one to two channels, but less improvement is observed from two to four channels. Vowel recognition improves significantly from one to three channels. Sentence recognition improves with the number of channels, approaching 100% correct with four channels. These results indicate that relatively little spectral detail is sufficient for recognition of speech. Results will be discussed in terms of speech processing strategies for cochlear implants. [Work supported by NIDCD.]
Phonetic structures of endangered languages: Observations and findings
Authors: Maddieson Ian , Ladefoged Peter
Published: October 2005

A project to describe the phonetic structures of selected endangered languages was outlined in the preceding abstract. Under this project, data have been collected on a number of languages from East Africa, the Indian subcontinent, North America, Taiwan, and the South Pacific, and further fieldwork is planned. The analysis of these data continues to demonstrate that the languages of the world display greater diversity in their phonetic structures than many linguists anticipate. Illustrations will include Dahalo, a Cushitic language with lateral ejective affricates of unusual type and clicks that are invariably nasalized; Toda, a Dravidian language with an unusually rich inventory of consonants confined to code position, including three different coronal trills (each of which may also be palatalized); and Iaai, and Austronesian language with an explanded set of vowels and an unusually large set of voiceless sonorants. Without knowledge of the kind provided by our research, linguists and speech scientists would be in danger of assuming that the phonetic patterns known from familiar languages demonstrate the full range of sounds that can be utilized in possible human languages. [Work supported by the National Science Foundation.]
ETCvérif, a prototype of a cooperative automatic speech recognition system
Authors: Cochard Jean-Luc , Langlas Philippe , Chollet Gérard
Published: October 2005

ETCvérif a speech processing environment under development at IDIAP, serves as a testbed for experimental design issues in the development of robust and efficient speech recognition applications. This works stems from the strong intuition that a probable solution to the general problem of speech understanding lies in the development of a system able to deal cooperatively with a large set of distinct, partial and even unreliable problem solvers, namely HMMs, NNs, APD (acoustic phonetic decoder), prosodic analyzers, and parsers. This paper will present the general architecture of ETCvérif focusing upon the main properties of ETC (environment for cooperative treatment). In particular, the core of the system is devoted to execution scheduling of speech and language processing modules; communication and negotiation over partial, compatible but not unifiable results. The design decisions regarding the requirements of cooperation between (modules such as) phonetic HMMs, an APD with lexical access and a bottom-up parser will be described as part of a prototype of speech recognition presently under realization. [Work supported by the Swiss National Funds for Scientific Research.]
Study of phonetic classification/recognition performance on data collected with different microphones
Authors: Chang Jane , Zue Victor
Published: October 2005

The goal of this study is to seek an understanding of the effects of microphone variations on the MIT segment-based speech recognition system, summit. Specifically, phonetic classification and recognition performance are evaluated on utterances extracted from the timit corpus. The timit corpus offers phonetically-transcribed and time-aligned data for three different microphones—a Sennheiser close-talking, noise-canceling microphone, a Bruel and Kajar (B&K) far-field pressure microphone, and a telephone handset (plus channel distortion). These transducers cause different convolutional, additive, and bandwidth effects in the speech waveform. Experimental procedures are established to measure and analyze system performance under variable training and testing conditions. Classification uses Gaussian models on a feature vector consisting of Mel-frequency cepstral coefficients and their time derivatives, plus duration. The experiments show that performance in phonetic classification and recognition degrades from the Sennheiser (27% classification error) to the B&K (29%) and the telephone (43%). Performance further degrades when training and testing conditions are unmatched. Closer examination reveals that these degradations are concentrated in specific phonetic classes. For example, confusions between voiced and unvoiced fricative pairs account for a large percentage of the additional errors when training on the Sennheiser and testing on the B&K.
A database of environmental sounds
Authors: Boyce Suzanne E. , Goldhor Richard
Published: October 2005

A database of 272 common environmental sounds, recorded for use in the development of a recognition system is described. These recordings represent 5–8 repetitions of 50 individual types of sound. The repetitions span a variety of recording conditions such as microphone positions, location of the source in the recording space, etc. The types of sound include airplanes, sirens, whistles, pouring small objects, telephones, motors, household appliances, footsteps, animal vocalizations, and speech. Simultaneous near- and far-field recordings were made of each sound. Unlike many commercially available collections, our database was produced via a consistent format of equipment use, equipment type, standardized microphone positions, etc. all of which has been exhaustively documented. Beyond those features mentioned above, the database includes information on the proportions and salient features of the recording spaces, the dimensions and salient features of the sources, and details of the recording and computer entry process. Currently, the database is comprised of four parts: (1) digitized audio waveforms; (2) information and notes on recording setup, recording space, source type, equipment, etc.; (3) sketches indicating shape and proportions of the recording setup, and (4) photographs showing sound sources and recording setups. This information is available in machine-readable format on a personal computer. [Work supported by NIH.]
Three-dimensional effect on modal travel time
Authors: Wang Y. Y. , Shang E. C. , Lee Ding
Published: October 2005

In a previous paper [D. Lee etal., J. Acoust. Soc. Am. 91, 3192–3202 (1992)], a three-dimensional effect on acoustic wave propagation has been examined by using a propagation model with azimuth-coupling capability (FOR3D). It has been found that a significant three-dimensional effect on transmission loss (TL) takes place at the Atlantic shelf area with bottom topography variations given by the Harvard open ocean model (HOOM). In this talk, some results are reported of the investigation of the three-dimensional effect on modal travel time under the same oceanic environment by using the scheme of modal spectrum of the PE field (MOSPEF).
Island diffraction and other three-dimensional effects in global acoustics
Authors: Collins Michael D. , Heckel Scott P. , McDonald B. Edward , Heaney Kevin D. , Kuperman W. A.
Published: October 2005

The adiabatic mode parabolic equation (PE) is an efficient approach for solving three-dimensional ocean acoustics problems [M. D. Collins, J. Acoust. Soc. Am. 94, 2269–2278 (1993)]. This method is used to solve the outgoing wave equations for the mode coefficients, which depend on range and azimuth. The range- and azimuth-dependent wave numbers appearing in these equations correspond to the eigenvalues. The adiabatic mode PE is applied to solve global-scale problems involving diffraction around islands and other three-dimensional effects. The significance of azimuthal coupling is illustrated by comparing solutions generated both with and without azimuthal coupling. The coupling azimuth solutions predict that the shadow zone behind an island is wider than uncoupled azimuth solutions predict. Since the modal phase velocities increase as ocean depth decreases, this result is consistent with the expected result based on ray theory. Adiabatic mode PE solutions are in qualitative agreement with observations from the Perth–Bermuda experiment [Heaney etal., J. Acoust. Soc. Am. 90, 2586–2594 (1991)] and the Heard Island Feasibility Test [Baggeroer and Munk, Phys. Today 45(9), 22–30 (1992)]. a)Permanent address: Planning Systems, Inc., McLean, VA 22102.
State estimation using ATOC measurements: A numerical study
Authors: Menemenlis Dimitris , Wunsch Carl
Published: October 2005

A numerical study carried out in anticipation of acoustic thermometry of ocean climate (ATOC) measurements is described. The aim of the study is to combine long-range tomographic measurements with numerical models and other apriori information in order to improve the ocean state estimate. A 4-yr integration of the Semtner and Chervin general circulation model is assumed to represent the real ocean. The integration was performed with a nominal resolution of a 1/4 deg in the horizontal and 20 layers in the vertical. Sound-speed profiles along the proposed North Pacific ATOC paths in conjunction with canonical spectra for the scales not resolved by the model are used to generate simulated tomographic data. The estimation problem is addressed using a lower resolution primitive-equation model of the North Pacific. The second model is assumed sufficiently realistic that perturbations from the Semtner and Chervin ocean may be treated as linear. A time-independent state transition matrix is computed for large-scale density anomalies about the mean of the second model. The estimation problem is then solved using a Kalman filter and the resulting state estimate is compared with the Semtner and Chervin ocean.
The sensitivity of global acoustic measurements to climate change
Authors: Dzieciuch Matthew A.
Published: October 2005

An ongoing experiment, Acoustic thermometry of ocean climate (ATOC), measures the integrated heat content of path through an ocean basin by recording the travel time perturbations of low-frequency sound waves. Anthropogenic CO2 in the atmosphere is expected to heat the oceans. Many physical processes affect the travel time signal in addition to the greenhouse signal. A partial list includes: internal waves, tides, mesoscale eddies, seasonal and interannual variability. A relevant question is: What is the signal-to-noise ratio of the CO2 warming signal versus the ambient signal? A synthetic experiment was therefore run to estimate the acoustic signature of the warming signal. Two coupled ocean-atmosphere general circulation models were run with no CO2 increase and with CO2 doubling. The GCM output was then used as an input to an acoustic propagation model. The acoustic time series can then be analyzed for climatic trends. The control run is used to estimate the ambient noise processes on long term scales. Shorter time scale processes can also be quantified and signal-to-noise ratios can then be constructed. The spatial properties of the CO2 signal are nonuniform and therefore maps of the signal strength are also constructed.
Mechanisms causing special features of ambient noise statistics in shallow water
Authors: George Jacob
Published: October 2005

Special features of ambient noise statistics, distinct from signal statistics, have been previously reported [J. Acoust. Soc. Am. 94, 1820(A) (1993)]. A possible mechanism for these features had been identified through simulation and from normal mode theory as depth fluctuations of a noise source close to the surface. Since then, the cumulative effect of multiple sources of this kind has been investigated following the techniques used by Dyer in analyzing distant shipping noise [J. Acoust. Soc. Am. 53, 564 (1973)]. The results show that while a single source assumption does fit experimental results, assumption of multiple sources does not. Because several merchant ships were known to be present during the data collection, it is unlikely that ambient noise was dominated by a single source. This suggests that additional mechanisms such as wind-generated noise may also be responsible for the observed features. The usefulness of the single source formula as an empirical fit to data, and as an important potential contributor to the observed statistical features will be discussed. [Work supported by ONR, with technical management provided by NRL/SSC.]
Recovering task-dynamic parameters from formant frequency trajectories: Results using computer ‘‘babbling’’ to form indexed files
Authors: McGowan Richard S.
Published: October 2005

Two indexed files of acoustic-articulatory pairs were created: one with 11 156 entries and another about ten times as large. Each individual entry was generated by running the Haskins articulatory synthesizer with randomly generated task-dynamic parameters. Each individual consisted of acoustic data in the form of the first three formant frequency trajectories and articulatory data consisting of the corresponding task-dynamic parameters. Each individual was indexed with keys denoting the direction of change in each formant frequency and the amount of change within 10% intervals. A genetic algorithm was used to recover the task-dynamic parameters from the formant frequency trajectories of four synthesized utterances. One recovery condition had a random initial population and the others used indexed files. Two factors were studied: the size of the indexed file and the criterion for an individual to be including into the initial population. One criterion was that the direction of formant change had to match that of the data, and the other was that the amount of formant change also had to match. The results were that using the larger indexed file provided faster convergence, but the results on matching criterion were ambiguous. [Work supported by NIH Grant No. DC-01247.]
Components of integrality in the perception of voice quality and tongue root position
Authors: Thorburn Rachel , Macmillan Neil A. , Kingston John
Published: October 2005

Vowels in this study varied in voice quality and tongue root position. Crosslinguistically, a lax voice quality typically co-occurs with advanced tongue root, whereas a tenser voice quality cooccurs with nonadvanced tongue root. Both articulations concentrate energy at low frequencies: laxing the voice increases the relative prominence of the first harmonic, and advancing the tongue root lowers F1. The Garner paradigm was used to test, for vowels in CVC context, whether these acoustically similar effects also integrate perceptually. Detection theory was used to assess the degree and type of perceptual integration. Meanintegrality (the mapping of harmonic amplitude and F1 onto a common perceptual dimension) was inferred by comparing two-stimulus classification tasks in which stimuli differed on either one or two dimensions. Varianceintegrality (an increase in perceptual uncertainty) was inferred by comparing classification and identification tasks with and without irrelevant variation. The source of variance integrality (sensory or context memory [Durlach etal., Percept. Psychophys. 46, 293–296 (1989)]) was determined by comparing the two-response Garner tasks with conditions using more than two stimuli and responses. [Work supported by NSF and NIH.]
Perception of stops by Norwegian listeners
Authors: Halvorsen Berit
Published: October 2005

This paper is part of a larger study on timing relations in Norwegian stops. The goal of this study is to describe the effects of vowel context, place of articulation, and speech type on stop perception. CV continua /b/–/p/, /d/–/t/, and /g/–/k/ in the vowel contexts /e/ and /u/, in modified natural speech (MNS) and synthetic speech (SS) were presented to 36 native Norwegian listeners for identification and discrimination. The pooled identification data from the speech types did not correspond well for the labial continuum +/u/ for the dental/alveolar continuum in both vowel contexts. The results were subjected to a five-factor ANOVA; controlling for speech type (natural and synthetic), vowel quality (/e/ and /u/), place of articulation (labial, dental/alveolar, velar), order of presentation of speech type (natural/synthetic or synthetic/natural), and listeners’ dialect (Tro/nder, Eastern Norwegian, and Bergen). All factors showed main effects, except dialect. Speech type and vowel quality interacted with the remaining factors. The results correspond well with VOT in spoken Norwegian [see Halvorsen, J. Acoust. Soc. Am. 91, 2388 (1992)]. [Work supported by the Norwegian Research Council and the University of Bergen.]
Voice variability effects on phonetic processing
Authors: Wannemacher C. A.
Published: October 2005

A growing body of research has begun to explore the relationship between the processing of phonetic information and that of talker voice information. To this end, the present work employed a series of experiments designed to further investigate potential phoneme and voice integrality in the acoustic-phonetic coding process. In a preliminary experiment, listeners rated the similarity of pairs of tokens from several female talkers. Stimuli chosen from this corpus were used in subsequent selective attention experiments to assess interference in phonetic processing under conditions of high and low vocal similarity. Phonetic classifications were made for isolated vowels, nonword CVs, and word CVs. As a further constraint of voice variability, multiple tokens from a single talker were also used in a selective attention task. The results will be discussed with respect to the nature and degree of voice-phonme integrality as a function of vocal variability, as well as to issues of talker normalization in speech perception. [Work supported by NIDCD Grant No. DC-000219 to SUNY at Buffalo.]
The role of voice pitch in the perception of glottal stops
Authors: Houde Robert A. , Hillenbrand James
Published: October 2005

Glottal stops that occur in VCV context are often not realized as stops at all, but rather show voicing that is continuous throughout the ‘‘occlusion’’ interval. Glottal stops that are realized in this way are apparently marked by reductions in intensity and fundamental frequency. In the present study recordings were made of speakers producing glottal stops in utterances such as ‘‘oh–oh’’ in which a glottal stop separates two identical vowels. The utterances were resynthesized in three ways: (1) with a flat F0 contour and an intensity contour matching the original utterance, (2) with a flat intensity contour and an F0 contour matching the original utterance, and (3) with intensity and F0 contours matching the original utterance. Results suggest that either cue alone is sufficient to signal the presence of the glottal stop. The impression of most listeners was that the signals cued by intensity alone were quite similar to those cued by F0 alone. The apparent equivalence of loudness and pitch in signaling the presence of a glottal stop would seem to present difficulties for a straightforward auditory account of phonetic recognition.
The representation of voice information in spoken word recognition: Differential effects of repetition in lexical decision and recognition
Authors: Luce Paul A. , Lyons Emily A.
Published: October 2005

Much recent research [e.g., Goldinger (1993)] has suggested that representations of spoken words in memory are veridical exemplars that encode specific information, such as characteristics of the talker’s voice. This exemplar account of phonetic form representations was examined in an experiment in which subjects heard two lists of words. The words in the first list were spoken by both a male and a female speaker. The words in the second list were (1) repetitions of words in the first list spoken by the same talker, (2) repetitions of words spoken by a different talker, or (3) new words. Subjects always made lexical decision responses to the stimuli in the first list. For the second list, subjects in one condition again made lexical decision responses; subjects in another condition performed a recognition task. Results indicated that changing the voice of the talker across lists had no effect on the repetition priming effect when subjects made lexical decisions to the second list. However, in the recognition task, reaction times were faster if the voice of the talker was preserved. These results are consistent with an exemplar-based account for recognition but also implicate the role of more abstract representations in identification. [Work supported by NIDCD.]
Spoken language identification with phonological and lexical models
Authors: Kadambe Shubha , Hieronymus James L.
Published: October 2005

A language identification (LID) system that uses phonemotactic models in addition to phoneme models to identify languages is described. The proposed LID system is trained and tested using the OGI multilanguage telephone speech database. The continuous density second-order ergodic variable duration hidden Markov phonemic models are trained for each language using a high accuracy phoneme recognition system developed at Bell Laboratories. The phonemotactic models for each language are trained using a text corpora of about ten million words and grapheme to phoneme converters. The language Li of an incoming speech signal x is hypothesized as the one that produced the highest likelihood f(x‖λi)f(λi‖Li) for all the phonemic models λi of a given set with the phonemotactic constraint. Initially, this LID system was trained and evaluated for English/Spanish language identification and the language identification was 83% correct (79% on English and 88% on Spanish). Results for four languages will be presented. The discriminative power of this LID system can be improved by mapping the phoneme lattice onto a syllable or a word sequence using a lexical analyzer and a trigram syllable or word language model The language identification results with and without interfacing the lexical analyzer will be presented.
An articulatory-feature-based HMM for automated speech recognition
Authors: Erler Kevin , Freeman George H.
Published: October 2005

Traditional HMM-based speech recognizers model the acoustics as a set of disjoint segmental units. Little meaning is attached to internal states of the model and there are difficulties in representing context-dependency effects observed in the acoustic waveform. In this work, an articulatory-feature model is defined as an HMM in which each state represents a point in a space of quantized articulator configurations. A single large ergodic model then contains all feasible configurations and is sufficient to represent any vocabulary. Each word in the vocabulary is described by a sequence of target articulatory configurations, currently derived from the phonetic transcription. Context dependency is explicitly handled by those states visited between articulatory-target states. Each model state has some meaning in terms of the physical state of the vocal production process, so linguistic and physiological knowledge can be incorporated to restrict the model evolution and improve recognition performance. The development of the quantized articulatory space, target articulatory-feature sequences, and feature-evolution constraints is described for a large-vocabulary word recognition system. Recognition results show it to be competitive with the traditional phoneme model. [Work supported by NSERC, Canada and ITRC, Ontario.]
A stochastic framework for articulatory speech recognition
Authors: Ramsay Gordon , Deng Li
Published: October 2005

One of the major difficulties in incorporating knowledge of speech production contraints into speech recognition lies in the problem of adequately characterizing the relationship between an articulatory description of speech and its lexical and acoustic counterparts, and in developing procedures for recovering such an articulatory description from acoustic input. In this paper, a new stochastic framework for articulatory speech recognition is presented aimed at addressing these issues. Utterances are described in terms of overlapping gestural units that are built into a Markov state structure. Each gestural combination is identified with a set of acoustic/articulatory correlates embodied in a targetdistribution on an articulatory parameter space, while articulatory motion is represented by a stochastic linear dynamical system whose parameters and input are indexed on the Markov state. A piecewise-linear approximation to the articulatory-acoustic mapping derived from an explicit production model transforms the articulatory distribution into acoustic space. The key innovation of the model is that articulation is described probabilistically as the response of a system to a sequence of target distributions produced by a gestural sequence. Derivation of the articulatory-acoustic mapping is discussed, and algorithms are proposed for recognition and training based on Kalman filtering and an approximate EM procedure.
Speaker verification via nonsupervised and supervised neural nets
Authors: Homayounpour M. Medhi , Boyanov B. , Chollet G.
Published: October 2005

A text-independent speaker verification approach based on nonsupervised (SOM=self-organizing map of Kohonen) and supervised LVQ3 algorithm [T. Kohonen, Proc. IEEE 78, 1464–1480 (1990)] is proposed. During the training phase a general prototype SOM is formed by using the input vectors of 20 speakers from a polyphone-like telephony database. Individual SOMs for every speaker are built by passing the vectors of each speaker through the general SOM prototype. Every individual SOM is fine tuned by means of the LVQ3 algorithm. In the verification phase, a short parametrized speech interval of 3-s duration from the speaker under verification is compared to both the general prototype and the speaker’s individual SOM. So two accumulated distortions are calculated. The decision to accept or reject a speaker depends on whether the speaker’s voice matches to the claimed speaker’s individual SOM or to the general prototype SOM. The MFCC, LPCC, and LSP coefficients were used in these studies as distinctive features. Personal SOMs were trained using 100 s of speech from each speaker. A conventional LBG algorithm was also applied. LVQ3 initialized by the SOM codebook was always more efficient than LVQ3 initialized by the LBG codebook using all three kinds of above-mentioned features.
Gemination of Italian stops
Authors: Rossetti Riccardo
Published: October 2005

An acoustic study of geminated and nongerminated Italian stops was carried out. The analysis was performed on 648 VCV syllables, where V is one of the vowels [i, a, u] and C is one of the consonants [p, t, k, b, d, g] (in geminated and nongeminated form). The syllables were uttered by 6 speakers, 3 repetitions. The analysis of the segments durations corresponding to the different articulatory phases of the VCV structure showed a strong correlation between the presence of gemination and the first vowel and occlusive silence durations. Statistical analyses using an aposteriori classifier yielded to a classification score of 96.1%. An analysis of the properties of the burst (energy, power, and spectral shape) showed that there is no correlation between these and the presence of gemination. An automatic software system, which allows the modification of a single/geminated VCV utterance into a geminated/single one, was developed. Informal perceptual tests, in which the stimuli were obtained by using the previous program, validated the significance of the first vowel and the occlusion durations as acoustic correlates of gemination.