bluuebunny/crossref_metadata_2025 huggingface dataset:
Conditioned pitch change and pitch adaptation
Authors: Peters Robert W. , Hall Joseph W.
Published: October 2005

In previous studies we have reported long-lasting changes in the pitch of a complex sound following its association with a sound of a different pitch. The present study concerned whether a sound altered in pitch would show pitch adaptation effects commensurate with its pitch or frequency status. The test stimulus was the sixth, seventh, eighth, and ninth harmonics of F0 196 Hz. The conditioning tone was the first five harmonics of F0 204 Hz and the adapting stimulus was the first five harmonics of F0 200 Hz. Stimuli F0 196 and 204 Hz were heard in association for 30 min each day for 6 days. This resulted in a shift of F0 196 Hz to a pitch near 204 Hz. The results indicated a pre-conditioning adaptation of the pitch of F0 196 Hz downward and a post-conditioning upward shift. While the pitch change was in process and near 200 Hz, no adaptation occurred. The results will be discussed with reference to plasticity of the auditory system.
Interaction between spectral and periodicity pitch perception in a songbird
Authors: Cynx Jeffrey
Published: October 2005

Six European starlings were trained to discriminate between upper harmonics of 400 Hz (2400–3600 Hz) and upper harmonics of 652 Hz (1304–3260 Hz). Analysis of the acquired discrimination showed that all birds maintained the discrimination on the basis of whether the signals contained frequencies below 2400 Hz. There was no evidence of discrimination on the basis of periodic information. The results parallel earlier work [S. H. Hulse and J. Cynx, J. Comp. Psychol. 99, 176–196 (1985)] showing that songbirds have a bias to discriminate between frequency information on the basis of spectral cues—a kind of absolute pitch. Stimuli then were added that produced the same 400- and 652-Hz periodicities, but contained new harmonics. The starlings transferred the discrimination to these new stimuli, now showing evidence for periodicity pitch perception. Thus, although starlings (and a number of other species of songbirds) appear to be predisposed to using spectral information, they can discriminate between complex frequencies on the basis of other information if necessary. [Work supported by NIH and NSF.]
An articulatory model
Authors: Baer Thomas , Rubin Philip
Published: October 2005

In this paper, we will review the principle features of the articulatory model implemented at Haskins Laboratories [P. Rubin, T. Baer, and P. Mermelstein, J. Acoust. Soc. Am. 70, 321–328 (1981); P. Mermelstein, J. Acoust. Soc. Am. 53, 1070–1082 (1973)], and we will discuss its design considerations. In particular, the Haskins model is intended to identify the key articulators and their linguistically relevant control dimensions, not necessarily to accurately reproduce physiological details. Two steps in the modeling process will be focused on: the transformation from articulator “positions” to midsagital shape, and the transformation from midsagital shape to area function. We will discuss the present implementation of the model and proposed refinements. Issues to be dealt with include the degrees of freedom associated with control of the articulators, and the collection of data to improve width to cross-section rules. Design of control structures for synthesizing continuous articulations will also be discussed. [Work supported in part by NIH Grants HD 01994 and NS 13617.]
Two-mass model of the larynx: Vocal fold vibration as a negative differential resistance oscillation
Authors: Conrad William A. , McQueen David M.
Published: October 2005

The vocal folds were analyzed as a two-mass model assuming zero glottal resistance and no supraglottal pressure recovery. Steady longitudinal flow calculations similar to those of Ishizaka and Matsudaira [SCRL Monograph No. 8 (1972)] were made, but assuming subglottal pressure varied through a sequence of steady states. A negative differential resistance was found when the dc displacement of the upper mass was small compared to that of the lower mass. Dynamic motion of the masses was represented by a pair of series resonant circuits within the glottis and transverse to the steady flow. The transverse circuits are isolated from the trachea and vocal tract by the resistances at the entrance and exit of the glottis. Sustained, self-excited, small-amplitude oscillations can be obtained when the magnitude of the negative differential resistance was equal to the real part of the circuit impedance. The oscillation frequency and phase difference between the masses depended only on myoelastic properties of the larynx. By contrast, oscillation frequency and phase in Ishizaka and Matsudaira's analysis also depends on dc aerodynamics. The calculations are therefore not contained in Ishizaka and Matsudaira's analysis. The difference in predictions should be resolved by experiment on linear controllable mechanical models.
A hybrid model of vocal fold vibration illustrating perturbations
Authors: Wong D. , Ito M. R. , Titze I. R.
Published: October 2005

This study of vocal fold vibration examines the physics of vocal fold vibration and some possible causes of perturbed vibration. A multiple mass-spring computer model is used to simulate the vibratory motion of the normal vocal fold and a fold with asymmetric parameters. Speech, displacement, acceleration plots, and a mathematical discussion reducing the model to a function of lateral tissue motion are presented. Acceleration-displacement (energy exchange) diagrams from the simulation model are compared to the behavior of the mathematical equations, showing that the system driving forces can be associated with physical phenomena such as vocal tract loading, vertical phasing, Bernoulli forces, and negative damping. Simulations have produced perturbed vibrations, suggesting that 1/2 and 1/3 subharmonics are present. The 1/2 subharmonic results from vocal tract resistive and inertive loading, while the 1/3 subharmonic arises due to inertive loading and nonlinear tissue stiffness. It is necessary to derive a fourth-order system using vertical phasing of the upper and lower tissue margins, in order to drive the system into modes other than the fundamental. Irregular random perturbations also appear when parameter changes are localized.
Oceanic ambient noise from collective oscillations of bubble clouds
Authors: Prosperetti A.
Published: October 2005

Data show a rather substantial amount of ambient noise in the ocean in the frequency region from several hundred Hz to 1 kHz. Emission in this frequency band from isolated oscillating bubbles appears to require unrealistically large bubbles. It is proposed that the noise is produced by bubble clouds (caused, e.g., by wave breaking) oscillating in a collective mode. It is shown that a cloud of N equal bubbles has a lowest collective mode at a frequency of the order of N−1/3 times the natural frequency of each bubble in isolation. A mathematical model for the description of the process is set up and preliminary results are discussed.
Oceanic noise generation by bubbles in breaking waves
Authors: Hollett Reginald D.
Published: October 2005

A model of oceanic noise generation is described for wind-induced noise in the frequency range from a few hundred Hz to several kHz. The noise is attributed to the free oscillations of bubbles formed in wave breaking. The breaking part of the wave is treated as a uniformly moving region within which bubbles are continually generated at a constant rate. The free oscillations are excited by the excess energy imparted to the bubbles at the instant of formation. The duration of the oscillatory motion is sufficiently short that bubble extinction by buoyant rise to the surface is not considered. The bubbles are assumed to oscillate independently of their neighbors so that each bubble radiates at its natural frequency which depends on the radius. A calculation of the energy spectrum of an individual breaking wave is presented using a bubble radius distribution taken from published results of breaking wave simulations. The noise spectrum, obtained by summing over the sea surface, compares favorably with Wenz's curves.
Oscillation of a sphere in a saturated poroelastic medium using singular solutions
Authors: Dlubac James J. , Chwang Allen T.
Published: October 2005

The Biot equations [J. Acoust. Soc. Am. 28, 168–178, 179–191 (1956)] for a saturated poroelastic medium, modified to include the mass and momentum sources, are solved for the fundamental monochromatic singularities. The oscillatory point source (sourcelet) and force (forcelet) and their higher-order poles are directly applied to solve the problems of a rigid sphere embedded in an infinite medium in various modes of vibration. These closed-form solutions include the torqued and pulsating sphere, as well as the sphere in rectilinear oscillation. The sourcelet, being radially symmetric, emits only dilatational waves and represents the total volume of solid and fluid supplied. The forcelet, in addition to the two compressional waves, induces a rotational wave and is applied to the solid and fluid constituents in proportion to the porosity.
Computation of radiation impedance of stepped piston radiator
Authors: Simon Ralph
Published: October 2005

A computer program was devised for applying the Helmholz integral formulation to determine the pressure distribution on the surface of a stepped piston radiator, which consists essentially of a series of frustrums of cones and sectors of plane circles. The same program can also be used for various axially symmetrical modes of vibration of the sphere and of the plane piston by suitably specifying the values of the input parameters that determine the shape of the surface and the angle between the normal to each surface element and its direction of vibration. Good agreement is obtained (usually < 1% error) with the known analytic solutions for these latter radiators for ka⩽ 2, using only ten node from pole to equator. The complex radiation impedance as a function of ka was computed for Lubell Laboratories model 98 underwater loudspeaker that vibrates by separating along its flexible equator. The results are compared with those of a sphere that vibrates likewise. [This work was supported by Lubell Laboratories, Inc.]
Magnitude estimation of motion sickness in an operational environment
Authors: Bittner A. C. , Guignard J. C.
Published: October 2005

Magnitude estimation of the subjective distress of seasickness has not been used for scaling in operational environments, although it has previously been utilized in laboratory studies [e.g., O. L. Bock and C. M. Oman, Aviat. Space Environ. Med. 53, 773–777 (1982)]. During sea trials, the method was used by one of us (ACB) to scale the provocativeness of five shipboard areas: Communications Center (CC), Communications Support Center (CSC), mess, bunk, and other areas. In addition, prediction weights for incremental changes in a “general motion illness” factor were independently derived by regression analysis on the responses of 16 crewmen during the same trials [A. C. Bittner, Jr. and J. C. Guignard, “Motion Sickness Evaluations in an At-Sea Environment: Seakeeping Trials of a USCG Cutter (WMEC 901), Naval Biodynamics Laboratory Rep. NBDL-86R002 (manuscript under review)]. These were: 0.12 (CC); 0.04 (CSC); − 0.09 (mess); − 0.35 (bunk); and 0.00 (other). Using a mess location as a comparison stimulus (assigned modulus 4), the comparable geometric means of (3–5) magnitude estimates for the same areas were: 10 (CC); 6 (CSC); 4 (mess); 2 (bunk); and 5 (other). The correlation between the direct (magnitude) and indirect (factor) scales was substantial and significant (RHO = 1.00; p < 0.017). These results demonstrate the utility of magnitude estimation of motion sickness for operational comparisons. [Work supported by DTNSRDC MIPR Z70099-4-00758 under an agreement with the U. S. Coast Guard. Views not necessarily those of USCG or Department of the Navy.]
An historical review of the strategies and struggles for incorporating acoustics in the undergraduate curricula
Authors: Hemond Conrad J. , Celmer Robert D.
Published: October 2005

Beginning over 25 years ago, when the College of Engineering of the University of Hartford was in its infancy, the faculty authorized the development of an undergraduate program in Acoustics within the framework of the then Engineer's Council for Professional Development (ECPD now ABET). Hence a unique course in Engineering Acoustics was developed and offered as a requirement for the BSME degree. Problems of offering such a course were the same then as they are now. Finding a textbook with engineering applications and at an undergraduate level was and to some extent is a limiting factor. Acquisition of acoustic instrumentation and continually updating it is a limiting economic factor. Finally, securing a professor who has had some practical experience could be the greatest hurdle to overcome. This is not a necessary criterion but it is helpful to the students. This paper will trace the strategies of overcoming classical resistance to the offering of this topic to undergraduate engineers and not reserving it for a graduate program. Contrary to this idea, experience has demonstrated that the course arouses student interest and results in continuing study in graduate acoustic programs.
Undergraduate acoustics at Florida Atlantic University. A success story
Authors: Davidson J. Blaine
Published: October 2005

Undergraduate acoustics has been an integral part of the Ocean Engineering program at Florida Atlantic University since its inception in 1965. Only Underwater Acoustics courses were offered in the original curriculum, but as the program matured, courses in the fundamentals of acoustics and in mechanical vibrations were added. Growth permitted more elective offerings, and in this way a course in acoustic signal processing and one in vibration, shock, and nose control were added. Today, Florida Atlantic University offers one of the strongest undergraduate acoustics programs available, and there are probably more graduates from this program entering the workplace in the field of underwater acoustics than from any other school.
Effects of sentence context on identification of whispered words
Authors: Jakimik Jola
Published: October 2005

In a previous study [R. Sturm and J. Jakimik, J. Acoust. Soc. Am. Suppl. 1 76, S29 (1984)], positive and negative effects of linguistic context on identification of whispered words were found. Relative to neutral contexts, appropriately biased contexts improved identification, whereas inappropriately biased contexts reduced correct identifications, and led to misperceptions, even for words accurately perceived in neutral contexts. The present study follows up on these results using more carefully controlled stimulus materials, and more sensitive response measures. Acoustically identical whispered words were presented in various contexts, and listeners rated their confidence in the words they identified. The accuracy and confidence results are examined to establish the scope of the effects of context. The implications of the findings for models of the integration of acoustic-phonetic and higher-level information in speech perception are discussed.
Matching of “physical” and “perceptual” spaces for vowels
Authors: Zahorian Stephen A. , Jagharghi Amir J.
Published: October 2005

An algorithm for matching physical and perceptual spaces for psychological stimuli will be described. Target points for each stimulus class must be chosen in a multidimensional perceptual space. The physical space consists of a multidimensional measurement space, in which measurements are made of each stimulus for a large number of subjects. A linear transformation from the measurement space to the perceptual space is determined such that the mean square distance between target points and transformed measurement points is minimized. There is no requirement that the dimensionality of the measurement and perceptual spaces be the same. Thus the algorithm can be used to redefine the measurement space with fewer dimensions such that the correspondence with predefined stimulus categories is maximized. This procedure has been tested using vowels spoken in an /hVd/ context, six principal components for measurement parameters, and a three-dimensional perceptual space. Target positions in the perceptual space were based on published data from multidimensional scaling experiments for vowels. The resultant transformation has been used to map vowels to colors for use in a speech training aid for the hearing impaired. Experimental results will be given. [Work supported by the The Whitaker Foundation.]
Pattern recognition in acoustic signature analysis
Authors: Haran S. , Finch R. D. , Jansen B. H.
Published: October 2005

The use of a pattern recognition approach in signature analysis is discussed. The application considered here is the inspection of railroad wheels by the analysis of acoustic signatures. Previous work [S. Haran and R. D. Finch, J. Acoust. Soc. Am. Suppl. 1 75, S76 (1984)] has established that the best detection method was a comparison of signatures from the two wheels on an axle. Pattern recognition techniques were used to detect the difference in the signatures of the two wheels on an axle, and to classify the wheels accordingly. After filtering and preprocessing, various features were computed from the signals for input into a pattern recognition module. Both time and frequency domain features were considered. The features include: power in various frequency bands, spectral moments, and spectral ratios. The pattern recognition package ISPAHAN was used, in which several classification schemes are available. Results from the analyses using some of the classification schemes will be presented.
Two strategies for acoustic localization in air using a compact sensor
Authors: Gonzales J. , Deems S. A. , Netzorg O. B. , Garrett S. L.
Published: October 2005

Two complementary methods for acoustic localization of an airborne sound source using a pair of microphones will be described. One approach develops a real-time solution by applying a dual time delay to the microphone outputs to form two oppositely directed cardioid beam patterns. The ratio of the difference to the sum of the cardioid signals determines the azimuthal location of the source. The second approach, which is slower, but more discriminating, uses a cross-correlation routine to find the relative time delay between the two microphone outputs. The characteristics of both methods were tested in an anechoic chamber using a synthesized helicopter signal and the helicopter signal plus an independently located white noise source. [Work supported by the Office of Naval Research.]
Holographic interferometry applied to the study of in-water dynamic behavior of acoustic transducers
Authors: Fein Howard
Published: October 2005

Holographic interferometry has been shown to be a successful method for characterization and testing of the behavior of acoustic transducers operating in air. Characteristics of motion and displacement are known to model the ideal behavior of these devices by defining specific geometries and symmetries in time-averaged holographic interference fringe patterns [H. Fein, J. Acoust. Soc. Am. Suppl. 1 73, S25 (1983)]. These techniques have been extended to the study of the actual dynamics of acoustic transducers operating in the water medium. Successful comparisons of the differences between motion characteristics of transducers operating in water and air are shown to be of great value in defining their true dynamic behavior. Holographic data and comparisons are presented and discussed.
Continuous measurement of complex modulus from 10–30 kHz
Authors: Madden Peter E. , Totten Timothy B.
Published: October 2005

The paper describes a technique which measures the complex modulus of materials in the range 10–30 kHz. The apparatus measures the input impedance of a sample of the material which is approximately 0.25 in. long and 0.7 in. in diameter. A shaker with an impedance head is used to drive one end of the sample which has a large mass at the other end. A noise drive and analysis by a dual channel spectrum analyzer connected to a desktop computer, gives a continuous frequency spectrum of the real and imaginary parts of the modulus. The method has been used over a frequency range of 10–30 kHz to measure moduli from 107–1010 Pascals with loss factors as low as 0.02 and it is felt that these ranges could be extended. The technique enables a measurement over the complete frequency range to be made in less than 5 min and the results are repeatable.
Modal analysis of a handbell
Authors: Hansen Uwe J. , Rossing Thomas D.
Published: October 2005

Modal analysis with impact excitation offers an effective way to study the modes of vibration of percussion musical instruments [Sessions O and U, 109th ASA meeting (April 1985)]. The vibrational modes of a handbell can be conveniently classified into groups according to the number and location of their nodal circles [T. D. Rossing and R. Perrin, J. Acoust. Soc. Am. Suppl. 1 78, S75 (1985)]. The modes of a G4 Malmark handbell will be described, and computer animations of modes from various groups will be shown.
A study of air loading of timpani
Authors: Rickey Martin E. , St-Amant Guy , Smith Wayne R.
Published: October 2005

Unlike vibrating strings, which require carefully designed coupled surfaces to propagate sound, drum heads are intrinsically strongly coupled to the surrounding atmosphere. Recent studies [R. S. Christian et al., J. Acoust. Soc. Am. 76, 1336–1345 (1984)] have compared measurement made in a normal atmospheric pressure environment with theoretical analysis. We have measured time-dependent Fourier spectra of the sound output of a kettle drum at many absolute pressures between 0.01 and 1 atm. The drum was excited in a precisely controlled repeatable manner using a remote controlled mechanical striker attached to a 75-cm-diam drum. This assembly was placed in an extensively sound dampened vacuum chamber 2 m in diameter and 2 m high. The results of these measurements will be presented and compared to the previous work.
Reflection functions of musical air columns
Authors: Hoekje P. L. , Benade A. H.
Published: October 2005

Characterization of musical air columns by their input impedance Z(f) has been used extensively in frequency domain discussions of the nonlinear mechanisms which sustain oscillations and has been of use to instrument makers. The McIntyre-Schumacher-Woodhouse time domain description of oscillators (1983) offers new insights into nonlinear phenomena, but requires characterization of the air columns in terms of a reflection function r(t). Though r(t) and Z(f) are directly related, conversion by transform methods, between these parametrizations, of finite experimental data can be cumbersome or misleading. We show theoretical and experimental examples of r(t) for the following terminations of a cylindrical duct: inertial, compliant, resistive, blunt open end, single side hole in infinite duct, semi-infinite tone hole lattice; these examples are compared with data for actual musical instruments. [Work supported by NSF.]
Source-filter model applied to pipe organ tones
Authors: Lemay Gerald J.
Published: October 2005

In the 15 years since the landmark paper by B. S. Atal and S. L. Hanauer [“Speech Analysis and Synthesis by Linear Prediction of the Speech Wave,” J. Acoust. Soc. Am. 50, 637–655 (1971)] there has been considerable interest and application of the source-filter construct for modelling the time-varying waveforms present in speech. This paper reports on the appropriateness and limitations of this model for analyzing and synthesizing different events present during the generation of musical waveforms. The first part of the paper includes a brief tutorial on the various assumptions inherent in the source-filter model. This is followed by results from analysis and synthesis experiments conducted on digitized tones from a Stuart tracker pipe organ. [Work supported by NSF Grant ECS-8408997.]
Comparison of LPC estimation algorithms from noisy speech
Authors: Ahmed Mohammed S. , Abbasi Asad K.
Published: October 2005

It is well known that LPC parameters are severely distorted when the speech waveform is corrupted by noise. This paper compares the performance of different LPC estimation algorithms from noisy speech. Direct estimation of LPC parameters from noisy speech as well as the use of preprocessor speech enhancement algorithms are used. The direct estimation algorithms considered are: the autocorrelation subtraction method, shifted Yule-Walker method, and an instrumental variable method. The preprocessor speech enhancement algorithms considered are: adaptive filtering (AF), adaptive noise cancellation (ANC), and linear maximum a priori (LMAP) estimation. The algorithms are applied on speech corrupted by white noise. Their performance is compared through Monte-Carlo simulation conducted on voiced, unvoiced, and v/uv mixture of frames. The criteria chosen for comparison are bias, standard deviation, and average euclidean distance. The above algorithms are also applied to a speaker recognition scheme based on orthogonal LPC parameters. Comparison of the recognition results using speech corrupted by white noise is reported. Our results indicated that of the above algorithms, preprocessing by AF and ANC can improve the LPC estimation in a noisy environment. Other algorithms produce poorer results compared to the conventional autocorrelation method.
Very low bit rate speech coding based on joint segmentation and variable length segment quantizer
Authors: Shiraki Yoshinao , Honda Masaaki
Published: October 2005

This paper presents a very low bit rate LPC vocoder based on a joint segmentation and quantization method using spectral segments having variable time length. The method exploits the nonuniform distribution of speech characteristics in the time and spectral domains. A measure of the spectral distance between a variable-length input speech segment and a fixed-length segment template is introduced based on linear time warping. The optimum segment boundaries and templates for a spectral sequence are efficiently determined using a dynamic programming technique so that the total spectral distortion in a voice interval is minimized. The segment templates are obtained by a sub-optimum pattern learning method, which guarantees a monotonic decrease in distortion, using a combined segmentation and clustering technique. Experimental results for a single male speaker show that this method reduces the initial distortion by 20% and yields a sound articulation score of 78%.
Calibration of computed features of isolated vowels using synthetic vowel waveforms
Authors: Cox N. B. , Ito M. R. , Morrison M. D.
Published: October 2005

A number of computer-extracted features of isolated vowels have been defined in the literature for use in quantification of the effects of laryngeal pathology. This paper describes the use of synthesized vowel waveforms for “calibration” of these features. The relative effects of jitter, shimmer, and additive noise perturbations on the following vowel features will be discussed; the relative average perturbation of pitch period duration and amplitude, a time domain harmonics to noise ratio [E. Yumoto, W. J. Gould, and T. Baer, J. Acoust. Soc. Am. 71, 1544–1550 (1982)] and a frequency domain harmonics to noise ratio [H. Kojima, W. J. Gould, A. Lambiase, and N. Isshiki, Acta Otolaryngol. 89, 547–554 (1980)]. It will be shown that the noise features are sensitive to jitter and shimmer as well as noise. A new noise feature will be introduced and shown to be effective at isolating noise perturbations from jitter and shimmer in the synthetic vowel waveforms.
An inverse filtering technique for efficient vowel classification
Authors: Cooper Beth A. , Sibul Leon H.
Published: October 2005

Classification of steady-state vowel sounds by formant frequency clustering is well documented in the literature [G. E. Petersen and H. L. Barney, J. Acoust. Soc. Am. 24, 175–184 (1952)]. A more efficient and equally effective scheme for classifying steady-state vowels has been developed using partial correlation (PARCOR) coefficients that arise as intermediate parameters in a least squares adaptive lattice inverse filter. A sixth-order autoregressive time series was synthesized from three formant/bandwidth pairs for each vowel utterance of the Petersen and Barney data. The PARCOR coefficients of the inverse filtered processes exhibited the same graphical clustering behavior as the formant frequency data, and numerical distance measures showed the two sets of parameters to be equivalent. Because the PARCOR coefficients are used directly to classify vowels and because a pth order lattice filter simultaneously generates PARCOR coefficients for all lesser order filters, PARCOR coefficients provide an efficient, minimal parametrization. These results motivate the study of this technique for processing and classifying more complex and time-varying speech signals. [Work supported by Naval Sea Systems Command.]
Speaker-independent speech recognition using vocabulary-dependent clustering techniques
Authors: Bernstein George
Published: October 2005

The statistical technique of template clustering for speaker-independent speech recognition have typically used the same number of clusters to represent each word in the vocabulary. However, linguistic idiosyncracies and word confusability are contributing factors in determining the number of clusters required to adequately represent a given word. Based on this notion, an algorithm has been developed to assign the number of clusters for each word automatically, using a rule based on average cluster width. The objective is to reduce the computational load without undue degradation of recognition accuracy. Experiments have been carried out with a data base of 20 male and female speakers and a 40-word vocabulary that includes the alpha digits. The range of cluster levels was 3 to 8. Thus far, experimental results have shown that, with a constant cluster density and a total of 320 centroid templates, a recognition score of 82.75% was achieved with the alpha digits. At 291 templates allocated as a variable number of centroid templates per word, a recognition accuracy of 81 percent was achieved. Additional experiments with variations of the heuristic rules will be carried out to optimize performance.
Speaker-independent connected digit recognition with Texas Instruments data base
Authors: Bocchieri E. , Doddington G.
Published: October 2005

We report about speaker-independent connected-digit recognition experiments performed on Texas Instruments multi-dialect database. Our algorithm features two components. (1) A front-end “word hypothesizer” detects the possible presence (hypothesis) of vocabulary digits in the input speech. This operation uses distance measures already evaluated for isolated digit recognition [“Frame Specific Statistical Features For Speaker Independent Speech Recognition,” to appear in IEEE-ASSP]. Dynamic time warping (DTW) continuously provides cumulative digit distance scores as function of input frame (time). Distance function local minima and initial/final times of relative DTW optimum paths characterize the digit hypotheses. (2) A network-based sentence recognizer selects the hypothesis sequence by minimizing a recognition score function of hypothesis distance values and initial/final times. Test and hand-labeled training data sets were collected from two distinct groups of 28 male speakers, respectively. Each set contains 7075 digit tokens in 2155 utterances. The number of digits in each utterance (from one to seven) is not provided to the recognizer. A word error rate (insertions, deletions and substitutions) of 1.1% was obtained with a ten-digit vocabulary.
Automatic recognition of isolated Spanish CV syllables
Authors: Univaso Pedro , Rosso Enrique , Franco Horacio
Published: October 2005

Due to the difficulties in the recognition of quite similar utterances, such as CV syllables with the same vowel, a two-step approach was proposed. In the first step the normalized log energy, 32 spectral band log energies, and the spectral change were used through a DP algorithm to determine: (a) one of the five broad acoustic classes of the consonant involved (voiced stop, unvoiced stop, nasal, liquid, and fricative) into which the best matched syllable fell and (b) the warping functions. As a second step the test pattern was compared with the reference patterns of the previous recognized class, emphasizing the differentiating regions so as to realize the final recognition of the syllable. The patterns were matched over the warping function taking into account only the frames around the transitional region. The final distances were calculated using only the spectral bands which focused the acoustic class distinctive features. Speaker-dependent performance over the ten more frequent Spanish CV syllables was improved from 78% to 99% with the two-step procedure instead of considering only the first step as final recognition of the syllable.
Shadowing fluent speech: The effects of mispronouncing word-initial consonants
Authors: Small Larry H. , Bond Z. S.
Published: October 2005

A shadowing task was employed to investigate normal listeners' ability to detect alterations in word-initial consonants. The same 600-word prose passage, containing 14 two-syllable target words with syllable-initial stress was employed in three separate experimental conditions so that the word-initial consonant was either altered by (1) a one distinctive feature change [Chomsky and Halle, The Sound Pattern of English (Harper and Row, New York, 1968)], (2) a three distinctive feature change, or (3) completely deleting the word-initial consonant. The target words were selected to be predictable from context and were examined for frequency of occurrence in English. There were 16 subjects in each condition. The subjects' shadowed responses were recorded on tape and analyzed to determine whether certain lexical or phonetic information was particularly salient in guiding the listener's shadowing behavior. Specifically, subject's responses were analyzed as to whether the part of speech, morpheme boundary, number of syllables, stressed vowel, and lexical stress were maintained when shadowing the mispronounced target words. Each of these items will be discussed as to their relative importance in lexical access while shadowing fluent speech.
Acoustics of Caribbean steel drums
Authors: Rossing Thomas D. , Hampton D. Scott , Boverman Joshua
Published: October 2005

As part of an ongoing study of Caribbean steel drums, the acoustical behavior of three alto/double tenor steel drums, each having 14–17 notes in the range of F13 to C6 (185–1047 Hz) has been studied. All except for the highest notes have three prominent harmonic partials, each associated with a mode of vibration of the local area of the drum tuned to that note. Additional harmonic partials (up to eight) are identified when a note is struck strongly, suggesting a nonlinear behavior characteristic of shallow spherical-cap shells [N. H. Fletcher, J. Acoust. Soc. Am. 78, 2069–2073 (1985)]. By means of holographic interferometry, some modes of vibration have been able to be identified as “local” and some as “global,” encompassing several note areas on the drum.
Tonal design and tuning of Caribbean steel drums
Authors: Alexis Clifford , O'Connor Al , Rossing Thomas D.
Published: October 2005

Steel drums were invented in Trinidad in the 1940s, but they soon became popular throughout the Caribbean islands. The drums in a steel band are usually fabricated from 55-gal oil drums, and they are known by various names, such as soprano, ping pong, alto, double tenor, guitar, cello, and bass. Each drum may have from 4–32 different notes. A skilled steel drum maker tunes at least one overtone of each note to a harmonic of the fundamental. Whenever possible, a second harmonic is tuned as well.
Precision of sound power measurements with intensity technique
Authors: Tichy Jiri
Published: October 2005

The precision of the sound power determination by measuring acoustic intensity depends on the precision of the measurement system, selection of the measurement area, and the number of measurement points. The decision on the optimum choice of these parameters depends on the properties of the radiated sound field and on the environmental effects including simultaneous sound radiation from other sources. To obtain qualifying information on the sound field properties, the measurement of the complex acoustic intensity may be necessary. To quantify the judgment on the measurement parameter selection, a series of indicators has been recently suggested. This presents an analysis of the mutual effects of the measurement parameters and their effect on the precision of the sound power determination.
Errors introduced by intensity probes
Authors: Brüel Per V.
Published: October 2005

The paper describes the errors introduced in intensity measurements on account of the finite size of the probe. The reflections between the microphones cause errors not only in the pressure amplitude, but also give rise to uncontrollable phase changes which result in significant errors. This is illustrated by systematic measurements on some of the most commonly used two-microphone probes, and measures to be taken for minimizing these errors are indicated. Furthermore, the types of errors caused when the points of measurement of the sound pressure and the particle velocity are not coincident, are also shown.
Instantaneous intensity
Authors: Heyser Richard C.
Published: October 2005

Expressing time as a fourth space coordinate under the rule x4 = ict, where i is the imaginary operator, allows the interpretation of a complex velocity which is the four-dimensional gradient of a potential. The three-conventional space components of this gradient yield particle velocity, while the fourth component yields sound pressure divided by the product of density and sound speed c. In accordance with an energy theorem of Heyser [J. Audio Eng. Soc. 19, 902 (1971)], this four-dimensional velocity is interpreted as the local component of an energy function whose global support is represented by a complex Hilbert transform component. This leads to a total energy density which contains a term identifiable as the reactive component of an instantaneous complex intensity. Coinciding with the well-known expression for the time averaged intensity of monochromatic signal dependence, this instantaneous intensity applies to any time dependence and has a time average identical to the conventional expression. It is shown that, in the case where instantaneous reactive intensity vanishes, the instantaneous active intensity is numerically identical to the energy-time curve (ETC) which is now used in acoustical analysis.
Exterior boundary-value problems for the Helmholtz equation with a generalized impedance boundary condition
Authors: Dallas Allan G.
Published: October 2005

An exterior boundary-value problem for the Helmholtz equation is considered, the boundary condition being of the (generally nonlocal) form ∂u/∂v + Bu = g, B a bounded linear operator in the Hilbert space L2(Γ), Γ denoting the boundary of the exterior domain, and g given in L2(Γ). It is also required that B satisfy a certain dissipativeness condition. The basic existence, uniqueness, and continuous-dependence results are stated. Various techniques for construction of the solution are described, which are extensions of the methods known for the Neumann or “classical” impedance boundary-value problems. It is argued that such an exterior problem is the appropriate one to study when modeling the more difficult interior-exterior interface problem corresponding to the scattering of time-harmonic acoustic waves by a penetrable body.
A model for transverse wave sensitivity to poor adhesion in adhesively bonded joints
Authors: Rose J. L. , Pilarski A. , Da-Le J. , LeCuru D.
Published: October 2005

Poor surface preparation in adhesive joints of metal-to-metal type usually results in a weak bond. The adhesive strength prediction problem has been studied extensively with little success to date. The authors of this paper are attempting to explain changes in the reflection and transmission coefficients of oblique incident longitudinal and transverse waves accompanying changes in adhesive bond quality, based on a model of a bond with finite rigidity. The reflection transmission characteristics for a solid-solid imperfectly bonded interface were based on an assumption that displacement across the interface was not required to be continuous. This discontinuity was taken to be linearly related to the stresses which are continuous across the interface. A comparison of cases “welded” and “smooth” boundary conditions gave us the optimal angle of incidence of transverse waves with the best sensitivity to areas with poor adhesion. Theoretical and experimental results are compared with excellent agreement.
Nasal co-articulation in child and adult speech
Authors: Flege James Emil
Published: October 2005

The present study was motivated by the finding [Thompson and Hixon, Cleft Palate J. 16, 412–420 (1979)] that the percentage of subjects showing emission of nasal airflow at the midpoint of prenasal vowels increased through childhood. An acoustical method was used to test the hypothesis that the domain of velar co-articulation increases with age. Adults and children aged 5 and 10 years (ten of each) produced /dVn/ and /nVd/ words eight times each at normal and fast speaking rates. Abrupt changes in nasalance scores (based on the ratio of the signal transduced by nasal and oral microphones separated by a thin plate) were used to determine what part of vowels (/i,I,U/) were nasalized. There was no difference between groups in either the percentage of the vowel intervals in /dVn/ (78%) or /nVd/ (82%) that were nasalized, or in the absolute duration of the nasalized portion of the vowels (389 and 342 ms, respectively). This was true even after normalizing for between-group differences in vowel duration. These results suggested that the domain of velar co-articulation either does not change with age, or shows a mature pattern by the age of 5 years. In addition, the subjects in all three groups manifested the same substantial difference in nasalance scores in the context of /d-d/ (17.9) versus /n-n/(58.9). Both adults and children showed significantly greater nasalance scores in the context of /i/ than /u/ or /I/, presumably as the result of greater oral resistance to airflow in /i/. [Work supported by NIH.]
Laryngeal compensation following sudden oral perturbation
Authors: Munhall Kevin , Lofqvist Anders , Kelso J. A. S.
Published: October 2005

Recently, there have been a number of demonstrations of rapid goal-directed compensation in remote articulators when an articulator is perturbed unexpectedly during ongoing speech. Much of this research, however, has involved articulators that cooperate to produce a single vocal tract constriction (i.e., tongue-jaw, lips-jaw). In this experiment, spatially more separate articulators are examined that are simultaneously active in the production of a phonetic segment (i.e., the larynx and the lip-jaw complex). Three subjects produced the nonsense utterance “/ipip/ again” and the lower lip was unexpectedly loaded in the first vowel or during the closure for the first /p/ on a small proportion of the trials. Laryngeal behavior was monitored using transillumination and fiberoptic videotaping. In addition, intraoral air pressure just above the folds was monitored. Oral movements were recorded with a Selspot system. Acoustically, the first /p/ closure in “ipip” was shorter in duration and the voice onset times were longer in the perturbed trials. An examination of the laryngeal gestures during the production of the first /p/ revealed longer adduction durations in the perturbed conditions than in the control trials and a longer overall abduction/adduction cycle. The results will be discussed in terms of task-dependent articulatory coupling. [Work supported by NINCDS.]
The effects of microphone placement and azimuth on peak sound pressure level of type A impulses
Authors: Shotland Lawrence , Robinette Martin , Brey Robert
Published: October 2005

Single, repeatable type A impulses were generated by colliding steel spheres in an acoustically reverberant room, and measured as A-weighted sound levels by a sound level meter having four exponential time averaging modes with time constants ranging from 2 μs to 1 s. Measurements using peak hold (2-μs time constant) were then obtained on KEMAR at four azimuths and four microphone placements at each azimuth. Comparative measurements showed peak hold to give the most accurate estimation of A-weighted sound level, as verified electrically. The remaining three underestimated this value. Maximum underestimation of 34.8 dB was demonstrated using the slow time constant. KEMAR measurements demonstrated peak A-weighted sound levels to be sensitive to both microphone placement and azimuth. Measurements made in the reference field and at external microphone placements tended to underestimate peak A-weighted sound level at the tympanic membrane (TM). Two conclusions were reached. Time constant of the exponential time averaging must be shorter than the duration of the signal. Peak A-weighted sound level cannot be predicted by measurements made at sites other than at the plane of the TM.
Speech sound identification influenced by adjacent “restored” phonemes
Authors: Ohala John J. , Feder Deborah
Published: October 2005

When listeners' identifications of speech sounds are influenced by adjacent sounds, do they operate only on the physical phonetic characteristics of there sounds or could they also use just their linguistic identity? We tested this by leading subjects to restore or induce the noise-obliterated medial consonant in VCV utterances by first presenting them with several prior utterances where this medial consonant was consistently the same, either a /b/ or /d/. Included as V1 were synthetic vowels from the /i-u/ continuum. As expected, more /u/s were identified out of this continuum in the environment of physically present /d/s than /b/s [Ohala et al., J. Acoust. Soc. Am. Suppl. 1 64, S18 (1978)]. But restored /d/s and /b/s had the same effect (though with slightly less intensity) thus indicating that the influence of context need not operate only via physical phonetic features. We discuss the implications of this finding for “direct realist” theories of speech perception and for claims of the existence of invariant phonetic elements. [Supported by the Cognitive Science Program, UCB.]
Optimum time domsin signal transmission and source location in a waveguide: Ideal waveguide experiments
Authors: Clay C. S. , Li Saimu
Published: October 2005

Experimental measurements that were made in an ideal waveguide in air are reported. The experiments confirmed the theoretical technique. Signals transmitted in a waveguide are distorted by the multiple arrivals. A waveguide matched filter has an impulse response that is the time reverse of the impulse response of transmission between a source and receiver. This filter is an optimum filter for the reduction of transmission distortion. The waveguide matched filter is sensitive to source and hydrophone locations, and it can be used to locate a source that transmits an unknown signal. Each signal channel is match filtered for a trial source location. Pairwise cross correlations have a maximum when the trial location is the same as actual source position. [Supported in part by the Shangdong College of Oceanography, People's Republic of China, the Geophysical and Polar Research Center of the University of Wisconsin, and NORDA.]