bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic and perceptual measurements of hoarseness in children’s voices
Authors: McAllister Anita M. , Hibi S. H. , Sundberg Johan
Published: October 2005

According to a previous investigation of children’s voices, hoarseness is a stable concept with three main predictors: hyperfunction, breathiness, and roughness [McAllister et al., J. Voice 8(3), 230–239 (1994)]. The present investigation analyzes the relation between various acoustic measures and these perceptual voice characteristics, complemented by instability and gratings. The analysis was carried out on a material of vowels sustained by 50 ten-year-old children representing different degrees of hoarseness. The acoustic measures were a frequency perturbation quotient, an amplitude perturbation quotient, and two versions of a harmonics-to-noise measure as proposed by Kasuya et al. [J. Acoust. Soc. Am. 80, 1329–1334 (1986)]. Highly significant correlations were found between hoarseness, breathiness, and roughness, on the one hand, and frequency perturbation quotient as well as one of the harmonics-to-noise measures, on the other.
Relationships between time and frequency measurements of harmonics-to-noise ratio in human voice signals
Authors: Qi Yingyong , Hillman Robert E.
Published: October 2005

The quantity, harmonic-to-noise ratio (HNR), has been used to estimate the level of noise in human voice signals. HNR estimation can be accomplished in two ways: (1) Time-domain techniques, in which HNR is computed directly from the acoustic waveform; and (2) frequency-domain techniques, in which HNR is computed from a transformed representation of the waveform. It is unclear, however, how to operationally relate time- and frequency-domain estimations of HNR because of the quasiperiodicity of human voice signals. This work demonstrates the relationship between time- and frequency-domain estimations of HNR experimentally. Results indicate that the time-domain estimation of HNR approximates the average levels of harmonics measured above a reference spectral envelope, which is obtained by cepstral analysis. [Work support by NIH.]
Characterizing speech timing and speaking rate through subword parse trees
Authors: Chung Grace Y. , Seneff Stephanie
Published: October 2005

A sublexical parse tree is used to model and study characteristics of speech timing in English. This representation is based on the ANGIE system [Seneffet al., Proc. ICSLP (Philadelphia, PA, 1996)], a hierarchical framework which captures morphological, syllabic, and phonological phenomena probabilistically. The duration of a unit in the tree is measured as a percentage of the total duration of its corresponding parent unit. This framework can be used both to conduct statistical studies to characterize temporal phenomena and to create duration models to aid speech recognition. A strategy has been developed in which unit durations in upper layers are successively normalized by their respective realizations in the layers below. This reduces the variance at each unit by enabling the sharing of statistical distributions to overcome sparse data problems. The normalized duration of a word node at the top of the tree can be taken as a parameter for speaking rate. The data were used to examine within-speaker rate variability and to study the second-order effects of speaking rate on relative durations of sublexical units. The experiments are being conducted in the ATIS domain [Glass et al., Proc. ARPA Spoken Language Technology Workshop, TX (1995), pp. 252–256].
Modeling segment durations with artificial neural networks
Authors: Kibre Nicholas J.
Published: October 2005

Prediction of segment duration in TTS systems has in the past generally been accomplished under arithmetic approaches such as multiplicative and incompressability models [D. Klatt, J. Acoust. Am. 54, 1102–1104 (1973); R. Port, J. Acoust. Soc. Am. 69, 262–724 (1981)], and sums-of-products models [J. van Santen, Comput. Speech Lang.8 , 95–128 (1994)]. Other research, however, suggests a more complex speech timing system than is captured by such models [H. Gopal, J. Phon. 18, 497–518 (1990)]. In this study, a limited domain of vowel duration phenomena are modeled under several designs of simple feedforward networks. The networks’ performance is then examined by using their output in our TTS system, and evaluating the naturalness of the resulting utterances in a perceptual experiment. Preliminary results indicate that simple two-layer perceptrons are able to learn the basic patterns of environmentally conditioned variations in segment duration, while more sophisticated networks are required to capture the complexities of these factors’ interactions.
Optimal Chebychev interpolation filters for accurate beam steering
Authors: Putnam William L. , Smith Julius O.
Published: October 2005

This work addresses the problem of the implementation of fractional sample delays necessary for accurate digital beamforming with small steering delay quantization. Past work in this area has employed techniques which upsample by an integer factor, providing an intermediate higher sampling rate at which more accurate delays can be implemented. The signal is then downsampled to the original rate. This results in a minimum delay resolution equal to the original sampling interval divided by the integer valued upsampling factor. The approach presented herein uses existing signal processing techniques to perform resampling without requiring a conversion to an intermediate higher sampling rate [J. O. Smith and P. Gossett, Proc. Intl. Conf. on ASSP (March 1984), Vol. 2, pp. 19.4.1–19.4.2]. This method allows for continuously variable delay values. This technique is extended by the inclusion of interpolation filters which are optimal in a Chebychev or min–max sense. It is shown that the design of fractional delay filters is a complex-valued optimization problem. Specifically, the design of Chebychev optimal filters can be posed as a quadratically constrained quadratic program. Techniques for designing Chebychev optimal fractional delay filters, as well as several examples, will be presented.
Deformation of impulse response estimates by the time window in the cross-spectral technique
Authors: Ono Takahiko , Suzuki Hideo , Fukushima Manabu , Kido Ken’iti
Published: October 2005

The estimation of the impulse response of a transfer system by the cross-spectral technique is very popular and most FFT analyzers are equipped with that function. However, the deformation of the impulse response in the process of estimation is not yet known. The deformation of the impulse response is caused by the use of the time window in the FFT as the DFT of a windowed waveform is the convolution of the spectrum of the waveform and the spectrum of the time window. This paper first describes the theoretical analysis on the deformation. Next, the result is verified by the computer simulation. The estimation errors are also investigated for various types of time windows. According to the investigation, the deformation is significant if the duration of the impulse response (the reverberation time) is longer than 1/4 of the window length. Finally, the use of the rectangular window in order to not deform the impulse response by shortening the length of window for the source signal is described.
Analysis of blood flow in the heart chambers by the stream-function method based on the ultrasonic Doppler technique
Authors: Sugawara Shigeo , Katahira Yoshiaki , Nitta Keiko , Tanaka Motonao , Ohtsuki Shigeo , Nitta Shin-ichi
Published: October 2005

For visualizing blood flow in the human heart noninvasively, the stream-function method based on the ultrasonic Doppler technique was introduced. In this method intracardiac blood flow is displayed as a two-dimensional streamline distribution pattern. In normal hearts, blood flow in the left ventricle during systole is displayed as a fan-shaped streamline pattern converging from the whole ventricular wall toward the outflow tract, and in diastole the streamline distribution shows fan-shaped divergence from the left atrium toward the whole ventricular wall. The fan-shaped streamline distribution is considered to reflect the uniform contractability and extensibility in the whole ventricular wall. On the other hand, in infarcted hearts with regional asynergy in the left ventricle, the streamline pattern is remarkably different. In systole, the streamline pattern at the normokinetic portion is the same as that in normals, but in the infarcted portion, the streamlines show a crooked pattern caused by the inflow into and the outflow from the ventricular aneurysm. In diastole, streamlines toward the infarcted portion disappeared. Changes in streamline patterns in both cardiac phases show the regional asynergy in the infarcted heart. From these results, it can be said that the intracardiac blood flows are visualized by the stream-function method, and this method is considered to be very useful for analyzing the intracardiac blood flow.
Optimal frequency for receiving an amplitude-modulated stochastic acoustic process in a medium with absorption
Authors: Gelman Leonid M. , Burau Yuri V. , Krytsyn Vladimir I.
Published: October 2005

The reception of amplitude-modulated narrow-band signals of a noisy carrier is considered for propagation in a medium with absorption which depends on the frequency of a background of noisy interference. The purpose of the work is to optimize analytically the frequency for receiving the amplitude-modulated stochastic acoustic process. The efficiency of reception depends very much on this frequency. An optimization problem was formulated and solved. The power criterion, namely, the overall signal-to-interference ratio at the detector output, was used as a criterion of optimality. Two important practical cases were considered: (1) The carrier is narrow-band; and (2) the carrier is broadband. New expressions have been obtained for optimum frequencies.
Relaxation techniques for finding horizontal eigenrays for global scale propagation
Authors: LePage Kevin D.
Published: October 2005

Prediction of the localization performance of a global hydrophone network requires the rapid computation of horizontal eigenrays connecting receivers to a grid of hypothetical source locations. The computation of these eigenrays using shooting techniques can be time consuming. For many of the paths, the refracted geodesic can be quite close to the horizontally refracted ray. In these cases relaxation techniques can be successfully used to bend the geodesic trial solutions onto the refracted path. This technique, which involves the iterative solution of a block diagonal system of equations, offers substantial performance benefits. Work by Collins et al. [J. Acoust. Soc. Am. 95, 3167–3170 (1994)] also suggests that relaxation techniques are useful for avoiding shooting problems associated with sensitivity to initial conditions, a common problem when shooting rays near shoaling bathymetry. Comparisons between traditional shooting eigenrays and the relaxed eigenrays are made and conclusions regarding the range of applicability of the relaxation technique are presented. [Work supported by DOE and Air Force Phillips Laboratory.]
A numerical investigation of the validity of a ray-based acoustic wave field expansion
Authors: Brown Michael G. , Ianniruberto Marco
Published: October 2005

The purpose of the work reported here is to determine the domain of validity in frequency and range of ray-based acoustic wave field expansions under typical deep ocean conditions. This is accomplished by comparing transient wave fields computed using the ray-based MaCh1 propagation model to those computed using the mode-based SNAP propagation model. Although MaCh1 is ray-based, it correctly treats caustics of arbitrary complexity. Attention is confined to range-independent environments so that potential ray theory deficiencies unrelated to ray chaos can be isolated; the latter problem will be investigated separately. Both high lattitude (constant gradient) and temperate (canonical) environments are considered. Two types of errors associated with the ray-based wave field expansion are identified. These errors may be attributed to the omission of: (1) complex rays (tunneling phenomena) near the source; and (2) beam displacement effects for near-grazing rays. Ray-based errors, which are generally isolated in time, grow with decreasing frequency and increasing range. Overall, however, the agreement between the ray- and mode-based wave fields is surprisingly good with, say, f0=20 Hz, Δf=25 Hz, r=300 km, or f0=80 Hz, Δf=100 Hz, r=1000 km. [Work supported by ONR.]
Coupled-mode solutions in generalized ocean environments
Authors: Stotts Steven A. , Knobles David P. , Levinson Scott J.
Published: October 2005

A two-way coupled-mode model has been developed to solve propagation problems in fluid-layered, range-dependent ocean environments. The model incorporates piecewise-linear depth-dependent, range-varying sound-speed profiles (SVPs). A fast and efficient normal-mode model which is based on a propagator matrix method is used to obtain the eigenvalues and eigenfunctions. The coupling coefficients are expressed in terms of depth integrals of products of Airy functions and are solved analytically. The coupled equations are solved exactly by an approach referred to as the U–K method, which incorporates the Lanczos method, previously used in solving similar coupled-channel equations in nuclear theory and successfully applied to nuclear response problems in heavy nuclei [Knobles, ‘‘Solutions of coupled-mode equations with a large dimension in underwater acoustics,’’ J. Acoust. Soc. Am. 96, 1741–1747 (1994)]. Highlights of the formalism will be presented which expand on the overview given previously [Stotts et al., ‘‘Solutions of coupled-mode equations with a large dimension in underwater acoustics,’’ J. Acoust. Soc. Am. 95, 2909(A) (1994)]. Several benchmark comparisons will be presented including propagation over a hill, upslope wedge propagation, and an analytic depth and range-dependent SVP in a flat waveguide.
Modeling shallow-water propagation with an appropriate sea-surface roughness spectrum
Authors: Norton Guy V. , Novarini Jorge C.
Published: October 2005

Predictions for shallow-water acoustic propagation are usually accomplished assuming a deep-water spectrum for the sea surface roughness. In addition, the effect of surface roughness is normally included through a loss mechanism. However, for waveguide propagation, the surface roughness mainly causes a redistribution of energy and therefore, the use of an accurate model to handle roughness is essential to obtaining accurate results. In order to properly model shallow-water surface roughness, the parametric spectrum GONO [Sanders and Bruinsma, WaveDynamics and Radio Probing ofthe Ocean Surface, edited by O. M. Phillips and K. Hassemann (Plenum, New York 1986), pp. 615–637] has been implemented. This spectrum is appropriate for wind-driven surfaces in waters ranging in depth from 15 to 100 m. To investigate the impact that different spectra has on the propagating field, predictions using GONO and Pierson–Moskowitz spectra are compared for different environments and at frequencies typical of active and passive sonars. An acoustic propagation model, which includes a conformal mapping algorithm to properly handle sea surface roughness [Norton et al., J. Acoust. Soc. Am. 97, 2173–2180 (1995)] is used to generate the numerical results. [Work supported by ONR.]
On reciprocity and energy conservation for one-way acoustic wave equations
Authors: Godin Oleg A.
Published: October 2005

Acoustic reciprocity in stationary media at rest and energy conservation are widely recognized as fundamental properties of the sound field to be inherited by any successful paraxial approximation. Rigorous validity of the reciprocity principle for a parabolic equation (PE) solution in motionless media is of particular importance for the PE to be capable of modeling subtle nonreciprocal acoustic effects due to ocean currents. In this paper, reciprocity and energy conservation are considered for a class of two-dimensional PEs for fluids with piecewise continuous range dependence. In the case of lossless media, equivalence of the reciprocity and the energy conservation is demonstrated. Uniqueness of reciprocity and energy conservation corollaries of a PE and their relation to respective properties of the true acoustic field are discussed. A wide-angle PE for moving fluid and appropriate boundary conditions are derived which lead to the exact fulfillment of a reciprocity principle and a conservation law. In a motionless layered fluid of constant density the PE reduces to the Claerbout PE. Some features of the new PE are analyzed. Its ability to model the nonreciprocal component of the acoustic field in the ocean is compared to that of other existing paraxial approximations. [Work supported by NSERC.]
Vibrating object sound field within a closed water area
Authors: Oboznenko E. I. , Oboznenko I. L.
Published: October 2005

Using the methods of boundary integral equations, the radiating object’s field within the closed water area is defined. A Green’s function for the closed water area with different boundary impedances is built. The sound-pressure field around the object is calculated on the basis of predicted (already measured) discrete pressure levels on the object’s surface that is irritated by the a priori unknown inner vibration sources of various nature. The algorithm of computation is used to find out the potentials of a single (or double) layer at the separate points (spots) on the object surface on the basis of sound-pressure amplitude and relative phase measurements in these very points. The experimental data within the frequency range of 2–256 Hz corresponded well enough to the suggested theoretical model of the problem. The differences between calculated pressure amplitudes and measured ones did not exceed the value of 6 dB. The 128-channel phase measurements have been run within the narrow frequency band after the signals’ time processing. The phase shift at the separate points has been defined with respect to the reference hydrophone located at the bow side of the object. The methods of noisy object’s sound field prediction (recomputation) from the closed water area measurements to the open sea deep water waveguides’ conditions are suggested. The methods are based upon the modified reciprocity principle.
Physical parameters for on-stage acoustics based on optimal design for a rehearsal room
Authors: Kishinaga Shinji , Kawakami Fukushi
Published: October 2005

Architectural and acoustical conditions of an auditorium stage in comparison with a rehearsal room were studied in order to obtain a physical parameter for on-stage acoustics. The basic assumptions are as follows: (1) The auditorium is acoustically divided into two spaces: the stage and the seats. (2) The final aim of designing the rehearsal room is to realize the acoustic properties of the stage. (3) The properties of the sound field for the room are determined by the room’s materials, shape, and volume. It is an important fact that their fandamental properties are influenced by the basic architectural conditions. As a result four parameters have been proposed: α (EDT), ST, RT, and the ts-ratio (between stage and seats). They are shown as representative measures relating to sound absorption and shape in the stage area, volume of the seats, and acoustical similarity between the stage and the seats. In order to check the validity of these parameters, they were applied to the acoustical design of the rehearsal room aiming for the reproduction of the sound field of the stage. The results of these practical examinations show that not only the intended acoustical properties but also the aural impression of the stage were realized by means of sound field control technology ranging from architectual acoustics to electronic acoustics.
Subjective experiment on the effect of hall response on a violin player in a simulated sound field
Authors: Ueno Kanako , Sato Fumiaki , Tachibana Hideki , Ono Takahiko , Senju Mariko
Published: October 2005

In order to investigate the effect of hall response on players, a subjective test was performed in an artificially simulated sound field in an anechoic room. By referring to the data obtained in actual concert halls, the sound field was modeled and synthesized by using 14-channel reproduction systems: 12 channels for the ambient reverberation using simple digital reverberators, and two channels for late reflections from the rear part of the audience area using real time convolvers. In the test, the reverberation was fixed and the level and delay time of the late reflections were changed in three steps, respectively. Under each condition, the subject, a professional violinist, imagining that he/she was playing on a stage, played various pieces in a variety of playing styles and responded to questionnaires. As a result, it has been found that favorable stage conditions for a player differ as to the style of playing; solo, with piano, with orchestra, and in a quartet, and that the late reflection from the audience area is, to some extent, necessary in order to get ‘‘hall reaction.’’
Questionnaire survey on the subjective impression of stage acoustics of European concert halls through the Japanese Philharmonic Orchestra Tour
Authors: Toyota Yasuhisa
Published: October 2005

The acoustics of stage space has many more complex factors than that of audience space. It can be studied only through the view of musicians, and its evaluation depends on various factors such as their personality, their educational background, size of ensemble, type of music, conductor, and so on. Therefore, it is too risky to evaluate a stage space from the response of a limited number of musicians. It is desirable to gather objective and subjective data on the acoustics of stage space from as many musicians as possible. Using the opportunity of the Japanese Philharmonic Symphony Orchestra’s European Tour in April and May 1996, a questionnaire on acoustical impression of the stages was carried out in seven concert halls in Europe. The questionnaire asking for a total subjective opinion on stage acoustics, was distributed to each orchestra member at each concert hall. The results showed a wide variety of responses, but some tendencies could still be found depending on the concert hall. The detail of the survey will be presented and discussed in the meeting.
Effects of multiple scattering by trees in a forest as a music-performance space
Authors: Sakai Hiroyuki , Sato Shin-ichi , Ando Yoichi
Published: October 2005

In order to examine the effects of multiple scattering by trees, acoustical measurements were carried out on a pathway in a forest in Kirishima, Japan. Four physical parameters of sound fields, i.e., the listening level, the initial time delay gap, the subsequent reverberation time, and the IACC, were measured. It was found that such an outdoor space in a forest has excellent acoustic properties, especially regarding factors of the subsequent reverberation time and the IACC. It is interesting to notice that this kind of characteristic due to scattering by trees may be taken into consideration in designing a sound field by use of a number of columns in a concert hall.
The autocorrelation function of a sound field in an existing room as a time-variant system
Authors: Atagi Junko , Ando Yoichi , Ueda Yasutaka , Taguti Tomoyasu
Published: October 2005

The sound field in a concert hall is, rigorously speaking, a time-variant system due to changes in temperature and air current. These time-variant factors cause sound propagation that results in the fluctuation and sound-pressure level. This paper shows that such a fluctuation may be identified by the autocorrelation function (ACF). As is widely known, effective duration of ACF (τe) of a sound source, which is defined by the delay at which the envelope of the normalized ACF becomes 0.1, describes several important subjective attributes [Y. Ando, Concert Hall Acoustics (Springer-Verlag, Heidelberg, 1985)]. In this paper, values of τe of a music signal with changing tempo, dynamics, and articulation produced by a computer-controlled piano were analyzed. Results indicate that the measured values of τe in an existing hall are shorter than those of the source signal itself, while τe values of reverberant sound field simulated as a time-invariant system are 1.2–1.6 times longer than that of the source signal. Therefore, it is considered that the reverberance perceived in the existing concert hall might be much shorter than that of the time-invariant simulated sound field.
An equivalent phase error on the acoustical impedance measurements using a single microphone and pseudorandom sequence
Authors: Li Jing-Fang , Hodgson Murray
Published: October 2005

An experimental system has been developed for in situ measurement of the acoustical impedance of surfaces using a single microphone, pseudorandom sequences, and transfer-function techniques. It is well known that the impedance determined from two transfer functions is very sensitive to phase errors. The biased impedance has been expressed as a function of equivalent errors [J.-F. Li and J.-C. Pascal, J. Acoust. Soc. Am. 99, 969–978 (1996)]. In the system used here only one channel is used; thus the error does not include the electronic phase mismatch. However, since no experimental system is perfectly linear and repeatable, an equivalent phase error occurs. This error can be represented as the difference in phases between two average transfer functions measured sequentially at two locations. The objective of this work is to show how this error affects the impedance. The error causes a large bias at low frequencies. The results suggest that to reduce errors the distance between the two microphone locations should be large enough at low frequencies, and the number of transfer functions averaged should be as large as possible.
Sound propagation through the near-surface ocean bubble layer
Authors: Burch Holly , Buckingham Michael
Published: October 2005

Experiments have been conducted to investigate sound propagation through two types of bubble layers. The first is a motor boat wake, and the second is the near-surface ocean bubble layer beneath breaking waves. The bubble distribution over depth in both cases produces an upward-refracting sound-speed profile which acts as an acoustic waveguide. The motor boat wake is very stable and provides a controlled environment in which to study waveguide propagation, and, in contrast, the ocean surface bubble layer which is naturally generated by the injection of bubbles from breaking waves is much more variable. A broadband source was used to insonify the wake layer, while the bubbles formed by breaking waves were the acoustic source in the naturally generated ocean bubble layer. Inverse-square theory will be used to interpret spectra from both experiments, and comparisons will be drawn between the spectra seen for the stable wake experiment and the variable ocean bubble layer. [Work supported by ONR.]
Bubble size distributions in the wind driven surface layer
Authors: Farmer David M.
Published: October 2005

Bubbles injected by breaking waves move under the influence of buoyancy, near-surface turbulence and Langmuir circulation and are subject to gas transfer across the bubble skin. In order to study these effects a resonator has been developed for in situ measurement of bubble size distributions and has been deployed with other instrumentation in the open ocean at wind speeds up to 18 m/s. The resonator differs from the Medwin design in its use of two identical reflector plates to form a 1/2λ cavity immune to flow-induced or hydrostatic pressure fluctuations. Separate inversion of both sound speed and attenuation over a broad frequency range provides a useful consistency check on results. Measurements with a coherently operated array of self-contained resonators in drifting or moored configurations have yielded a database for detailed investigation of bubble size distributions. Turbulent diffusivity in the wave enhanced layer, vertical and horizontal velocity structure, and dissolved N2 and O2 were measured simultaneously with the bubble size distributions. Data are analyzed and interpreted with the aid of a two-dimensional model of the flow dynamics, incorporating representative turbulent diffusivity and advection and accounting for both primary gas constituents.
Bubble size distributions obtained from ambient noise measurements
Authors: Glegg Stewart A. L. , Auffret Yvan
Published: October 2005

A model has been developed for ambient noise in the ocean generated by breaking waves. The model has three parts: a bubble pulse spectrum based on the probability density distribution of bubble size, a propagation function and a wind speed-dependent factor specifying the number of events. The probability density of the bubble size is based on a theoretical distribution developed by Longuet-Higgins for the shattering of a large bubble into many small bubbles. The theoretical model has been compared with measurements of ocean ambient noise in the Tongue of the Ocean [R. M. Kennedy and T. V. Goodnow, IEEE J. Ocean Eng. 15(4) (1990)] for a range of wind speeds. The bubble size probability distributions are obtained as a function of wind speed using the best fit of the theory to the measured data. Mean-square errors of less than 1.5 dB are obtained for all the spectra considered. [Work supported by ONR.]
Imaging the acoustical sources beneath individual breaking waves
Authors: Epifanio Chad L. , Buckingham Michael J.
Published: October 2005

Breaking surface waves produce oscillating bubbles which generate high-frequency acoustic energy in the ocean. An experiment was conducted in May 1996 to examine the spatial and temporal evolution of the acoustically active regions beneath breaking surface waves within the frequency band 8–80 kHz. A multibeam, high-resolution, passive sonar, originally designed for acoustic daylight imaging, was aimed at the surface with a slant range of 45 m. The surface footprint was 9 m in diameter and contained 126 beams, each 0.75 m wide between the −3-dB points at the highest operating frequency. With this resolution, pictorial images have been obtained which show the spatial distribution and the spectral structure of the acoustic sources within individual breaking events. The spectral structure of the sources evolves on a time scale that is at least as fast as 40 ms, which is the temporal resolution of the system. The experimental site was 25 miles from the coast of southern California, with wind speeds rarely exceeding 15 kn. Examples of breaking-wave images will be presented and interpreted in terms of wind speed, wind direction, and horizontal anisotropy in the ambient noise field. [Work supported by ONR.]
A novel method for sorting piano note partials
Authors: Girolami G. , Rossi L.
Published: October 2005

A search procedure which takes into account the possible frequency fluctuations of the partials of piano notes is presented in this paper. This technique treats polyphonic sounds (as many notes as desired, provided that a sufficient number of their partials is present in the spectrum considered), in the same way as monophonic sounds and the computational time is linearly dependent on the number of partials recorded in a database. The identification of piano signals is difficult because of the problem of processing polyphonic signals. Depending on the model chosen, a note can be considered as harmonic [J. C. Brown, J. Acoust. Soc. Am. 92, 1394–1402 (1992)] or nonharmonic [Chafe and Jaffe, Proc. ICASSP, Tokyo, 1289–1292 (1986)] and the identification procedure is directly dependent on this model. The new sort procedure developed by the authors allows the sorting of harmonic, nonharmonic, or missing partials of a piano note.
Extraction of fundamental frequencies from the sound of a violin duet
Authors: Shirado Tamotsu , Yanagida Masuzo
Published: October 2005

An algorithm for extracting fundamental frequencies from the sound of a violin duet is presented. The proposed algorithm is composed of two processes: (1) framewise enumeration of candidates for fundamental frequencies using the cepstrum method, and (2) grouping among candidates over its adjacent frames using a hierarchical clustering method. Process (1) includes a subprocess of discriminating between the fundamental component of a tone and harmonic components of the other for duet tones in an octave relation, in particular. This algorithm differs from most of the traditional methods for extracting fundamental frequencies in the fact that it introduces loose constraints on target sounds, but no acoustical model is assumed for the instruments concerned. Thus the algorithm is expected to be applicable to many more types of musical instruments than conventional methods. The correct identification rate evaluated by musical notes was 93% for violin duet sounds in the best case.
Spectral energy balance in wind instruments
Authors: Hoekje P. L.
Published: October 2005

The nonlinear flow control mechanism in cane- and lip-reed wind instruments is explored by comparing the pressure p across the reed and the volume flow u exciting the air column under playing conditions. The pressures pu upstream and pd downstream of the reed are measured over time and their waveforms are subtracted to obtain the pressure difference p. The input impedance Zd of the instrument mouthpiece downstream of the reed is also measured, and the flowu estimated as the ratio pd/Zd. In this case, u includes both the airflow through the reed aperture and the air displaced directly by motion of the reed. The relation of u as a function of p gives the flow control characteristics. At low playing levels, u(p) is nearly linear, but becomes progressively nonlinear at louder playing levels, as predicted. For each spectral component, the variation of u with p is then separated into a linear part, which can be considered to describe the linear feedback mechanism, and a nonlinear part that is ultimately responsible for spreading energy among different components. Viewed in this way, individual spectral components can be identified as net energy producers or consumers.
Measurements of a thermoacoustic pin stack
Authors: Gibson Rodney J. , Nessler F. Scott , Keolian Robert M.
Published: October 2005

A comparison of the pin stack geometry with the conventional rolled geometry has been made in a prime mover which uses low-pressure neon gas held between room and liquid nitrogen temperatures. The pin stack was constructed by hand sewing a 75-μm Constantan wire back and forth about 2300 times between hot and cold heat exchangers. The wire forms a hexagonal lattice with wires along the acoustic axis spaced 750 μm apart with a 3.2% error rate. Preliminary results are that the onset of sound occurs at a mean pressure which is 40% lower for the pin stack compared to the rolled stack, the sound produced is generally 18% louder for the pin stack, and the efficiency of the pin stack is about 25% better than the rolled stack, although this last result has the most uncertainty due to the difficulties of reliably measuring the heat flow. [Work supported by ONR.]
Measurements with reticulated vitreous carbon stacks in thermoacoustic prime movers and refrigerators
Authors: Adeff Jay , Hofler Thomas J. , Atchley Anthony A. , Moss William C.
Published: October 2005

Reticulated vitreous carbon (RVC) is a porous carbon sponge. It is a low-cost, lightweight, machinable material that is manufactured easily. RVC has three important advantages compared to thermoacoustic ‘‘stack’’ materials that are used currently in prime movers and refrigerators: RVC is impervious to high temperatures; it has an extremely low thermal conductivity; and it has a high specific heat. In addition, there are no ‘‘straight through’’ optical paths like those in parallel-plate and conventional pin stacks, so that RVC may inhibit acoustic streaming. Theoretical thermoacoustic models for these RVC stacks do not exist, so the approach is empirical. Measurements are reported of the performance of several RVC stacks with different pore sizes, in a thermoacoustic prime mover and a thermoacoustic refrigerator. Initial results indicate a very good onset temperature and amplitude performance that in some cases exceeds the performance of traditional plastic roll and wire mesh stacks. [Work supported by Office of Naval Research.] a)Work performed under the auspices of U.S. Department of Energy by Lawrence Livermore National Laboratory under Contract No. W-7405-Eng-48.
Entrance effects in acoustic ducts: Experiment and theory
Authors: Mehl James B. , Gillis Keith A. , Moldover Michael R.
Published: October 2005

Nonuniform flow in the transition region where a circular duct is coupled to a large chamber has been modeled to lowest order by numerical solution of the Helmholtz equation describing the acoustic field. The associated end effects are described as a complex acoustic impedance Zend, which plays an important role in the Greenspan acoustic viscometer [Gillis et al., Rev. Sci. Instrum. 67, 1850–1857 (1996)]. The effects of geometry on the end effects, including the effects of chamfering sharp edges, have been calculated for a variety of orifice geometries and tested experimentally with Greenspan viscometers. A new formalism for modeling end effects by numerical solution of the coupled PDEs for the viscous and acoustic fields will also be described. The formalism can be extended to calculate entrance effects in other geometries, including thermoacoustic stacks. [Work supported by the Office of Naval Research.]
Convective heat transfer from a cylinder in a resonant acoustic field
Authors: Gopinath Ashok , Harder Donald R. , Bridenstine Mark
Published: October 2005

Experimental measurements of heat transfer rates from a cylinder in a zero-mean oscillatory flow will be presented. The cylinder is representative of a heat exchanger tube, while the oscillatory flow is typical of the strong acoustic field present in a thermoacoustic engine. A steady-state measurement technique has been used to determine the heat transfer coefficients. The effect of different flow regimes on the heat transfer behavior is considered. Correlations for the Nusselt number have been developed in terms of suitable governing dimensionless parameters for the different flow regimes. The goal is to develop a basic knowledge base of working correlations for the Nusselt number which would cover the wide operating parameter space of thermoacoustic engines. [Work supported by the Naval Postgraduate School.]
Thermoacoustic oscillation of a surface burner for premixed gas
Authors: Kwon Young-Pil , Lee Joo-Weon , Lee Dong-Hoon
Published: October 2005

A strong thermoacoustic oscillation of a surface burner made of a perforated metal fiber plate on the exit of a tube was investigated. First, the thermoacoustic power generation by a plane heater was analyzed theoretically based on the one-dimensional analysis of the heat transfer response of the heater to the oscillating flow and the result showed that the maximum power generates when the gas velocity is around (ωα)1/2, where ω is the angular frequency and α is the thermal diffusivity of the gas. Second, an experiment was performed for the frequency characteristics and the standing wave pattern in the tube for various fuel rates and air ratios. It was found that the modal frequency of the tube associated with the maximum power generation is driven by the heater and that the effect of turbulence lowers the frequency by increasing the effective thermal diffusivity. In addition, the measured impedance of the perforated plate on the exit of a tube showed that the thermoacoustic oscillation can occur although the heat source is not at the lower half of a tube corresponding to the condition for the Rijke thermoacoustic oscillation.
Integrated acoustic and language modeling of speech disfluencies
Authors: Shriberg Elizabeth E. , Bates Rebecca A. , Stolcke Andreas
Published: October 2005

This work investigates the use of prosodic features in modeling disfluencies (filled pauses, repeated words, and self-repairs) in spontaneous speech. The main goal is to automatically detect and correct disfluencies, so that a ‘‘fluent’’ version of a disfluent utterance can be used as input for speech understanding and other applications. A second goal is to develop explicit acoustic and language models for disfluencies to improve speech recognition performance for spontaneous speech. The prosodic features examined include duration, fundamental frequency, amplitude, and features correlated with voice quality. Decision trees serve as the acoustic models that relate these prosodic features to disfluency events. To integrate the disfluency model into speech recognition, decision tree probabilities are combined with standard acoustic model scores and probabilities from a ‘‘Clean-up’’ language model to rescore N-best hypotheses. The Cleanup language model represents disfluencies as hidden events, and predicts words following a disfluency from the corresponding fluent word sequence. A linguistically hand-annotated version of the Switchboard corpus is used for model training and evaluation. [Work supported by the National Science Foundation under Grants No. IRI-9314967 and No. IRI-8905249.]
Stochastic models of prosody for automatic spoken language systems
Authors: Veilleux Nanette M.
Published: October 2005

The prosody of an utterance, especially the placement and type of prominences and phrase breaks, is important in human understanding of speech. Prosody provides listeners with cues to factors such as syntactic structure, discourse information, and semantic content. Although prosody provides helpful and, sometimes, necessary cues for human listeners, it has not been used to a significant degree in automatic speech recognition or understanding systems. Some of the difficulties stem from an incomplete understanding of the relationship between prosody and higher level linguistic structures, e.g., discourse. Other barriers simply result from the intrinsically variable nature of prosodic production. In both cases, statistically modeling the relationship between prosody and the acoustic signal on one hand, and between prosody and higher level syntactic and semantic structures can be useful. The work here describes such a model, which expresses the mapping between the acoustic signal and syntactic/semantic structures, using prosody as an intermediary representation. Although the current model mainly uses syntax and word class information, extensions are described for handling some discourse phenomena. Analysis of the results using this approach in the context of the ATIS and other speech understanding tasks will be discussed. [Work supported by ARPA and NSF.]
The automatic generation of French declarative prosodic structures
Authors: Keller Eric
Published: October 2005

The fully automatic generation of a prosodic structure for a text-to-speech system has remained an elusive task. Most systems still depend on extensive user-specified timing and F0 markers to improve the approximation of a natural prosody. A three-stage processing system that provides fully automatic prosodic control for a wide variety of French declarative sentences has been developed. Perceptual testing has demonstrated a high degree of acceptability, as well as an excellent resistance to degradation induced by telephone transmission. The initial phonological processing phase, comprising phrase identification, liaison, chaining, and syllabification rules, produces a phonemic chain marked for prosodic, lexical, and syllable boundaries. The second processing phase calculates the temporal structure on the basis of a statistical model incorporating some 25 segmental and suprasegmental parameters. The final phase specifies accented and nonaccented syllables and applies Fujisaki modeling for the generation of the F0 contour. The system has been integrated into a TTS system that functions in real time on high-end personal computers. Examples are available on the laboratory’s ftp site (ftp://eliot.unil.ch/pub/LAIP/speech/LAIPTTS). This report will concentrate on the derivation of the relevant control parameters. [Work supported by FNRS, OFES.]
Modeling of pitch, loudness, and segmental durations in Finnish using neural networks
Authors: Altosaar Toomas , Vainio Martti , Karjalainen Matti
Published: October 2005

Several facets of the man–machine interface, such as speech synthesis and recognition in the spoken language realm, can be modeled using neural networks. Here neural networks have been applied to model the lexical prosodic parameters: segmental duration, loudness, and pitch, for the Finnish language. The prosodic models that were generated can be used in currently viable applications such as speech synthesis to further improve their naturalness. The text input stream was first converted into a phoneme sequence from which the input representation for the nets was generated. Inputs included: phoneme position in word, number of phonemes in word, and context in terms of previous and future phonemes. Optimal input representations for each type of prosodic net were searched for by varying the size of the input vector. The number of hidden nodes was also varied to determine the complexity of the problem. Estimating duration required class specific nets for the error to drop below 20%, the difference limen. For loudness it was 2.2 phon (1 phon is just noticeable), while pitch networks performed well with an error of 3.5% (equals 0.6 semitones at 100 Hz which is less than the 1.5 semitone perceptual intonation threshold).
On the model of global F0
shape for Japanese text-to-speech systems
Authors: Ishikawa Yasushi , Ebihara Takashi , Nakajima Kunio
Published: October 2005

A model of F0 control is one of the most important problems for the naturalness of synthesized speech in Japanese TTS systems. In general, a two-stage model which consists of a global model and a local model is used as a Japanese F0 control model. A local model generatesF0 contour for each accent phrase, a global model generates parameters of a local model from the linguistic information of an accent phrase. The parameter based on tree structure which is obtained from syntactic analysis is a typical parameter for the global model. However, in such a global model, it is difficult to express syntactical context of phrases, and syntactical analysis is also a difficult problem. A global model is proposed which has integrated F0 shape generation and syntactic analysis. This model is presented as a network of those states which show syntactical and prosodic states of sentences. In the model a linguistic class of input accent phrase decides a state to move, and generates a phrasal accent parameter for a local model when taking the transition. The training method of this network is also proposed. The predicted results showed that this model can predict the phrasal accent parameters with satisfactorily high accuracy. It strongly suggests that high quality synthesized speech can be obtained with the model.