bluuebunny/crossref_metadata_2025 huggingface dataset:
Enhancing second formant identification in speech by listeners with significant hearing loss
Authors: Nelson Peggy B. , Revoile Sally G.
Published: July 2002

Listeners with significant hearing loss have consonant recognition abilities that appear limited by their inability to use spectral peak information in the higher frequencies, specifically in the highly informative second formant (F2) frequency region. For them, ensuring the audibility of speech may not result in full recognition of consonants and vowels that differ by F2 characteristics. In the current study, listeners with moderate to severe hearing loss identified naturally spoken consonant-vowel-consonant stimuli (CVCs) with a wide range of F2/F1 formant frequency characteristics at levels that ensured the audibility of the F2 region. CVCs were presented in three conditions: (a) unmodified, (b) filtered to enhance the F1 and F2 regions, and (c) frequency-modulated to transpose the F2 peak to a lower, more detectable frequency region. The amount of modulation for condition (c) was determined by measuring individual spectral peak detection thresholds for each listener using the rippled-noise method described by Summers and Leek (1994). Closed-set identification of the CVC stimuli was measured and consonant confusion matrices generated. The amount of improvement due to enhancing the spectral contrast and to transposing the F2 region will be reported. Correlations between listeners characteristics and enhancement due to contrast sharpening and/or F2 transposition will be reported. [Research supported by NIDCD K08-000124 and by the University of Maryland School of Medicine.]
Continuous digit speech recognition using subword unit HMM juncture
Authors: Youn Jeh-Seon , Hong Kwang-Seok
Published: July 2002

In this paper, a new recognition system is described for digit strings of various length in Korean under fluent speech conditions using subword HMM juncture. A robust automatic recognition system for connected digit strings under spontaneous speech conditions is crucial for a number of applications such as voice dialing and credit card or personal identification numbers. It is thought that the recognition of connected digits is a relatively easy task, but the output of the recognizer must be a very exact transcription of the input speech. The process of recognition is to construct the database of digit string models by concatenating subword clusters not trained on target application through the typing task, avoiding the recording of training data each time the digit changes. Thus, the acoustic segments divide into N subword clusters, and each subword cluster generates the HMM model. The goal is to give users a flexible number recognition that can add new connected digits without collecting a new corpus or training new models, namely that can be easily modified for different applications with minimum amount of training and modification effort.
Fractal modeling of a glottal waveform for high-quality speech synthesis
Authors: Aoki Naofumi , Ifukube Tohru
Published: July 2002

From several psychoacoustic experiments, it is well known that the oversimplified glottal waveform degrades the quality of synthesized speech. For the enhancement of the quality of synthesized speech, it is desired to generate a glottal waveform in detail using an appropriate model which requires a few control parameters. In this study it is supposed that the differential glottal waveform, which is estimated by linear prediction filter, is modeled as a random fractal signal. The differential glottal waveform is regarded as the first integral of white noise whose frequency characteristic is approximated as −6 dB/oct. This signal can be interpreted as Brownian motion in terms of fractal theory. In order to generate random fractal signal, a kind of wavelet synthesis method called the midpoint displacement method is employed. Psychoacoustic experiments were conducted to examine whether or not the proposed method is effective. Experimental results indicate that the quality of original speech is fully preserved even if more than 90% of wavelet coefficients are replaced with a random fractal signal generated by the proposed method. The wavelet coefficients that cannot be replaced are to be the control parameters in this model.
Perceptual effects of spectral envelope and F0 manipulations using the STRAIGHT method
Authors: Kawahara Hideki , Akahane-Yamada Reiko
Published: July 2002

A versatile speech manipulation method, STRAIGHT (Speech Transformation and Representation using Adaptive Interpolation of weiGHTed spectrogram) [Kawahara, ICASSP (1997)], was applied to natural speech in order to test the effects of speech parameter manipulations. STRAIGHT consists of three procedures called STRAIGHT-core, TEMPO (Temporal domain Excitation extraction using a Minimum Perturbation Operator) and SPIKES (Synthetic Phase Impulse for Keeping Equivalent Sound). STRAIGHT-core is a method to extract a smoothed time-frequency representation, which is free from interferences due to the source periodicity. TEMPO is used to extract F0 and other source-related information. SPIKES provides artificial ‘‘naturalness’’ to the synthetic speech. As a base line, resynthesized speech, using the STRAIGHT method, was found to provide equivalent ‘‘naturalness’’ compared to the original speech when no parametric modification was introduced. Simultaneous manipulation of the spectral envelope and F0 illustrated that stretching the frequency axis of the spectral envelope, using the quantity of the cubic root of F0 stretching, preserved ‘‘naturalness’’ the best. Perceptual effects of group delay used in the SPIKES procedure were also investigated. [Work supported by CREST (Core Research for Evolutional Science and Technology) of Japan Science and Technology Corporation (JST).]
Measurements of temperature and velocity oscillations in acoustic waves
Authors: Huelsz Guadalupe , Lopez-Alquicira Francisco , Ramos Eduardo
Published: July 2002

Experimental techniques using cold and hot wire anemometry have been developed to measure acoustic temperature and velocity oscillations in gases, respectively, especially in the case where there is no imposed mean flow. The calibration of both instruments was made through the measurement of the pressure oscillation. Using these techniques the structures of thermal and viscous oscillatory boundary layers produced by an acoustic standing wave in air with 130-Hz frequency and pressure oscillation amplitude up to 200 Pa are determined. Experimental results are satisfactorily compared to theoretical results of a linear theory. The advantages and the limitations of these experimental techniques are discussed. [Partial support for this work has been provided by DGAPA-UNAM IN103197 and CONACYT G0044E projects.]
Absolute magnitude estimation of pitch
Authors: Rakowski Andrzej , Miskiewicz Andrzej
Published: July 2002

The pitch of 24 pure tones covering a frequency range of 63–12 500 Hz in 1/3-oct. steps was measured by absolute magnitude estimation. The tones were presented monaurally at a loudness level of 60 phons. Tone duration was 1 s. Sixteen music students, not possessing absolute pitch served as subjects. The experiment was carried out in individual listening sessions. The subjects were instructed to assign a number to the tone in such a way that their impression of how large the number was matched their impression of the pitch of the tone. The results show that the pitch function plotted in log–log coordinates against frequency is considerably steeper below than above 250 Hz. A similar shape of the pitch function was found in a previous study in which narrow-band noises were used [A. Rakowski, Proc. Fechner Day ’97, pp. 91–96]. It is hypothesized that the change of the pitch function steepness at 250 Hz may also be observed in the perception of musical intervals. Intervals in low pitch registers may be perceived larger than those corresponding to the same frequency ratios in high-pitch registers.
New experiments beyond the traditional Fourcin Pitch range
Authors: Raatgever Johan , Bilsen Frans A. , Mungra Reshma
Published: July 2002

The Fourcin Pitch is a dichotic pitch phenomenon occurring when a white noise signal and its interaurally delayed replica (delay T1) are presented simultaneously with a second (uncorrelated) noise signal with interaural delay T2. Former experiments dealt with T1 between 2 and 10 ms, while T2 was kept at a value between 0 and 2 ms [Bilsen and Wesdorp, 8th ICA (1974)]. With proper polarities of the noises and the delays at opposite ears, the pitch would typically have a value corresponding to 1/(T1+T2) and the (lateralized) pitch image would correspond to the shortest delay (T2). In the present study, both delay times were chosen about equal and with a value above about 5 ms; again the delays were presented at opposite ears, but with equal polarities. It turned out that a pitch is perceived with a saliency estimated between the Fourcin Pitch and the dichotic repetition pitch (one noise source only), with a value corresponding to 2/(T1+T2), and with a (lateralized) pitch image corresponding to an interaural delay of (T1−T2)/2. This new result, also, can be explained by the central spectrum theory [Raatgever and Bilsen, J. Acoust. Soc. Am. 80 (1986)].
Listeners’ spectral weighting functions for pitch judgment
Authors: Dai Huanping
Published: July 2002

Two experiments were carried out to determine how listeners use spectral information in judging the pitch of harmonic complex tones. In particular, spectral weighting functions were obtained using a correlation technique in which random, independent perturbations were added to the frequencies of the spectral components. The correlation coefficient between the perturbations applied to a given frequency component and listeners’ responses provides an estimate of the weight associated with that component. In the first experiment, all complex tones had the first 12 partials; the fundamental frequency was varied from 100 to 800 Hz. The obtained weighting function tended to show a peak for partials close to 600 Hz regardless of their harmonic numbers. In the second experiment, the fundamental was fixed at 200 Hz, and the number of partials was reduced by omitting the first two, four, or six partials. As these partials were excluded, the weights redistributed, with the maximum weight consistently going to the lowest partial present in the stimulus, i.e., at 600, 1000, or 1400 Hz. Overall, these spectral weighting functions reaffirmed the presence of a frequency dominance region, showing prominent influences on pitch judgment from low-ranked partials. [Work supported by NIH/NIDCD.]
Pitch mechanisms for unresolved harmonics I: Duration effects
Authors: White Louise J. , Plack Christopher J.
Published: July 2002

The experiment measured the effects of duration, fundamental frequency (F0), spectral region, and relative phase of harmonics on F0 discrimination for complex tones consisting of unresolved harmonics. The F0 discrimination was measured for complexes with durations between 20 and 160 ms, and with F0s of 62.5, 125 and 250 Hz. The complexes were bandpass filtered between either 2750 and 3750 Hz or between 5500 and 7500 Hz. Finally, either all the harmonics were in sine phase (SINE) or alternating harmonics had a cosine phase shift (ALT). The pitch of an unresolved ALT complex is one octave higher than that for an unresolved SINE complex with the same F0 [T. M. Shackleton and R. P. Carlyon, J. Acoust. Soc. Am. 95, 3529–3540 (1994)]. For the SINE stimuli the improvement in F0 discrimination with duration was dependent on F0, with a larger effect of duration for the lower F0s. However, for the ALT stimuli the improvement with duration was the same as that for a SINE complex with twice the F0. Overall the results suggest that pitch, rather than F0 per se or spectral region, is the main determinate of the duration effect and also possibly of the integration time of the pitch mechanism.
Incorporation of inhomogeneous, atmospheric turbulence into numerical propagation calculations
Authors: Wilson D. Keith
Published: July 2002

Homogeneous, atmospheric turbulence is often synthesized for use in numerical propagation calculations (the parabolic equation, for example) by multiplying Fourier modes by random phases and then inverse transforming. This method can be generalized to inhomogeneous turbulence by multiplying the empirical orthogonal functions (EOFs) by random phases; in homogeneous turbulence the EOFs become the Fourier modes. The EOF technique is applied in this paper to the case of turbulence in a vertically inhomogeneous, atmospheric shear layer, where the integral length scale is proportional to the height. The EOFs are determined numerically, and then the turbulence is synthesized and incorporated into calculations of scattering into a shadow zone. The results are compared to previous calculations involving homogeneous turbulence.
Design and acoustic characteristics of an anechoic chamber with high-power heating or cooling and sound field free transform system
Authors: Yang Weicheng , Yang Jiahua
Published: July 2002

In this paper, the design and the acoustic characteristics of an anechoic chamber with high-power heating or cooling condition and sound field free transform system are summarized. Moreover, during the construction of the anechoic chamber, the following technical problems have been solved successfully. (1) Under 15 kW heating or cooling energy load being released by the sound source in the anechoic chamber, the adjusting to the temperature and the humidity can be completed. (2) Through the moving of the mobile sound absorption plane, two anechoic chambers can be combined into one anechoic chamber, one semi-anechoic chamber, or one octant semi-anechoic chamber. Its sound field characteristics conform to the standard ISO 3745. (3) During the normal state, the background noise in the anechoic chamber is less than 23 dBA.
Hand-held analyzer systems for building acoustics and sound intensity/sound power
Authors: Larsen Peter
Published: July 2002

Two new applications have been developed for a hand-held analyzer: Building acoustics and sound intensity/sound power. The building acoustics application is designed for measurements of airborne and impact sound insulation and reverberation time. The sound intensity application is used for sound power measurements, noise source location, noise mapping, and building acoustics. Both applications we used with complete systems consisting of all the necessary parts to simplify the measurement task and provide the user with all results immediately at the measurement site. There is no need for postprocessing. The measurements can be done according to a wide selection of international and national standards. The paper will describe the use of both applications in solving a problem in building acoustics. The building acoustic application will be used to detect a problem with poor sound insulation of a wall. Then, the intensity technique is used to determine the cause of the problem. Both applications use a built-in noise generator in the hand-held analyser with wireless transmission to an amplifier/loudspeaker. The paper will focus on the various design features that have been incorporated to ensure easy, secure, and flexible use.
Precision measurements of thermoacoustics in a single pore
Authors: Petculescu Gabriela , Wilen Larry A.
Published: July 2002

Continuing experiments designed to look at thermoacoustic effects in a single pore will be described. A volume-modulation technique is employed to measure the complex compressibility of a gas in the pore directly. When a temperature gradient is imposed on the pore, the complex compressibility is sensitive to thermoacoustic effects which can be measured to high precision. These effects depend on both thermal and viscous properties of the gas. Measurements on different gases and mixtures of gases will be presented and compared to predictions of the theory. [Work supported by Ohio University Research and Sponsored Programs.]
Detecting irregularity of envelope modulation
Authors: Sheft Stanley
Published: July 2002

The ability to detect periodic irregularity in the temporal envelope of a modulated stimulus was measured with a 2IFC procedure. Irregular amplitude modulation (AM) was generated with a periodic phase modulation (PM) of the sinusoidal modulator of wideband noise. The modulation index of the PM was adaptively varied in terms of delta envelope rate to determine the threshold to just detect the PM of the sinusoidal amplitude modulation (SAM). Base SAM rate ranged from 4 to 256 Hz. For each SAM rate, the PM rate ranged from 2 to half the SAM rate. The PM waveshape was either sinusoidal or had linear ramps with slope varied across conditions. At low SAM rates, PM waveshape had little effect with thresholds across conditions rising with PM rate. At higher SAM rates, an effect of waveshape was obtained; thresholds were highest at lower PM rates with a steeply sloping linear-ramped PM. For the other PM waveshapes at the higher SAM rates, thresholds were roughly constant across change in PM rate. Results suggest two rate regions in processing AM with the regions separated at roughly the corner frequency of the temporal modulation transfer function. [Work supported by NIH and AFOSR.]
The influence of different speech segments on spatial localization
Authors: Karlsen Brian L.
Published: July 2002

Numerous researchers have investigated to what degree humans are able to localize different sounds in the horizontal plane. Much of this work has focused on human response to simple sounds, like pure tones, white noise, and click trains, the underlying assumption being that responses to more complex stimuli could be inferred from a combination of responses to the simple stimuli. It was decided to investigate this for speech segments. Specifically, it was hypothesized that subjects would display a difference in localization ability for CVs containing some consonant manner groups rather than others. A number of Danish CVs were used as target stimuli in a horizontal localization task. The CVs were simultaneously masked by a diffuse white noise field. All subjects were normally hearing. The subjects were asked to determine from which clock position they heard a speaker, and to respond as fast as possible by pressing the corresponding button on a clock face. Final results are expected to confirm the hypothesis in full and indicate that there is indeed a difference in localization ability between the different consonant manner groups.
On the difference between absolute and relative auditory distance perception
Authors: Kim Hae-Young , Suzuki Yôiti , Takane Shouichi , Ozawa Kenji , Sone Toshio
Published: July 2002

Absolute distance perception in the absence of information about intensity and reflections from boundaries is quite difficult [B. G. Haustein, Hochfrequenztech. u. Elektroakustik 78, 46–57 (1969)] but is possible to achieve if a sound source exists within the distance of around 1 to 2 m from a listener. In the presence of multiple sound sources, however, some cues can be derived by comparing information from different sources. From this point of view, it is important to study relative distance perception between sounds from multiple sources to investigate general distance perception in auditory system. Thus using the same experimental setup, two kinds of listening experiments about the absolute and relative distance perception were conducted. In the experiment for relative distance perception, a subject was asked to report the perceived distances of two sound sources presented successively with an interval of 2 s by using a rating scale with seven categories. The results of absolute distance perception showed that the perceived distance of a sound image significantly increases with the increase of actual distance of the sound source up to around 1.2 m. On the other hand, the results of the relative auditory distance perception showed that the perceived distance significantly increases up to 2 to 3 m.
Echolocation by moving and stationary listeners
Authors: Gordon Michael S. , Rosenblum Lawrence D. , Jarquin Luis
Published: July 2002

It has long been known that human listeners can echolocate a sound reflecting surface as they walk towards it [e.g., Supa et al., Am. J. Psychol. 57, 133–183 (1944)]. There is also evidence that stationary listeners can determine the location, shape, and material of nearby surfaces from reflected sound [Rice, Science 155, 656–664 (1967)]. The current research tested whether there is an advantage of listener movement for echolocating as has been found for localization of emitted sounds [Ashmead et al., J. Exp. Psychol.: Hum. Percept. Perform. 21, 239–256 (1995)]. Two groups of blindfolded subjects were asked to echolocate a 3×6 ft wall while either moving or stationary. After echolocating, the wall was removed, and subjects were asked to walk to where the wall had been. Subjects in the moving and stationary groups echolocated for the same amount of time and for each of four distances. Results showed that moving group subjects were more accurate than stationary group subjects for some distances. These results might be a function of useful time-to-arrival acoustic information available to moving listeners.
Nonlinear wave propagation from a discrete annular array: Experiments
Authors: Edson Patrick , Roy Ronald A. , Kargl Steven G.
Published: July 2002

To date, efforts to model nonlinear wave propagation from a bounded beam source have chiefly been concerned with unfocused and weakly focused circular piston sources [Lee and Hamilton, J. Acoust. Soc. Am. 97, 906–917 (1995)]. When the peak pressure does not exceed approximately 150 MPa, the Khokhlov–Zabolotskaya–Kuznetsov (KZK) nonlinear parabolic wave equation yields numerical solutions in agreement with experiments. However, little research has been reported on nonlinear wave propagation from an array of discrete sources. Theoretical and experimental results for an eight-element annular array will be presented. The source signal is typically a ten-cycle sine wave tone burst with a center frequency of 500 kHz. The outer diameter of the array is 0.14 m, and the surface areas of the elements are equal. While theoretical predictions will be shown, the experimental aspects of this research will be emphasized. [Work supported by ONR.] by Kargl
The parametric efficiency of a bifrequency focusing Gaussian nonlinear source
Authors: Özyar M. Serhat
Published: July 2002

The field distribution from a bifrequency focusing nonlinear source was calculated. The simple formulation of Novikov et al. [‘‘Nonlinear underwater acoustics,’’ ASA, 86–90 (1987)] for the Gaussian wavefront was adopted for simulations. Radius of aperture or beamwidth (a), radius of curvature (D), mean primary frequency (fm=(f1+f2)/2), and difference-frequency (fd=f1−f2) are essential parameters describing the performance of a parametric focusing source. The geometrical ratio D/a and the down–shift were varied between 5 and 200 by changing fd and the resulting difference–frequency pressure wave and primary intensity distributions were plotted against the propagation axis for D/a=2. A similar calculation was repeated for fd=50 kHz by varying fm. Separate effects of D and a were also investigated where the down–shift ratio was fixed to 20. Results were plotted for D/a: 1–4. Finally, an efficiency was defined as η=(parametric pressure)/(primary intensity). It was presented that η made a peak in the prefocal region and reached to a minimum value at the geometrical focus. The variation of η in the postfocal region was also discussed.
Determination of roots of eikonal equation for WKB solution in cochlear models
Authors: Parthasarathi Anand A. , Grosh Karl , Nuttall Alfred L.
Published: July 2002

The WKB-theory-based asymptotic methods have been used to obtain solutions to analytical cochlear models [E. de Boer, Phys. Rep. 105(3), 141–226]. The local wave number for this solution is obtained as the root of the eikonal equation or the dispersion equation. For a three-dimensional rectangular model, multiple complex roots exist, which could give rise to additional waves on the basilar membrane. However, previous results were based on a single root locus determined using initial estimates obtained from the long-wave approximation. In this study, the winding-number integral technique [P. R. Brazier-Smith and J. F. M. Scott, J. Sound Vib. 145(3), 503–510] is used to obtain a definitive identification of the multiple roots and their dependence on the number of acoustic cross modes. A priori information about the characteristics of the dispersion relationship was used to obtain an initial estimate of the search region. Thereafter the region was refined and the roots located using moments of the winding integral and secant iterations. It was found that higher-order roots are strongly evanescent away from the resonance location. Also, as the number of acoustic cross modes of the duct increases, more roots appear. The influence of these roots on the solution will be discussed.
Detection of an increment in a single-frequency component of a noise background as a function of increment frequency and duration
Authors: Formby C. , Heinz M. G. , Aleksandrovsky I. V.
Published: July 2002

Increment detection thresholds for a single-frequency component were measured over a range of increment frequencies (250–6000 Hz) and durations (T=10–500 ms) in a 500-ms, 6000-Hz-wide noise. The noise spectrum level was either constant (N0=40 dB) or randomly varied (between N0=20 and 40 dB) from interval to interval. Preliminary results reveal performance for each increment frequency to be similar for the fixed- and random-level conditions. Detection thresholds for T=10 ms are largely independent of frequency and correspond to S-N0 values of 17–18 dB, whereas corresponding S-N0 values measured for long duration increments increase at a rate of 3 dB/oct for increment frequencies between 500 and 4000 Hz. The latter finding is consistent with changes in the critical ratio. This ratio is assumed to be based on an absolute energy cue, which is inconsistent in the random-level condition. Temporal integration (TI) functions fitted to the normalized detection data reveal less integration with increasing increment frequency from 500 to 4000 Hz. When the detection results are corrected for the apparent ‘‘critical ratio’’ effect, the TI functions are largely independent of increment frequency. [Work supported by ONR.]
A unified theory of two-tone suppression and upward-spread of masking
Authors: Allen Jont , Sen Deep
Published: July 2002

Low-frequency two-tone suppressors (TTS) [Sachs and Abbas], and the upward spread of masking, have traditionally been treated as separate phenomena. We postulate that both these well-known response measures are due to nonlinear (NL) cochlear compression. Furthermore, we attribute this NL compression to a changing basilar membrane (BM) stiffness, which varies inversely with hair cell stimulus. A 65 dB SL low-frequency tonal masker/suppressor excites a broad basal region of the cochlea uniformly, raising the threshold of high-frequency probes throughout this portion of the cochlea. Sufficiently intense low-frequency maskers/suppressors cause a reduction in the stiffness in the base of the BM, resulting in the probe characteristic frequency shifting towards the base. This basal NL shift of the BM response, combined with high-pass filtering by the tectorial membrane, results in an attenuated basal hair-cell response. This change in gain accounts for the expansive slope of the upward spread of masking, having a maximum slope of 2.4 dB/dB. This slope implies a compression proportional to pressure to the 0.8≊2/2.4 power. The same model explains TTS if we treat the CF tone as the probe and the suppressing tone as the masker. The rate of growth of suppression, in this case, is also known to reach a maximum of 2.4 dB/dB [Delgutte].
Sound scattering by a fluid-loaded finite cylindrical shell
Authors: Klauson Aleksander , Metsaveer Jaan , Touraine Nicolas , Decultot Dominique , Maze Gerard
Published: July 2002

Let a thin-walled finite cylindrical shell with the hemispherical end caps be immersed in a fluid medium. The shell contains structural inhomogeneities like radial solders and internal ribs and is excited by an axisymmetric pressure load. The paper deals with the numerical and experimental investigation of the sound-pressure field scattered by the shell and its internals. The numerical solution of the problem is based on the finite element method (FEM) with the use of the thin shell theory elements. The Green’s function of the structure is obtained by the FEM and is further used in the boundary element solution scheme using both physical and modal coordinates. Flexural and membrane type eigenmodes of the fluid-loaded structure can be determined and identified. A time-domain solution to the problem obtained by the inverse Fourier transform gives echo-signals of the sound scattered by the structure. Individual pulses in the echo-signal help to localize structural inhomogeneities and to appreciate their importance in the scattered sound field. The membrane type vibrations are shown to be substantially modified by the internal solders and substructures. Experimental investigation is performed for a steel target immersed in a water tank and the numerical results correlate well with the experiment.
Visualization of the sound field generated by a plate-cavity coupled system using acoustic holography
Authors: Kim Sea-Moon , Kim Yang-Hann
Published: July 2002

In most cases of structural-acoustic problems, it is reasonable to assume that structural vibration is not influenced by the surrounding fluid. In these cases, the vibration of the structure is solved first, then the radiation sound field is calculated by simply applying the Kirchhoff–Helmholtz integral equation. However, this assumption is no longer satisfied when structural stiffness is small or fluid impedance is comparable to it. In this situation the vibration and acoustic fields are to be solved simultaneously. Although many researchers have studied this structural-acoustic coupling problem, there are still difficulties in solving the problem analytically or even numerically. In this study, visualization of sound field by a geometrically simple system (plate-cavity coupled system) is performed experimentally in order to figure out the coupling mechanism between fluid and structure. The system is excited by a speaker, and both internal and external sound pressures are measured. The acoustic holographic method is used to estimate the sound field. The results exhibit that there are frequencies where both plate and cavity are strongly coupled as well as ones where the plate can be considered rigid. Visualization that shows acoustic power flow between the internal cavity and external field enables us to understand the fluid-structure coupling mechanism.
Recent advances in speech recognition for spontaneous speech translation
Authors: Sagisaka Y.
Published: July 2002

Speech recognition research for speech translation at ATR will be reviewed. For spontaneous speech recognition, new methods for parameter extraction will be shown that yield acoustic measures to allow discriminative acoustic model training. A new HMM state-sharing algorithm (ML-SSS), fine-acoustic modeling using stochastic segment models, and recurrent neural networks will also be described. The MAP-VFS speaker adaptation techniques will be presented in the context of speaker clustered acoustic models. As for language modeling, variable order N-gram models and task adaptation of language modeling will be described. In addition, further research efforts on pronunciation modeling and training using error characteristics will also be reviewed. Finally, an outline of the continuous speech recognition system (ATR-SPREC) for spontaneous speech will be presented, along with a review of recent recognition experiment results.
Global autofocusing techniques for acoustic imaging
Authors: Gough Peter T. , Lane Richard L.
Published: July 2002

When images of acoustic fields are computed from measured data, the quality of the image is sometimes impaired by the acoustic wavelength limits or by turbulence in the media. However, more often, the image is impaired by random displacements in the position of the detecting array. Techniques exist to optimize the image on the basis of a priori knowledge. Since this knowledge is often statistical rather than deterministic, the image computed is the most probable image given the a priori knowledge. There can be no guarantees that what is computed is the exact image since there is an infinite combination of acoustic fields and tracks able to produce the same measured field. Using examples from some simulations of an existing synthetic aperture sonar, it is shown (by appealing to Bayesian statistics) that statistical information about the towfish behavior can be used to improve the quality of the image. Inherent in this technique is a cost function to be minimized and it is the precise nature of the cost function chosen that produces the most probable image. It is techniques such as this that hold out the prospect of SAS and passive towed array sonars without the need for on-board navigation units.
The next steps: Four examples of Artec’s evolving room acoustics philosophy
Authors: Johnson Russell , Wolff Robert W. , Doria Damian J. , Goodall Ashley
Published: July 2002

This paper examines four Artec projects that have opened in 1997, and relates them to a room acoustics philosophy that has been evolving in Artec’s designs since 1970. An illustrated discussion of the Winspear Centre (Edmonton, Alberta), the Living Arts Centre in Mississauga (Ontario), the Chan Centre (Vancouver, British Columbia), and the renovations to the stage acoustics at Music Hall, Cincinnati, highlights the importance of seat-count, complete freedom from exterior noise, room shaping, and a considerable amount of adjustability in order to achieve acoustical success in performing arts facilities.
Evening fish choruses near coral reef systems in the Great Barrier Reef, Australia
Authors: McCauley Robert D. , Cato Douglas H.
Published: July 2002

The nighttime ambient sea noise on and around coral reef systems in northern Australia is characterized by choruses produced by nocturnally active fishes. These choruses are produced year round, from dusk to dawn. Choruses are composed of many ‘‘pops’’ of between 5 and 28 ms length, with a decay envelope suggestive of a lightly damped swimbladder and with source levels calculated at 154–157 dB re: l μPa2 at 1 m peak to peak. Sources are found mostly concentrated over 6–20 m water depth. Choruses have a mean spectral peak at 550 Hz (normal range 500–750 Hz) but may influence sea noise from 100 Hz to 3 kHz. The highest chorus levels occur just after dusk or in the early evening, reaching up to 35 dB above background levels. The recorded levels of choruses displayed distinct daily patterns, lunar patterns with higher levels around new moon periods, and seasonal patterns of highest level during the austral summer. Choruses occur over large spatial scales, with active sources heard at up to 15 km from reef systems and chorus influence on ambient sea noise extending considerably further. It is believed that nocturnal planktivorous fishes, such as those from the families Holocentridae and Priacanthidae, are responsible for choruses. [Work supported by the Australian Defence Science and Technology Organization.]
Nonlinear response of microbubbles to pulsed diagnostic ultrasound
Authors: Hilgenfeldt Sascha , Lohse Detlef , Zomack Michael
Published: July 2002

The sound radiation from micrometer-size bubbles driven by short ultrasound pulses emitted by diagnostic ultrasonography devices is investigated. The frequency spectrum is analyzed in order to predict useful parameter ranges for the application of the second harmonic method.
Acousto-mechanical imaging for cancer detection
Authors: Sarvazyan Armen
Published: July 2002

One of the hot areas of ultrasonic medical diagnostics is elasticity imaging (EI). Recently, an alternative method of imaging tissue structures in terms of their elasticity, the method of mechanical imaging (MI), has emerged. The essence of MI is the solution to an inverse problem using the data of stress patterns on the surface of tissue compressed by a force sensor array [A. P. Sarvazyan, Proc. Tenth IEEE Symposium ‘‘Computer based medical systems,’’ Maribor, Slovenia, 120–125 (1997)]. A prototype of the device for mechanical imaging of the prostate has been developed and feasibility of MI technology has been proven in recent clinical studies. The present paper describes a modality of medical imaging, acousto-mechanical imaging (AMI) comprising features of both EI and MI. The data on the stress pattern measured by the force sensor array of MI complement the strain data obtained by ultrasonic EI device. An embodiment of the AMI device comprises a force-sensing array made of sound transparent film inserted between an ultrasonic scanner probe and the tissue. Synergy of combining these two complementary methods results in the fact that for certain applications the AMI technology has a diagnostic potential much superior to that of EI and MI separately.
Sensor engineering at the microscale
Authors: Gabrielson Thomas B.
Published: July 2002

Modern techniques for microfabrication have led to visions of significant new sensor applications. However, the translation of devices from centimeter scales to micrometer scales can cause effects that are unimportant at the larger scales to dominate the behavior at the smaller scales. When sensor structures approach material grain sizes or ferroelectric or ferromagnetic domain sizes, design philosophies need to be adjusted; when clearances are very small, viscous and thermal effects can dominate the dynamic response. While microfabrication has made dimensional downscaling of several orders of magnitude possible, intuition and ‘‘engineering practice’’ developed on the macroscale may lead to unsound devices on the microscale. Successful miniaturization of devices and structures depends on an understanding of the changing roles of various forces, energies, and interactions on the appropriate scales. Even with this level of understanding, optimum performance may not be achieved by simple shrinking of a macroscale device. Nowhere is this more evident than in biological organisms where markedly different solutions to sensing, searching, navigation, locomotion, and thermal regulation are used by organisms of different sizes.
Active control of sound transmission through an industrial sound encapsulation
Authors: Sas Paul , Dehandschutter Wouter , Boonen Rene , Vecchio Antonio
Published: July 2002

In this paper, an active control system is developed to enhance the sound transmission loss of a sound encapsulation at low frequencies. The sound encapsulation is used to shield a noise source, such as an air compressor. The control systems comprise two control loudspeakers and two error microphones. The Filtered-X LMS adaptive feedforward control algorithm is used to drive the signals at the error microphones to a minimum. The reference signal for the control algorithm is taken from the noise source, i.e., the tacho signal of the compressor. This study deals with the optimization of the location of control sources and error sensors in view of the achieved control performance. From this perspective it is important to note that minimizing the sound-pressure level at the error microphones, which are constrained to be located inside the sound encapsulation, does not necessarily reduce the acoustic intensity measured in the far field of the encapsulation (or at least, not to the same degree). Relevant issues which are considered are: the coupling of control sources with the acoustic modes of the enclosure, changes to the interior sound field due to active control, etc.
Non-native listeners’ representations of within-word structure
Authors: Otake Takashi , Yoneyama Kiyoko , Maki Hideki
Published: July 2002

Human listeners may form conscious representations of potential within-word structure in which lexicon is represented by some phonological units. An earlier study examining monolingual speakers of Japanese and English with native inputs suggests that levels of representation by Japanese speakers may be involved with richer knowledge of word-internal structure, while English speakers are sensitive to syllables [Otake et al., Proceedings of EUROSPEECH 95 3, 1703–1706 (1995)]. The present study investigated how monolingual speakers of English learning Japanese could form conscious representations of potential within-word structure in Japanese. Three groups of subjects (N: 36, 33, and 40 for three different levels) were presented with 150 Japanese spoken words and asked to mark on a written transcript of each word the second natural division point from the onset in the word. The statistical analysis showed that all groups exploited syllables to represent Japanese words irrespective of Japanese proficiency. These results suggest that the exploitation of phonological unit to represent within-word structure in foreign inputs may be the same as the one in the native input as proposed by our recent pilot work (Otake and Yamamoto, 134th Meeting of the ASA 101). [Work supported by Fulbright senior research grant and TAF.]
Mediacoustics, a software package on CD for teaching acoustics using multimedia techniques
Authors: Wahlrab Michael
Published: July 2002

Mediacoustics uses multimedia techniques on CD for teaching acoustics. It is divided into four study areas: basic physics, noise and man, room acoustics, and noise control. Sound clips, text, pictures, photos, and video animation illustrate each module. The index and chapters may be edited. Individual text passages may be annotated and an included acoustical dictionary provides definitions of common terms and formulas. Any contemporary multimedia computer equipped with a sound card and amplified loudspeaker can be used to access Mediacoustics and its hundreds of video clips.
Study on acoustic index variations due to small changes in an observation point
Authors: Sekiguchi Katsuaki , Hanyu Toshiki
Published: July 2002

Acoustic index variations due to small changes in an observation point in a concert hall were studied. Three kinds of methods were used in order to measure the acoustic indexes: (1) one point measurement using a single channel microphone; (2) binaural measurement using a dummy head microphone; and (3) the regular tetrahedron peak point method using a 4-channel microphone system. The acoustic index variations in each of the methods were analyzed. In addition, subjective experiments were performed in order to evaluate the effects of the variations. Compared to these measurements, problems with the acoustic indexes which were analyzed by each of the methods were found. One point measurement may incorrectly evaluate a concert hall.
Subjective assessment of the uneven distribution of reflections within a concert hall
Authors: Collados Eugenio
Published: July 2002

Most physical variables other than RT are known to vary significantly within a concert hall. Namely, early to late energy ratio, lateral energy fraction, and sound strength. Less well known are the subjective correlates to the above variations. A series of experiments were carried out in order to assess the number and meaning of subjective dimensions throughout a single hall. Testing environments were both an anechoic chamber array and an actual hall with artificial sources, where subjects were asked to judge classical music samples. In both cases the standard spatial distribution was modified by the addition of discrete reflections while keeping level and RT constant. At least four subjective dimensions were found, as many as found in hall-to-hall comparisons. Up to 59% of variance was related to the spatial image of the source. After comparing results between the anechoic chamber and the actual hall, the visual impression seems to partially compensate for the variations due to distance. These results raise some doubts on the validity of listening tests performed without a visual stimulus. It is also suggested that these effects should be considered in designing halls with active systems.
An experiment to identify preference groups among concert hall listeners
Authors: Halstead M. Miklin , Nielsen Johan L. , Marshall A. Harold
Published: July 2002

The task of specifying the acoustical behavior of a concert hall depends upon knowing the preferred subjective acoustical characteristics for the intended program material. Several studies have suggested that the listening population might be divided into groups with respect to their preferred acoustic, with different weightings applied to different subjective factors in each. Our project endeavors to identify and describe these groups, and to make their composition an integral part of hall design. This paper describes a subjective experiment designed to identify and measure individual listening preferences in a simulated concert hall setting, and presents the experiments first results. Following from these results, an attempt is made to place tested individuals into preference groups. The applicability of the experiment is considered in terms of the quality of hall simulation and of the impact of the simulator and experimental design on the listeners discrimination.
Variations in the surface textures of choral reflectors
Authors: Lee Elizabeth J. , Yule Campbell J.
Published: July 2002

The commission to design a portable choral reflector seemed a simple process of finding a balance between acoustical and structural requirements. Initial inquiries, however, found a lack of information on the acoustical effect of different surface textures normally used. An experiment was developed to assess both the subjective and objective performance of varying degrees of surface roughness. Eight possible surfaces were chosen and applied to standardized panels. The impulse response of each panel was measured in an anechoic room using a maximum length sequence signal. Early results indicate significant differences in the reflected energy and frequency response of the respective panels. An attempt to obtain a subjective evaluation of the reflections by convolving choral and solo voice recordings with the impulse response of the panels will be reported. A prototype choral reflector based on this work will be described. [The support of the Acoustics Research Centre for this project is acknowledged.]
Acoustic and system parameters affecting destruction of ultrasound contrast agents
Authors: Chang Peter P. , Makin Inder Raj S. , Crum Lawrence A.
Published: July 2002

In order to effectively use ultrasound contrast agents (UCAs), it is necessary to tune the ultrasound scanner parameters to a particular agent or change the characteristics of the UCA to target-specific applications. Both efforts are being sought by ultrasound manufacturers and pharmaceutical companies. The destruction of UCAs is usually undesirable, although it can sometimes be used advantageously such as in harmonic imaging mode. In the present work, changes in acoustic and system parameters such as frequency, transmitted acoustic pressure, pulse repetition frequency, flow rate, diluting media, concentration, and type of contrast agents were measured at both first and second harmonic. A prototype Doppler system previously described [Chang et al., Ultrasound Med. Biol. 22, 1205–1214 (1996)] has been adapted to carry out these measurements. Results with Albunex show that bubble destruction increases with an increase in flow rate, acoustic pressure, PRF, and decreases with an increase in the concentration of the contrast agent and oxygen content in the diluting medium. In conclusion, the rupture of UCAs are influenced by several imaging parameters. The threshold values when destruction of the agents occurs should be known in order to optimize the use of UCAs in clinical settings. [Work supported by DARPA-TRP.]