bluuebunny/crossref_metadata_2025 huggingface dataset:
Geometric and dynamic effects of stages in concert halls
Authors: Lee James B.
Published: October 2005

Wallace Clement Sabine's place in the history of concert halls is definitive, so much so that many have forgotten the seminal work of Joseph Henry, even though they use it implicitly in their practice. But Henry, writing from 1856, still has much to tell us: His famous echo experiment quantified the boundary between perception in the time domain and the frequency domain in terms of architectural scale—about 8 m—which implicitly has been made the boundary between large and small rooms. Concert halls are rightly thought of as large rooms. But look at the stage of Boston Symphony Hall: 8 m deep, bounded by an open trapezoid of very hard and very plane walls! If it were smaller, an orchestra would not fit; if it were larger, players would hear echoes and ensemble would suffer. This “Henry” scale stage works in the frequency domain too: Bass viols, with their low E of 41 Hz, λ = 8 m, are less than 1/4 wavelength from a wall; not only is their sound reflected geometrically, but also its source strength is augmented due to the enhanced impedance of radiating in a pressure zone.
Acoustical design of Tokyo Geijutsu Bunka-Kaikan. Part 2. Room acoustical design of the large hall
Authors: Nagata Minoru , Ikeda Satoru , Oguchi Keiji
Published: October 2005

The large hall of the Tokyo Geijutsu Bunks-Kaikan is planned as a concert hall for classical music with 1887 seats. At the first stage of the design, the room shape and seating arrangements proposed by the architect were studied by computer simulation, in which the energy distribution of the early reflections over the seating area was observed. The room shape was repeatedly studied and re-designed considering the architectural design policy and acoustical characteristics. At the final stage of the design, a ray experiment in a 1/50 scale model was introduced. Acoustical experiments in a 1/10 scale model are planned in order to predict the room acoustical parameters and to check on detrimental echoes. The room acoustical design policy and the details of the new auditorium will be presented.
A study of the characteristics of early reflections in concert halls
Authors: Toyota Yasuhisa , Oguchi Keiji , Nagata Minoru
Published: October 2005

Among many room acoustical parameters except RT60, clarity and room response (the C-RR chart) have been adopted and measured in several halls. But C is relatively dependent on reverberation time and the evaluation by the C-RR chart is not always satisfactory enough to explain the acoustical features of concert halls. As a new method for evaluating concert hall acoustics, “RECC” (reflection energy cumulative curve excepting direct sound) has been studied. The results are as follows. (1) RECC, up to about 80 ms, shows various shapes depending on the hall and measuring points. But, after about 80 ms, it shows a smoothly increasing convex shape for all halls. (2) Total reflection energy is almost determined by the ARE80 (accumulated reflection energy up to 80 ms). (3) The values of the ARE80 level are comparatively large at seats where one can feel intimacy in the hall. Room acoustics of concert halls is evidently very dependent on its room shape. The ARE80 is considerably influenced by room shape and independent of RT60. Optimum level, density, direction, frequency characteristics, etc., of the reflections up to 80 ms are problems for further study.
Study on a “spatially selective” microphone system
Authors: Date Hikaru
Published: October 2005

Spatial selectivity of a microphone system is defined as a function that rejects all sound signals coming from the region specified as external by the system. It can be effectively realized by signal processing of the multiport output signals of a receiving system that comprises several microphone pairs located on a virtual spherical boundary surface and one center microphone. Because the signal processing part of the system consists of adders, constant multipliers, differentiators, and a single delay, it is essentially linear and can be realized in either analog or digital form according to requirements of precision. The practical advantage of the spatially selective microphone system is that it has the capability of suppressing howling in public address systems and conference telephony, of improving the signal-to-noise ratio of microphone output to automatic speech recognition machines, and of improving the separability of the signal picked up from a particular musical instrument from those of other instruments played simultaneously in the same studio. The design method is presented after the simulation results for the system.
Estimation of position and waveform of individual sound sources using many sensors
Authors: Kido Ken'iti , Nagata Yoshifumi , Abe Masato
Published: October 2005

This paper describes a method for estimating the position and waveform of individual sound sources using many sensors when there are many sound sources. First, an imaginary sound source is introduced, and the distance between the imaginary sound source and every sensor is calculated. Next, the output of every sensor is compensated both in time and in magnitude so that the compensated output is the same for the sound radiated from the imaginary sound source. Then, the compensated outputs are averaged, and the power of the averaged waveform is calculated. When the position of the imaginary sound source just coincides with one of the positions of the real sound sources, the power has a large value. Otherwise, it has a small value. The position of the imaginary sound source is scanned within a specified space. The position, at which the power yields a peak, is the estimated location of a real sound source, and the averaged waveform is the estimated waveform of the sound source. Then, the perturbation of the estimated locations and waveforms is carried out. The final estimated values are those obtained when the error function reaches its smallest value. Computer simulations and experiments were carried out using two sound sources and 16 sensors, and good results were obtained.
Dolphin sonar target detection in noise
Authors: Au Whitlow W. L.
Published: October 2005

Several different types of experiments have been conducted to determine the sonar detection capability of the Atlantic bottlenose dolphin. In one experiment, a dolphin's performance in detecting the presence or absence of a 7.62-cm-diam (− 28.3-dB target strength), water-filled, stainless steel sphere as a function of range was determined. The threshold range in Kaneohe Bay, Oahu, Hawaii was approximately 113 m. In another experiment, the same sphere was fixed at a range of 20 m, and the detection capability of two dolphins was determined as a function of the level of artificial masking noise. In a third experiment, a simulated phantom electronic target was used to simulate a target at 20 m, and the animal's detection performance as a function of noise was determined. Using the transient form of the sonar equation, it was found that the dolphins' performances as a function of the echo signal-to-noise ratio were very similar. Dolphins process echoes like an energy detector with an integration time of approximately 264 μs. The results from the various experiments conformed to the Urkowitz [Proc. IEEE 55, 523–531 (1967)] energy detector model. The dolphin detection sensitivity was also found to be approximately 6 to 8 dB lower than that of an ideal or matched-filter receiver.
Detection threshold calculations for Weibull interference
Authors: Young Joel W. , Cable Peter
Published: October 2005

The Weibull probability density function (pdf) has been used to represent the noise-only output of an envelope detected matched filter when the input noise contains non-Gaussian, high amplitude components. This situation may occur in active sonar when the total interference is dominated by reverberation. The familiar Rayleigh pdf associated with Gaussian input noise is a member of the Weibull family with a shape factor of 2. If the threshold for detection is based on an assumed Rayleigh density and the noise changes to Weibull with a shape factor less than 2, the false alarm rate will increase dramatically even if the average noise power is unchanged. Therefore, in order to maintain a constant false alarm rate (CFAR), the threshold must be raised. This will result in an increase in the signal to noise ratio required to achieve a specific probability of detection (Pd). It is necessary to determine the signal plus noise pdf to quantitatively evaluate this effect on detectability. An integral formulation of this function has been derived by Eckstrom, and it has been numerically integrated for a few cases by Nilsson and Glisson. The present authors have developed a much simpler approximate procedure for computing receiver operating characteristic (ROC) curves for a linear detector in Weibull noise that does not require numerical integration. This method is based on the premise that the signal plus noise pdf is not strongly affected by the Weibull shape parameter, at least over the 10% < Pd < 90% range. In this paper, the details of this approximate technique are presented, and some comparisons with more exact calculations are provided.
The use of the trans-spectral coherence technique, to separate a signal from noise
Authors: Vaidya P. G. , Anderson Mike
Published: October 2005

A new concept of the trans-spectral coherence has been developed. Consider a narrow-band signal A, with a center frequency f1, and a signal B, with a center frequency f2. Now, if a nonlinear filter could be found that converts the signal A into a signal C at f2, the trans-spectral coherence between A and B is the ordinary coherence between C and B. If the two signals A and B are derived from a common source through a nonlinear interaction, this kind of coherence is nonzero. Most speech signals, signals from musical instruments, and some signals from industrial noise sources, can be expected to possess nonzero trans-spectral coherence. In this paper, it has been conjectured that, if sound from a nonlinear source is mixed with random noise, the trans-spectral technique can be used to separate the signal from noise.
Acoustics of Eastern and Western bells, old and new
Authors: Rossing Thomas D.
Published: October 2005

Bells have belonged to nearly every culture in history. The oldest surviving bells from the Near East and China are thought to be over 3000 years old. Many ancient Chinese bells, by virtue of their oval shape, have two distinctly different strike notes, separated in pitch by about a major or minor third. Bells developed as Western musical instruments in the 15th and 16th centuries when bell founders learned how to tune their partials harmonically. The acoustical behavior of Western church bells, carillon bells, and handbells are compared to those of ancient Oriental bells. Recently carillon bells with a major third replacing the traditional minor third partial have been developed [A. Lehr, Music Percept. 4, 267 (1987)].
Pitch of a set of ancient Chinese bells: Piao-shi Bian Zhong
Authors: Takahashi Junji
Published: October 2005

The precise pitch and size are measured for a set of 12 ancient Chinese bells, named “Piao-shi Bian Zhong” and made probably in 404 B.C., which have been preserved at the Sen-Oku Hakko Kan Museum in Kyoto. The complete set consists of 14 bells, including the 12 bells measured in this paper and two other bells preserved at the Royal Ontario Museum in Canada. Zhong, a kind of Chinese bell with an almond-like cross section, emits two tones according to the position struck. The bells are all constructed in a similar shape except for the thickness. The frequencies of the 12 lower tones were found to be inversely proportional to the squared distance between the two tails of the lateral spines, called “xian,” of the bells. This was the same with the upper tones. This relation suggests that ancient craftsmen should probably have taken advantage of such an empirical formula to determine the size of each bell to produce two specified pitches of a musical scale.
Velocity of sound in biological materials
Authors: Carstensen Edwin L.
Published: October 2005

The phase velocity of longitudinal waves in biological fluids and tissues with high water content exceed the velocity of sound in water roughly in proportion to the solid content of the material. In general, fatty tissues have lower and bone higher sound speeds than water. With the exception of water, the sound speeds of most of the components of tissues have negative temperature coefficients. Dispersion in the velocity of sound is related to the distribution of relaxing elements in the material and, hence, to the absorption. Since the velocity is a function of the pressure, sound propagation is a nonlinear process. Biological materials, in general, are more nonlinear than water. High-amplitude sound waves become distorted as they propagate and may form shocks that have been associated with biological effects such as excess absorption and fragmentation of kidney and gall stones.
An ultrasonic scattering characterization of tissue
Authors: Waag Robert C.
Published: October 2005

Average differential scattering cross section is obtained from measurements by a normalization that accounts for temporal waveforms and spatial apertures. A model is employed to express the average differential scattering cross section as a linear combination of the power spectrum of the medium compressibility variations, the power spectrum of density variations, and the cross power spectrum of compressibility and density variations. Since the scale factors depend on scattering angle and temporal frequency, known values of the average differential scattering section at different scattering angles and temporal frequencies corresponding to the same spatial frequency, which is considered to be the argument of the average differential scattering cross section and the power spectra, are used to calculate each of the three power spectra. Results are shown for model random media and calf liver. The results indicate that determinations of average differential scattering cross sections and the power spectra of scattering medium variations can be made under practical conditions and also imply that density variations contribute significantly to scattering by calf liver. [Work supported by NSF, NIH, NATO, and Industrial Associates.]
Application of the acoustic nonlinearity parameter to tissue composition prediction
Authors: Apfel Robert E. , Everbach E. Carr
Published: October 2005

The objective of this research is the prediction of the composition of tissues (percent water, protein, and fat) from the results of ultrasonic measurements (sound speed and nonlinear parameter) and reasonable mixture laws (for the compressibility, density, and nonlinear parameter). These laws are a redundant set allowing for the use of an optimization routine to predict the volume fractions of a three-component mixture. When applied to the data of others for fluid mixtures and tissues, reasonable agreement with known compositions has been achieved. This methodology has now been applied to tissue phantoms composed of known amounts of gelatin, water, and fat. One crucial requirement for the theory's application is a convenient apparatus that gives accurate, repeatable data for the nonlinear parameter, as has not always been possible with finite amplitude and traditional thermodynamic methods. In these experimental procedures, a commercial instrument is adapted employing a phase-locked loop to a modified thermodynamic method first used by Emery et al. [J. Phys. (Paris) 11, 231–234 (1979)]. This apparatus permits rapid and precise measurements. Results (and limitations) of recent measurements and comparisons with theoretical predictions will be presented. [Work supported by the U. S. National Institutes of Health through Grant R01GM30419.]
Measurement of the nonlinearity parameter B/A of tissue by the time difference method
Authors: Endoh Nobuyuki , Asahina Tetsuya , Igawa Masayuki , Hamamoto Masahide
Published: October 2005

A method of measuring the nonlinearity parameter B/A of soft tissue in vitro was developed for thermodynamic phenomenon. Using a gated ultrasonic rf pulse, the propagation time through the tissue was measured at two different thickness values. The B/A of the tissue could be estimated by the thickness difference, and the propagation time difference between atmosphere and a pressurized (3 atms) condition. A fundamental experiment was carried out to confirm the validity of the method. The phase (time) difference was measured by the interference method with a two-channel synthesizer. The B/A of water was proportional to its temperature from 10 to 40 °C and was 5.1 at 20 °C. A B/A of 9.2 for a chicken liver in vitro was also obtained.
Ultrasonic spectroscopy in bovine serum albumin solutions
Authors: Choi Pak-Kon , Bae Jung-Rim , Takagi Kenshiro
Published: October 2005

Ultrasonic absorption was measured in bovine serum albumin aqueous solutions (50 g/l) in the frequency range from 100 kHz to 1.6 GHz at pH of neutral, acid, and alkaline regions. Three experimental techniques were used to cover the wide frequency range: plano-concave resonator, conventional Bragg reflection, and high-resolution Bragg reflection (HRB) methods. At pH 7, the absorption spectrum fitted the relaxation curve well using the distribution function of a mirror image of Davidson-Cole. This suggests that the dominant relaxation mechanism is due to hydration. In the acid region, the relaxation peak observed at 2 MHz was attributed to the proton transfer of the carboxyl group on the glutamic and aspartic acid residues. Another relaxation peak at 200 kHz was associated with the conformational change in the albumin molecules. The two relaxation peaks in the alkaline region were ascribed to the proton transfers on the tylosyl and lysyl residues. A relaxation due to the conformational change was also indicated.
New technique for spot-poling piezopolymer membrane hydrophones
Authors: DeReggi Aime' S. , Harris Gerard R.
Published: October 2005

One characteristic that poses a lower limit to the size of existing spot-poled membrane hydrophones is that the active region is always larger than the electroded region. The difference is due in part to the fringe poling fields that surround the electrodes and proximal portions of the leads. This effect makes it difficult to construct hydrophones having dimensions less than ≃0.5 mm. A potential way around this problem is to reverse the standard poling and plating process. That is, the film is first poled and then the metal electrodes and leads are added. This approach has been applied using small (< 0.1-mm tip diameter) needles as poling electrodes. Piezopolymer film was mounted on support rings and a central spot was poled at room temperature. Electrodes and leads were vacuum deposited, and sensitivity and directional response measurements were made. In this presentation, details of the poling process will be described and measurement results will be discussed.
Cascaded likelihood vector compression of linear predictive models
Authors: Shoham Yair
Published: October 2005

Compression of linear predictive spectra has applications in acoustics research, particularly, in communication of acoustical information. Cascaded likelihood vector compression (CLVC) is proposed for coding the spectral parameters of linear predictive models at rates of 20 to 26 bits per model. The system is based on representing the LP all-pole model as a cascade of two lower-order models. The partitioning of the LP polynomial is done in the root domain by clustering the roots into two distinct groups. The compressor uses two codebooks to quantize each of the lower-order models. However, the quantization process is done so as to jointly optimize the overall performance. The likelihood ratio distortion measure is used as a performance criterion. Splitting the LP model into two subsystems dramatically reduces the complexity while the efficiency of vector compression is essentially preserved. Experimental results show an average performance of 1.59, 1.41, 1.24, and 1.10 dB of log-spectral distortion at the rates of 20, 22, 24, and 26 bits per model, respectively. These results indicate high-quality model compression particularly for communication applications where the number of bits assigned to the LP model is very limited.
Temporally overlapping segment quantization for speech coding
Authors: Honda Masaaki
Published: October 2005

A method for the efficient segment quantization of LPC spectral parameters is presented for very low-bit-rate speech coding. Recent methods of segment quantization for speech coding have used a concatenation segment model, in which spectral parameters are represented as a concatenation of variable-length code spectral segments. The present method uses a temporally overlapping segment model to capture more efficiently the variance in the spectral parameters due to coarticulation effect in speech. Spectral parameters are represented as a sequence of temporally overlapping segments, in which each code segment is decomposed into fixed-length temporal patterns and their weighted patterns. A dynamic programming procedure is used to determine the optimum code segments and the segment positions of the overlapping model so as to minimize the spectral distance between the input and the model spectral parameters in the analysis window. An iterative algorithm for code segment design in this model is also presented. Comparison of the overlapping model with the concatenation model for speech coding [Shiraki and Honda, to appear in IEEE Trans. ASSP] is described in terms of spectral distortion and computational complexity.
Dynamic perception of tones simultaneously modulated in both frequency and amplitude
Authors: Tsumura Takashi
Published: October 2005

Auditory processing in the dynamic perception of tones with simultaneous frequency modulation (FM) and amplitude modulation (AM) was investigated. Thresholds for the detection of target modulation (FM or AM) were measured with the respective nontarget modulation (AM or FM) kept well above threshold. The carrier frequencies of tones below 1 kHz were used. For FM and AM having the same sinusoidal modulation frequencies below 20 Hz, the thresholds of the target modulation changed as a function of the phase difference between the target and nontarget modulations. When the frequencies of the FM and AM were different from each other, the thresholds of the target modulation were nearly the same as those obtained for a tone with the target modulation alone. When the frequency and amplitude continuously increased or decreased in a short tonal duration, there was little difference in the threshold of the target modulation between the cases for increase and decrease in nontarget modulation. These results suggest the existence of channels selectively tuned to the periodicity of FM and AM in the auditory pathways. [Work supported by a Grant in Aid for Scientific Research (No. 62510059) from the Ministry of Education.]
Effects of manipulating the spectral shape of reproducible noise samples on detection judgments
Authors: Gilkey R. H. , Simpson B. D. , Hammoud A. M.
Published: October 2005

The single-interval/yes-no detection judgments of subjects to a subset of ten noise-alone and ten signal-plus-noise samples from the 300 waveforms examined by Gilkey and Meyer [J. Acoust. Soc. Am. Suppl. 1 82, S92 (1987)] were investigated further by manipulating the spectral shape of the samples. The spectral shape was “modulated” by either incrementing or decrementing the level in a 47-Hz band centered around one of seven center frequencies ranging from an octave below to an octave above the signal frequency. The waveform on each trial within a block was randomly sampled without replacement from a large set of modulated and unmodulated waveforms. Thus the experience of the subject was assumed to be comparable to that with truly random noise. Detection judgments are affected by changes in the shape of the spectrum that are well outside the critical band centered at the signal frequency. The results are discussed in relation to models of auditory masking. [Work supported by NSF and AFOSR.]
Critical band distortion analysis
Authors: Cudahy Edward , Levitt Harry , Link H. Cynthia
Published: October 2005

Current measurement techniques typically focus on distortion produced by clipping or intermodulation of pure tones. There is little information on the relation between these measurements and the perception of distortion. The present study had two purposes. First, to develop a general method for measuring distortion. Second, to determine the association between this physical measurement and the perception of distortion for normal hearing listeners. Results suggest that a critical band analysis of distortion can be useful for predicting thresholds for distortion, at least for simple stimuli. [Research supported by NIDRR.]
A minimum discriminable bandwidth test for critical bandwidth estimation
Authors: Celmer Robert D. , Bienvenue Gordon R.
Published: October 2005

A test procedure has been developed through the use of a modified method of limits technique for the direct measurement of critical bandwidth. A tonal complex with an initially subcritical bandwidth is presented to a listener. The test signal then discretely widens in bandwidth with time; each bandwidth has a duration of approximately 800 ms. Subjects indicate at which bandwidth they perceive a change in the sound. The minimum bandwidth at which a listener can discriminate a difference in the tonal complex is taken to be that subject's critical bandwidth (cf. Scharf, 1970). The tonal complexes were generated using a digital signal processing algorithm, with the value of the initial bandwidth randomized to reduce learning effects. Results of the test performed on normal and sensorineural hearing impaired listeners are presented. Implications for evaluating sensorineural hearing impairment are discussed.
Frequency effects for multicomponent maskers with high spectral uncertainty
Authors: Nef Donna L. , Callaghan Brian P.
Published: October 2005

Simultaneous maskers composed of a few sinusoidal components can produce considerable masking if the component frequencies are unpredictable. This experiment examined how components in particular frequency regions influenced performance. Thresholds were measured for a 1000-Hz signal presented simultaneously with a masker composed of 2, 4, 6, 10, 50, or 100 components. Components were drawn from a 300–3000-Hz range, excluding the critical band around 1000 Hz. When sufficient masking was produced by uncertainty, limiting components to the high-or low-frequency side of the signal, or widening the notch around the signal, had little effect. Decreasing uncertainty reduced thresholds for maskers limited to high-frequency components, but not for those limited to low frequencies. Paradoxically, forcing components into narrower bandwidths around the signal could reduce masking by about 5 dB. Large individual differences on this task were not well predicted by measures of auditory filter shape with notched-noise maskers. [Work supported by AFOSR.]
Detection of a silent gap in single components of a multitonal complex
Authors: Green David M. , Forrest T. O.
Published: October 2005

Complex multitonal waveforms were generated using 21 equal-amplitude sinusoids. The tones range in frequency from 200–5000 Hz, and the frequency ratio of the successive components was constant (equal-logarithmic intervals). The listener's task was to detect the silent gap in a single component of the complex. The gap occurred either in the temporal center of a 500–ms burst or at a randomly chosen temporal location. For both conditions, gap threshold decreases with component frequency from about 20–50 ms at the lower frequencies to about 2–5 ms at the highest frequency. It is somewhat easier to hear a gap if it is present in more than a single component. Synchronized gaps are slightly easier to hear than asynchronous gaps, even if the asynchronous gaps occur at predictable times. [Research supported by AFOSR and NSF.]
Stream segregation—source grouping versus peripheral channeling
Authors: Hartmann William Morris , Johnson Douglas
Published: October 2005

The prevailing view is that auditory stream segregation should be understood as fundamentally teleological. Two tones are segregated into the same stream if, according to Gestalt grouping rules, they probably derive from the same acoustical source. A problem with this point of view is that there are many different stimulus features that might serve equally well to identify and group sources, but not all of these features are equally effective in promoting stream segregation. Using an interleaved melody identification paradigm, the ability of listeners to segregate streams based upon ten different kinds of stimulus features was measured. The results show that tones that excite different peripheral channels of the auditory system, based upon differences in octave range, ear presentation, or spectral envelope, can be separately streamed with very high efficiency. Tones that excite primarily the same peripheral channels, but that may be identified as different sources based upon level, duration, amplitude envelope, or interaural phase difference, can be separately streamed, but with markedly less efficiency. Tone sequences that employ irregular rhythm or added noise cue bands do not promote stream segregation. [Work supported by the NIH.]
Resonance frequencies of the human skull in vivo
Authors: Brandt Anders , Carlsson Peder , Hakansson Bo , Tjellström Anders
Published: October 2005

In an investigation on patients with skin penetrating titanium implants, the free-resonance frequencies of the human skull have been determined in seven subjects in vivo. Frequencies ranging from 100 Hz–7.5 kHz were investigated and a total of 13 to 17 resonances were identified for each subject. From the point and transfer characteristics, the lowest free-skull resonance was estimated to 990 Hz (mean) with a standard deviation of 120 Hz. Relative damping factors were found to be between 2%–15%. Intersubject variation was large, probably due to individual variations in skull geometry and mechanical properties. It is clear from both the point and transfer characteristics that the free-skull resonances have little influence on the vibrational levels of the skull, and thus should have little influence on bone conducted sound. In the transfer characteristics between the mastoids, however, narrow dips (antiresonances) were found, more than 20 dB below the average level. These antiresonances may affect bone conducted sound transmission to the contralateral ear. [Work supported by P&B Sound System Inc., Göteborg, Sweden.]
Improved articulatory models
Authors: Maeda Shinji
Published: October 2005

After a brief historical review of articulatory models, a factor analysis of the lateral shapes of the vocal tract is described. In the analysis, the profiles are specified by the sum of a small number of linear components. Studies at CNET indicate that, by using an arbitrary factor analysis in connection with the principal component analysis, it is possible to derive a linear model in which each component can be interpreted in articulatory terms, such as jaw position, tongue-body position, etc. The statistical model, therefore, can be regarded as an articulatory model. The analysis of radiocinematographic data exhibits that the same vowel can be produced differently depending on speakers and phonetic contents, suggesting a compensatory articulation. Such a model is also useful in investigating the articulatory-acoustic relationships. For example, F1–F2 trajectories are calculated by varying a specific parameter value, while the remaining parameter values are fixed appropriate for a vowel. The results show that the different jaw positions can be compensated only by the tongue-body position for front vowels, and only by the lip aperture for back vowels, to obtain a similar F1–F2 pattern. Finally, limitations of current articulatory models are discussed.
Acoustic radiation from line excited plates—Asymptotic series
Authors: Hsu Hong-Yuan , Hayek Sabih I.
Published: October 2005

The theoretical analysis for predicting the acoustic radiation from an infinite fluid-loaded, elastic plate excited by a line force is presented. The acoustic pressure radiated by this coupled plate-fluid system is given in an integral representation. A uniformly asymptotic series solution, obtained by the use of the modified saddle point and subtraction of poles method, is valid in the farfield and the nearfield, and is especially accurate at the coincidence angle where the first-order approximation by the saddle point method overpredicts the acoustic pressure. The uniformly asymptotic series solution also predicts a decaying but nonvanishing surface acoustic pressure while the approximate solution from the saddle point method vanishes at the grazing angle. The solution was also obtained by numerical integration along the steepest descent path. Results obtained from the numerical integration are compared with the uniformly asymptotic series solution and excellent agreement has been observed.
Analysis of vibration transmission through periodic structures using a power flow approach
Authors: Cuschieri J. M.
Published: October 2005

The analysis of the parameters controlling the transmission of vibrational power through coupled structures can be performed using a structural power flow approach. In this paper, a periodic structure is considered with equal length spans between the stiffeners or the supports. With the power flow technique, the global structure is divided into subelements for which structural mobility functions can be defined. Also, structural mobility elements can be introduced at the interfaces between the subelements to represent different types of supports or stiffeners or some other interface characteristic. Thus, using this approach, the influence of such parameters as translational and rotational inertia, stiffness, and damping of the supports or stiffeners, and the stiffness, damping, and other structural characteristics of the separate spans, can be determined using this technique. This approach is applied here to a thin periodic beam with different types of stiffeners and supports, and some of the analysis results are presented. [Work supported by NASA.]
Gaussian beam propagation on cylindrical thin shells
Authors: Norris A. N.
Published: October 2005

The flow of acoustic energy on cylindrical thin shells can be understood by looking at how disturbances in the form of Gaussian beams propagate. Gaussian beams are also useful tools for considering radiation from a given source, if the source can be easily represented as the sum of a few distinct beams. The recent work of Pierce [A. D. Pierce, Proceedings of IUTAM Symposium on Elastic Waves, Ireland, 1988 (to be published)] showed that the equations are very similar to those for an unbounded two-dimensional anisotropic and dispersive medium. Using this analogy and a recently developed formalism for Gaussian wave packets in anisotropic solids [A. N. Norris, Wave Motion 9, 509–532 (1987)], the equation governing the change in wave front curvature is derived. Gaussian beams are obtained by taking the curvature to be complex valued initially. The change in curvature then determines the narrowing or broadening of the beam. The anisotropy causes the broadening to occur in a direction that is not perpendicular to the direction of propagation. The strong dispersion of the flexural wave introduces effects not previously seen in Gaussian beams. The influence of fluid loading will also be mentioned.
Application of the Wigner distribution to speech analysis
Authors: Howitt Andrew Wilson
Published: October 2005

The Wigner distribution, a time-frequency signal representation similar to the spectrogram, has recently been applied to speech analysis. It offers higher resolution than the spectrogram, but introduces artifacts that do not correspond to the components of the signal. These artifacts seem to be characterized by negative values of the Wigner distribution. If so, skilled experimental subjects should be able to visually identify and disregard the artifacts. Subjects examined spectrograms and Wigner distributions of a set of synthetic speech utterances, and performed a simple formant-tracking task (initial slope estimation) on them. The subjects performed this task marginally better using the Wigner distribution than the spectrogram. This performance advantage of the Wigner distribution held over the range of formant trajectories that the synthesizer could reliably produce. Performance depended critically on the subjects' understanding of artifacts. A theoretical prediction of a limit on formant slope, previously untested, was consistent with both spectrograms and Wigner distributions of natural speech. The overall conclusion is that the Wigner distribution is a viable alternative to the spectrogram for analysis of rapid spectrum changes. However, the burden of its greater complexity seems to outweigh its potential performance advantages.
Shallow water acoustics on the Canadian East Coast Continental Shelf
Authors: Staal Philip R. , Hughes Steven J. , Ellis Dale D. , Chapman David M. F.
Published: October 2005

The Canadian East Coast Continental Shelf is a large area (roughly 600 000 km2) of complex bottom types geologically related to the glacial cover and low water level of the recent Ice Age. The bottom types include thickly sedimented sand banks under less than 100 m of water, thickly sedimented silt/clay basins under more than 200 m of water, thinly covered or bare glacial till, and outcropped rock. This paper describes DREA measurements of underwater acoustic propagation and ambient noise in the 1- to 1000-Hz frequency band, and attempts to explain these measurements. Acoustic propagation over thickly sedimented seabeds can be explained by models that assume “fluid” sediment layers (i.e., no shear waves). However, over thinly sedimented rocky seabeds the acoustic models need to include shear waves to explain the measured acoustic propagation, in particular the high losses between 10 and 100 Hz. Ambient noise is affected by the seabed, mainly by the difference in acoustic propagation over unlike seabed types, from distant sources.
Developments in shallow water seismoacoustics
Authors: Muir T. G. , Akal T.
Published: October 2005

Since the 1970s, when low-frequency acoustics in shallow water became a priority topic, developments have progressed from propagation loss in the frequency domain, to waveguide phenomena dominated by bottom interaction, to ocean bottom seismology and studies of interface waves involving sediment rigidity. Emphasis has now been placed on the entire propagation medium, including the water column, the sediment interface, the volume of unconsolidated sediments, and the underlying bedrock strata; as well as on the entire spectrum of wave types, which in reality are part and parcel of a single unified process. Sophisticated research tools have been developed for these studies, including remotely operated, three-axis seismometers, machines to deploy them at depth within the sediments, and various computer models, such as the SAFARI code that can address the creation and propagation of seismoacoustic waves. Results obtained with these tools in research done in the shallow seas surrounding Italy, Norway, and Spain are presented.
Pulse response measurements over three 45-km propagation paths in the Florida Straits
Authors: DeFerrari Harry A. , Monjo Charles
Published: October 2005

Three point reciprocal transmission experiments, undertaken in the Florida Straits, have yielded a 35-day time history of shallow water pulse responses for 45-km ranges in a depth of approximately 550 m. The transmitted pulses consist of 4 cycles of a 460-Hz carrier. Several distinct and separable arrivals persist throughout the 35-day period for all three ranges. The sound-speed profile has a 90-m surface duct above a negative gradient of approximately 0.15 (1/s). Both the sound speed and bathymetry have complicated range dependence and the medium is highly variable in time owing to meanders of the core of the Florida Current. Yet, the arrival patterns are remarkably stable. The observed time spread of all arrivals is nearly a full second. Travel times of the earliest arrivals are consistent with surface duct transmission, whereas later arrivals are consistent with ray model predictions for refracted bottom reflected (RBR) rays. The RBR arrivals are spread over approximately 0.3 s. Ray models, normal mode, and PE models are used to predict and identify intermediate arrivals. [Work supported by ONR.]
Shallow water propagation under the Arctic ice canopy
Authors: Wegmann David , Medwin Herman
Published: October 2005

Propagation in a shallow water waveguide, covered by a layer of “ice,” is being studied in a 10-m-long laboratory waveguide that models both the scale and the physical properties of the Arctic. Both smooth ice and rough ice effects are investigated. Comparisons are made with predictions of the Naval Research Laboratory computer model, KRAKEN. Although the ice canopy thickness is only a small fraction of a wavelength, the sound field is distinguishable from the usual Pekeris waveguide behavior by evidence of the presence of solid-borne modes in the ice. The observed effects are related to the stimulation of ice head waves, which has been reported previously for deep water regions [J. Acoust. Soc. Am. 83, 1794 (1988) and J. Acoust. Soc. Am. Suppl. 1 82, S31 (1987)]. [Work supported by the office of Naval Research.]
Factors affecting the integration of auditory and visual information in speech: The effect of vowel environment
Authors: Green Kerry P. , Kuhl Patricia K. , Meltzoff Andrew N.
Published: October 2005

In the McGurk effect, observers typically report the illusory syllable /da/ when they hear the auditory syllable /ba/ presented in synchrony with a video display of a talker saying /ga/. While the effect itself has been well established, there is still little research on the conditions under which the effect occurs. In the experiment reported here, the number of illusory /d/ responses to the auditory /b/-visual /g/ combination is examined in three vowel environments: /a/, /i/, and /u/. The results of this study indicate that the magnitude of the illusion is not the same across different vowel environments. It appears to be strongest for the /i/ vowel, moderate for /a/, and almost nonexistent for /u/. The results thus show that vowel environment is an important factor in determining the magnitude of the McGurk effect, which needs to be considered in accounts of auditory-visual integration during speech perception. [Work supported by NIH.]
Can speech envelopes' modulation spectra be used to support segmental decisions?
Authors: McCormick M. M. , Porter R. J. , Seitz F. , Watson I. M. C.
Published: October 2005

The auditory system senses the rate, the magnitude, and the phase of temporal modulations of sounds' spectra and amplitudes. Such “modulation sensations” have been proposed as a possible basis for perception of speech segments and their features [see R. J. Porter, in Language Perception and Production, edited by Allport et al. (Academic, London, 1987), Chap. 5]. This preliminary study extends a Russian attempt to analog-model speech segmentation decisions using speech-envelope modulation magnitude [Malinnikova et al., Fiziol. Zh. SSSR im. I. M. Sechenova 66, 139–145 (1980)]. In the current study, comparisons are made between human segmentation decisions and a multistage, computer-based, spectral analysis of the modulations of the amplitude envelopes within different speech spectral regions. [Research supported in part by Alvey Grant No. MMI092.]
Naturalness and intelligibility of amplitude modulated time-varying sinusoidal speech
Authors: Carrell Thomas D.
Published: October 2005

Synthetic speech has been used for decades to test theories regarding human speech perception. Typically, the speech has been constructed to sound as natural as possible given the constraints of the experiment. An alternative strategy has been to examine the perception of intentionally impoverished stimuli such as time-varying sinusoid (TVS) replicas of speech [Remez et al., Science 212, 947-950 (1981)]. The TVS signals consist of three tones whose frequencies mimic the formant center frequencies of a natural sentence. They exhibit few of the acoustic properties of natural speech. It has been demonstrated that TVS signals sound extremely unnatural although they are surprisingly intelligible. The goal of the present experiment was to determine some of the general acoustic characteristics of signals that are important for speech perception. This was accomplished by examining the perceptual consequences of adding simple temporal and spectral information to TVS sentences. The TVS signals were amplitude modulated at 100 Hz in order to give them more speechlike acoustic characteristics without giving them fundamental frequencies or harmonic structures. The modulation greatly improved the phonetic intelligibility of the acoustically sparse TVS signal. The modulated signal was also significantly more natural sounding to listeners than the unmodulated TVS signal. Performing this operation on natural speech, however, caused a decrement in intelligibility.
Acoustic correlates of intervocalic stop confusions
Authors: Bunnell H. Timothy , Martin James G.
Published: October 2005

As part of the study of signal processing algorithms that automatically enhance speech intelligibility, the acoustic factors underlying place confusions in intervocalic stop consonants have been examined. The speech database in these studies consists of nine repetitions of each of eighteen nonsense utterance frames from five talkers. The utterance frames were of two forms: “say a CVwuh” and “say a wVCuh” with C = {/b/,/d/,/g/} and V = {/i/,/a/,/u/}. In producing the repetitions of each frame, talkers were under instructions to vary their speaking style from enunciating very clearly to speaking in a casual, relaxed fashion. Confusion matrices and percentage correct consonant identification have been obtained for these utterances from both normal-hearing listeners (using broadband noise as a masker) and from hearing-impaired listeners. Acoustic measures related to place of articulation cues and general spectral/temporal contrast have been obtained from each utterance. The relations between these acoustic measures and consonant confusability will be described and implications for algorithms intended to enhance speech intelligibility will be discussed. [Work supported by NINCDS.]
Unsupervised speaker adaptation of spectra based on a minimum fuzzy vector quantization error criterion
Authors: Matsumoto Hiroshi , Yamashita Yasuki
Published: October 2005

Reference spectral code vectors are adapted to an input speaker by adding a distance-dependent linear combination (an interpolated vector) of speaker difference vectors (unknown adaptation vectors) at given typical points in the reference spectral space. The unknown adaptation vectors are estimated so as to minimize the total fuzzy vector quantization errors for training vectors using an adapted reference codebook. This minimization is iteratively resolved under a restriction on the sum of the norms over all the unknown adaptation vectors. The above adaptation process is applied in a stepwise manner by increasing the number of typical spectral points. The fuzziness, interpolation parameters and the limits of the norm were examined through 28 word recognition tests for 4 male and 4 female speakers, using reference patterns from a male speaker. Under the best conditions, with 16 adaptation vectors, this method achieved the same recognition accuracy as a supervised speaker adaptation for male speakers with training samples as short as 3 s.