bluuebunny/crossref_metadata_2025 huggingface dataset:
The DARPA spoken language systems program: Past, present, and future
Authors: Sears J. Allen
Published: October 2005

The long-range goal of the DARPA strategic computing speech program is to develop speech recognition and understanding technology sufficiently robust to support a wide range of DoD applications. The application environments may require systems capable of handling up to 10 000 words, with habitable grammatical constraints measurable by a perplexity of 100 to 200. Such a system should have a word accuracy of better than 95%, and should operate in real time. Specific short-term goals for the first phase include the development of a 1000-word continuous speech recognition system for use in acoustically benign environments by the end of 1988. Research activities supported by the DARPA strategic computing speech program include the development of speech recognition systems, related component technology, and supporting infrastructure. For the next 4 years, the program will build on accomplishments of phase one of the speech recognition and natural language programs in order to achieve spoken language understanding. Specifically, the program will move from speech transcription and textual language processing towards interactive problem solving using voice input. In this talk, the objectives and the structure of the DARPA-sponsored program will be described in greater detail.
Program of the European communities for the European speech technology
Authors: Roukens J.
Published: October 2005

Considering the strategic importance and potential size of the speech technology market, the European Commission helps the technological advance of laboratories and industries in this area by financing a series of projects within the ESPRIT program. These projects bring the best European laboratories and advanced industries together for cooperation in this area, and are oriented in four directions: (1) acquisition of basic knowledge about languages and human factors in human-machine speech dialogue; (2) advancement of existing techniques, e.g., speaker-dependent and independent recognition, coding, synthesis, speaker identification, etc.; (3) development and integration of knowledge-based techniques; and (4) development of speech assessment technology. These projects put emphasis on the multilinguism of recognition and synthesis. Multilinguism is considered to be prerequisite for the success of products on the international market. Some projects have just started; others are almost completed. For example, the project “Linguistic Analysis of European Languages” will reach completion at the end of 1988 and has built a knowledge base containing the necessary rules to solve ambiguities in phoneme-to-grapheme and grapheme-to-phoneme conversion for eight European languages. In addition, and by way of example, synthesizers and recognizers have been built for French and Italian, and a speaker verifier. This work will continue to be strongly encouraged in the future.
Sensitivity of matched-field processing to sound-speed profile mismatch for vertical arrays in a deep water Pacific environment
Authors: Tolstoy A.
Published: October 2005

In this paper, the sensitivity of matched-field processing (MFP) to sound-speed profile mismatch (based upon archival profiles resulting in various degrees of mismatch) will be examined with emphasis on sources within 50 km. A 10-Hz source will be considered whose field is generated by a normal mode model where only the water-borne energy is used, thereby eliminating issues relating to the estimation of bottom parameters. Array parameters, i.e., number of phones and array depth, will be examined for their effects on range and depth localization for various degrees of mismatch. In particular, it will be indicated where an array is most and least sensitive to sound-speed mismatch as a function of depth from the surface and range from the source. Also, the issue of selecting the range increment for the minimum variance (Capon) processor will be briefly addressed and it will be pointed out how too crude an increment can lead to erroneous results. Finally, an example will be presented of coherent broadband (BB) MFP illustrating improved robustness and sidelobe suppression relative to both single frequency and incoherent BB results.
Modal amplitude filtering of vertical array data in a shallow water environment
Authors: Frichter George M.
Published: October 2005

Current research at NORDA into new matched-field processing techniques based on nonlinear matching of complex modal amplitudes filtered from noisy vertical array data shows that very low SNR sources may be detected and localized when spatially correlated (modal) noise dominates. All mode-matching algorithms require knowledge of the local ocean environment to calculate depth eigenfunctions for the waveguide. This modal structure is then incorporated into a mode filter that deduces complex modal amplitudes from the complex acoustic pressures measured on the array. However, discretely sampling a noisy incident acoustic field means that the modal amplitudes will not be recovered exactly. These modal amplitude errors can cause degradation of both detection and localization performance of modal matched-field processors. In this study, three techniques for mode filtering are examined from the point of view of processor detection degradation. Computer simulations demonstrate how array configuration and noise combine to produce mode filter errors and how these errors translate into processor degradation. In addition, an analytic description of the relation between mode filter error and processor performance is discussed.
Modal beamforming array gain
Authors: Yang T. C.
Published: October 2005

In a previous paper [T. C. Yang, J. Acoust. Soc. Am. 82, 1736–1745 (1987)], range and depth were successfully estimated by modal beamforming. In this paper, the modal beamforming array gain in closed form for a vertical array of N hydrophones is derived. For the case of white noise and a long well-populated vertical array, the array gain nearly equals 10 log N. For the case of colored noise and a finite aperture, finite element vertical array, the array gain degrades from the theoretical by an amount determined by the mode-depth amplitudes evaluated at the source and receiver depths, and the modal covariance matrix of the noise. Numerical estimates of array gain are given for the FRAM IV vertical array for a colored noise environment in the Arctic sound channel. Approximately 1-dB array gain degradation is found at 47 Hz. Proper array configuration and deployment depth is not only important for optimizing array gain but also critical for maximizing the input signal-to-noise ratio on a vertical array, which together with the array gain determines the array output signal-to-noise ratio.
Overview of telephone interpretation research at ATR
Authors: Kurematsu Akira , Shikano Kiyohiro
Published: October 2005

The Automatic Telephone Interpretation system is a facility that enables a person speaking in one language to communicate readily by telephone with someone speaking another. Basic research on the component technologies, speech recognition, machine translation, and speech synthesis is being carried out at ATR. In speech recognition research, the goal is to recognize continuous speech containing a large vocabulary of approximately 3000 words. Based on phonemes as the subunit of speech, words and phrases are recognized. The main approach is a hybrid combination of feature-based segmentation and phoneme and word recognition by hidden Markov and neural network models. A large-scale speech database with phonetic transcriptions has been developed for use in speech research. The incorporation of speaker adaptation has been chosen as our approach to speaker independence. Recognition of conversational speech on a specific domain will be explored. Integrating speech and language processing is an important area to be tackled. A technology that uses information about languages and background knowledge to understand the speech content is being studied. As for speech synthesis, rule-based speech synthesis with high quality and high naturalness is under development. In the area of machine translation, several approaches to the translation of spoken dialogue between Japanese and English are under study. Intention extraction for dialogue interpretation, a plan recognition model for intersentential analysis, and knowledge processing in a specific domain are intended to be investigated.
Complex impedance measurements in guided wave tubes
Authors: Harrison R.
Published: October 2005

A pulsed guided wave tube operating over 2 decades of frequency was constructed for the measurement of the acoustic properties of materials. The impedance (Z) is calculated from the coefficient of reflection and the difference in phase shift between waves reflected from a steel reference plate and waves reflected from the material. The phase shift is obtained from the angle formed by the real and imaginary components of the Fourier transform of the reflected waveforms. Additional points for plotting in the Z plane are found by interpolation between the data points defined by the Fourier transform zeros in the reflection coefficient-phase angle space. Well-formed trajectories in the Z plane can now be plotted. Using the FFT has the advantage that (1) only a few accurately known data waveform points are needed as compared to traditional techniques that require a more complete waveform description of the waveforms to be compared and (2) the storage of data for calibration is facilitated. The entire system has been completely automated. Data will be presented showing the change of impedance with frequency for silicone rubber.
Design of a three-layer acoustic window made from plastic
Authors: Nakanishi Toshiyuki , Kikuchi Toshiaki , Hasegawa Akio , Takahashi Sumio
Published: October 2005

Theoretical and experimental studies are conducted on the design of an acoustic window for the acoustic imaging sonar to be equipped on “SHINKAI 6500.” Frequencies used for this sonar are 100 kHz for PPI and 300 kHz for acoustic imaging. This acoustic window made from plastic material of good physical strength and low transmission loss at these frequencies is being developed. Generally, a single layer plastic plate has difficulty since it has both the longitudinal and the transverse waves in itself in the case of oblique incident acoustic waves in liquid, and, consequently, the transmission loss increases remarkably under the influence of these waves. The control for reducing the losses for the required frequency bands is accomplished by fitting matching layers on both sides of the plastic plate that constitutes the main layer. From calculations based on the theory, adequate acoustic velocity and thickness for the matching layers can be successfully obtained.
Development of off-line, deep-towed seismic profiler
Authors: Nishimura Kiyokazu , Kishimoto Kiyoyuki , Miyazaki Teruki , Joshima Masato , Murakami Fumitoshi
Published: October 2005

A deep-towed seismic reflection profiler has been developed by the Geological Survey of Japan. The aim of the system is to obtain detailed profiles of subbottom structures in the deep seas. The system consists of deep-towed seismic recording packages with a deep-sea hydrophone streamer, a shipboard data logging and processing system, and an acoustic navigation system for the deep-towed packages. The packages are towed near the seafloor, and a sound source (an air gun) is towed near the sea surface. After the recovery of the packages on board, the seismic data in the package are transferred to the data logging and processing system through a high-speed serial link. The system has the following features. (1) Off-line towing: the hydrophone streamer and the recording packages are towed by an ordinary wire without a conducting cable. (2) No depth limitation: all electrical packages are kept in pressure-proof housings (depth limit 10 000 m), and unlimited-depth hydrophone elements are used for the streamer. (3) Digital recording and nonvolatile, solid-state memories: an A/D converter with a floating-point amplifier is used. Digitized seismic data are recorded in 16-megabyte magnetic bubble memories. Field tests of the system were successfully carried out. Some preliminary profiles of deep-towed digital seismic data were obtained.
Low-frequency seismoacoustic ambient noise levels
Authors: Kibblewhite A. C. , Wu C. Y.
Published: October 2005

Studies of the infrasonic ocean noise induced by the interaction of surface waves have shown that the noise levels are dependent not only on the ocean-wave field but also on the geoacoustic properties of the environment. A full understanding of the processes involved requires an examination of the role of the inhomogeneous components of the induced acoustic field and the reflection coefficient of a multilayered seabed. An analysis has shown that, at infrasonic frequencies, the inhomogeneous component can be important in waters with depths less than a few hundered meters. It has also been established that the reflection loss for particular models shows energy conversion from interface waves to trapped normal modes, when a low-speed layer is present within the geoacoustic sequence. This conversion leads to an apparent periodicity in the frequency dependence of the reflection loss.
Towed array measurements of bottom time/angle spreads during Pacific Echo
Authors: Dicus Ronald L.
Published: October 2005

Pacific Echo was a joint United States/Canadian experiment conducted in the Pacific in June 1986. As part of the exercise, explosive charges (MK-61 SUS) were detonated along a closing-range track near broadside to a towed array. A high-level source (MK-V1, 193 dB at 15 Hz) transmitted linear frequency-modulated signals, sweeping from 5–15 Hz and back in 50 s. Measurements reported here are for a thin-sediment site just south of the Tufts Abyssal Plain. Previous omnidirectional measurements in thin-sediment areas are characterized by bottom time extensions on the order of a second. Time (angle) spread is defined here by identifying a two-dimensional area in time/angle space containing an arrival, integrating over angle (time), convolving the resulting one-dimensional sequence with a rectangular window, and determining the length of the window giving a peak −3 dB down from the total energy of the particular arrival. Observed time and angle spreads increased with grazing angle and frequency to values on the order of 0.5 s and 15° over a frequency and grazing angle range of 10–300 Hz and 7°–70°, respectively. The dependence of reflectivity, time/angle spreads, and signal gain on frequency and bottom grazing angle will be discussed.
Ultrasonic relaxation in GeO2 glass
Authors: Sakai Keiji , Choi Pak-Kon , Takagi Kenshiro
Published: October 2005

Ultrasonic velocity and absorption were observed in GeO2 glass over the frequency range of 20 MHz–1.5 GHz. High-resolution Bragg reflection (HRB) technique was successfully utilized at the higher frequency range, where the conventional pulse method loses accuracy due to the very strong attenuation. A pulse apparatus equipped with a PVDF polymer transducer was useful over the 20- to 90-MHz range. It is known that GeO2 has a large absorption maximum with respect to temperature if measurements are made in the MHz range. In the observed absorption spectrum, it was found that the corresponding maximum was centered at around 1 GHz at room temperature. The excess absorption was ascribed to a relaxation mechanism, in which the molecular structure was deformed in a double-well potential. The present results suggest that the height of the energy barrier between the two states has a distribution of the Gaussian type centered at 0.13 eV and with a width of 0.09 eV. Assuming this energy barrier, the ultrasonic absorption was predicted to be a function of temperature. The result of prediction was in good agreement with the present experiment made in the low-temperature region.
Viscosity sensor using a surface acoustic wave delay line
Authors: Nomura Tooru , Iino Syu , Yasuda Tsutomu
Published: October 2005

A surface acoustic wave (SAW) sensor for sensing viscosity is presented. SAW devices are commonly used in electronic signal processing. SAW delay lines offer many attractive features for applications as vapor and liquid sensors. In this paper, a SAW viscosity sensor utilizing the shear horizontal displacement of a leaky SAW mode is presented. The sensor is based on a dual SAW delay line oscillator. The leaky SAW with the shear horizontal displacement is excited on a 36°-rotated Y-cut X-propagation LiTaO4 substrate using interdigital transducers. The sensitivity of the SAW oscillator frequency to the viscosity of liquids has been measured. It is shown that the viscosity sensor has a sensitivity of about 100 Hz/cP at an operating frequency of 35 MHz with a threshold sensitivity of about 2 cP. Test results to include response time, hysteresis, linearity, and temperature effects will also be presented.
Experimental investigation of the SAW harmonic reflection characteristics of a metal strip array
Authors: Kojima Toshihiro , Obara Hiroyuki , Shibayama Kimio
Published: October 2005

It has been reported by R. C. M. Li and J. Melngailis [IEEE Trans. Sonics Ultrason. SU-22 (3), 189–198 (1975)] that a metal strip array on Y-Z LiNbO3 shows a strong reflection of the surface acoustic wave (SAW) at the frequency of the second harmonic component due to the energy storage effect. To apply this properly to reflectors for a SAW resonator, the following characteristics for both shorted and open metal strip arrays on LiNbO3 (128° rot. Y, X-prop.) have been experimentally investigated. The dependence of the second harmonic reflection coefficient on the metalization ratio and the comparison between the reflection coefficient for the second harmonic and the fundamental frequency regions have been examined in detail. The relations between the frequency at which the maximum reflection coefficient occurs and the center frequency of an interdigital transducer (IDT) have also been determined. The characteristics obtained here are useful for the optimum design of second harmonic, one- and two-port resonators.
Modal beamforming in a range-dependent environment
Authors: Yang T. C.
Published: October 2005

Modal beamforming, previously developed for a range-independent case [T. C. Yang, J. Acoust. Soc. Am. 82, 1736–1745 (1987)], is extended to a range-dependent environment. The range-dependent sound channel is partitioned into consecutive waveguides as a function of range. The normal modes in each waveguide are determined by the sound-speed and bottom profile in that waveguide. The mode coefficients in adjacent waveguides are related by matching the boundary condition at the vertical interface. Modal beamforming is conducted for each waveguide. The array outputs from the waveguides are combined and plotted as a two-dimensional surface with respect to range and depth. The source location is determined by the peak of this surface. A numerical example is given based on simulated data for a source propagating through a series of different sound-speed profiles. The simulated data are created using an improved coupled-mode program that was initially developed by R. B. Evans. Range and depth are estimated using a long vertical array of 3000-m aperture, containing 50 phones. Modal beamforming for a range-dependent environment can also be conducted directly using the coupled-mode program or any other propagation code.
Acoustic guided waves in periodical axisymmetric structures
Authors: Ching G. , Lagier M.
Published: October 2005

Acoustic wave propagation in axisymmetric periodic waveguides concerns many different application fields: acoustic waves in ducts, acousto-mechanic waves in bore hole, and noise propagation in towed arrays. A periodic guide with an intricate elementary cell and coupling with an external medium (possibility of leaky waves) makes the theoretical analysis of such waveguides more difficult. The present work is based on the combination of three analysis tools: finite element analysis for the elementary cell dynamics, Bloch-Floquet wave decomposition for the periodic structure processing, and the perturbation method applied to harmonic space in order to take into account the external medium interactions. Dynamic matrices partitioning and the introduction of input-output propagation conditions lead, in the case of zero coupling (with the external medium), to the typical secular equation: [P + ΓQ + Γ2R]x = 0, where Γ is the unknown propagator and x is the associated eigenvector; P, Q, and R are matrices deduced from the initial dynamic matrices. The external medium boundary condition introduces an additional nonpolynomial term and leads to a new nonlinear secular equation that is difficult to solve. The perturbation method allows, on the one hand, this difficulty to be overcome and, on the other hand, leads to a more physical approach, particularly in the method state where coupling leads to a Bloch-Floquet harmonic wave degeneracy overtaking. Preliminary numerical tests on well-known waveguides are presented. The results agree very well with analytical solutions. [Work supported by GERDSM.]
Angular distribution of ice cracking events in the Arctic
Authors: Zakarauskas P.
Published: October 2005

The measured vertical directionality of ice events in the polar ice pack is presented. Concurrent 345-s samples of ambient noise measured with four vertically spaced hydrophones suspended over the Arctic continental shelf were searched for corresponding peaks in the envelope of the acoustic pressure. In a previous analysis [P. Zakarauskas, J. Acoust. Soc. Am. Suppl. 1 82, S30 (1987)], the ice events were shown to be of thermal origin. The resulting angular distribution of the 765 ice events observed during the sample has its maximum at 15 deg below the horizontal, with most of the events arriving between 0 and 25 deg below the horizontal. An explanation for this observation is given in terms of a simple ray-based model of the propagation of ice cracking noise. The model includes the seafloor and under-ice angular reflection coefficients measured at a neighboring location.
A standard definition for wind-generated, low-frequency ambient noise source levels
Authors: Carey William M. , VonWinkle William A. , Browning David G. , Kewley Douglas J.
Published: October 2005

Low-frequency, wind-generated ambient noise source levels are important input parameters for newly developed ambient noise prediction models such as DUNES. However, there has been a significant variation among recently reported source levels. An analysis is made of these values. Although the actual noise measurements, the assumed source model, and a small angle approximation are similar for all papers, differences arise due to additional geometrical factors or a further approximation to shift the source level to the surface. A standard source level definition and evaluation method are proposed. Standard values of low-frequency, wind-generated source levels are presented based on this strategy. [Work supported by NUSC.]LLL2. Abstract withdrawn.
The hydroacoustics of a drop impact
Authors: Nystuen Jeffrey A.
Published: October 2005

Rainfall has consistently been one of the most difficult geophysical quantities to measure, especially in oceanic regions. Recently, it has been demonstrated that rainfall can be detected at sea by monitoring for the unique underwater sound generated by the rain impacting onto the ocean surface [Nystuen and Farmer, Atmos-Ocean (1988), in press]. The goal of quantifying rainfall rate through underwater ambient sound measurements requires a more complete understanding of the hydroacoustics of the drop splash and the influence of wind upon these physics. A numerical model is used to explain the hydroacoustics of a drop impact as an acoustic waterhammer modified by the emergence of a fluid jet at the base of the drop. The drop is an acoustic source until the radius of the contact circle between the drop and the underlying water surface reaches the maximum cross-sectional radius of the drop. The duration of the pulse depends on drop size, shape, and speed. Using realistic raindrops, the observed unique underwater acoustical signature of rain, an increase of ambient sound energy at 15 kHz, is explained.
Characterization of anisotropic elastic wave behavior in media with parallel fractures and aligned cracks
Authors: Hsu Chaur-Jian , Schoenberg Michael
Published: October 2005

Ultrasonic experiments were performed to investigate the anisotropic properties of an isotropic background medium with a system of long closely spaced parallel fractures. Immersion experiments at frequencies from 0.1-0.7 MHz were performed on Plexiglas that was cut into thin sheets and then pressed together. Measurements of phase and group velocity were obtained for compressional and shear waves both normal to and parallel to the “fractures” and for quasicompressional and quasishear waves at a variety of oblique angles. The data were analyzed to determine the anisotropy parameters due to the fracturing, based on modeling the fractures as a set of infinite, linear, although not necessarily rotationally symmetric, slip interfaces [M. Schoenberg, J. Acoust. Soc. Am. Suppl. 1 82, S88 (1987)]. For the most part, the experiments fit the theory quite well even though losses that can occur in the fractures have been ignored in this first attempt to match theory and experiment. The anisotropy attributable to the fracture system depends on the geometry of the fractures, the roughness, the static closing stress across the fractures, and the nature of the infilling medium, if any. From a knowledge of the anisotropic elastic moduli of the fractured medium and the isotropic background moduli, the average compliance matrix may be deduced. In turn, this gives the average behavior of each fracture plane.
Dispersion curves for thermomechanical waves
Authors: Rudgers Anthony J.
Published: October 2005

When mechanical deformation and heat conduction are both accounted for, two coupled partial differential equations (PDEs) describe wave propagation in a medium. For one-dimensional wave propagation in a medium without mechanical dissipation, these PDEs yield a dispersion relation of complex quadratic form, which shows that two kinds of waves can propagate. Each wave has a mechanical and a thermal component. The speeds and the attenuations of such waves are illustrated in two cases. First, it is shown that the “classical” form of the thermomechanical PDEs provides an interesting but nonphysical description of wave propagation. Second, a Vernotte thermal relaxation time is introduced into the PDEs. When the value of the Vernotte parameter is set by applying the Debye theory of specific heats in a rudimentary way, it is shown that a physically plausible description of wave propagation can be obtained. In this latter connection, the consequences of using a “phonon-gas” model of a thermomechanical medium is considered briefly. [Work supported by ONR.]
Experimental simulation of a pressure-release finite cylinder using a closed-cell neoprene coating
Authors: Caille Gary W. , Rogers Peter H. , Jarzynski Jacek
Published: October 2005

The scattered field of a coated finite cylinder with flat end caps is compared with the theoretically obtained field for a pressure-release cylinder. The cylinder is stainless steel with a radius of 2.0 in. and length of 12.75 in. The wall thickness is 0.25 in. and the end cap thickness is 0.75 in. The ka range is approximately 1.5–15. The coating is 0.25 in. of closed-cell neoprene rubber. The backscattered pressure is calculated for sound incident normal to the axis using a modified Naval Oceans Systems Center CHIEF program and for end-on incidence using the Naval Research Laboratory SHIP program. The intent of the coating is to simulate the pressure-release boundary condition. This condition is, in fact, achieved because: (1) Agreement between the theoretical and experimental results is within 1 dB; (2) the coating removes various elastic effects including a return signal from the back end (observed for the uncoated cylinder, for the end-on case); and (3) the scattered signal from the coated target is phase inverted relative to the uncoated case. [Work supported by Office of Naval Research, Code 11250A.]
Playing with hornists
Authors: McCue Edward
Published: October 2005

An examination of the acoustical requirements of the orchestral horn section during rehearsal and performance serves as a point of departure for the design of performance platforms and orchestra pits. In addition to displaying an asymmetrical radiation pattern that is disadvantageously oriented to the right and rear of the player, the spectrum of the horn is easily masked by other sections of the orchestra. The instrument's timbre and physical placement within the orchestra are reviewed with respect to the historical evolution of orchestration and the architectural design of opera houses and concert halls. Contemporary difficulties with excessive loudness and ensemble are discussed. Architectural design criteria leading to a good performance environment for the horn section are suggested.
Free-field nonlinear pulse propagation outdoors
Authors: Sparrow Victor W. , Raspet Richard
Published: October 2005

As an intermediate step toward developing a two-dimensional finite difference model of outdoor sound pulse propagation containing both cumulative finite amplitude effects and the local effect of a finite impedance ground, a program addressing the nonlinear steepening has been implemented. Previous studies [J. Acoust. Soc. Am. Suppl. 1 80, S104 (1986) and J. Acoust. Soc. Am. Suppl. 1 82, S64 (1987)] have involved linear propagation only. The vectorized code has been running on a CRAY X-MP at the National Center for Supercomputer Applications in Champaign, IL because of its intensive CPU and memory requirements. Details of the time stepping algorithm, as well as those of the numerical boundary conditions, will be given. The results of this program will be compared with that of SHOCK.F, a one-dimensional acoustic propagation code including finite amplitude effects [J. Acoust. Soc. Am. 74, 1514–1517 (1983)]. As an example of some typical color graphics output, a short videotape will conclude the presentation.
Across-talker acoustic properties for place of articulation in nasal consonants
Authors: Seitz P. F. , McCormick M. M. , Watson I. M. C. , Bladon R. A. W.
Published: October 2005

Several recent investigations of nasal consonants have focused upon the nature of the spectral change that takes place over the acoustic boundaries between nasal murmurs and vowels. Existing findings that use an algorithmic classification of place of articulation in nasals are based upon data from only a small numbers of talkers. A new method for parametrizing spectral change in nasal-to-vowel transitions was developed and was used to develop classification criteria that were applied to a larger talker base of 20 female and 20 male talkers who produced a total of 854 nasal-initial tokens. Male-female differences of a predicted kind (approximately 1 Bark spectral shift) were detected. However, the automatic classification criteria were sufficiently robust that, even ignoring such spectral shifts, the technique yielded approximately 79% correct classification of place of articulation in initial nasals. [Research supported by Alvey Project MMI 092.
Distributed processing of finite state grammars for connected word speech recognition
Authors: Glinski Stephen C.
Published: October 2005

An algorithm is presented for performing real-time connected word speech recognition under the constraint of a finite-state grammar, and executed in a multiprocessor pattern recognition computer. Each processing element of the computer consists of a 40-MIP one-chip pattern recognition processor and its associated local memory. [S. Glinski et al., “The Graph Search Machine (GSM): A VLSI Architecture for Connected Speech Recognition and Other Applications,” Proc. IEEE 75 (9), 1172–1184 (1987)] Aspects of the algorithm discussed include: the choice of models for speech sound units, words, and grammars; the problems of partitioning the models across processing elements; and the problem of removing redundancy in the finite state grammar. Extensions of the algorithm to more powerful grammars is also discussed. The capacity of the recognizer (i.e., vocabulary size) as a function of grammatical complexity is presented. Results indicate that high-accuracy medium vocabulary speaker-trained connected word recognition is readily attainable.
Acoustic segmentation and phonetic classification in the SUMMIT system
Authors: Glass James , Kassel Robert , Kaufman David , Phillips Michael , Seneff Stephanie , Zue Victor
Published: October 2005

This paper describes recent work on acoustic segmentation and phonetic classification as part of an effort in speech understanding system development. The signal representation is based on an auditory model that incorporates known properties of the human auditory system, including critical-band filtering, saturation, adaptation, forward masking, and synchrony detection. Acoustic landmarks are determined using a measure of local similarity. These landmarks are embedded in a multilevel structure in which information ranging from coarse to fine is represented in an organized fashion. An analysis of the acoustic structure, using 500 utterances from 100 different talkers, shows that it captures over 96% of the acoustic-phonetic events of interest with an insertion rate of less than 5%. Phonetic classification is achieved by defining a set of generic property detectors based on the knowledge of acoustic phonetics. The settings of the parameters are obtained by a search procedure, using a large body of training data. The resulting features are selected through an optimization process. Classification is achieved using conventional pattern classification algorithms. Classification performance was evaluated against hand-aligned phonetic transcriptions provided by phoneticians. The evaluation was based on 225 sentences from 45 talkers using a set of 38 context-independent phone labels. The correct label was the top choice over 70% of the time, and within the top three over 90% of the time. [Work supported by DARPA-ISTO under contact N00039-85-C-0254.]
Analysis, perception, and recognition of isolated Korean vowels
Authors: Chung Hyun-yeol , Makino Shozo , Kido Ken'iti
Published: October 2005

Distribution of the formant frequencies of eight isolated Korean vowels were analyzed and compared to the results of perceptual experiments. Recognition experiments were carried out based on these formant distributions. In the distributions of vowels on the F1-F2 plane, overlappings were found between the vowels [u], [o], [ə],and [Ɯ]. Here. [ɛ] and [e] overlapped heavily as if they were the same vowel. Using perceptual domains defined by the results of listening lests, the dialectal differences and iterspeaker differences were investigated. In a recognition experiment, a speaker-independent recognition with the Bayes decision method was conducted for 1760 isolated vowels spoken by 20 speakers. A recognition rate of 81.2% was obtained with the formant frequencies F1 and F2, while the rate with the LPC cepstrum coefficients was only 76.4%. Normalization for eliminating interspeaker differences showed good results for vowels uttered by speakers of the same dialect. The highest recognition rate of 99.8% was obtained using a supervised learning method.
Automatic speech recognition based on findings of the human processes of speech perception
Authors: Udagawa Hirofumi , Ohno Sumio , Fujisaki Hiroya
Published: October 2005

One of the shortcomings of coventional methods of ASR is that they operate on principles that are quite different and far less flexible than those of human speech perception. Previous studies on the perception of continuous speech have revealed that (1) a linguistic message may have multiple perceptual representations in various units which differ in size, and that (2) a representation of the input speech in larger units is derived earlier in the process of speech perception than that in smaller units such as phonemes. Hence, a word is often recognized on the basis of global characteristics without being subjected to detailed phonemic analysis. Based on these findings, a novel system for continuous speech recognition is proposed. The front end analyzes the input speech with various degrees of temporal and frequency resolution, and the results are matched against templates of various linguistic units that differ in size. The recognition of smaller units takes place only when the matching of larger units fails to produce a unique output. The capability of the approach was demonstrated by preliminary recognition experiments of weather forecast sentences in Japanese.
Measures of phonological rule set coverage
Authors: Cohen Michael , Bernstein Jared , Murveit Hy
Published: October 2005

Some speech recognition systems use alternative lexical forms to cover various pronunciations encountered when people speak. However, excess forms in the lexicon degrade recognition performance. Thus lexicons should maximize coverage of actual pronunciations that occur, while minimizing the number of alternative pronunciations. A set of phonological rules for English as been developed at SRI to cover a large body of transcriptions of sentences read by American speakers. This phonological rule set covers 98.5% of the segments in a separate set of transcriptions. Rule efficiency can be defined as the ratio of the number of times forms generated by a rule were spoken in a corpus of test data to the number of forms generated by the rule for the sentences in that test data. By this definition, palatalization of /t/ and /d/ before tautosyllabic /r/ is efficient, while flapping of /t/ that follows /l/ is not. There is a trade-off between the coverage of a rule set and the number of alternative pronunciations generated by the rule set. Rule efficiency can be used to rank rules in a rule set, to derive a family of rule sets that maximize coverage for a given number of alternative forms represented, or to select among alternative rule sets. [Work supported by DARPA.]
Recognition of speech produced in a simulated noisy environment
Authors: Hattori Hiroaki , Yoshida Kazunaga
Published: October 2005

Pronunciation manner varies depending on the noise of the speech environment. This is known as the Lombard effect. It affects the acoustic features of speech such as intensity, pitch, duration, and spectral shape. Because vowels play an important role in Japanese speech recognition, special attention was paid to the variability of the acoustic features of vowels uttered in noise. First, the variability of the five Japanese vowels was examined using speech uttered while hearing noise through headphones. It was observed that the higher formants are unstable and that the energy in the middle frequency range of the vowel spectrum increases depending on the noise level. Based on these observations, a normalization method combining band limitation and spectral tilt compensation was proposed, and its effectiveness in the recognition of vowels was shown by experiment. Next, a new word recognition method utilizing this normalization technique was proposed. This method performs vowel normalization at the portion where both test and reference patterns are vowel-like sounds. The experimental results show the significant effectiveness of the proposed method. The recognition rate obtained by the proposed method with normalization is 96% for 50 words uttered by two male speakers, while the recognition rate without normalization is 80%.
Approach to speaker-independent connected word recognition for telephone application
Authors: Hirose Kei Miki
Published: October 2005

This paper describes a new speaker-independent connected word recognition method for telephone use. For this recognition, it is necessary to deal with the following four problems: utterance speed variations in speakers; spectrum variations caused by a speaker's sound source and vocal tract characteristics; interference between words in continuous speech; and telephone circuit noise. For the first problem, a new nonlinear time division method is proposed using the spectral transition ratio. The result shows a phoneticlike segmentation. This method is more effective than DTW (dynamic time warping) in the case of speaker-independent systems. For the rest of the problems, a set of features will be used that give a maximum recognition accuracy, for example, the spectral local peak, the ratio between the higher and lower frequency components, and spectral transition vectors. The method is quite robust in the face of spectral changes because the features cooperate with each other. This method shows a high performance in speaker-independent connected word recognition for telephone use.
Scattering and reflection of acoustic waves by striated surfaces depicting the Arctic Ocean ice cover
Authors: McClanahan Theodore P. , Diachok O. I.
Published: October 2005

Acoustic reflectivity from plastic models of sea ice ridges on a thin plate were investigated for 0.67 < ka < 101 (where k is the wavenumber and a is the mean ridge depth). Models were constructed of lucite, which has acoustic velocities consistent with analytically estimated velocities of sea ice ridges and of “fast cast,” a material with much lower velocities. For ka > 1 measurements from randomly oriented and parallel lucite and fast cast ridges on a thin plate and on an infinite half-space were nearly identical, and in good agreement with Twersky's theoretical predictions. At ka < 1 and at grazing angles less than 20°, measured reflection losses were significantly greater than predictions: twice as large for randomly oriented lucite ridges on a thin plate. The phase of the reflected signal was strongly dependent on the orientation and the material property of the ridges. Results for randomly oriented lucite ridges were found to be consistent with previously reported inferences of reflection loss and phase based on matched field processing of actual Arctic data and consistent with reported resonance effects.
Wave scattering from rough surfaces using the facet-ensemble approach
Authors: Ma Yushieh , Ro Ruyen , Varadan Vasundara V.
Published: October 2005

The facet-ensemble method has been widely used to investigate scattering from long crested surfaces whose roughness and slopes are, in general, fairly small. In addition to its simple physics, this method also takes the full advantage of modern computing facilities to provide numerical results in a very efficient manner under any change of surface conditions. In order to extend this method to any irregular surfaces of different roughness and slope scales, multiple scattering as well as wedge diffraction should be carefully incorporated. Therefore, a modified facet-ensemble approach that does not restrict the grazing angles is reported here. In order to describe the scatttering characteristics of random rough surfaces, the statistical functions can be built, but have not been considered before, into the formalism that saves even more CPU time in performing the ensemble averaging. A comparison is made using this approach with the experiment conducted on scattering from random Gaussian surfaces. Results agree very well and also presented is the effort towards the two-dimensional random rough surface scattering. [Work supported by the Research Center for the Engineering of Electronic and Acoustic Materials.]
Multiple scattering of radiation by rows of parallel cylindrical obstacles in water
Authors: Matsuda Kazuhisa , Kikuchi Toshiaki , Takahashi Sumio , Hasegawa Akio , Nakanishi Toshiyuki
Published: October 2005

Experimental forward scattering pressures for a line array of ten cylinders of various sizes and materials were obtained to compare with calculated values. An investigation was made for kd from 1 to 10 (k is the wavenumber and d is the center-to-center spacing) and kɑ < 1 (ɑ is the cylinder radius). The calculations were made by Twersky's theory of multiple scattering. When kd is larger than 3, the pressure level of the second-order scattering wave is 10–20 dB lower than that of the first-order scattering wave. In this range, the experimental results agree well with the calculated results. Large discrepancies are found, however, for values of kd smaller than 2. Also discussed are the effects of boundary conditions.
Simple approximate formulas for backscattering of sound by spherical and elongated objects
Authors: Stanton T. K.
Published: October 2005

The scattering of sound by objects is a complex process and is dealt with both analytically and numerically. Regardless of method, calculation of the exact scattered field is quite laborious and may require much computer time as well as human time to program the computer. In order to simplify calculations of the backscattering by a fluid sphere, Johnson combined low- and high-frequency limits in a heuristic manner to obtain a simple closed-form solution (named the “high-pass” solution) [J. Acoust. Soc. Am. 61, 375–377 (1977)]. In this present work, his approach is refined and extended to the prolate spheroid, straight finite cylinder, and bent finite cylinder. The high-pass solutions are formulated in a general way so as to describe scattering by idealized objects where their shape is simple (i.e., sphere, spheroid, etc.) and their material is lossless and by nonideal more realistic objects where their shape may be irregular and their material lossy. Comparisons are made of the scattering of these objects using both modal series solutions and the “high-pass” solutions for fluid, elastic, rigid and fixed, and gaseous materials over a wide range of frequencies. In addition to the numerical simulations, some of the high-pass solutions and corresponding modal series solutions are compared with data involving “ideal” objects (machined Dural) and “nonideal” objects (marine organisms: shrimp, euphausiids, and fish). The numerical and experimental results show promise for use of the high-pass models for quick estimates of backscattered sound of many types of objects. [Work supported by the ONR.]
Formant-frequency estimation by linear transformation of the LPC-cepstrum
Authors: Broad David J. , Clermont Frantz
Published: October 2005

Correlations near 0.98 between vowel formant frequencies and linear combinations of 1/3-octave-band spectrum levels reported by Pols et al. [“Perceptual and physical space of vowel sounds,” J. Acoust. Soc. Am. 46, 458–467 (1969)] suggest that the formants might be estimated by linear transformation of either a low-resolution log spectrum or of the first M cepstral coefficients ci, which are linear functions of a log spectrum. Applying this idea with M = 14 to four speakers results in pooled intraspeaker prediction errors of 41, 126, and 134 Hz for F1, F2, and F3, respectively. These become 52, 164, and 218 Hz when regressions on data from three speakers are used to measure formants from a fourth. The method is therefore not very accurate, but it is robust in that large errors from misidentified peaks are rare. A study of single-resonance waveforms explains why the method works: The ci are roughly cosines in the resonance frequency F, with i half-cycles over the analysis bandwidth; these ci (F) form a basis set for approximating the function Fesl = Freal.