bluuebunny/crossref_metadata_2025 huggingface dataset:
Sintering temperature dependence of the photoacoustic spectra for ceramic CdS and Cd S0.5Se0.5.
Authors: Toyoda T. , Aikawa N. , Shinoyama K.
Published: October 2005

CdS and CdSe are compound semiconductors which have bandgaps of 2.52 and 1.75 eV, respectively. The bandgap of mixed crystals such as CdS1−xSex (0≤x≤1) is proportional tox . Reliable characterizations of them have become important for applications in solar cell technologies. Although several investigations of the single crystals have been carried out, optical absorption measurements of ceramic CdS and mixed crystals have been a difficult problem. The purpose of this report is to show the results of photoacoustic spectra for ceramic CdS and CdS0.5Se0.5 for different sintering temperatures. Ceramic plates were obtained by sintering over the temperature range from 600 to 950 °C in N2 gas. The body of the cell was an aluminum cylinder with a small channel in its periphery in which the microphone was inserted. Modulation frequencies for the measurements were 33, 133, and 257 Hz and the measurements were carried out in the wavelength range of 330–1200 nm. The signal intensity and phase spectra shift toward the high-energy regions and the slope of the absorption edge becomes steeper as the sintering temperature is increased. Moreover, the phase spectra show minima whose positions depend on sintering temperature, because of the increase of the recombination rate.
All-pole representation of the cochlear input impedance, modal representation of the ear canal signal, and reflectance in the ear canal in the presence of spontaneous otoacoustic emissions
Authors: Talmadge Carrick L. , Tubis Arnold , Long Glenis R.
Published: October 2005

The cochlear input impedance associated with the linear active component of cochlear mechanics is represented as an all-pole function of frequency. This allows the dynamics of the ear canal signal to be expressed in terms of the behavior of coupled damped harmonic oscillators. Spontaneous otoacoustic emissions correspond to negative damping and, consequently, unstable oscillations. The introduction of a compressive nonlinearity, which stabilizes the spontaneous emissions, gives an interesting and computationally simple model for investigating the response of the auditory periphery in the case where there are one or more spontaneous emissions. The model is used, in particular, for studying the behavior of the reflectance in the ear canal. The magnitude of the reflectance will normally be less than one due to the entrainment of the emissions.
Dynamic behavior of the human middle ear: A finite-element method (FEM) analysis
Authors: Koike Takuji , Wada Hiroshi , Kobayashi Toshimitsu , Takasaka Tomonori
Published: October 2005

A few investigations of finite-element method (FEM) application to the analysis of the middle ear have already been reported [Funnell et al., J. Acoust. Soc. Am. 81, 1851–1859 (1987); K. R. Williams and T. H. J. Lesser, Br. J. Audiol. 24, 319–327 (1990)]. However, in the models of these studies, ossicles and middle ear cavities were neglected, and their effects on the dynamic behavior of the middle ear have not been analyzed. In this study, first, applying the authors’ own FEM program, a three-dimensional FEM model of a human auditory periphery was established. Then, vibration modes and transfer functions of the middle ear were obtained. The following conclusions can be drawn. (1) The vibration mode of the tympanic membrane and ossicles changes considerably due to the elastic ligament support of the ossicles. (2) This change enables the transfer function of the middle ear to keep flat. (3) The effect of the cavities on the dynamic behavior of the middle ear is large; the amplitude of the tympanic membrane vibration after opening the cavities is nearly two times larger than that before opening them. [Work supported by Research Fellowships of the Japan Society for the Promotion of Science for Young Scientists.]
Middle ear effect on otoacoustic emissions (OAEs): A finite-element method (FEM) analysis
Authors: Wada Hiroshi , Koike Takuji , Kobayashi Toshimitsu , Ohyama Kenji , Takasaka Tomonori
Published: October 2005

Otoacoustic emissions (OAEs) are believed to originate from outer hair cell movement and are considered to be transmitted to the external auditory meatus through the middle ear in a retrograde fashion. Therefore, the effect of the middle ear on OAEs seems to be large. However, few investigations of middle ear effects on OAEs have been published. In this study, first, applying the authors’ own finite-element method (FEM) program, a three-dimensional FEM model of a human auditory periphery, which included the external auditory meatus, the tympanic membrane, ossicles, and middle ear cavities, was established. Then, the retrograde and anterograde transmission factors of the normal and pathologic middle ear were examined. The following conclusions can be drawn. (1) As expected, the value of the retrograde transmission factor is lower than that of the anterograde one. (2) This is mainly caused by the dynamic behavior of the tympanic membrane. (3) The levels of OAEs obtained from the patients such as atelectatic tympanic membrane, I–S joint separation, and otosclerosis are lower than the noise level of the OAE measurement system, because the value of the middle ear transmission factor of these patients is smaller than that of the normal subject.
The half-octave temporary threshold shift
Authors: Bies David Alan
Published: October 2005

A model is proposed which provides a simple explanation for the noise-induced half-octave temporary threshold shift. The cochlear partition is modeled as a series of lightly coupled mechanical oscillators in which the outer hair cells are displacement responsive and the inner hair cells are velocity responsive. The outer hair cells are assumed to take an active role in providing negative damping in a region of stimulation of the cochlea at low-sound pressure levels. In the model, each oscillator is linear to very-high-sound-pressure levels except in a region of stimulation where the damping ratio varies with sound-pressure level. The model predicts that the passive viscous damping ratio of each oscillator is 1/√2. At low-sound-pressure levels negative structural damping reduces the total damping ratio to about 0.055. At high-sound-pressure levels negative damping ceases and the response in a region of high-level stimulation is then passive. The variable damping ratio provides an explanation for all cases of nonlinear cochlear response which have been investigated.
Expression of basic fibroblast growth factor and transforming growth factor β1 in the inner ear of the human fetus
Authors: Chen Xiangdong , Dong Mingmin , Dong Xuelui , Dong Minsheng
Published: October 2005

The expression and localization of bFGF and TGF-β1 in the inner ear of ten 10-29 week human fetuses were studied using the LSAB immunohistochemical technique referred to the method of Shi. The bFGF was strongly expressed in Corti’s organs, vestibule macula and ampullar crista. The color of the upper cells of Corti’s anlage was a little deeper than that of the basal cells. The hair cells and supporting cells in the matured Corti’s organ continued to have distinct expression. Sites such as spiral ganglion (SG) neurons, stria vascularis (SV) had positive expression. Some cartilaginous cells and osseous cells in the developing osseous labyrinthus had light expression. TGF-β1 was expressed in the epithelium of Corti’s organ, vestibule macula, and ampullar crista. Hair cell color had a tendency to become weak with the maturity of the sensory epithelium, SG neurons, SV and spiral prominence (SP) had the TGF-β1 expression too. The results indicated that bFGF and TGF-β1 had participated in the course of inner ear development. The bFGF is a strong mitogen. However in this experiment it was found that it had the effect to promote differentiation. TGF-β1 had double effect of promotion and inhibition on the proliferation and differentiation of cells. The expression of bFGF and TGF-β1 in SG neurons indicated that both factors had participated in the development of the nervous system of the inner ear.
Tuning in primary saccular afferents of the toadfish (Opsanus tau) revealed by REVCOR analysis
Authors: Fay R. R. , Edds-Walton P. , Highstein S.
Published: October 2005

Male toadfish vocalize (e.g., the ‘‘boatwhistle’’ -f0 from 100 to 250 Hz) to attract females during mating season. This experiment investigates whether the frequency response of primary saccular afferents is adequate to encode these vocalizations. Whole-body oscillation (0.1–300 nm displacements) was delivered by a three-dimensional shaker system driven by noise with a flat spectrum between 50 and 1000 Hz. Stimuli were presented along three orthogonal axes (side–side, front–back, and up–down). Spikes were time stamped with 0.1-ms resolution. REVCOR functions were averaged accelerometer recordings of the noise triggered by spikes for 51.2 ms before and after each spike. This estimated the impulse response of the linear filtering that preceded spike generation. An FFT of the impulse responses estimated an afferent’s filter shape. Impulse responses obtained were damped oscillations having a major peak indicating the magnitude and direction of excitatory acceleration. In general, filter shapes are low-pass within the frequency range investigated with corner frequencies varying between 120 and 350 Hz. These and other data [P. Edds-Walton and R. Fay, Biol. Bull.189 , 211–212 (1995)] show that the saccule’s response to acoustic particle motion can account for the reception of ‘‘boatwhistle’’ vocalizations. [Work supported by the NIH, NIDCD.]
Single unit recordings from directionally sensitive cells of the torus semicircularis of the goldfish (Carassius auratus)
Authors: Ma Diana , Fay Richard R.
Published: October 2005

This experiment characterized the directional responses of cells of the auditory midbrain of goldfish. Single units in the midbrain were characterized by their response to whole-body vibratory stimuli of various directions and amplitudes. The animal’s head was fixed in a rigid dish that was accelerated using two orthogonal pairs of shakers in the horizontal plane, and one shaker oriented vertically. Three orthogonally oriented accelerometers resolved the dish’s motion in azimuth and elevation. Computer-synthesized sinusoids activated the shaker channels with relative phases and amplitudes required to produce linear motion along 12 axes equally spaced over 180° in each of the horizontal, midsagittal, and frontal planes. Spike rate was plotted as a function of stimulation axis angle in both Cartesian and polar coordinates. Most cells showed strongly directional response patterns with a primarily vertical orientation. Response patterns were classified with respect to the symmetry, sharpness, and the amplitude dependence of these directional features. Models were developed to give possible explanations for the observed responses. Many of the response shapes can be explained by combinations of simple peripheral effects (threshold and saturation) and inhibitory interaction. [Work supported by a Program Project Grant (NIH, NIDCD) to the Parmly Hearing Institute.]
Chemistry of the hamster cochlear nucleus
Authors: Godfrey Donald A. , Mikesell Nikki L. , Fulcomer Andrea B. , Godfrey Timothy G. , Kaltenbach James A.
Published: October 2005

The chemistry of the hamster cochlear nucleus was studied by microdissection of freeze-dried sections combined with microassays. Tissue sample dry weights usually were between 0.1 and 0.5 μg. Amino acid concentrations were measured by high-performance liquid chromatography (HPLC) and activities of enzymes of acetylcholine metabolism by radiochemical procedures. Expressed per dry weight of sample, aspartate concentration was higher in the ventral than in the dorsal cochlear nucleus, whereas glutamate, glutamine, y-aminobutyrate (GABA), taurine, glycine, serine, and alanine concentrations were highest in the molecular layer of the dorsal cochlear nucleus. In general, amino acid concentrations were similar to those in cats and rats. Activity of choline acetyltransferase, the enzyme of synthesis for acetylcholine, was relatively high in granular regions and low in the molecular layer of the dorsal cochlear nucleus, a pattern similar to what has been found in rats and cats. The activity in the anteroventral cochlear nucleus was low compared to the value in rats, however. Activity of acetylcholinesterase, the enzyme of degradation for acetylcholine, was generally lower than in rats and cats, especially in granular regions and the molecular and fusiform soma layers of the dorsal cochlear nucleus. [Work supported by the American Tinnitus Association.]
On the sound environment for the right and left human hemispheric tasks
Authors: Mouri Kiminori , Ando Yoichi
Published: October 2005

Considering the human cerebral-hemispheric specialization, specific tasks can be performed under a certain sound environment without any interference effects. Under other conditions, strong interference effects occur between the task performance and the sound environment [Y. Ando, Acustica64 , 110–116 (1987)]. For example, the left hemispheric tasks can be interrupted by the verbal sound, and the right hemispheric tasks can be interrupted by nonverbal sounds like noise. In order to identify hemispheric specialization for some important tasks, which are often performed in an office, the effective durations of the autocorrelation function of the α-brain wave range from both left and right hemispheres are analyzed here. Results may be utilized to design the recommended sound environment for each task in the office.
Electrical stimulation of the cochlea improved word recognition under noises in tinnitus patients
Authors: Matsushima Jun-Ichi , Sakai Noboru , Miyoshi Shigeki , Uemi Norihiro , Sakajiri Masatsugu , Ifukube Tohru , Kamada Tsutomu
Published: October 2005

Hearing impaired patients sometimes complain of poor understanding of speech under noise. It was shown that electrical stimulation of the cochlea in tinnitus patients improved their word recognition. The objective of this study is to show that electrical stimulation of the cochlea in tinnitus patients improved their word recognition under noises. Twenty four-segment sentences mixed with multiple talk recorded on a CD were delivered to tested ears at a comfortable level via a headphone. The signal-to-noise ratios were 0-, 5-, and 10-dB. Patients were requested to repeat what they heard. Improved score was calculated at each segment in 20 sentences. A sinusoidal wave of 10 kHz at the intensity of 200 μA was delivered to the cochlea. The stimulating Pt–Ir electrode was placed on the middle ear and the return electrode, which is made of a plate electrode for ECG, was placed on the skin behind the ear. Alternatively, an external electrical stimulation was applied to patients. Improved word recognition was shown in most patients with tinnitus relief following electrical stimulation of the cochlea, showing that electrical stimulation of the cochlea improved S/N in the auditory system.
The optimization of an implantable tinnitus suppressor using an extracochlear stimulation method
Authors: Sakajiri Masatsugu , Miyoshi Shigeki , Ifukube Tohru , Matsushima Junichi
Published: October 2005

Tinnitus patients have been treated using simple electrical stimulation to the promontory of the cochlea. With this treatment about 60% of the patients reported that the tinnitus was suppressed. This method, however, suppresses tinnitus for only a short period of time, so many tinnitus sufferers have long been waiting for an implant system which will be able to suppress tinnitus whenever it appears. For this reason, an implantable tinnitus suppressor has been developed using extracochlear stimulation technologies. The implantable tinnitus suppressor consists of a wave generator, a primary coil set behind the auricle, a secondary chip coil implanted in the mastoid cavity, a Pt–Ir electrode covered with a polyvinyl alcoholic gel placed on the promontory, and a return electrode fixed to the subcutaneous tissue near the mastoid tip. The wave generated by the electric circuit of the wave generator is sent to the primary coil. The current that is induced in the secondary coil by using electromagnetic coupling is transmitted to the Pt–Ir electrode. The main purpose of this study is to optimize the function of this implantable tinnitus suppressor. [Work supported by the Ministry of Education, Science and Culture, Grant-in-Aid for Scientific Research (A)(2) No. 06402067.]
A robust speech recognition algorithm
Authors: Nakata Kazuo , Matsumoto Khoji
Published: October 2005

The very important point of applications of speech recognition in a real field is its robustness against noise disturbances. A new robust algorithm is proposed, which has the following three features: (1) It uses only one microphone for input and noise-free speech as references; (2) it simulates a human hearing process in the following three functions: frequency analysis by the critical bandwidth, physical to sensory level transformation, and lateral inhibition; and (3) the speech is enhanced by passing a noisy one to a filter made of the spectrum envelope derived by the linear predictive analysis of the noisy speech itself. Outputs of the analyzing filter are vector-quantized and recognized by VQ and HMM composed of the noise-free reference speech processed in the same way as the noisy ones. The method can improve recognition scores from 15% to nearly 30% in the range of SNR of 15–10 dB for additive white random noise, compared to those of the noisy speech by HMM of noise-free speech.
Effects of noise masking on the identification of fricative noise frequency
Authors: Schoenburg Diane L. , Mann Virginia A.
Published: October 2005

This study examined how signal degradation affects the use of natural formant transitions and following vowel quality (rounded versus unrounded) in the identification of a fricative noise continuum varying from /s/ to /sh/. The stimuli were made by combining a synthetic /s/-/sh/ continuum with vocalic segments extracted from natural tokens of /sa/, /sha/, /su/, and /shu/. These were presented in a clear condition and with the vocalic segment masked by signal-correlated noise at 1.8 dB SNR. Listeners labeled the fricative consonant that began each stimulus as ‘‘s’’ or ‘‘sh.’’ In general, the results replicated previous findings that /s/ transitions and the presence of /u/ gave rise to more ‘‘s’’ responses than /sh/ transitions or the vowel /a/ [V. A. Mann and B. H. Repp, Percept. Psychophys. 28, 213–228 (1980)]. It was further found that noise masking significantly decreased the effect of the transitions, but not the effect of vowel quality. Discussion of this interaction considers the effect of noise on (1) the salience of brief versus sustained speech cues and (2) the relative weighting of weak versus strong speech cues [Gordon et al., Cognit. Psychol. 25, 1–42 (1993)].
Recognition of first words of isolated spoken word pairs with ambiguous word boundaries
Authors: Swiergosz Matthew J.
Published: October 2005

A series of experiments was conducted to test whether acoustic-phonetic information elucidates word boundaries. Word pairs were presented to subjects in gated format, where discrimination of the first word was marked by the first or last phoneme of the second word (e.g., RAN–PUMP or RAN–DUMP, respectively). Subjects were asked to provide either a written perceptual identification or use a numeric keypad to indicate the number of words at each gate (lexical enumeration). Results were consistent across response measures; perceptual identification of first words and lexical enumeration required less acoustic-phonetic information when first words were marked early in second words. In addition, the interval of silence between the word pairs was manipulated in two experiments to ascertain whether interword pause (30, 90, 150, or 210 ms) would result in differential perceptual identification and lexical enumeration. The interval of silence was not a reliable cue for first-word recognition or word counting and, thus is at odds with theories of auditory word recognition that incorporate pause as a factor influencing the activation levels of lexical candidates.
Phonetically balanced cepstrum mean normalization for telephone speech
Authors: Morishima Masatoshi , Isobe Toshihiro , Koizumi Nubuo
Published: October 2005

Cepstrum mean normalization is an effective method for recognizing distorted telephone speech. This method compensates for the difference of bias on cepstrum coefficients (CC) between training data and test data by subtracting the mean value of the CC calculated from a certain amount of given speech data. Such adaptation data are not phonetically balanced, which makes it difficult to get an accurate mean value for the CC. In this paper, a new approach to resolve this problem is proposed. Before recognizing speech, not only the mean value of the adaptation data itself must be calculated, but also the mean value from Gaussian distribution of continuous density HMMs must be calculated, whose phonemes appear in the adaptation data. When recognition occurs, the difference of these two mean values is subtracted from speech data. This proposed method using various telephone speech data has been investigated, i.e., speech data from an ordinal analog telephone, a code-less handset telephone, and a digital cellular phone (Japanese full-rate digital phone based on VSELP), which was recorded through the public switched telephone network. These experimental results show an advantage of the new method.
Robust word recognition in an automobile by using cepstral normalization and an improved acoustic model
Authors: Ohshima Y. , Sakamoto M.
Published: October 2005

Multiple approaches are needed to achieve robustness in speech recognition in an automobile due to various noise sources, relatively enclosed cabin acoustics, and the manner of speech of the driver under stress. A HMM-based word recognizer was developed for automobile applications, by integrating cepstral normalization into the front end and by training model parameters in multiple conditions according to a few driving scenarios mainly in city traffic. Speaker-dependent models were built and evaluated for a male and a female speaker using samples recorded in an automobile, and conducted a road test in city traffic. It was found that the system works well with several percentage points degradation in recognition accuracy compared with the laboratory experiments. The difference was attributed to the errors in word boundary detection.
Affine transformation-based feature normalization for speaker-independent speech recognition
Authors: Luo Ping , Ozeki Kazuhiko
Published: October 2005

In statistical speech recognition, speaker-independent models are usually trained by using speech samples from a large number of speakers. Those models have a problem in that they have wider feature distributions and hence greater overlaps between different phones than adequately trained speaker-dependent models. In order to cope with the interspeaker variability, a method of speech feature normalization based on affine transformation has been presented [P. Luo and K. Ozeki, Tech. Rep. of IEICE, SP96-10 (1996)]. Prior to HMM training, feature vectors of each speaker are mapped to those of a reference speaker by an affine transformation estimated with a small amount of training data. The transformation, which is phone independent and speaker dependent, is also applied to feature vectors of unknown speakers in the recognition stage. It has been shown experimentally that this method is effective in reducing interspeaker variations in the cepstral domain. In this paper, further discussions about the performance and limitations of the method are given within the framework of continuous HMM. Practical issues related to the selection of an appropriate reference speaker will also be discussed.
Noisy speech recognition using HMM-based cepstral parameter generation and compensation
Authors: Kobayashi Takao , Masuko Takashi , Imai Satoshi , Tokuda Keiichi
Published: October 2005

This paper proposes a technique for adapting continuous density HMM speech recognition systems trained on clean speech data to make it robust to additive noise. The approach is based on cepstral parameter generation from speech and noise HMMs and parameter compensation using generated parameters of speech and noise. For clean speech and noise HMMs including cepstral and dynamic parameters, cepstral parameter vector sequences are generated in such a way that the probability of observing the parameter vector sequence from the given HMM is maximized using dynamic parameters. Then the generated cepstral vector sequences of speech and noise are combined to yield a noisy speech cepstral vector sequence. Compensated means of cepstral and dynamic parameters for noisy speech HMM are obtained from statistics of the noisy speech parameter sequences. Variances for noisy speech HMM are also estimated using the relationship between the clean and noisy speech parameter sequences. As a result, the technique provides cepstral and dynamic parameter compensation for noisy speech HMM. Moreover, it does not require the lognormal approximation used in cepstral parameter compensation based on the parallel model combination technique. Word recognition experiments based on phoneme HMMs show the effectiveness of the proposed technique.
Isolated word recognition in a noisy environment
Authors: Suzuki Tadashi , Abe Yoshiharu , Nakajima Kunio
Published: October 2005

In noisy environment, performance of speech recognition systems trained in quiet environment is degraded. One of the reasons is acoustic phonetic modification caused by the Lombard effect, another is noise contamination of speech signal. This paper presents a new method for isolated word recognition in noisy environment. The method is based on two techniques. One of them is based on variability models for acoustic phonetic modification in Lombard speech, and another is to estimate additive noise spectrum frame by frame. The acoustic phonetic variability models represent the spectral difference between normal speech and Lombard speech. Each model is comprised of a nonlinear warping function on spectral domain and two spectral filters. The warping function represents formant shift. Two filters do the changes of formant bandwidth and of spectral tilt. The noise estimation is executed for each frame of noisy input with noise models and speech models made with clean speech data. These techniques were applied to speaker-dependent word recognition based on continuous density HMMs of subphoneme. Experimental evaluations were executed with the noisy Lombard speech data of 100 isolated words. From the experiments, the effectiveness of the proposed method has been confirmed.
Beyond a ‘‘short-term’’ analysis of speech
Authors: Hermansky Hynek
Published: October 2005

A short-term analysis of speech, inherited from speech coding, describes a time-varying speech signal as a sequence of vectors, each vector reflecting properties of a relatively short (10–20 ms) segment of the signal. Each analysis vector provides a single sample from the underlying time-varying speech process. Typically, the whole feature vector is used as one entity for a subsequent processing. For use in ASR, the short-term analysis may have some deficiencies. (1) From a single vector alone, there is no way to differentiate between components with different rates of change. (2) When one or a few components of the vector get corrupted, the subsequent ASR result may be corrupted. This is inconsistent with human speech perception where (1) many auditory phenomena seem to span at least a length of a syllable, and that (2) decoding of a linguistic message does not have to be severely impaired by a partial degradation of the speech signal. Emerging ASR techniques are discussed which attempt to alleviate the above-mentioned deficiencies of the short-term analysis by (1) doing temporal processing on trajectories of short-term features of speech, and (2) selectively merging information from several subsets of the short-term representation of speech. [Work supported by DoD (MDA-904-94-C-6169) and NSF/ARPA (IRI-9314959).]
Evaluation of a rapid environment adaptation algorithm in adverse environments
Authors: Hattori Hiroaki , Takagi Keizaburo , Watanabe Takao
Published: October 2005

This paper reports results for a rapid environment adaptation algorithm in adverse car environments. It is known that the difference between training and testing environments degrades speech recognition performance. This degradation becomes serious especially in applications such as telephone speech recognition and speech recognition inside a running vehicle, where the testing environment may drastically change. To solve this problem, a rapid environmental adaptation method (hereafter referred to as REALISE) is proposed and its performance is measured in telephone speech recognition. REALISE estimates the differences in multiplicative and additional noise in the spectral domain between the training and testing environments and uses them to adapt acoustic features of reference patterns to the testing environment. Utterances in a car were also recorded to investigate the performance of REALISE under more severe conditions. The testing data were 100 city names uttered by three males and three females under three running conditions (idling, 50 kph, and 100 kph). Whereas the average recognition rate achieved by a conventional spectral subtraction technique was 75.6%, REALISE achieved 85.7%. This result proves the effectiveness of REALISE in a very noisy vehicle environment.
Hands-free speech recognition by a microphone array and HMM composition
Authors: Nakamura Satoshi , Yamada Takesi , Takiguchi Tetsuya , Shikano Kiyohiro
Published: October 2005

Hands-free speech interface is one of the final goals of human–machine interface. This paper introduces two methods for distant talking speech recognition in noisy and reverberant rooms. The first method is speech recognition using a microphone array. The microphone array enables one to enhance a speech signal using spatial phase differences even in environments where unstationary noises exist. The proposed method is composed of two modules: (1) a high SNR signal retrieval by a delay-and-sum beamformer and (2) localization and trace of the speaker’s direction by extracting a signal power and pitch harmonics. The second method is speech recognition based on the HMM composition. The proposed HMM composition is obtained by extending the HMM composition method of an additive noise to that of the convolutional acoustical transfer function. The HMMs are prepared beforehand for clean speech, noise, and acoustical transfer function. Then the HMM composition is conducted twice for a speech HMM and an acoustical transfer function HMM in the cepstrum domain and for the distorted speech HMMs and noise HMM in a linear spectral domain. The speaker-dependent/independent word recognition experiments using tied-mixture monophone HMMs are carried out and have clarified the effectiveness of the proposed methods. Furthermore, an effective coupling of these methods is also discussed.
Model-based unsupervised instantaneous speaker adaptation
Authors: Furui Sadaoki , Matsui Tomoko
Published: October 2005

Unsupervised instantaneous adaptation, which uses the input utterance itself for adaptation, is the ideal speaker adaptation method for speech recognition, and is expected to be very useful for a wide range of applications. Since voice individuality is phoneme dependent, speaker adaptation must be performed model dependently. However, it is impossible to obtain a complete model sequence, that is, what is spoken, for each input utterance, especially for speakers who have many recognition errors when using speaker-independent models. Therefore, how to perform model-dependent adaptation without knowing the correct model sequence is a crucial issue. If all possible model sequences were hypothesized and used for adaptation, the amount of calculation would become enormous. This paper proposes a new adaptation method, in which N-best hypotheses are created by applying speaker-independent phone models to each input utterance, and speaker adaptation based on a constrained MAP estimation technique is then applied to each hypothesis. Using this method, the likelihood of a correct hypothesis existing in a low rank with speaker-independent models rises, and, as a result, recognition accuracy increases. Experimental results for several continuous speech recognition tasks show that recognition accuracy is increased by this method, even for speakers who have very low accuracy with speaker-independent models.
Acoustical imaging from the perspective of the acoustical imaging symposia
Authors: Powers John P.
Published: October 2005

The Acoustical Imaging Symposium series began in 1968. The 23rd meeting of the series will be held in Boston in April 1997. The symposium has focused on the engineering and physical-science aspects of acoustic imaging. This presentation will survey a selected number of topics presented at these symposia, including acoustical holography techniques, phase-only imaging, imaging with backward and forward propagation techniques, acoustic microscopy imaging, underwater imaging, seismic imaging, and other imaging techniques.
Medical ultrasonic imaging: Present status and future prospects
Authors: Jones Joie P.
Published: October 2005

For over 30 years, ultrasonic imaging, based on the same pulse-echo principles as radar and sonar, has played an important role in diagnostic medicine. In many ways ultrasound is an ideal diagnostic tool—noninvasive, nontraumatic, capable of producing real-time images, and as all available data indicate, apparently safe at the acoustical intensities and duty cycles encountered in existing diagnostic equipment. Here the objective is to provide a snapshot of the present state of medical ultrasound and to assess the future prospects of this technology from ongoing research activities. Innovations such as Doppler analysis and cross-correlation techniques to measure flow, phased array transducers, contrast agents, tissue characterization, and, in particular, quantitative imaging methods based on unique reconstruction techniques will, it is believed, greatly increase the clinical utility of ultrasound as well as maintain a continued high rate of growth in the marketplace.
Tomographic reconstruction techniques for pulse-echo imaging
Authors: Akiyama I.
Published: October 2005

Imaging is possible based on tomographic reconstruction techniques such as computed tomography, although ultrasonic pulse echo imaging is based on focusing techniques. It is possible to obtain three-dimensional imaging data by the tomographic method. When divergent waves are insonified toward a target by a part of a spherical surface, backscattered waves which are generated at the target are received by the transducer, and the echo signal on a delayed time is expressed as the integral of scattered waves at the surface of the radius proportional to the delay. If the transducer has both convex and concave parts of the spherical surface, both divergent waves and convergent waves can be generated. These waves are focused onto the arc on the spherical surface; thus the amplitude at the delay of the received echoes is expressed as the product of the backscatter coefficient on the arc and the diffraction term. Those values are equivalent to the projection data as the tomographic reconstruction technique. Therefore the distributions of the backscatter coefficient on the spherical surface are reconstructed from the projection of the echoes. In this study the effects of the angle dependency of the amplitude on the spherical surface due to the diffraction term are discussed and then the method for correction is described.
Ultrasonic mapping of blood flow through the microvasculature
Authors: Ferrarra K.
Published: October 2005

An overview of the use of high-frequency ultrasound to map blood flow through the microvasculature will be presented. The correlation of scattered echoes from blood at frequencies from 20 to 100 MHz are shown to demonstrate a smaller probability of error than the correlation at a lower transducer frequency. Experimental three-dimensional (3-D) maps of the velocity profile within vessels with a diameter of 100 μm or less will be presented. Blood velocity within vessels as small as 40 μm can be mapped using a transducer with a nominal center frequency of 50 MHz. Estimation of flow within these small vessels at the appropriate low velocities is challenging due to the small signal levels, the effect of cardiac and respiratory motion, and the unknown rate of fluctuation of the received signal from the small number of red blood cells contained within these sample volumes. Reliable in vivo estimation of the velocity within the microvasculature requires realignment of the signal from a single line of sight, and this method is presented. Recognition of 3-D continuity of a vessel tree is required to eliminate spurious noise. The rate of fluctuation in the received signal from small blood vessels is evaluated as a function of the sample volume and vessel size. A minimum time window for reliable detection of flow through a small vessel is determined. Finally, the use of acoustic contrast agents to improve the visualization of flow within small vessels will be introduced.
Acoustic-optic hybrid wave filter development
Authors: Matthews Anthony D.
Published: October 2005

Digital filters are developed for mitigation of wave-induced image degradation in laser vibrometer data taken from the water surface. The target is ensonified by a submerged acoustic projector operated near 10 kHz. The filters are evaluated for effectiveness in restoration of the image. Unbeamformed (raw) target data are acquired from a surface with waves of variable height. Images are formed with and without the filters to determine restoration performance. [Work sponsored by Coastal Systems Station under Internal Research funding.]
High-frequency scattering from a target in proximity to a randomly rough interface
Authors: Bishop Garner C. , Smith Judy
Published: October 2005

A T-matrix formalism for free-field target scattering and a Helmholtz–Kirchhoff integral equation in the Kirchhoff approximation for rough interface scattering are used to calculate high-frequency plane-wave scattering from a stationary target in proximity to an interface with random surface roughness. Scattering from targets located above and below the interface are considered: When the target is located above the interface, it is assumed that the scattered pressure field is simply the superposition of the target and rough interface scattered fields. When the target is located below the interface, it is assumed that target-interface multiple scattering can be ignored and that the scattered pressure field above the interface is given by the superposition of the field scattered from the interface and the field scattered from the target and transmitted through the interface when the field incident on the target is the transmitted plane-wave field. Numerical calculations indicate some of the effects of surface roughness and target physical and geometric parameters on the ability to ‘‘see’’ the target.
A composite model for rough bottom scattering
Authors: Dorfman Yevgeniy Y. , Dyer Ira
Published: October 2005

To be valid and accurate, any model for mid-frequency bistatic scattering must account for the different scales of roughness found on the bottom. A composite scattering model was first introduced by Kuryanov. An extension for this model, similar to the one by Kuryanov, was applied to data acquired during the 1993 ARSRP cruise. In this model, the scattering surface is represented by a random surface composed of facets. Each facet is parametrized by a mean slope to account for large scale roughness and a perturbation to account for small scale roughness. First, partial coherence of the field over the facet surface is used to determine the contribution of each individual facet. Then, the scattered field is calculated via incoherent addition of the contributions from all facets. The choice of a Gaussian distribution for the mean slope of the facets results in a good match between the large scale roughness spectrum and the empirical power law. A simplified model for scattering from the individual facets assumes that all scatter is directed into the specular direction. This allows a completely analytical solution and is in reasonable agreement with experimental data. [Work supported by ONR.]
Probability distribution functions for shallow water acoustic bottom scattering at high frequency
Authors: Farwell Robert W. , Meredith Roger W.
Published: October 2005

When designing signal processors to discriminate against underwater sound background noise and scattering, it is important to know the statistics of the unwanted signal disturbances. A study was undertaken for the Naval Research Laboratory, Stennis Space Center, Mississippi to review the literature reporting on a number of bottom scattering experiments that were conducted during the past 15–20 years. The scope of the investigation is limited to high frequencies (10 to 200 kHz) and shallow water of depths less than 100 m. Many of these tests were performed by the Naval Research Laboratory, Stennis Space Center; the Applied Research Laboratories, University of Texas at Austin; and the Applied Physics Laboratory, University of Washington. Other experiments reviewed were performed by Thomson-Sintra ASM, France and by the University of Wisconsin. Each experiment will be briefly described, and the statistical results discussed and summarized for various parameters, such as frequency, beamwidth, pulse length, grazing angle, range, and environmental factors. The statistic that was primarily examined was the probability of false alarm. The PFAs were found to follow a number of probability distribution functions including Gaussian, Rayleigh, log-normal, etc. The PFAs were quantified to systematically determine any trends with the parameters.
Fast speaker-independent acoustic modeling and speaker adaptation
Authors: Matsunaga Shoichi , Tonomura Masahiro , Kosaka Tetsuo
Published: October 2005

A fast acoustic modeling method for speaker-independent speech recognition and a speaker-adaptation method, which is effective even with only a small amount of speech data, is described. The speaker-independent phoneme models are generated by composing representative speaker-dependent phoneme models, which are selected from among all speaker-dependent models by clustering the models without Baum–Welch parameter re-estimation. This generation method greatly reduces the computational cost needed to create the speaker-independent HMMs to much less than that of the Baum–Welch method, i.e., by a factor between approximately 1/20 and 1/50. This speaker adaptation algorithm unifies two conventional techniques, i.e., a maximum a posteriori (MAP) estimation and transfer vector field smoothing.A priori knowledge from initial models is statistically combined with a posteriori knowledge derived from the adaptation data to complement the sparse adaptation data. Transfer vector smoothing is used to interpolate the untrained parameters. Furthermore, in order to obtain a suitablea priori knowledge concerning speaker characteristics, a speaker-clustering model, generated by using speech of a selected speaker cluster, is used as an initial model. The cluster selection is performed with a tree-structured speaker clustering technique that determines the number of speakers and the members in the cluster based on speaker similarity. [Work supported by ART-ITL.]
Measurements and modeling of low-frequency bottom-scattering strengths from the Scotian Continental Rise
Authors: Gauss Roger C. , Holland Charles , Fialkowski Joseph M. , Neumann Peter
Published: October 2005

A series of low-frequency (190–310 Hz), bottom-backscattering strength measurements were made on the Scotian Continental Rise in August 1993 using combinations of short-duration cw and HFM signals. Grazing angles realized were between 5 and 30 deg. A characteristic of many sites were strong (on the order of 10 dB) oscillations with a grazing angle superimposed on generally Mackenzian backscatter. The oscillatory behavior is due to caustics within the sediment, i.e., a propagation effect. Previous modeling of this general area [P. D. Mourad and D. R. Jackson, J. Acoust. Soc. Am. 94, 344–358 (1993)] predicted such an oscillatory behavior but only for unrealistic sediment sound–speed profiles. An alternative bottom-scattering model is proposed with the goal of predicting the observed scatter using a more realistic geoacoustic model. Measurement/model comparisons will be presented for several representative sites. [Work supported by SPAWAR, NRL, and ONR.]
Acoustic radiation or scattering: A multifrequency analysis by boundary element method
Authors: Vanhille Christian , Lavie Antoine
Published: October 2005

In the field of underwater acoustics, the boundary element method is usually used to model acoustic radiation or scattering from an immersed structure impinged by an acoustic wave. The EQI boundary element code uses Helmholtz integral equation. The Sommerfeld radiation condition is implicitly included in the formulation, and the pressure field in the surrounding fluid is evaluated from the mesh restricted to the surface of the body. At some frequencies called irregular frequencies, this integral formulation presents an infinite number of solutions. This problem is solved with the help of the Jones method: The integral equations system is overdetermined by null-field equations. For a multifrequency analysis, integral equations matrix assembling is very computation time consuming. A new boundary element algorithm based upon a frequency interpolation is proposed. It is adapted to quadratic isoparametric discretization for any geometry. This technique is original for two reasons: the automation of its utilization between 0 Hz and the top frequency of the spectrum; its adaptation to axisymmetrical problems by decomposition of circular integration surfaces. This algorithm allows an important saving of computation time which increases with the size of the problem without loss of precision.
Scattering from an axisymmetric structure insonified at any incidence using a coupled finite-element/boundary element method
Authors: Lavie Antoine , Dubus Bertrand
Published: October 2005

In the field of the scattering of acoustic waves from structures immersed in an infinite fluid medium, a coupling between the EQI boundary element code and the ATILA finite-element code is available. The boundary element method used is the Helmholtz integral equation. At some frequencies, called irregular frequencies, this integral formulation presents an infinite number of solutions. This problem is solved with the help of the Jones method: The integral equations system is overdetermined by null-field equations. Many structures are axisymmetric. So, it is attractive to develop a modeling that is able to keep the advantages of the axis of symmetry and to take into account an incident wave of any direction. This numerical tool is performed using a harmonic cylindrical decomposition. In this case, the mesh is restricted to the line of the structure included in the meridian plane. Compared with three-dimensional modeling, this new possibility provides the following advantages: either a reduction of the computation time, the memory size, and the disk storage space and an improvement of the numerical behavior; or an increase of the maximum frequency. Comparison between numerical results and experimental data is presented.