bluuebunny/crossref_metadata_2025 huggingface dataset:
Inversion of wave field data by simulated annealing
Authors: Basu Atanu , Frazer L. Neil
Published: October 2005

Simulated annealing is a new Monte-Carlo optimization method that (if properly used) does not get stuck in local minima of the objective function. This feature is attractive for the inversion of wave field data where the lack of low-frequency energy makes cycle skipping a problem. To gain experience with the method, it was applied to a simple problem: determining a sound-speed profile c(z) from recordings of two shots at unequal distances from a vertical array. For a nearly stratified ocean this problem is equivalent to a single shot, recorded on two vertical arrays. A profile c(z), propagating data from the near array to the far array, is assumed and then the propagated data are compared with the actual data at the second array. If the propagated and actual data agree then c(z) is a good profile. If the sound-speed profile has, say, 15 parameters with, say, 10 possible values for each parameter then the number of possible profiles is 1015. In tests with synthetic data, simulated annealing gave a nearly correct profile with a few hours of iteration (on a desk-side workstation). Experiments are currently going on with different annealing schedules. Results so far suggest that critical temperature (Tcr) is difficult to determine, and so annealing strategies which do not require a prior estimate of Tcr are more efficient than those which do. [Work supported by ONR.]
Numerical experiments with time-domain localization
Authors: Pecholcs Peter I. , Frazer L. Neil
Published: October 2005

Time-domain localization is of interest because: (a) ambiguity surfaces constructed using frequency-domain methods have large sidelobes unless many frequencies are used; (b) time-domain Green's functions can be rapidly constructed using ray theory; and (c) in theory, time-domain methods can be used to directly determine the velocity, as well as the position, of the source. Here some natural time-domain algorithms are applied to a homogeneous waveguide with a soft top and hard bottom. Clay's matched filter approach, and a technique known in the oil industry as reverse time migration, are attractive because neither requires a knowledge of the source waveform. If the source waveform is available (even though its time origin is not) then one can localize using a Bartlett, or a maximum likelihood, formula directly in the time domain. A concurrent search is needed to determine the best time origin for the source waveform. If the source waveform is available, then one can localize in velocity space by use of a ray theoretical Green's function in which the source waveforms of different rays are compressed or expanded according to the component of velocity parallel to each ray at a candidate source location. [Work supported by ONR.]
An application of the broken mirror approximation to modeling bistatic reverberation in the ocean
Authors: Weinberg Henry
Published: October 2005

The Navy Interim Surface Ship Model (NISSM II), a monostatic reverberation model, was applied to data measured by the Canadian Defence Research Establishment Pacific (DREP). NISSM II could not explain the behavior of certain discrete arrivals. The source of the discrepancy was eventually attributed to the bistatic nature of the experiment. In 1986, the Generic Bistatic Reverberation Model was developed and applied to the data without much success. Other modelers were already aware that direct transmitter-receiver arrivals, the counterpart of vertical fathometer returns in the monostatic case, could be significant in bistatic reverberation measurements. When these direct arrivals were added, it became obvious that the source of discrepancy between the model and the data had been isolated, but not resolved. The implementation of a “broken mirror” model provided the missing link for explaining the DREP data. This paper highlights the results of the bistatic investigation.
Electrochemical aspects of calcium ions in the stria vascularis
Authors: Ikeda Katsuhisa , Morizono Tetsuo
Published: October 2005

The dc potential and the Ca concentration of the stria vascularis were measured using double-barreled Ca-selective microelectrodes with fine tips. Healthy chinchillas were anesthetized with ketamine hydrochloride and artificial ventilation was maintained under conditions of muscular relaxation. The microelectrode was advanced through the spiral ligament by an autodrive manipulator. The dc potential from the spiral ligament and the Ca concentration (mean ± s.d.) are respectively as follows: 0 mV and 1.7 ± 0.3 mM in the spiral ligament (n = 11); 6.3 ± 7.8 mV and 2.4 ± 2.3 μM in basal cells (n = 10); 55.2 ± 5.8 mV and 0.68 ± 0.20 mM in the intrastrial space (n = 5); 72.9 ± 9.8 mV and 0.38 ± 0.28 μM in marginal cells (n = 11); 78.3 ± 2.3 mV and 17 ± 7 μM in endolymph (n = 11). The transport mechanism of calcium ions in relation to the localization of Ca-ATPase is discussed. [Work supported by NINCDS.]
Development of auditory function and cochlear morphology in the rat
Authors: Rybak L. P. , Whitworth C. , Scott V.
Published: October 2005

Auditory function was studied in the rat pup using cochlear potentials [endocochlear potential (EP) and compound action potential (CAP)] and auditory brainstem response (ABR). EP development followed a sigmoid curve. CAP and ABR threshold responses to dicks were detected at about 13 days of age and showed a progressive decrease in threshold until adultlike levels were attained at 30 days. Parallel studies of organ of Corti structural development were carried out using scanning electron microscopy (SEM). Interesting changes in the width of the organ of Corti and stereocilia configuration were observed. Correlation of onset of detectable auditory threshold and morphologic changes will be presented. [Work supported by DRF.]
Human hair cell-VIII nerve transmission deficit: A case study
Authors: Starr Arnold , McPherson David , Patterson Julie , Walker Cam
Published: October 2005

A 12-year-old girl with bilateral moderate pure tone hearing loss (30–50 dB) and profound speech comprehension impairment (0%–30% correct discrimination) was tested with auditory-evoked potentials. No neural components occurred in the brainstem (1–10 ms), mid (10–50 ms), and long-latency (50–250 ms) time domains even though the patient could “hear” the clicks. A cognitive P300 component was evoked during the discrimination of an infrequent tonal signal indicating normal stimulus classification function. Cochlear microphonic activity of normal amplitude was recorded from an ear-canal electrode. Thus, the absence of neural components of auditory-evoked potentials in the presence of preserved receptor (hair cell) function is interpreted as due to an alteration of transmission between hair cell and VIII nerve leading to a loss of synchrony of VIII nerve activity and an absence of evoked potentials. The finding that the patient's temporal discrimination abilities were severely impaired supports such an interpretation: binaural time or intensity cues could not be used for localization; binaural “beats” were absent; monaural resolution of two clicks was markedly slowed.
Neuroanatomical basis for cochlear-cochlear interaction
Authors: Thompson Glenn C. , Thompson Ann M.
Published: October 2005

Using neuroanatomical tract-tracing methods in guinea pigs, four separate pathways were observed that connect one cochlea with either the other cochlea or with itself all via brainstem olivocochlear neurons. Since each of these pathways involved a two-neuron circuit including the posterior ventral cochlear nucleus and its projection to either ipsilateral or contralateral olivocochlear neurons, the mediating structure exists for feedback control (to the same cochlea) or feedforward control (to the opposite cochlea) with only 3-ms delay. With this short latency, these pathways have the potential to modulate the sensitivity of either cochlea after the initial onset of sound, or to modulate the neural cues resulting from differences in interaural intensity and thus may be important in localizing long duration sounds. Whatever the function, these pathways constitute the most direct connection between the cochleas thus described. [Work supported by NIH and DRF.]
Tonotopic representation of the auditory cortex of the rat
Authors: Horikawa Junsei , Ito Susumu , Hosokawa Yutaka , Murata Keiichi
Published: October 2005

Representation of the best frequencies (BFs) was studied by a tungsten microelectrode method in the auditory cortex of Wistar-strain rats (200–350 g) anesthetized with Nembutal (40 mg/kg of initial and 5 mg/kg/h of supplement doses). BFs of 0.5–63 kHz were represented in the auditory cortex concentrically, with high BFs in the center and low BFs in the periphery. From the tonotopic representation, this area can be separated into two areas: the primary auditory area (AI) (posterior half), and the anterior area (A) (anterior half). However, the response latency distributions in these areas (6–14 ms were not significantly different. Other smaller areas with concentrations of high BFs were located anteroventrally (AV) and posterodorsally (PD) to the main area (AI and A). The response latencies of the neurons in AV and PD (12–24 ms) were significantly longer than those in the main area. Sixty-seven percent of the 33 tuning curves obtained from the neurons in the main area were closed (showing nonmonotonic intensity functions); 33% were open; 55% were multi-peaked; and 45% were single-peaked.
Frequency dependency on the I-V interval using high-pass noise-masking derived ABR's
Authors: Ponton C. W. , Eggermont J. J. , Coupland S. G. , Dom M.
Published: October 2005

In normal-hearing adults, there is little variation in the I-V interval as a function of stimulus intensity or the rate of stimulus presentation for the standard click-evoked auditory brainstem response (ABR). However, in a population of infants, Teas et al. [Hear. Res. 7, 19–54 (1982)] found that the I-V interval decreased as the center frequency (CF) of the filtered click decreased down to 1 kHz CF. The present investigation evaluated the duration of the I-V interval for frequency specific ABR's using the narrow-band responses generated by the high-pass, noise-masking procedure [Don and Eggermont, J. Acoust. Soc. Am. 63, 1084–1092 (1978)]. In an adult population, it was found that the I-V interval was constant for all derived bands down to 1.4 kHz CF. However, a significant increase in the I-V interval was found at both 0.7 and 0.2 kHz CF. Although an explanation for the I-V interval increase is not obvious, the possibility exists that a central masking mechanism similar to that suggested by Burkard and Hecox [J. Acoust. Soc. Am. 81, 1050–1063 may account for the findings of this study. [Work supported by the Alberta Heritage Foundation for Medical Research (AHFMR).]
Inhibition and altered cortical processing
Authors: Daly D. M. , Wada J. A.
Published: October 2005

When balance of inhibitions declines/fails, patients with seizures in auditory cortex report certain synthetic stops as nasals, “bleats,” and then undifferentiated buzzes. Effects of increased inhibition can appear with auditory cortex in volume surrounding disinhibited areas (even contralaterally): patients report same range of stimuli as clearly identical [Daly et al., J. Neurophysiol. 140(2), 141–162 (1972)]. A case is reported with seizures in anterior temporal cortex (bilateral loci; spikes/sharp waves) and post-ictally impaired comprehension with normal peripheral acuity. Protocol included 9 h of simultaneous EEG/video monitoring and auditory testing with prerecorded sets of [be]-[de]-[ge] and [ge]-[ye]. Patient's ability to respond appropriately remained remarkably consistent during localized, bilateral, and even generalized discharge in EEG. Instances of no-response occurred when stimulus was embedded in bilateral/generalized discharge (patient once noted “missing” the stimulus). Patient also experienced episodes (20–200 s) when [de] bounds for [be]-[de] and [de]-[ge] increased approximately 200 Hz, or when [ge] bound for [ge]-[ye] increased more than 30 ms (during one 15-min interval left ear [ge] boundary exceeded right by approximately 50 ms). Episodes can occur without changes in response time or EEG activation and thus likely arise through increased infield inhibition.
Functional-activation asymmetries in normal humans studied with quantitative EEG (QEEG): First tests in the CNS Project
Authors: Lauter Judith L.
Published: October 2005

Work with behaviorally defined asymmetries such as relative ear advantages has led to the CNS Project, designed to apply noninvasive techniques such as psychophysics, EPs, QEEG, MEG, MRI, and PET to study human brain responses during functional activation. Preliminary results of a QEEG test series involving both auditory and hand-movement conditions indicate that QEEG power asymmetry patterns reflect at least two types of asymmetry organization: (1) “side of space,” e.g., right-hand movement elicits a power asymmetry favoring the left hemisphere, and v.v; and (2) asymmetries based on “higher-level” principles of organization, e.g., coordination during bilateral hand movement, or differential activation based on the physical characteristics of test sounds. As with behavioral patterns of relative ear advantages, QEEG shows individual differences in detail but group agreements in overall patterns of response. It is believed that this is the first report of QEEG used for studying functional activation in healthy human subjects, and illustrates its potential usefulness for studying human neurophysiology. [Work supported in part by AFOSR.]
Transient radiation and scattering from fluid-loaded spherical and cylindrical shells
Authors: Stepanishen Peter R.
Published: October 2005

Transient radiation and scattering from fluid-loaded spherical and cylindrical shells has been the subject of numerous investigations during the past 30 years. A new time domain method of addressing these problems is presented here where the velocity field of the shell in fluid is represented as an in vacuo modal vector expansion with time-dependent coefficients. For the two-dimensional problems of present interest, sets of two coupled convolution integral equations are developed for the time-dependent modal velocity coefficients. These equations are readily solved by a “marching forward in time” method. The often used plane wave and DAA methods are shown to be special cases of the present formulation. Numerical results are presented for the velocity and pressure field response of a point-excited spherical shell and a plane-wave-excited cylindrical shell. These results will illustrate the transition from light to heavy fluid loading on the transient response of the shells. [Work supported by ONR.]
Sound radiation from an accelerated vibrating body
Authors: Wu Xiao-Feng , Akay Adnan , Ingard K. Uno
Published: October 2005

The problem of sound radiation from a moving body is reexamined with particular emphasis on vibrating objects with time-dependent velocity profiles. Specific examples of a rectilinearly accelerating sphere and a vibrating sphere in circular motion are considered. Numerical examples of radiation patterns are given. [Work supported by NSF and WSU Institute for Manufacturing Research.]
Modeling of the radiation and the transmission of a double stiffened finite plate
Authors: Nicolas J. , Laulagnet B.
Published: October 2005

The vibroacoustic behavior of a double finite plate with or without stiffeners and with or without absorbing material is modeled using the modal approach. The excitation of the first plate may be a point force or an acoustic plane wave. The analytical method relies on the following steps. (A) Find the Hamilton's functional of each finite plate including stiffeners effects. (B) Minimize the functional through the projection in the simply supported basis, such as W(x,y) =  ∑ m = 1ω  ∑ n = 1ω Amn sin (mπxlx) sin (nπyly). (C) write the equation of continuity at the vibroacoustic junctions. (D) Describe the coupling between the two plates using propagating waves in the sandwich medium which may be a double layer. (E) Finally, establish the matricial system that permits one to find the displacement amplitude of the second plate. The formulation allows the extraction of some key parameters such as quadratic velocity of each plate, radiated sound power, and transmission loss. Preliminary results for the classical situation will be presented and compared with the case of infinite double plates.
Coupled finite element/boundary element approach for acoustic radiation and scattering
Authors: Everstine Gordon C. , Henderson Francis M. , Schuetz Luise S.
Published: October 2005

A new computational capability is described for calculating the sound-pressure field radiated or scattered by a harmonically excited, submerged, arbitrary, 3-D elastic structure. This approach, called NASHUA, first calculates the fluid pressures and velocities on the structural surface by coupling a NASTRAN finite element model of the structure with a boundary element model (based on a discretized form of the Helmholtz surface integral equation) of the surrounding fluid. Farfield radiated pressures are then calculated from the surface solution using the Helmholtz exterior integral equation. The overall capability is very general, highly automated, and requires no independent specification of the fluid mesh. A new out-of-core block equation solver was written so that very large problems could be solved. The use of NASTRAN as the structural analyzer permits a variety of graphical displays of results, including computer animation of the dynamic response. The overall approach is illustrated and validated using known analytic solutions for submerged spherical shells subjected to both incident pressure as well as uniform and nonuniform applied mechanical loads.
Acoustic resonances of thin cylindrical shells: Plate waves and the resonance scattering theory
Authors: Talmant Maryline , Quentin G. , Rousselot J. L. , Subrahmanyam J. V. , Überall H.
Published: October 2005

An experimental and theoretical study of acoustic backscattering from thin cylindrical shells immersed in water is described. Previous studies showed no general agreement for acoustic scattering from thin elastic shells. In the low-frequency domain (ka < 150) and for shells such that the inner-outer radius b/a tends towards unity, two surface waves have been observed: a fast wave that exhibits a dispersion curve similar to that of the S0 plate mode, and a slow wave. This latter is observed in a limited frequency domain (outside the domain of the S0 wave), whose extent depends on the value of the ratio b/a for a given material. The previous studies interpreted this wave as the A0 plate mode, or as Stoneley-type mode. Presented here are relevant experimental results for duraluminum shells with b/a = 0.96. A comparison with theoretical calculations from the resonance scattering theory shows good agreement, as well as with the simple model of a solid plane plate. A theory of the physical properties of the slow wave, analyzed as a water-borne wave on a solid plate bounded by water on one side, and by air on the other has been developed.
Burst point location in stop consonants using backpropagation neural networks
Authors: Kitazawa Shigeyoshi , Fourati Mourad , Ichikawa Susumu
Published: October 2005

A three-layered neural network approach for burst point location is presented, which can be used to extract consonant segments in a speaker-independent continuous speech recognition system. By using neural networks trained with the backpropagation algorithm [Ruemlhart et al., Nature 323, 533–536 (1986)], nonlinearity is introduced into the articulatory event detection decision making. The system can detect the burst point location in French voiced stop consonants /b,d,g/. For the experiments, a neural network structure of 12–20 units in the hidden layer, 50 units in the input layer, and 1 unit in the output layer is used. The input patterns represent the time series values of the speech power transition. The network was trained in several steps initially using a smaller set of training data and then using larger sets. The results of the burst location detection were encouraging, especially for the syllables “ba,” “dou,” and “ga.” Generally, the detection rate for /b/ was two times better than that for /d/ and /g/. There was no remarkable difference between the detection rates in the training data and in the unknown data.
Improving the performance of backpropagation-trained vowel classifiers
Authors: De Haan Gregory R. , Egecioglu Omer , Wakita Hisashi J.
Published: October 2005

Classification experiments using nine steady-state vowels were performed to compare artificial neural networks trained via backpropagation and K-nearest neighbor (KNN) classifiers. Normalized critical-band filterbank outputs served as input patterns in all experiments. Initial experiments used prototypical feedforward networks [R. Lippmann, IEEE ASSP Mag. 4(2), 4–22 (1987)], with fully interconnected adjacent layers of units. Once the critical number of hidden units [D. J. Burr, J. Acoust. Soc. Am. Suppl. 1 83, S46 (1988)] was established for a given experiment, the networks compared favorably to KNN. Significantly, while networks with two hidden layers did better than networks with one hidden layer for (binary) front-back vowel distinctions, they performed worse for (nine-class) vowel classification. It appears that backpropagation may be particularly powerful for binary classification [e.g., R. P. Gorman and T. J. Sejnowski, Neural Networks 1, 75–89 (1988)]. Experiments were run comparing prototypical networks with partitioned networks, where each hidden unit was connected to only one of the output units. Preliminary results indicate that these partitioned networks, which in effect do nine independent binary classifications in parallel, outperform the prototypical networks, suggesting that the critical number limit on performance can be overcome in a straightforward manner.
Comparison of theoretical and experimental form functions for a viscoelastic coated infinite shell
Authors: Caille Gary W. , Jarzynski Jacek , Rogers Peter H.
Published: October 2005

The experimentally observed backscattered farfield form functions (normal incidence) for a simulated infinite cylindrical shell coated with (a) closed-cell nitrile rubber and (b) corprene are compared with the theoretically derived form functions. The shell is 304 stainless steel with a radius ratio of 0.97 and a length greater than 8 ft so as to avoid end effects. Air is the inner fluid. The coatings have specific acoustic impedances less than water and the nitrile coating is highly attenuating. The experiment was conducted for an approximate ka range of 1.5–15. The viscoelastic constants for the coatings were determined from the experimentally measured Young's and plane-wave moduli. The experimental procedure for measuring the form function was validated by the excellent agreement of the observed form function with the theoretical form function for the shell only. Significant observations are that (1) these coatings increase the form functions relative to the shell at low ka values; (2) the coatings reduce or eliminate the resonances observed in the shell-only baseline; and (3) the nitrile-coated shell is an excellent simulation of an ideal pressure release cylinder. [Work supported by Office of Naval Research, Code 11250A.]
Radiation from an array of simple sources
Authors: Akay Adnan
Published: October 2005

General closed-form expressions for the farfield intensity and power radiated by finite arrays are derived for a class of linear arrays made up of simple point sources. The closed-form solutions are made possible by a special trigonometric relationship that simplifies the intensity expression. Earlier work on out-of-phase sources [J. Acoust. Soc. Am. Suppl. 1 79, S30 (1986)] has been extended to include in-phase sources with axes either transverse or coincident with the axis of the array. [Work sponsored by NSF.]
Oblique reflection of a finite-amplitude dilatational wave in an elastic half-space
Authors: Shu Kun-Tien , Ginsberg Jerry H.
Published: October 2005

A ray theory for two-dimensional, finite-amplitude acoustic waves forming a mode within a hard-walled rectangular waveguide was described previously [K. T. Shu and J. H. Ginsberg, J. Acoust. Soc. Am. Suppl. 1 83, S1 (1988)]. The present paper extends those developments to the treatment of oblique reflection and mode conversion of a finite-amplitude dilatational wave (P wave) at the stress-free boundary of an elastic half-space. Due to nonlinear self-action, cumulative growth of second harmonics occur in the incident and reflected P waves in proportion to the square of the amplitude of the first-order incident and reflected P waves, respectively, but such growth is not encountered in the reflected vertically polarized shear wave (SV wave), nor in the many waves arising from nonlinear interaction between dilatational and shear waves. Uniformly valid expressions for strain are obtained by using renormalization techniques along the rays. The analysis indicates that the mode conversion between the nonlinear P and SV waves can be described by linear reflection theory. As a consequence of the reflection process, the nonlinear effect in the reflected P wave corresponds to a simple planar wave that originated from a weaker source at a longer range, even though the phase of that wave is governed by the actual propagation distance from the source. [Work supported by NSF and ONR.]
Classification of pitch periods using expert knowledge and neural net classifiers
Authors: Cole Ronald A. , Barnard Etienne , Vea Mathew , Alleva Fil
Published: October 2005

A pitch tracking algorithm that uses an artificial neural net (ANN) classifier to identify pitch periods has been developed. The algorithm consists of a peak detector that locates candidate peaks in the filtered waveform (0–700 Hz), and an ANN classifier that uses feature measurements to decide if the candidate peak begins a pitch period. The feature measurements include (a) the amplitude of the candidate peak, (b) the amplitude of the 7 peaks before and the 7 peaks after the candidate peak, (c) the location of these 14 peaks relative to the candidate peak, and (d) the median pitch observed so far in the utterance, based on the prior output of the classifier. The classifier was trained and evaluated on hand-labeled pitch periods in utterances in the DARPA TIMIT database. The classifier contained one hidden layer with six units, and was trained using backpropagation. Initial results on 10 test utterances using small amounts of training data revealed 98% correct classification of candidate peaks. Results of subsequent experiments will be presented.
Location and classification of plosive consonants using expert knowledge and neural net classifiers
Authors: Cole Ronald A. , Barnard Etienne , Hou Lily
Published: October 2005

A rule-based segmentation and broad classification algorithm [R. A. Cole and L. Hou, Proc. ICASSP 88, 453–456 (1988)] located over 95% of segments labeled [b], [d], [g], [p], [t], [k], [ch], [jh], [dr], and [q] (glottal stop) before sonorants in utterances of the DARPA TIMIT database. Artificial neural net (ANN) classifiers were trained to discriminate among the labels using perceptually motivated features. In one condition, 37 feature measurements were used to describe (a) the averaged spectrum during the 15 ms following the release burst, (b) zero crossings and peak-to-peak amplitude contours in the region of the segment, (c) the duration of the segment, and (d) the amplitude of the plosive burst. In a second condition, an additional 16 features were used to characterize the averaged spectrum during the first 30 ms of the sonorant following the plosive. The ANN classifiers consisted of either 37 or 53 input units, 30 hidden units, and 1 output unit for each category. Classifiers were trained using backpropagation and tested on 2000 segments provided by 20 speakers. With different amounts of training, classification accuracy was consistently 3%–5% better when vowel spectra were used, suggesting that ANN classifiers are able to learn coarticulatory relationships between consonant and vowel spectra. Classification accuracy was 70% for the 10 plosive consonants.
Incremental learning of large phonetic neural networks from smaller subnets
Authors: Waibel A.
Published: October 2005

Time delay neural networks (TDNNs) [Waibel et al., ATR-TR-0006 (1987)] have recently been shown to achieve excellent recognition performance for difficult phonetic discrimination tasks (e.g., voiced stops). This was achieved in part by the TDNNs' ability to not only activate the correct output category, but also to inhibit all incorrect outputs. The disadvantage of this property is that training larger networks with many more output categories (e.g., all consonants) becomes nontrivial, as all categories have to be incorporated in the learning process requiring excessive amounts of training. Several techniques are presented that overcome this problem by exploiting the hidden structure of previously learned smaller neural nets to train larger nets incrementally in comparatively short training runs. Experimental results show that the resulting larger networks aimed at stop consonants, at voiced stops and nasals, and at all consonants achieve recognition scores (95%–99%) as high as the smaller subnetworks from which they were constructed.
Phoneme recognition using Kohonen's LVQ
Authors: McDermott Erik , Katagiri Shigeru
Published: October 2005

Teuvo Kohonen has recently developed an algorithm similar to that used in his feature map classifiers but in which learning is supervised rather than unsupervised. This algorithm, known as learning vector quantization (LVQ), is similar to a K-nearest neighbor algorithm and allows a system to learn the vector quantization of the inputs to different categories. This algorithm is very simple, does not require a large number of training trials, and is capable of forming complex decision regions. As a recognition task, the speaker-dependent recognition of the phonemes /b/, /d/, and /g/ in different phonetic contexts is considered. The training procedure is applied to speech patterns that are stepped through in time, thus providing the system with a measure of shift invariance. Preliminary results indicate that LVQ can yield a recognition rate of 98.3% for 1880 testing tokens from three speakers. The simple vector operations that constitute the core of LVQ allowed for very easy parallelization and thus high learning speed, i.e., less than an hour.
A study of English word category prediction based on neural networks
Authors: Nakamura Masami , Shikano Kiyohiro
Published: October 2005

This paper describes word category prediction based on neural network models for constructing an accurate word recognition system. It is difficult to represent hidden linguistic structure and make an N-gram word prediction model using traditional stochastic approaches. In this paper, two neural network models that can learn hidden linguistic structure are proposed. These models can easily be expanded from Bigram to N-gram networks. They were tested by training experiments with an open English text database. The Trigram word category prediction rates show that neural network models are comparable to stochastic models. Trigram neural network models compress information about 150 times, which is the ratio of the Trigram stochastic model free parameters (893 = 704 969) to the neural network model link weights (4649). In addition, this paper proposes a new method that dynamically controls the training parameters, updating step size and momentum. These techniques are effective for calculating the efficiency of this system.
Connectionist techniques for speaker-independent recognition of isolated utterances
Authors: Franzini Michael A.
Published: October 2005

The performance of previous speech-recognition systems has been limited to a large extent by the incomplete structural theories of speech upon which the systems were based. The work reported here suggests that connectionist learning procedures provide an effective method for generating speaker-independent recognition systems without the need for any a priori model of human speech production. The backpropagation learning algorithm is applied to the problem of isolated-utterance speech recognition, and the resulting recognition rates approach those of the best conventional systems. Several studies were peformed using computer simulations of various networks trained by backpropagation. With 1 s of digitized speech as input, the networks had to generate as output the appropriate labels, which in these studies were letters of the alphabet. The accuracy for speaker-independent recognition of the whole alphabet reached 89%, which is higher than the accuracies achieved by several more conventional recognizers using the same database [K. F. Lee, Carnegie Mellon Univ. Tech Rep. 85–181 (1985)]. Recognition rates for speaker-dependent recognition of the whole alphabet reached 99% and, for speaker-independent recognition of confusable letter sets such as b-p-e-v-d, the networks achieved 94%. These studies demonstrate that networks with simple task-independent learning procedures can perform as well as systems that explicitly implement procedures such as dynamic time warping and vector quantization of inputs.
Speaker identity feature combination with a neural net
Authors: Velius George
Published: October 2005

This study compares the performance of a neural net approach with conventional linear methods as applied to the problem of feature combination in the domain of speaker identity verification (SIV). The experiment endeavors to combine features consisting of LPC-cepstral coefficient differences and pitch differences for isolated words in a template-matching scenario. The signal features are analyzed for 30-ms frames every 10 ms. The pitch estimate is based on the cepstrum of the LPC residual. Previous work [G. Velius, ICASSP 88, 583–586 (1988)] showed that the Fisher linear discriminant (FLD) was better at feature weighting (for cepstral coefficients only) than several other common linear methods. Results show that, when feature combination is done by the neural net, the SIV task is performed significantly better than when the feature combination (i.e., weighting) is done by the FLD. The neural network architecture used in this experiment was in no way “optimized” for the specific task at hand. An additional finding is that the pitch feature used here, in conjuction with the cepstral coefficients, contributes significantly to the SIV task; that is, the error rate is reduced by 13%.
The LPC trace as an HMM development tool
Authors: Doddington George R. , Picone Joseph , Godfrey John J.
Published: October 2005

Hidden Markov models have gained wide acceptance in speech recognition due to the ability to construct optimum (maximum likelihood) models automatically from speech data. Understanding how recognition performance is related to a model's structure is not a simple matter, however, especially where the structure is complex. Thus analysis of errors, especially in terms of the properties of the speech data, is not often undertaken. This paper describes a method for relating speech recognition performance to HMM structure. The basic tool is a best-path (Viterbi) trace of the model through the input data, which is represented using well-known LPC parameters. Linear predictor parameters are used as auxiliary model parameters, independent of the HMM output parameters used for recognition, in order to take advantage of established LPC speech synthesis and LPC speech spectrogram utilities. The trace can then be used to gain insight into error mechanisms, often leading to improvements in model structure and system performance. The description will include techniques for creating the LPC auxiliary model, spectrographic and auditory output from supervised and unsupervised model traces, and an evaluation of the impact of this technique on the development of an HMM-based speech recognition system.
Duration control methods for HMM phoneme recognition
Authors: Hanazawa Toshiyuki , Kawabata Takeshi , Shikano Kiyohiro
Published: October 2005

Two kinds of duration control for HMM (hidden Markov model) phoneme recognition are proposed: phoneme duration control for an HMM phone model and an event duration control for an HMM state. The phoneme duration control is carried out by combining an HMM output probability with a phoneme duration probability. The phoneme duration probability is calculated using a phoneme duration histogram obtained from training samples. Phoneme duration control is effective in discriminating phonemes with different durations such as /n/ and /N/. Event duration control is realized as a state duration penalty calculated from an HMM state duration probability of training samples. Event duration control is effective in discriminating phonemes with different event structures such as /s/ and /ts/. Recognition experiments are carried out using Japanese phonemes extracted from an isolated word database uttered by one male speaker. The phoneme recognition rate is improved from 84.8%–90.0% using these duration control techniques.
Speaker-independent phoneme recognition using hidden Markov models
Authors: Lee Kai-Fu , Hon Hsiao-Wuen
Published: October 2005

In this paper, the currently popular hidden Markov modeling to speaker-independent phoneme recognition is extended. Using multiple code books of various LPC-derived parameters and discrete HMMs, speaker-independent phoneme recognition accuracy of 58.8%–73.8% on the DARPA TIMIT database, depending on the type of acoustic and language models used, is obtained. In comparison, the performance of expert spectrogram readers is only 69% without use of higher level knowledge. The co-occurrence smoothing algorithm that enables accurate recognition with only a few training examples of each phone is also introduced. Since these results were evaluated on a standard database, they can be used as benchmarks to evaluate future systems. [Work supported by DARPA.]
A TRACE-like speech perception simulation model for the recognition of Japanese phonemes
Authors: Niemöller Meinrad , Makino Shozo , Kido Ken'iti
Published: October 2005

Based on the concept of the interactive activation model of speech perception (TRACE model) introduced by McClelland and Elman, a simulation model for the recognition of Japanese vowels and voiced plosives has been built up. The system currently consists of a feature and a phoneme level. For preprocessing, the output of a 29-channel filter bank is reduced by KL transformation to 16 dimensions and presented to the feature level every 10 ms. The phonemes are represented by nodes every 20 ms and have excitatory connections to five and to nine feature slots for consonants and vowels, respectively. In each processing cycle, there is lateral inhibition of nodes of the same level, feedforward and feedback excitation between the levels, and an activation update. A stable phoneme response output is reached after about 30–40 processing cycles. For the training of the feature-phoneme mapping by a gradient-descent procedure, a database of 212 words uttered by ten male speakers is used.
Sound power determination in reverberation chambers
Authors: Maa Dah-You
Published: October 2005

The determination of sound power in reverberation chambers is discussed in the light of normal mode theory. It is known that the sound power emission of a source depends on its position in the reverberation chamber and on interference pattern results in the sound field excited. As a consequence, the measured sound pressure varies widely from point to point. It is shown that the sound pressure is proportional to the average contribution of the product of the normal functions at the source and at the receiver. Based on this relation, the determination of sound power emitted by the source at a particular point in the room is devised through the measurements of average sound field as well as by a corner microphone. It is also shown that sound power determined in reverberation chambers by the standard method is always less than the free-field power, and the difference increases as the frequency decreases. This is just what happens in practice, and good agreement with the theory is obtained with earlier experiments. A statistical formula of the sound pressure in a reverberation chamber developed from the exact theory is used to this end.
Steady-state acoustic energy distribution in a reverberation chamber at low frequencies
Authors: Flynn Daniel R. , Bartel Thomas W. , Yaniv Simone L.
Published: October 2005

The spatial variation of the mean-square sound pressure in a hard-walled rectilinear reverberation chamber is analyzed by extending the “Waterhouse theory” (based on a free-wave model for an omnidirectional sound field impinging on the corner formed by three orthogonal planes) to apply to a closed chamber at low frequencies. This is done by expressing the mean-square sound pressure in terms of the contributions from the actual pressure microphone and from an infinite array of image microphones due to multiple reflections in the chamber walls, and then transforming this expression into a weighted sum of the normal eigenfunctions for the chamber. Experimental sound-pressure levels, measured along different linear paths in the 425-m3 NBS reverberation chamber, are compared with the predictions of this analysis. It is found that the spatial dependence of sound-pressure level is accurately predictable, at least in the NBS chamber, to significantly lower frequencies than has usually been thought possible. Expressions are given for relating the spatial average (over the volume of the chamber) mean-square sound pressure to that which is measured at a given position in the room. The implications of these findings to sound power determinations are discussed.
Kinetic and potential energies of a stationary sound field
Authors: Cook Richard K.
Published: October 2005

There is a continuing need for accurate measurement of radiated sound power by compact noisy sources. Examples are the noises of fans, motors, machinery, etc. One of the standard methods of measurement makes use of a reverberation chamber. The potential energy of the sound radiated into the chamber is measured by sampling the sound pressure at various points, from which the time-space-averaged potential energy is calculated. The kinetic energy is assumed to equal the potential energy; then in the standard method, the sum of the two is used as the total energy for calculation of the radiated sound power. In experiments on a source having an accurately known radiated power, we found that the power determined in a standard-method chamber was about 20% (0.9 dB) less than the true power. A likely source of the systematic difference is the possibly false assumption that the time-space averaged kinetic energy is equal to the potential energy. It is shown that, in general, in a sound field stationary in an enclosure, the two energies are not equal. Several examples are presented. Included are fields in enclosures having sound absorbers. An accurate ratio of the energies requires detailed knowledge of the field.
Analysis of sound fields in rooms by Bergeron's method
Authors: Shimoda Hidemaro , Yoshida Norinobu , Fukai Ichiro
Published: October 2005

The behavior of sound wave motion in rooms is so complicated that it is not easy to treat it theoretically unless simple geometrical shapes or simple boundary conditions are assumed. This study presents a formulation applying Bergeron's method, developed for an electromagnetic field simulation by computer [N. Yoshida et al., Trans. IECE Japan, J62-B, 6, 511–518 (1979)], to room acoustics. In this formulation, the sound field is represented by an electrical equivalent circuit composed of distributed lines corresponding to the acoustic equation, and nodal equations for each connection of the lines are derived. As an example, consider a cube-shaped room with uniform absorption by each wall. The transient responses to the sinusoidal waves and the 1/3-octave-band tone burst are calculated, and the stationary sound-pressure distribution and the reverberation time are obtained from these responses. The results show the validity of the formulation and prove the effectiveness of the application of this method to room acoustics.
Measurements of short reverberation times using a real time analyzer
Authors: Højbjerg Klaus
Published: October 2005

Classical measurements of reverberation time with use of the interrupted noise method is known to almost everybody working with acoustics. However, there seems to be less knowledge about the limitations of the method, i.e., the connection between measurement bandwidth and minimum averaging time and what that means for minimum reverberation time measured. Another method is impulse excitation followed by a backwards integration. Here, the analysis can be done in two ways, either direct frequency analysis of the response signal giving the same limitations as with the interrupted noise method or by recording the time signal and then feeding the signal reversed to the filter bank giving possibilities of measuring very short reverberation times. The methods of measuring reverberation time will be discussed and examples of measurements shown.
Frequency distribution of normal modes of vibration in rectangular rooms; A new look
Authors: Sepmeyer Ludwig W.
Published: October 2005

In a previous study on the frequency and angular distribution of the normal modes of vibration in rectangular rooms, published in 1965, the criterion for judging the frequency distribution of the modes was called the mode spacing irregularity MSI. The MSI, based on the ratio of the actual mode spacing between two adjacent modes to the ideal spacing between two modes predicted by Maa's equation, is quite insensitive to changes in room dimension ratios. In this paper, a new criterion called the modal volume distribution index, MVDI, is proposed. The MVDI is based on the difference in spherical volume enclosed by two adjacent modal vectors in frequency space and the difference in volume between two similar ideal vectors. In this way, larger and smaller than ideal spacings are weighted equally. Results, quite different from those inferred from the MSI, are given for a few selected dimension ratios. Revisions to National and International Standards that require sound-pressure level measurements in reverberation rooms are recommended.
Analysis of the sound field in a room with uneven surfaces by the hybrid method combining geometrical theory and wave theory
Authors: Fujiwara Kyoji
Published: October 2005

This study presents a new way of estimating the sound field in a room with walls that do not reflect the incident sound specularly. So far the reflection characteristics of these walls have been considered to be perfectly diffuse or a mixture of partially diffuse and partially specular. The aim of this study is to introduce the real reflection characteristics of the walls into the simulation procedure for the sound field. The estimation procedure is essentially based on the Monte Carlo simulation method. For simplicity, the uneven walls are assumed to be periodically corrugated. Prior to the simulation, the reflection characteristics of the periodically corrugated wall are analyzed by wave theoretical methods, and the result, which contains the directions and magnitudes of the reflected waves, is used in the simulation procedure. The energy impulse responses estimated by the method presented are compared with those obtained by the ordinary method based only on geometrical theory.