bluuebunny/crossref_metadata_2025 huggingface dataset:
Phone inventory optimization for multilingual automatic speech recognition
Authors: Melnar Lynette , Talley James
Published: October 2013

This paper describes a phone inventory optimization procedure for application in multilingual automatic speech recognition (ASR). The optimization procedure is based on three knowledge sources that act collectively to guide phonological reduction and selection processes: (1) abstract (language-independent) phonological universals and tendencies that are used in the construction of a hierarchical structure that specifies phone class reduction paths; (2) language-dependent knowledge that includes information of the targeted languages’ phone inventories and individual phone frequencies in language data resources; (3) acoustic data that provides phone discriminability and similarity metrics. Using the optimization procedure, the phone inventories of six languages, American English, Mandarin Chinese, Egyptian Colloquial Arabic, Japanese, German, and Spanish, were merged to create an inventory consisting of 64 distinct cross-phonological units. This reduced phone set was used in all training and testing procedures and resources for the recognition of the six targeted languages. Preliminary recognition results are very encouraging: while purely data-driven approaches to multilingual ASR fail to reach word-recognition rates comparable to monolingual applications, the use of the optimized phone inventory in our multilingual ASR program yields recognition rates approximating that of monolingual ASR.
A pulse length tolerant neural network-based detector for sector-scan sonar
Authors: Perry Stuart W. , Guan Ling
Published: October 2013

In this paper we present a neural network-based system to detect small manmade objects in sequences of sector-scan sonar images created using signals of various pulse lengths. The sonar system considered has three modes of operation to create images over ranges of up to 800 m using acoustic pulses of different durations for each mode. After initial cleaning and segmentation to extract objects, features are computed from each object. These features consist of basic object size and contrast statistics, shape moments, moment invariants, and features derived from the second-order histogram of each object. Optimal sets of 15 features from the total set of 31 are chosen using sequential feature selection techniques. Using these features a neural network is trained to detect manmade objects in any of the three sonar modes. The proposed detector is shown to perform very well when compared with detectors trained specifically for each sonar mode and a number of statistical detectors. The proposed detector achieves a 92.4% detection probability at a mean false alarm rate of 10 per frame averaged over all sonar mode settings. Finally, research into Recurrent Neural Network detectors is described and shown to further improve performance.
A neural network approach for data inversion, application to the characterization of marine sediments, and sonar targets
Authors: Zakharia Manell E.
Published: October 2013

The paper presents an innovative method for inversion based on the use of neural networks. The inversion procedure will be split in three major steps: the understanding and enhancing of the physical phenomena involved, the learning of the physics from extracted parameters, and the inversion itself. This separation allows a very convenient distribution of computation load: the first two steps (which are the most demanding in terms of computation time) can be done in a prior phase, while the third one (which becomes very simple) can be implemented on-line (real time). Another major advantage of the proposed method (with respect to conventional approaches such as conjugate gradient) is that it requires only poor a priori information thanks to a judicious choice of the characteristic physical phenomena and the selection of relevant signal processing tools for parameter extraction. Two examples of the proposed procedure are described: the inversion of geoacoustic parameters of seabed and the characterization of sonar targets. The inversion is applied to both simulated and experimental data sets. In both cases, the training will be achieved on simulated data and then applied to the experimental.
Real time identification of several targets by iterative time reversal
Authors: Montaldo Gabriel , Tanter Mickael , Fink Mathias
Published: October 2013

Finding punctual targets inside an aberrating medium is a problem of interest in nondestructive testing and medical research. Previous works have shown that the iteration of the time reversal method allows us to identify the strongest reflector of the medium in real time but the identification of several targets is not possible with this technique. Using the idea that each strong scatterer is associated to an eigenvalue of the time reversal operator, a method has been developed (the DORT method) to identify punctual diffusers in a medium. This technique requires the experimental acquisition of the whole time reversal operator and its numerical decomposition implies many calculations. We propose here a way to identify in real time all the punctual targets by modifying the time reversal iterative method. In a first step, we identify the strongest reflector by time reversal processing. In a second step, we subtract its contribution in order to find the second reflector contribution and so on. This method allows us to build an ‘‘eigen pulse’’ able to focus a pulsed signal on each target. We demonstrate experimentally its real time feasbility by identifying several targets inside a strongly aberrating medium.
Modeling the interaction of acoustic and internal wave fields in shallow-water environments
Authors: Finette Steven
Published: October 2013

Over the past decade, there has been considerable interest in the effect of solitary wave packets on the acoustic field. This talk will present an overview of modeling efforts to describe the interaction of these internal gravity waves with acoustic field propagation in shallow-water waveguides. The emphasis will be on the interaction of solitary wave packets with low-frequency acoustic fields propagating in a continental shelf/slope environment. Topics to be discussed will include both adiabatic and mode-coupling effects, differences between 2-D and 3-D propagation, as well as the influence of wave packets on horizontal array beamforming. Most research to date involves solitary wave propagation over flat bathymetry, but does not address the generation of internal tides over variable bathymetry and their subsequent evolution into wave packets. In this regard, recent nonlinear fluid dynamic simulations of both the generation and propagation of internal tides and wave packets are illustrated and their influence on acoustic propagation briefly addressed. [Work supported by ONR.]
Rapid fluctuations of the channel impulse response at midfrequencies in shallow water
Authors: Hodgkiss W. S. , Kuperman W. A. , Ensberg D. E.
Published: October 2013

A fixed source, fixed receiving array experiment was carried out to measure the stability of forward transmissions in shallow water (∼100 m deep) over a 6 km forward propagation path off San Diego, CA. The source was moored 6 m above the seafloor and the 12 aperture, 64-element vertical receiving array was deployed with the lowest element 4 m off the bottom. The source transmissions of interest here are the 2 kHz bandwidth, 1 s duration FM chirps which were transmitted continuously for 5 min at a time and have been matched filtered to yield the channel impulse response. In addition to CTDs taken in the region between the source and receiving array, a thermistor string at the receiving array site provided measurements of water column temperature fluctuations. The time-evolving structure of the channel impulse response clearly shows significant, environmentally induced fluctuations which also are evident in an arrival angle vs travel time spatial decomposition at the array. [Work supported by ONR.]
High frequency sonar variability in littoral environments: Irregular particles and bubbles
Authors: Richards Simon D. , Leighton Timothy G. , White Paul R.
Published: October 2013

Littoral environments may be characterized by high concentrations of suspended particles. Such suspensions contribute to attenuation through visco-inertial absorption and scattering and may therefore be partially responsible for the observed variability in high frequency sonar performance in littoral environments. Microbubbles which are prevalent in littoral waters also contribute to volume attenuation through radiation, viscous and thermal damping and cause dispersion. The attenuation due to a polydisperse suspension of particles with depth-dependent concentration has been included in a sonar model. The effects of a depth-dependent, polydisperse population of microbubbles on attenuation, sound speed and volume reverberation are also included. Marine suspensions are characterized by nonspherical particles, often plate-like clay particles. Measurements of absorption in dilute suspensions of nonspherical particles have shown disagreement with predictions of spherical particle models. These measurements have been reanalyzed using three techniques for particle sizing: laser diffraction, gravitational sedimentation, and centrifugal sedimentation, highlighting the difficulty of characterizing polydisperse suspensions of irregular particles. The measurements have been compared with predictions of a model for suspensions of oblate spheroids. Excellent agreement is obtained between this model and the measurements for kaolin particles, without requiring any a priori knowledge of the measurements.
Sediment tomography in the East China Sea: Compressional wave speed and attenuation inversions from Airy phase dispersion measurements and time series correlation
Authors: Lazauski Colin , Miller James , Potty Gopu , Chen Chuen-Song , Dahl Peter
Published: October 2013

This paper discusses ongoing data analysis results from the acoustic bottom interaction experiment conducted in May–June 2001 in the East China Sea as part of the Asian Seas International Acoustics Experiment (ASIAEX-2001). Using time-frequency scalograms of broadband signals, the modal arrivals and group speed minimums (Airy Phase) of several modes are clearly observed. The structure of the Airy Phase signal is used to match the dispersion curves which forms the basis of this inversion technique. Utilizing the Airy Phase group speed minimums and corresponding pressure amplitudes of each observable mode, the sediment compressional wave speed and attenuation as a function of depth are derived. The group speed minimum for each mode provides additional information on the compressional wave speed in the modal sediment depth penetration interval. To refine the sediment parameters, synthetic and measured time series are correlated for goodness of fit and used in the inversion process. The synthetic time series is generated from the scalogram values corresponding to the times and frequencies of the calculated dispersion curves. Inverted speeds and estimated modal penetration depths are then used to develop the sediment profile. The estimated resolution is dependent on the number and frequency span of the observable modes. Estimated sediment properties from several areas are presented with verification from coring results. [Work supported by ONR.]
Inversion of geoacoustic parameters in the South China Sea
Authors: Potty Gopu R. , Miller James H. , Lazauski Colin J. , Lynch James F. , Newhall Arthur
Published: October 2013

During the ASIAEX Experiment in the South China Sea, light bulb sources were deployed in the vicinity of the receive array in order to provide array element localization. We present inversions for bottom geoacoustic properties using these broadband light bulb data. The light bulbs imploded about 40 s after launch and were deployed approximately 1 km from the receive array. The receive array was at a depth of 125 m of water. The ground truth measurements include bathymetry data from ship board measurements, chirp sonar data, and sediment cores. SeaSoar surveys and CTD measurements provide sound-speed data in the water column. The inversion is carried out using a coherent broadband matched field processing technique [Chapman et al., Oceans 97, MTS/IEEE Conference Proceedings, Vol. 2, pp. 763–768]. The thickness of the top layer of the sediment and the compressional wave speeds in the top and bottom layers are estimated in the vicinity of the receive array and compared with ground truth measurements. [Work supported by ONR.]
Simulations of time spreading in shallow water propagation
Authors: Thorsos Eric I. , Elam W. T. , Tang Dajun , Henyey Frank S. , Williams Kevin L. , Reynolds Stephen A.
Published: October 2013

Pulse propagation in a shallow water wave guide leads to time spreading due to multipath effects. Results of PE simulations will be described for pulse propagation in shallow water with a rough sea surface and a flat sandy sea floor. The simulations illustrate that such time spreading may be significantly less at longer ranges than for the flat surface case. Pressure fields are simulated in two space dimensions and have been obtained using a wide-angle PE code developed by Rosenberg [A. D. Rosenberg, J. Acoust. Soc. Am. 105, 144–153 (1999)]. The effect of rough surface scattering is to cause acoustic energy initially propagating at relatively high angles but still below the critical angle at the sea floor to be eventually shifted to grazing angles above the critical angle. This energy is then lost into the bottom, effectively stripping higher propagating modes. The surviving energy at longer ranges is concentrated in the lowest modes and shows little effect of time spreading. Thus, the effect of rough surface scattering is found to produce a simpler temporal field structure than if the surface were treated as flat. [Work supported by ONR.]
Evoked-potential study of echolocation in whales and dolphins
Authors: Supin Alexander Ya. , Nachtigall Paul E. , Au Whitlow W. L.
Published: October 2013

During echolocation, whales and dolphins perceive faint echoes very shortly after loud ongoing pulses. How does the auditory system avoid the masking of the echo by the preceding pulse? To investigate mechanisms of release from masking, auditory brainstem evoked responses (ABR) were recorded in bottlenose dolphins and a false killer whale both during active echolocation and during paired external acoustic stimulation. When the animal ensonified a target, both the ongoing pulse and echo (2-ms delay) evoked ABRs of comparable amplitudes, in spite of echo intensity of 40 dB relative the ongoing pulse near the animal’s head. When a phantom echo was used (emitted pulses were replayed through an electronic channel with the same delay of 2 ms), the ABR to the echo was comparable to that to the emitted pulse at the echo intensity of 20 dB, but the ABR decreased when the echo intensity diminished to 40 dB relative to the ongoing pulse. An explanation of higher efficiency of the real than the phantom echo may be a difference between spectra of the real echo and the outgoing pulse. Paired-pulse stimulation confirmed that a moderate spectral difference between two pulses resulted in release from masking at short interpulse intervals.
A gypsum-based artificial stone for shock wave lithotripsy research
Authors: McAteer James A. , Williams James C. , Evan Andrew P. , Cleveland Robin O. , Bailey Michael R. , Crum Lawrence A.
Published: October 2013

Natural kidney stones are heterogeneous in structure, composition, material properties and fragility, and as such are problematic for use in determining the mechanisms of SW-action in SWL. A variety of model stones have been developed. We have adopted Ultracal-30 gypsum [Dahake and Gracewski, J. Acoust. Soc. Am. 102, 2138 (1997)] for in-vitro and in-vivo studies. U-30 stones (7.5×6.5 mm) cast in polystyrene molds were liberated with chloroform and stored in water. Drop-impact testing of dry stones showed a linear relationship between increase in surface area of fragments and energy applied. Breakage of hydrated stones in a research-electrohydraulic lithotripter, likewise showed a linear increase in fragment area with increased SW-number and SW-voltage. The density (1800 kg/m3) and transverse (1520 m/s) and longitudinal (3100 m/s) wave speeds of U-30 stones place them in the range determined for natural stones. U-30 stones implanted in pig kidneys exhibited cavitation erosion and spall fracture similar to stones in-vitro, and U-30 stones in-vitro and in-vivo showed equivalent response to SW-rate (200% higher fragmentation at 0.5 Hz compared to 2 Hz). U-30 stones softened with prolonged exposure to water and degraded during long-term implantation in-vivo. With these cavaets U-30 stones provide a useful model for SWL research. [Work supported by NIH P01-DK43881.]
Progression of crack formation in artificial kidney stones subject to shock waves
Authors: Van Cauwelaert Javier , Cleveland Robin
Published: October 2013

We used micro computed tomography (CT) imaging to follow the progressive development of cracks in artificial kidney stones. The artificial stones were made from U30 cement with a cylindrical shape (6.5 mm diameter and 7.5 mm long). The stones were held within a polypropylene vial in one of three different orientations: vertical, horizontal, and angled at 45 deg. The stones were treated with an electromagnetic lithotripter using between 50 and 150 shock waves. The initiation and growth of cracks was observed using microCT. We found that crack formation in the U30 stones was influenced by the orientation of the stone with respect to the shock wave (SW) propagation direction. Vertical stones developed a spall-like crack near the distal surface; horizontal stones had little internal damage for the number of shock waves applied; and angled stones were damaged primarily in the vicinity of the leading corner. The position of the cracks were in qualitative agreement with the numerical solutions of the pressure field inside the stones. The elastic properties of the U30 stones are being measured which will alow a quantitative analysis of crack growth to be performed and compared to the experimental data. [Work supported by the Whitaker Foundation.]
The reduced renal blood flow observed after exposure to shock wave lithotripsy involves intact renal nerves
Authors: Connors Bret A. , Evan Andrew P. , Willis Lynn R. , Simon Jay R. , Shalhav Arieh L. , Paterson Ryan F. , Kuo Ramsay L. , Lingeman James E.
Published: October 2013

Treating one kidney with high-energy acoustic shock waves reduces blood flow 65% in that kidney and 33% in the contralateral unshocked kidney. We tested the hypothesis that renal nerves were involved in this response. Six-week old pigs underwent unilateral renal denervation. Nerves along the renal artery of one kidney were cut and the artery was painted with 10% phenol. After 2 weeks, the pigs were anesthetized and bilateral renal function was determined using inulin and PAH (para-aminohippurate) clearance. Glomerular filtration rate (GFR, inulin clearance) and renal blood flow (PAH clearance) were measured both before and after lithotripsy to the lower pole of the innervated kidney (2000 shocks, 24 kV, unmodified Dornier HM-3) or sham lithotripsy. Both kidneys were then removed to measure norepinephrine content in the tissue. Norepinephrine levels were significantly reduced (more than 90%) in denervated kidneys indicating complete denervation. As expected, blood flow and GFR fell (approximately 50%) after lithotripsy in the innervated shocked kidneys, but did not fall significantly in denervated contralateral kidneys or in sham lithotripsy animals. Results suggest that renal nerves mediate the fall in contralateral renal function after lithotripsy. [Work supported by NIH P01 DK43881.]
Effects of the viscosity of the liquid in the microcapsule on deformation process of a bubble for developing DDS using shock waves
Authors: Tamagawa Masaaki , Yamanoi Ichiroh , Matsuo Toshiaki
Published: October 2013

This paper describes the fundamental investigations for developing new DDS using shock waves and bubbles, especially effects of the viscosity of the liquid in the capsules on the deformation process of the bubbles. The viscosity of the liquid in the microcapsules requires controlling the bubble position and motion, deformation process to optimize the disintegration of the capsules. Using plane shock wave in the water by shock tube apparatus, a bubble near the curved gelatin wall was observed by high-speed framing camera. The deformation process for a bubble was analyzed by image processing. From these experiments, it is found that the viscosity of the liquid around the bubble suppress the amplitude of the bubble oscillations and collapsing the bubble. This means that the viscosity of the liquid in the capsule should be as small as possible for developing DDS using shock waves. In spite of this result, the effects of curvature and elasticity of the wall on the deformation process is almost the same as low viscosity. [Work supported by a Grant-in-Aid for Scientific Research from the Ministry of Education, Science, Sports and Culture, Japan.]
Objective measurements for the assessment of hearing protectors attenuation at high level impulsive noise
Authors: Vergara Felipe , Gerges Samir N. Y. , de Lima Washington J. N. , Birch Robert
Published: October 2013

The evaluation of hearing protectors for the attenuation of high amplitude impulsive noise cannot be carried out using the conventional subjective Real Ear Attenuation at Threshold (REAT) technique. In the case for high level impulsive noise it is not possible to conduct subjective type tests with volunteer listeners, therefore, other methods based on using artificial human head forms need to be considered. In this paper we present an objective technique that uses an artificial head, ear-canal simulator and shock tube to assess earplug and earmuff protectors. The shock tube is used as a means of producing controlled repeatable high amplitude pressure pulses (>140 dB) of varying rise time, amplitude, and duration. The artificial head and ear simulator with a protector are mounted inside the shock tube and subjected to a high level sound pulse. Pressure levels are measured simultaneously outside of the protector (incident pulse) and inside at the ear simulator at the eardrum position. The attenuation of peak pressure levels in the time domain and the corresponding spectra of the pulses are determined. The technique offers pointers towards future standards of performance hearing protectors for applications with high level impulsive noise.
Communications earplug performance in tracked-vehicle crews
Authors: Ahroon William A. , Ostler Dale A. , Mozo Ben T. , Crossley Cynthia M.
Published: October 2013

Tracked-vehicle crewmembers operate in an acoustical environment that is one of the most inhospitable encountered in the U.S. Army. Overall noise levels in the M1A2 Abrams tank and M2A2 Bradley Fighting Vehicle can be well over 100 dBA and can exceed 130 dB in some octave bands. Due to the nature of the crewmembers’ duties, speech communication is essential for effective operations and this capability is seriously degraded because of the nature of the noise environment. The Communications Earplug (CEP), consisting of miniature receivers attached to replaceable foam earplugs, is used in Army aviation to provide superior noise attenuation with improved speech communication. The noise attenuation of the DH-132A Combat Vehicle Crewmans helmet (CVC) when used in conjunction with CEP (measured in accordance with ANSI S12.6-1997) is reported with an NRR of 27 dB vs 15 dB for the CVC alone. CEPs were installed in the helmets of 77 tracked-vehicle crewmembers. Following a 6-month trial period, crewmembers (n=36) reported increased noise attenuation, improved speech communication, and reduced stress when using the CVC-CEP integration. The CEP is a cost-effective alternative to expensive active noise reduction helmets currently being fielded for use by tracked-vehicle crews. [Work supported in part by USACHPPM.]
System for measurement of headband force in hearing protection devices and audiometric equipment
Authors: Llamas-Llamas Osvaldo , Razo-Razo Jose-Noe
Published: October 2013

Application force influences audiometric results and hearing protection devices (HPD) attenuation performance. In HPD attenuation testing it is neccessary to know the application force. Audiometric studies are being conducted with standardized values of the application force. Design and results of a measurement device for bands and headbands application force are presented. Error analysis in the range from 1N to 30N is carried out and the associated measurement uncertainty is estimated. The device provides settings for test distances from 70 mm to 140 mm (vertical axis), and 115 mm to 195 mm (horizontal axis), as required in most of the available standards related with HPD and audiometric equipment.
Correlations in the seismic coda and elastic Green’s functions
Authors: Campillo Michel , Paul Anne
Published: October 2013

The conjecture that the correlation of coda records at two stations may provide the Green’s function between these two stations, or an approximation of it, is discussed. Under the assumption of equipartition and/or considering that we use a set of sources that is evenly diffracted in the whole space, it can be shown that the average cross correlation between the records of every earthquake at the two stations is an approximation of the Green’s function between the two stations. Records from stations of the Mexican network are used to test this idea. The stack of the cross correlations indicates that a low-frequency coherent signal is present with a signal-to-noise ratio that was ∼0.2 for a single correlation. The Green’s function between two points at the surface is expected to be widely dominated by the Rayleigh wave. The signal that we extracted presents the characteristics expected for a Rayleigh wave: elliptical polarization in the radial–vertical plane and adequate group velocity. The coherent signal extracted from coda correlations has the symmetry properties of the Green tensor. In spite of the a priori limitations of this approach in seismology, the preliminary results presented here demonstrate its potential.
Coda wave interferometry in finite solids, recovery of the P to S conversion rate
Authors: Weaver Richard L. , Lobkis Oleg I.
Published: October 2013

Diffuse fields, which appear incoherent, nevertheless retain their deterministic character. Even wave fields that have scattered thousands of times are highly repeatable if temperature is held fixed. The temperature dependence of elastic wave speeds, and the great age of these signals can, however, lead to significant fluctuations. Our 1 MHz transient point source in a 10 cm aluminum block gives rise to a complex waveform that was observed to undergo an almost pure dilation of 0.0262% per degree C over the range from room temperature to 40°. The dilation rate was determined from the cross-correlation function between time-windowed responses at different temperatures. Theoretical calculations of the shift of the correlation function were constructed in terms of a weighted average of the temperature dependencies of the individual P and S waves that compose the full field. That the P and S wave speeds have different dependence on temperature leads to a degradation of the strength of the correlation function maximum. The rate of degradation should be a measure of the mean lifetime of P and S rays against mode conversion; comparisons with theory corroborate that suggestion. [Work supported by the NSF, Grant No. CMS 99-88645.]
Absorption coefficient and energy flow path identification by means of inverse local energy method
Authors: Hardy Pierre , Jezequel Louis , Ichchou Mohammed
Published: October 2013

Estimation of the single absorption rate and of acoustical power input is currently made in reverberant rooms where a diffuse field is established. In this paper we aim at describing a method dedicated to absorption coefficients and energy flow path identification within all type of acoustic fields in medium and high frequencies, by means of an inverse local energy method. Making use of an energy integral equation with diffuse reflection, an estimator of the wall various absorption rates is built up, while the cavity is excited by a standard spherical acoustic source. A similar formulation is used to characterize a wall continuous excitation. Then it is possible to go through the measurements (pressure, intensity) and supply a detailed analysis of the wall input energy flow. This study also includes a numerical comparison between optimization methods used when trying to match the calculated field and the reference field. Among those methods, the spheric gradient proves to be efficient when compared with estimators based on other cost-functions. Sensitivity to discretization, robustness versus statistic measurement errors, and relevance of identified parameters are dealt with.
Automatic phone segment alignment using statistical deviations from manual transcriptions
Authors: Hayakawa Toru , Shirai Katsuhiko , Kato Hiroaki , Sagisaka Yoshinori
Published: October 2013

For precise temporal characteristic description, disagreements between manual labeling and automatic labeling were quantitatively analyzed with respect to the spectral feature extraction, adoption of acoustic matchers (HMM models), and acoustic matcher by itself. Error analysis shows that boundaries are shifted at phone boundaries where the speech spectrum changes quite rapidly. This disagreement results from the spectral feature extraction averaged over a given window. For the adoption of model, big errors are found at phone boundaries where the spectrum changes slowly. The third model-dependent errors are seen at phones whose duration cannot be shorter than the frame increment period times the HMM state number. To take into account these error factors individually to reduce the amount of alignment errors, we modified the automatic alignment results context-dependently using statistical characteristics of phone boundary displacement. This post-processing of boundary modification reduces boundary errors from 14.79 ms to 11.07 ms. Supplementary experiment shows that this improvement of about 4 ms corresponds to eight times of error reduction obtained by speaker adaptation of acoustic matchers. [Work supported by TAO, Japan.]
Duration normalization for improved automatic speech recognition
Authors: Nedel Jon P. , Stern Richard M.
Published: October 2013

While hidden Markov models (HMMs) serve as the basic acoustic modeling framework for many automatic speech recognition systems, they are known to model the duration of sound units poorly. Phone duration normalization can be accomplished by adding and reconstructing missing frames when a phone is shorter than the desired duration, and by deleting frames when a phone is longer than the desired duration. If phone segmentations are known a priori, this technique achieves relative reductions in word error rate (WER) of up to 35%, confirming the conjecture that speech with normalized phone durations may be modeled better and discriminated more accurately using standard HMM acoustic models. Unfortunately, duration normalization using imperfect automatically generated phone segmentations has not yielded significant recognition improvements. A modification of the duration normalization approach has been developed. Three different feature streams are generated for each utterance using various combinations of expansion and contraction of hypothesized phone segments. Each stream is recognized using an acoustic model trained for that stream. While the resulting recognition hypotheses themselves are not significantly better than baseline, these hypotheses can be automatically combined to produce relative improvements in WER of up to 7.7% over several speech databases. [Work supported by DARPA and Telefónica.]
A descriptive analysis between syllables in English and Spanish
Authors: Martinez Licona Fabiola M. , Goddard Close John Ch. , Martinez Licona Alma E.
Published: October 2013

In recent years different attempts have been made to incorporate temporal information longer than the phoneme into automatic speech recognizers (ASR) for English. The reason for these approaches is related to limitations which arise with existing systems based on phonemes, such as the degradation in performance of ASRs under noisy conditions and variations in pronunciation due to phoneme omission. It is conjectured that humans naturally use longer time periods, corresponding, for example, to syllables, to perceptually integerate information. In the case of Spanish, little seems to have been done in this direction for ASRs. In the present paper, a preliminary comparison is made between syllables in Spanish and English with a view to their factibility in an ASR for Spanish. In particular, a descriptive statistical analysis is conducted with a Spanish speech database to derive the most common structures of the syllables and the most common monosyllables. This is contrasted with previously found results in English. Spectrograms are also used to illustrate pertinent characteristics of Spanish. These results suggest that syllables may indeed provide useful information for an ASR in Spanish and could provide greater success than their counterparts in English. [Work supported by CONACYT under Project 31929-A.] (To be presented in Spanish.)
Low bit rate speech coding using spectral trajectory modeling
Authors: Dusan Sorin , Flanagan James
Published: October 2013

In the U.S. Federal Standard coder for 2400 bps, a data frame containing 54 bits of encoded signal is transmitted every 22.5 ms. In each frame, 25 bits encode the spectral features (10 Line Spectrum Frequencies—LSF). In this paper we describe a method for reducing the transmission rate while preserving most of the quality and intelligibility. This method is based on modeling the spectral trajectories with polynomial functions and on encoding these functions for segments of speech extending over multiple frames. Here 10 polynomials are computed by fitting them to the 10 LSF trajectories in the least-squares sense. Then the polynomial coefficients are encoded for the whole segment instead of directly encoding the LSF vectors. The spectral parameters are thus reduced (compressed) to [(P+1)/N]×100%, where P represents the order of the polynomials and N the number of frames for each segment. Different compression rates can be achieved. For example, for P=5 and N=10 the spectral features are encoded using 40% less bits than those required to encode N=10 LSF vectors. For this example, the overall transmission rate is reduced to 1956 bps, yielding an average Itakura–Saito spectral distance of 0.14 between the original and reconstructed LPC parameters computed for a typical utterance.
Intelligibility tests for synthetic speech subjective evaluation: The semantically unpredictable sentences approach for European Portuguese
Authors: Braga Daniela , Coelho Luís , Moura António , Freitas Diamantino
Published: October 2013

In this paper a test proposal for European Portuguese (EP) synthetic speech quality evaluation as well as some of its application are presented. For this purpose we have built a semantically unpredictable sentences (SUS) corpus of 25 sentences with different lengths that were chosen according to the standard EP phonological, syllabic and prosodic specific features. The goal is to create a standard test linguistically focused that constitutes a base for a more accurate comparison and ranking of different synthesis techniques. The tests were conducted in two distinct conditions: one performed in a controlled environment with listeners ranging from 16 to 60 years old; the other performed through the web by a large group of listeners of different ages, each one in their own environment. The listeners had only one chance of listening to each sentence. Of course in the second case this could not be controlled. The evaluation parameters considered were the number of (1) totally well identified sentences; (2) nonidentified sentences; (3) misunderstood words in each sentence; (4) misunderstood words in the whole corpus; (5) misunderstood words according to the length of the sentence; (6) errors in the perception of a word according to its position in the sentence.
An estimation method for fundamental frequency and voiced segment in infant utterance
Authors: Nakatani Tomohiro , Amano Shigeaki , Irino Toshio
Published: October 2013

Fundamental frequency (F0) and unvoiced/voiced segment (U/V) estimation of infant utterances are important for investigating humans perception of prosodic information in an early stage of speech communication. However, this estimation process is difficult as infant utterances have several features that differ to those of adults: (1) F0 has a wide range in value (200 to 2000 Hz); (2) F0 is unstable, for example, it discontinuously changes to its double or half value; and (3) voiced segments may have high energy in the higher frequency regions degrading U/V decisions of existing methods. Additionally, infant utterance data is often collected in daily child care settings which lowers the signal-to-noise ratio (SNR). To cope with these problems, a robust F0 estimation method based on instantaneous frequency [Nakatani and Irino, ICSLP2002] is introduced, and a new U/V detection method is proposed. The former has a mechanism to extract accurate F0 avoiding double and half pitch errors in low SNR environments. Once accurate F0 is obtained, the latter method can reliably detect U/V just by examining the harmonic structure corresponding to the F0. The effectiveness of this method is examined using a database devised from infant utterances in daycare settings [Amano, Kato, and Kondo, ICSLP2002].
Auditorily motivated elastic spectral distance and its application to emotional morphing of portrayal speech
Authors: Matsui Hisami , Kawahara Hideki
Published: October 2013

An elastic spectral distance measure based on a F0 adaptive pitch synchronous spectral estimation and smoothing, that is developed for a high-quality speech modification procedure STRAIGHT [Kawahara et al., Speech Commun. 27 (1999)], is introduced to provide a basis for emotional morphing. The proposed measure is based on a smoothed nonlinear frequency mapping between the target and the original speech spectra on the ERB axis. A portrayal emotional speech database, which was recorded using professional actors and a recording studio for professional use, was designed for developing and evaluating morphing functions. Target words were recorded under four different contextual conditions (preceding, following and franking carrier sentences and isolated pronunciation), two sentence types (declarative and interrogative). Perceptual effects of constituent parameters of the proposed distance measure will be discussed based on morphing experiments using STRAIGHT procedure. [Work supported by JSPS and NIME, Japan.]
Evaluation of a strategy for automatic formant tracking
Authors: Nearey Terrance M. , Assmann Peter F. , Hillenbrand James M.
Published: October 2013

Variations on an automatic formant tracking strategy developed at Alberta will be compared to manual formant measurements from two databases of vowels spoken by men, women, and children (in Texas or Michigan). ‘‘Correct’’ vowel formant candidates for F1, F2, and F3 may be found roughly 85–90 percent of the time for adult male speakers using autocorrelation LPC with the following settings: F3 maximum at 3000 Hz, LPC order of 14, sampling rate of 10 kHz [J. Markel and A. Gray, Linear Prediction of Speech (Springer, New York, 1975)]. Experience shows good results are also often found with females’ and children’s speech, provided the sampling rate and F3 maximum are scaled appropriately for each speaker. Our new basic strategy involves analyzing each utterance at several distinct sampling rates and coordinated F3 cutoff frequencies with a fixed LPC order. Each scaling choice provides an independent set of candidates that is post-processed by a simple tracking algorithm. A correlation measure between a spectrogram of the original signal and one resynthesized from each estimated track set is defined. This measure is combined with other heuristic figures of merit (based on, e.g., continuity, formant ranges, and bandwidths) to choose the ‘‘best’’ analysis.
Speech enhancement with beamforming enhanced by parametric spectral subtraction
Authors: Cho Jaeyoun , Krishnamurthy Ashok
Published: October 2013

This paper describes a novel method of speech enhancement that combines multichannel beamforming techniques with single channel spectral subtraction. Spectral subtraction methods have been widely used for speech enhancement because they require only a single microphone and provide moderate improvements in the signal-to-noise ratio (SNR). But a major drawback of spectral subtraction is that it inevitably introduces musical noise, which is nonstationary and very annoying. Generally, the musical noise is reduced by smoothing its spectral variance in the frequency domain. Multimicrophone beamforming techniques, on the other hand, enhance the SNR by using spatial and temporal filters to introduce spatial selectivity towards the desired signal. It is shown that these two techniques are complementary, and can be combined to provide better performance than either method alone. One approach, for example, is to use spectral subtraction as a preprocessor in each channel before beamforming. In this case, the summing process of beamforming reduces the musical noise generated by spectral subtraction. We provide examples of using this approach in an automotive application.
Outdoor experiment results for tracking airborne high-speed broadband acoustic sources
Authors: Frazier William G. , Williams Chad , Williams Jay E. , Gilbert Kenneth E.
Published: October 2013

Recent results of tracking airborne high-speed (subsonic) broadband acoustic sources with an experimental system are presented. The system consists of three acoustic arrays, wireless telemetry systems, real-time signal processing and tracking processors, and a tactical operation center-style display unit. The acoustic signal processor consists of a coherence-based detector and generalized cross correlator. The real-time tracking processor uses a modified extended Kalman filter algorithm. In this particular work, the calculation of acoustic sensor-to-source-bearing information will be briefly discussed along with the operation of the source detector, which is adaptive and dependent upon both amplitude and signal coherence. Special emphasis will be placed on how the tracking algorithm is compensated to accommodate unknown and variable data latencies produced by acoustic signals propagating from distant sources that are moving at a significant fraction of the speed of sound. Experimental results include estimates of source position, speed, and heading versus time. Estimates of source state uncertainty and comparison to truth data are also provided. [Work supported by the U.S. Army Space and Missile Defense Command.]
Automatic acoustic mine detection using morphological perceptions
Authors: Ritter Gerhard X. , Gader Paul D. , Hocaoglu A. Koksal , Iancu Laurentiu
Published: October 2013

Recent developments in landmine detection based on the acoustic-to-seismic coupling phenomenon have demonstrated the feasibility to detect both metallic and nonmetallic mines. In this method, a loudspeaker above the ground surface insonifies the target region of the surface. Acoustic energy is coupled into the ground producing ground vibrations. The ground vibration velocity at the ground surface is measured with a laser Doppler vibrometer producing a ground surface image whose pixel values are the measured velocity amplitudes. Vertical particle velocity amplitudes directly above a mine contrast with those of the background (‘‘away from the mine’’). Image processing techniques are used in order to automatically detect regions of interest (‘‘possible mines’’). Further image analysis methods extract geometric as well as amplitude features in order to produce four-dimensional feature vectors. These vectors are input to a novel neural network based on mathematical morphology which classifies the regions of interest into mines and false alarms. After training the network, the network correctly identified all the mines on a given test set with an extremely low alarm rate.
A network, via www, linking laboratories of underwater acoustics
Authors: Gallego Vicente
Published: October 2013

The High Technical School for Naval Weapons Engineers is a teaching institution of the Spanish Navy dedicated to training advanced engineers in the related technologies to the systems of naval weapons. Some scientific disciplines, like the underwater acoustics, are also included in its curriculum. The tasks of research are being developed by means of collaborations with civil organisms such as Institutes of Scientific Research and departments of some universities. In this context, a proposal has been prepared for creating a network for laboratories of underwater acoustics. It was initially formed by the interconnection via www of the Laboratory of Underwater Acoustics in the Naval Weapon School and the Laboratory of hydroacoustics in the Institute of Acoustics, a center of scientifical research pertaining to the Science and Technology Ministry. The joint exploitation of the diverse capabilities from different laboratories, the rational integration of activities and the saving of efforts and costs are the basic ideas for this project. In this paper are presented the general lines that have been considered in the development of this proposal, the future possibilities of using for a network like this as well as the practical limitations addressed by this new concept of networks of laboratories.
Acoustical contamination: Effects on the population of mular dolphin and on their habitats
Authors: Hernandez Ricardo , De Stephanis Renau , Cueto Jose Luis , Neus Rivas
Published: October 2013

In this conference we present the part of the research which has been carried out since July 2002, inside the European Project LIFE-Nature 2002 Ref. LIFE 02/ NAT/E/8610. The research aims at the anthropogenic activities existing in the research area that could affect the studied species in a different way. One of the threats is the acoustical contamination, very strong in the research area due to intensive maritime traffic, especially in the area of Gibraltar. It would be extremely useful to establish the common data base in order to permit the researchers to exchange the data obtained, tests details, and experience. In our case it would be very interesting to follow this type of marine species.
New underwater acoustic tank facility at Georgia Tech
Authors: Gray Michael , Herkert Ralph , McCall George , Caille Gary , Biesel Van , Bogle John , Caspall Jayme , Hahn Steven , Lamb Adam , Logan Thomas , Martin James , Rogers Peter , Trivett David
Published: October 2013

A large underwater acoustic tank facility located in the Woodruff School of Mechanical Engineering at Georgia Tech has recently been completed. The facility includes a rectangular concrete water tank 25 feet deep, 25 feet wide, and 34 feet long containing around 160,000 gallons of water. There are three computer-controlled positioners: an x-y-z-θ positioner and a z-θ positioner mounted on carriages and a bottom mounted rotator. The facility has a large rectangular nearfield array which can be used either as a receiver or a transmitter. A single vertical nearfield line array can be translated by the x-y positioner to synthesize a cylindrical nearfield receiving array. The rectangular nearfield transmitting array and the synthesized cylindrical receiving array were designed to be used with the bottom mounted rotator to measure the true farfield bistatic target strength of any target up to one meter in length as a function of the target aspect angle. Such measurements can be done from 2 kHz to over 10 kHz. The tank is being used for transducer development, materials, and flow noise studies in addition to structural acoustics. Several available multichannel data acquisition systems will be described. [Work supported, in part, by a DURIP grant from ONR.]