bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of recording/playback technique and experimental method on assessments of noise
Authors: Çelik-Christensen Emine , Persson Waye Kerstin , Mo/ller Henrik
Published: May 2008

The study investigated possible effects of recording/playback technique and experimental method on assessments of annoyance, loudness and unpleasantness. A possible effect of exposure duration was also studied. Sounds were recorded with two different techniques: monophonic and binaural (dummy-head technique). In addition, they were reproduced with three different techniques: monophonic recordings presented through a loudspeaker system, binaural recordings presented through closed (circum-aural) and completely open (free-of-the-ear headphones. The study adopted three psychometric methods for collecting responses from test-subjects. Fifty-four subjects participated, and three types of sounds were used: everyday restaurant sound, road traffic sound and ventilation sound dominated by low frequencies. Each sound was played back at three different levels. The results show that there is no significant main effect of recording and playback technique for any of the three perceptual attributes; however, significant interactions between techniques and sounds were found. Since the effect of recording and playback technique differs depending on sound, this finding is of importance for future design of experiments and interpretation of results. The results also show that long-term annoyance and unpleasantness are poorly predicted by short-duration methods.
Speech perception in fluctuating noise with signals compensated for hearing loss
Authors: Festen Joost M.
Published: May 2008

For speech reception in noise normal-hearing listeners gain from masker modulations up to about 12 dB depending on rate, duty cycle, and depth of the modulations. Listeners with sensorineural hearing loss need a better signal-to-noise ratio to improve signal quality as a compensation for their auditory deficits. Generally, a larger compensation is needed for fluctuating interferences leading to reduction or even absence of release from masking for modulated maskers. With the Speech Intelligibility Index adapted for modulated maskers, SIImod [Rhebergen and Versfeld, J. Acoust. Soc. Am. 117, 2181-92 (2005)] it is shown that these elevated thresholds are needed to compensate not only reduced hearing sensitivity but also impaired auditory and non-auditory processing. After frequency-dependent compensation for hearing loss as offered by a hearing aid, more of the speech is presented at impaired frequency regions. As a consequence the effect of a hearing aid on speech intelligibility will be less than predicted by SIImod. Speech reception thresholds in noise are affected by peripheral spectro-temporal processing (bottom-up) and by cognitive processing capabilities, like working memory (top-down). Effects of cognitive processing on speech perception are found especially in fluctuating noise as this masker presents a more complex and demanding environment than steady noise.
Model-based objective assessment of noise reduction systems for hearing aids
Authors: Kollmeier Birger , Huber Rainer , Rohdenburg Thomas , Beutelmann Rainer , Hohmann Volker
Published: May 2008

Since the ultimative goal of hearing-aid development is the (subjective) judgment of the individual hearing-impaired listener, time-consuming tests with the end user are indispensable. However, time- and effort-saving objective methods to assess the potential benefit of different versions and parameter sets of hearing aid algorithms are gaining importance. This contribution reviews perception-model-based approaches to predict the hearing-impaired judgement and speech reception performance achieved with various noise reduction schemes. The perceptual similarity measure PSM evaluates the similarity between a tested condition and an "ideal" reference condition not on the physical level, but rather on the perceptual level at the output of a perception model for the individual hearing-impaired listener. The binaural extention of the SII approach uses a binaural preprocessing stage followed by a speech intelligibility index (SII)- based prediction scheme capable of predicting the relative benefit of binaural signal presentation and signal enhancement in complex spatial signal and noise source configurations. Both model-based schemes can be combined to assess the effect of noise reduction algorithms (such as adaptive beamformers) and to optimize their respective performance for different acoustical situations.
Quantifying and modeling the acoustic effects of compression on speech in noise
Authors: Rhebergen Koenraad S. , Versfeld Niek J. , Dreschler Wouter A.
Published: May 2008

In this presentation a method is proposed that is able to separate a speech signal out of a noise signal after processing of the signal through wide-dynamic-range compression (WDRC). This technique reconstructs the speech signal and noise signal sample by sample separately using the gain factor of the WDRC, and can be used to quantify the acoustic effects of WDRC in noise. It will be shown that this technique is more accurate than a frequently used inversion technique, because the method is not affected by phase shifts that introduce distortion products in the reconstructed speech signal. As a result, the acoustic effects of WDRC can be measured more accurately. In addition, this reconstruction method allows modeling the speech intelligibility after non-linear signal processing in the Speech Intelligibility Index. With the aid of Speech Reception Threshold data it will be shown that this approach can give a good account for most existing data.
Combining hearing aids and cochlear implants to solve the cocktail party problem
Authors: Zeng Fan-Gang
Published: May 2008

The cocktail party problem refers to the difficulty in speech recognition in noise that a hearing-impaired listener must face in daily life. Combining a hearing aid with a cochlear implant can provide complementary information that may have a great potential to solve this problem. On one hand, a hearing aid may provide low-frequency temporal fine structure cues that are not conveyed by a cochlear implant. On the other hand, a cochlear implant can provide high-frequency temporal envelope cues that are not effectively delivered by a hearing aid. This talk will provide psychophysical and speech recognition evidence for combining hearing aids and cochlear implants to solve the cocktail party problem. One interesting finding along this line of research is that in many important functional tasks, the hearing aid and cochlear implant combination provides a more effective solution than bilateral cochlear implants. Another interesting finding is that the fundamental frequency cue alone can significantly improve speech perception in noise, especially when the noise is a competing voice. The latter finding suggests that combining a tactile aid and a cochlear implant can potentially achieve the same benefit as combining a hearing aid and a cochlear implant in patients with no residual acoustic hearing.
Auditory localization with linear and compression hearing aids
Authors: Simon Helen J. , Yund E. William , Levitt Harry
Published: May 2008

The question of how well hearing-impaired individuals can localize sound (with or without amplification) is still not fully resolved. This study was designed to compare sound localization with two types of hearing-aid (HA) processing, wide dynamic range multichannel compression (WDRMCC) and linear amplification (LA) with compression limiting, during the first 32 weeks of HA use. HAs from two different manufacturers were included to compare different digital signal processing implementations, (1) fast Fourier transform (FFT), necessitating a 10 ms delay, and (2) non-FFT signal processing with a shorter time delay (̃1 ms). We found an initial degradation of sound localization, relative to original unaided performance, for both WDRMCC and LA in both FTT and non-FTT platforms. We found no difference between WDRMCC and LA processing. However, sound localization with non-FFT platform improved consistently throughout 32 weeks of HA use and was better than the original unaided measurements at 16 and 32 weeks. In contrast, localization with the FFT platform showed no consistent change throughout the 32-week test period and remained inferior to original unaided performance. The continuing localization problems present for the FFT, but not the non-FFT, implementations of LA and WDRMCC may be due to its 10-ms processing delay.
Effects of noise type and location on binaural benefit in asymmetric directional fittings
Authors: Hornsby Benjamin
Published: May 2008

The benefits of bilateral directional processing for improving speech understanding in noise are well documented. However, these fittings are not universally accepted by hearing aid wearers. Research suggests that an asymmetric fitting (omnidirectional in one ear/directional in the other) may provide benefit in noise comparable to symmetric directional fittings (directional in both ears). This study evaluated factors that may affect the relative benefit provided by an asymmetric directional fitting. Specific factors evaluated included noise configuration, reverberation and noise type. Twenty individuals with mild-moderate SNHL participated. Aided speech understanding in cafeteria babble was assessed in bilateral omnidirectional and directional modes and in an asymmetric mode in four (4) different noise configurations. Measures were made in both an anechoic and reverberant (RT ̃620 ms). In a second experiment the effects of noise type were evaluated by comparing performance in symmetric and asymmetric modes in both steady state noise and cafeteria babble. Results suggest that noise configuration has a significant effect on the relative benefit provided by asymmetric fittings. No significant differences due to noise type were observed. Individual differences in bilateral directional benefit also appear to have a significant effect on the reduction in benefit resulting from an asymmetric fitting.
Effect of hearing aids on distance perception
Authors: Akeroyd Michael A.
Published: May 2008

The two primary auditory cues to distance in rooms are the overall level of the sounds received by the listener and the ratio of the level of the direct sound to the reverberant sounds. In previous work we have demonstrated that hearing-impaired listeners showed no overall deficits in the ability to use the overall-level cue, but they did have deficits in the ability to use the direct-to-reverberant cue [M. Akeroyd, S. Gatehouse, and J. Blaschke, J. Acoust. Soc. Am., 121, 1077-1089 (2007)]. These deficits would be expected to contribute to the auditory disability suffered by the listeners. But both of these auditory cues are level cues, as they require some measurement of intensity, and so they would be expected to be affected adversely by the amplitude compression found in most modern hearing aids. Initial results from an experiment measuring the just-noticeable difference (JND) for changes in distance (at 2 m and 5 m) with experienced hearing-aid users suggest, however, that their JNDs are no different to those of unaided but impaired listeners. If confirmed by the final results, this surprising result may be interpretable if listeners have acclimatized to the effects of their aids on level.
On the comparison of symmetric and unsymmetric formulations for experimental vibro-acoustic modal analysis
Authors: Ouisse Morvan , Foltete Emmanuel
Published: May 2008

The classical u-p formulation for vibro-acoustic problems is very convenient for experimental vibro-acoustic modal analysis since the physical variables are directly those which are measured by operators. In this particular context, the objective is to identify from experimental measurements a reduced model which has the same behaviour as the measured one. The complex mode shapes which are identified using this technique must satisfy a properness condition. When they do not verify it, they should be modified to be able to represent the behaviour of a physical system. Some techniques have been proposed in order to develop a strategy to obtain the modified eigenshapes, but this is a quite difficult point because of the unsymmetric topology of the equations. In this paper, a symmetric formulation is used in order to be able to directly apply the classical methodology which has been developed for structural modal analysis to obtain the physical reduced system. The methodology is described and compared with the u-p formulation, in terms of efficiency and precision, in particular when some absorbing devices are considered. All results are first presented on an ideal numerical test-case, and applications on experimental data are finally shown.
Three dimensional orthogonality of the Lamb modes in layered plates of elastic and viscoelastic materials and their implementation to the far field evaluation
Authors: Zakharov Dmitry
Published: May 2008

The 3D guided waves in the linearly viscoelastic laminates are considered. On the plate surfaces any of the homogeneous boundary conditions are allowed, e.g., the Lamb waves, waves in clamped plates, etc. are taken into account. The fundamental property of these waves is their generalized orthogonality, which is deduced and discussed. The applications of the orthogonality relations for solving some particular boundary value problems are suggested. A method for the exact calculation of the far field caused by an acoustic source of a finite size is suggested. The only restriction is that the distance required must exceed the longitudinal radius of the source. The obtained results can be used for evaluating the fields radiated by ultrasonic transducers of arbitrary aperture and by other realistic sources.
Sound Radiation of a Large Truck Oil Pan: Estimation and Experimental Investigation
Authors: Heintze Olaf , Wittstock Volker , Hartung Carl Fredrik
Published: May 2008

The oil pan of large diesel engine trucks has been identified as a significant contributor to their external noise radiation. This undesired broadband noise is caused by the oil pan's structural vibration and can not be treated effectively by passive measures especially in the low frequency regime up to 500Hz. In order to address this challenge, an active structural acoustic control system consisting of structural sensors and actuators is suitable to alter the oil pans vibrations in a sound reducing manner. A first step is however to classify the broadband sound radiation such that it allows for a proper and efficient sound power estimation resulting from structural measurements. Therefore, an acoustical model was set up based on a geometrical surface scan of a serial production large truck oil pan mounted in a laboratory test stand. This model served for the numerical computation of a set of principle velocity patterns contributing independently to the active sound power, where its hybrid estimation has been performed employing additionally the measured structural response of the oil pan assembly due to a shaker excitation. Finally, the sound power radiation of the test stand has been measured in a reverberation room to validate this hybrid estimation.
Wavelet-based data processing for comparative study of noncontact measurement techniques for vibroacoustics
Authors: Borza Dan , Nistea Ioana
Published: May 2008

In vibration analysis of structures complex measurement information is required in order to perform modal analysis, therefore the choice of the measurement technique to be applied is essential. Non contact measurements are preferred to classical transducer based methods, mainly due to the absence of influence upon the structure under test. The various techniques available today in vibroacoustics produce results which are quite different in terms of spatial and temporal resolution or measured quantities and therefore a choice has to be made of the experimental tool best adapted for different fields of research (acoustics, mechanical structures, dynamics). In this paper, we present a comparative study of several optical, acoustical and numerical techniques for vibration measurement or simulation, namely Digital Speckle Interferometry, Laser Doppler Vibrometry and a FE model. The tests were made for the free and the forced vibrations of a highly dumped, non-metallic plate. In the data processing stage, discrete wavelet decomposition has been applied on the experimental data in order to match up the spatial maps of vibration amplitudes.
Weak radiator design using dimples
Authors: Cheng D. Chih-Chun , Cheng Wen-Nan , Koopmann Cary H.
Published: May 2008

A design method for achieving minimum radiation of sound from a beam by creating cylindrical dimples on its surface is presented. Two strategies of determining the dimple size and its location on the beam are presented. The first is based on the optimization method, in which the dimple size and its location are the design variables. The design variables that minimize the sound power are obtained directly using an optimization subroutine. The second is to synthesize the beam's weak radiator mode using a set of dimples, in which the mode shape of the dimpled beam is a close fit to the beam's weak radiator mode. As a comparison in sound power reduction between these strategies, numerical results for a simply supported beam in a rigid baffle excited by a harmonic force are presented.
Estimation for vibration mode of membrane by NAH method
Authors: Ibrahim Nabilah , Shibayama Hideo , Itakura Toru
Published: May 2008

Near Field Acoustic Holography (NAH) is the method of measuring sound information on the radiant surface in the near field as a hologram. By use the technique, we can visualize the vibration of the radiant surface. Thus, we can easily understand the object of the acoustic conditions in acoustic image processing. We made the measurement on one side of the drum using array microphone while the driven source is attached to the other side. In addition, the measurement is not touched on the vibrating membrane surface. This paper describes the estimated results of the vibration mode on the vibration membrane of a drum. And this is possible by apply it to the vibration controlling field for detecting the sound structure on the membrane.
Computing high frequency vibrations of polygonal plates by the Image Source Method
Authors: Cuenca Jacques , Gautier François , Simon Laurent
Published: May 2008

The aim of this paper is to show that the Image Source Method (ISM) can be used for predicting medium and high frequency vibrations of arbitrarily shaped polygonal plates with controllable precision. Modeling the vibrations of polygonal plates by ISM consists in superposing the contributions of elementary image sources, which allows the plate boundary conditions to be satisfied. For particular plate shapes leading to a spatially periodic pattern of image sources, it is shown that ISM allows the exact computation of the Green's function by means of a Poisson summation formula. For arbitrarily polygonal plates, a criterion for truncating the image source generation process is examined and used as a control parameter for the precision of the estimations. The given examples show that the results are in good agreement with exact analytical solutions for simple plate shapes and with results obtained by the finite element method for more complex shapes. The convergence towards reference solutions as the number of image sources is increased is investigated. The main advantage of the method is that the accuracy is improved with structural damping and with frequency for a given number of image sources.
Acoustical Effectiveness of Damping Coating
Authors: Fischer Raymond , Boroditsky Leonid
Published: May 2008

Damping coatings have again become popular in the shipbuilding industry to reduce onboard and underwater noise. Understanding of the actual effectiveness of damping tiles applied to ship structures is important for noise prediction during the design stage and for optimizing noise control. The commonly held opinion that energy loss is proportional to loss factor values is not confirmed by tests conducted on actual structures. This paper discusses differences between loss factors measured on the Oberst beam and on two-dimensional real structures. The paper shows how loss factor change may influence transmission loss, radiation efficiency and acceptance. The results of loss factor measurements in different conditions including water-loaded media are demonstrated. influences of resonant and non- resonant modes on damping tile effectiveness is discussed. Corrections for SEA noise prediction algorithms, taking into account results of this research, are proposed.
On the prediction of absorption coefficient of porous materials with Finite Elements
Authors: Langer Sabine , Wulkau Meike
Published: May 2008

The absorption coefficient gives the relation between incited and reflected sound power on a surface and is a measure for the damping properties of materials. Experimentally it can be investigated by using the impedance tube for perpendicular incitation and the echo chamber in the case of a diffuse sound field as is known. To optimize the sound absorption properties of materials in the pre-prototype stage, numerical simulation can support the design of materials. A detailed finite element simulation based on Biots Theory for poroelastic-media is used to predict the absorption coefficient of materials with open-pored surfaces. Prospects and limits of this strategy are discussed.
Structure-borne sound transmission from machines into ribbed structures
Authors: Gibbs Barry M. , Mayr Andreas R.
Published: May 2008

The total structure-borne sound power of an installed machine is a function of the source activity and mobility, and the receiver mobility, at each contact and for each component of excitation. The data and computational requirements for prediction therefore are large. Manufacturers view their products as single entities and desire corresponding single values of source strength. A laboratory reception plate measurement procedure has been proposed which yields single equivalent values of source strength and source mobility. The source data, in combination with an estimate of the single equivalent value of receiver mobility, yields the approximate total installed power. The accuracy of the estimate is dependent on the spatial variation in contact conditions over the connections. In addition, phase information has been lost in the simplification. Case studies are described for multiple contact sources on a non-homogeneous plate (a timber-joist floor) where the approximate estimates of structure-borne power are compared with exact values obtained from full mobility formulations.
Measurements of the bending moment at boundaries of a structure
Authors: Chesne Simon , Chomette Baptiste , Pezerat Charles
Published: May 2008

In the vibration transmission process, the part due to the moment excitation is often neglected, because of the difficulty to measure them, even if these terms are sometimes important. Indeed, several studies show that the influence of the moment in structure borne power transmission can be higher than simple force. Moment measurement or identification is an old problem especially at boundaries. It has been mainly investigated in mobility methods. In fact, bending moment expression can be seen as spatial derivatives of displacements. These derivatives can be approximated from measured displacements but two major difficulties appear: derivatives are highly sensitive to measurement errors and the usual methods used to obtain them (finite differences, modal approach, etc.) are not well adapted at boundary points. In this paper, three different moment identification methods are investigated where the considered structure is a beam. Both of these approaches are already developed by authors, the third method is based on the use of particular mechanical and electric setups of piezoelectric patches. These 3 methods are numerically and experimentally implemented allowing us to compare and discuss on results.
Measurement of Speech Intelligibility Using Low Level Output - Threshold Efficient S/N Ratios
Authors: Nestoras Christos , Dance Stephen
Published: May 2008

Excessive background noise levels or perhaps annoyance due to a high test signal level are the main reasons for having a low signal to noise ratio (S/N) or low level output during a measurement session. A significant error, in this sense, can be introduced in the resulting data since the measurement technique requires, among others, a minimum S/N ratio for an accurate result. A validation is presented of low level output intelligibility measurements in an attempt to establish a point of reference for the verification of data accuracy for a given space. An indication of the functions taking place in this respect could be obtained through practical experimentation. Results are reported here.
Vowels recognition using mellin transform and plp-based feature extraction
Authors: Jamaati Mahdi , Marvi Hossein , Lankarany Milad
Published: May 2008

Feature extraction for speech recognition is a subject of majer interest today. Different feature have been investigated in speech recognition systems. The scale transform is a particular restriction of the Mellin transform. The key property of the scale transform is the scale invariance.The mel frequency(MFCC) and perceptual linear predicate (PLP) have usually reported to have yielded good performance. In this paper, a new method is presented which combines feature extracted from mellin transform with the plp features. To evaluate the performance of the proposed features the comparative tests with the lpc features, MFCC and PLP were performed on the vowels based speech recognition system. Preliminary experiment show that this approach posses promissing result.
Modelling acoustic parameters of prosody for read and acted-speech synthesis
Authors: Rusko Milan , Trnka Marián , Darjaa Sakhia , Kováč Richard , Hamar Juraj
Published: May 2008

The prosody model is one of the most important parts of every speech synthesizer, influencing mainly its naturalness. The intonation contour and phoneme lengths (together with speech quality) bear a great deal of extralinguistic and paralinguistic information contained in the synthesized speech. The features reflecting personality, mood and emotions of the speaker are in strong interaction with those reflecting speech styles. Anyway the appropriate choose of prosody model and training material can make it possible to create special model for every speaking style. The paper presents our approach to modelling of acoustic parameters of prosody in two different speech styles in Slovak. Our model is based on Classification and regression trees (CARTs). It uses independent CART for phoneme lengths and three CARTs for fundamental frequency (F0) at the beginning, centre, and end of every syllable. Two hours of read speech were used for training a model of read speech. The recordings of a puppet player were used to train a model of acted speech. The models were implemented in the Kempelen 2.2 unit selection Slovak speech synthesizer. The listening tests have shown that the models are capable of modelling significant amount of the differences of the two speaking styles.
Action-based multisensory integration in striking events
Authors: Giordano Bruno L. , McAdams Stephen , Crosato Paolo , Avanzini Federico , Casciato Carmine , Sinclair Stephen , Wanderley Marcelo M.
Published: May 2008

Many theories of multisensory integration focus on action-independent perception. This approach has limited ecological validity, because under everyday conditions we frequently act upon multisensory events through motor behavior. We present a novel methodology for the study of multisensory integration in action-directed perception, focusing on striking events. Observers repeatedly hit a virtual object with a target striking velocity, and are presented with various types of sensory information about the striking event: auditory and/or haptic and/or visual. For each of the experimental trials, they initially receive feedback on the tracking of the target striking velocity. In a second phase, feedback is eliminated. In a third phase, the sensory properties of the striking event are changed. We quantify the extent to which a variation in each of the sensory modalities disrupts performance in the tracking of the target striking velocity. Multisensory dominance hierarchies are inferred from the comparison of the patterns of performance disruption in unimodal and multisensory contexts.
Cross-modal synchrony perception reveals aspects of categorical perception
Authors: Kohlrausch Armin , Van Eijk Rob L. , Van De Par Steven , Juola James F. , Vitevich Michael
Published: May 2008

The phenomenon of "categorical perception" has played an important role in speech research. When a specific (combination of) feature(s) of a speech stimulus is varied along a physical dimension, categorical perception is reflected by two observations: (1) the percept of the sound changes abruptly from one category to another, e.g., from "ba" to "da" to "ga" for changes in formant transition frequencies, and (2) physical changes of a given amount lead to more easily perceivable differences for stimuli close to a category boundary, compared to stimuli in the center of a category. In this talk, I present data about audio-visual synchrony perception, which indicate that perceived synchrony also reveals properties of categorical perception. Depending on the physical delay between the auditory and visual component, the percept changes from "audio first," to "synchronous" to "video first." When measuring sensitivity to changes in audio-visual delays, we observe that the thresholds are small for stimuli at the transitions between the earlier determined categories, while they are larger for stimuli within a category.
Do we perceive articulatory gestures when we listen to speech
Authors: Divenyi Pierre , Lammert Adam
Published: May 2008

Articulatory synthesis methods, classic and contemporary, have demonstrated that it is possible to generate speech from an ensemble of functions derived from articulatory gestures. Such gesture-to-waveform transforms suggest that, inversely, the speech signal should be also decomposable into the same set of gesture, or gesture-like, functions. These functions vary slowly in time and their association with the speech waveform (words as well as sentences) can be established by machine learning algorithms. In a recent study at our laboratory, listeners were asked to type the word or the sentence they heard, with speech (degraded in diverse ways) as the stimulus. The subjects' responses were synthesized, time-aligned with the stimulus, and decomposed into a set of eight gestures, as specified by the Haskins Laboratories TADA system (http://www.haskins.yale.edu/tada_download/index.html). When the running distance between input and response gesture functions is calculated, results indicate a significant degree of gesture information transmitted even during severely degraded speech segments, suggesting that the perceptual system may track speech via underlying functions similar to gestures. Epochs at which this running distance estimate fails, i.e., exceeds a certain threshold, may be considered to signal periods during which insufficient bottom-up information had to be supplemented using higher-order linguistic knowledge.
Exploiting confidence measures for missing data speech recognition
Authors: Cerisara Christophe
Published: May 2008

Automatic speech recognition in highly non-stationary noise, for instance with a competing speaker or background music, is an extremely challenging and still unsolved problem. Missing data recognition is a robust approach that is well adapted to this kind of noise. A standard missing data technique consists in marginalizing out, from the observation likelihoods computed during decoding, the contribution of the spectro-temporal fragments that are dominated by noise. However, such an approach can hardly be applied to advanced parameterization domains that do not separate speech from noise frequencies, such as the cepstrum or ETSI AFE. We propose in this work to extend this technique to such parameterization domains, and not only to spectrographic-like front-ends as it was the case before. This is realized by masking the observations that favor erroneous decoding paths, instead of masking the features that are dominated by noise. These new missing data "masks" are now estimated based on speech recognition confidence measures, which can be considered as indicators of the reliability of decoding paths. A first version of this robust algorithm is evaluated on the French broadcast news ESTER corpus.
An MTF-based blind restoration of temporal power envelopes as a front-end processor for automatic speech recognition systems in reverberant environments
Authors: Lu Xugang , Unoki Masashi , Akagi Masato
Published: May 2008

To reduce speech degradation in reverberant environments, we previously proposed a modulation transfer function (MTF) based method for speech dereverberation. It is based on the MTF relation that the sub-band temporal power envelope of reverberant speech can be represented as the convolution between temporal power envelopes of clean speech and the room impulse response. Therefore, the sub-band power envelope of clean speech can be estimated using inverse MTF filtering without measuring the room impulse response. We tested the effectiveness of this method as a front-end for automatic speech recognition (ASR) in both artificial and real reverberant environments. Reverberant speech signals were created by simple convolution of clean speech (AURORA-2J) and artificially-produced or real room impulse responses. The relative spectral filtering of the auditory-power-spectrum based method was used as a baseline. Compared with the baseline, our proposed method had 36.64% and 21.68% improvements in error reduction rate for artificial reverberant environments (reverberation times from 0.2 to 2.0 s) and real reverberant environments (43 reverberant impulse responses), respectively. These results indicate that our proposed method can be used as a robust front-end for ASR. [Work supported by a Grant-in-Aid for Science Research from the Japanese Ministry of Education (No. 18680017).]
Speaker adaptation combined with missing data reconstruction
Authors: Remes Ulpu , Palomäki Kalle J. , Kurimo Mikko
Published: May 2008

Missing data methods offer an attractive framework for automatic speech recognition (ASR) under time-varying and unpredictable noise conditions. The motivation for the methods comes from the human speech perception and auditory scene analysis, studies on which have suggested that the more intense sound components in an auditory scene dominate the neural response to the scene. The missing data methods developed for ASR divide the noisy speech signal to speech and noise dominated regions. Speech components in the noise dominated regions are viewed as missing but may be reconstructed based on the observed components. Missing data methods have performed well in previous studies with noise corrupted speech but have not become a standard in ASR. Especially considering ASR on large vocabularies, the methods have a disadvantage: it is not straightforward to combine them with common speaker adaptation methods such as constrained maximum likelihood linear regression (CMLLR). In this work, we investigate efficient ways to combine missing data reconstruction with speaker adaptation in large vocabulary ASR task with speech data recorded in noisy real-world environments. The experiments show that speaker adaptation combined with the missing data reconstruction can improve the missing data approach in noisy large vocabulary ASR.
Ocean acoustic time reversal
Authors: Kuperman William A.
Published: May 2008

A series of ocean acoustic time experiments have been jointly conducted in the last ten years by the NATO Undersea Research Centre and the Marine Physical Laboratory. The experiments demonstrated a robustness of the ocean acoustic time reversal mirror process similar to that found in more controlled, ultrasonic laboratory experiments. Results from these experiments have pointed to potential applications to SONAR signal processing methods and to acoustics communications. In addition, the experimental procedures themselves have yielded data that provide further understanding of the ocean acoustic environment and the increased stability of time reversal process over one way propagation in a fluctuating medium.
Model-based time reversal method for photoacoustic imaging of heterogeneous media
Authors: Burgholzer Peter , Gruen Hubert , Nuster Robert , Paltauf Günther , Haltmeier Markus
Published: May 2008

In photoacoustic (also called optoacoustic or thermoacoustic) tomography acoustic pressure waves are generated by illumination of a semitransparent sample with pulsed electromagnetic radiation. Subsequently the waves propagate towards the detection surface enclosing the sample. The inverse problem consists of reconstructing the initial pressure sources from those measurements. In certain applications of photoacoustic imaging one has to deal with media with spatially varying sound velocity, e.g. bones in soft tissue. Image reconstruction without any compensation of this effect leads to a poor image quality. It is therefore essential to develop reconstruction algorithms that take spatially varying sound velocity into account and are able to reveal small structures in acoustically heterogeneous media. A model-based time reversal reconstruction method is presented that is capable of reconstructing the initial pressure distribution despite variations of sound speed. This reconstruction method calculates the time reversed field directly with a second order embedded boundary method by retransmitting the measured pressure on the detector positions in reversed temporal order. Numerical simulations and experiments with phantoms consisting of areas with spatially varying sound velocity are presented.
The effects of transducers on the time reversal process in solids
Authors: Anderson Brian E. , Griffa Michele , Johnson Paul A.
Published: May 2008

Every experimental implementation of Time Reversal (TR) involves the use of transducers to convert wave motion, whether mechanical or acoustical, into electrical signals, and vice versa. Practical considerations of transducers are not included in the basic theory of time reversal, which is based on idealized point-like sources. These considerations include temporal ring down at a narrowband transducer resonance, the finite size of the transducer giving rise to directivity, and the impedance contrast between the transducer and the medium. The effects of these considerations on the TR process will be characterized by presenting data from various TR experiments.
Estimation of guided waves from cross-correlations of diffuse Wavefields for passive Structural Health Monitoring
Authors: Duroux Adelaide , Sabra Karim G. , Ruzzene Massimo , Sharma Vin , Ayers James
Published: May 2008

Recent theoretical and experimental studies in a wide range of applications (ultrasonics, underwater acoustics, seismicœ) have demonstrated that Green's functions (impulse responses) can be extracted from cross-correlation of diffuse fields using only passive sensors. The technique, whose validity is supported by a physical argument based on time-reversal invariance, effectively uses a correlation process between the point source and points located in the focal zone. Indeed, the coherent noise source distributions can be considered as a time-reversal mirror and the cross-correlation operations gives the field measured at one receiver after refocusing on the other receiver. Passive-only reconstruction of coherent Lamb waves (80-200 kHz) in an aluminum plate and thickness comparable to aircraft fuselage and wing panels will be presented. In particular, the influence of the noise source characteristics (location, frequency spectrum) on the signal-to-noise ratio the emerging coherent waveform will be investigated using a scanning laser Doppler velocimeter. This study suggests the potential for a structural health monitoring method for aircraft panels based on passive ultrasound imaging reconstructed from diffuse fields.
Separation of single and multiple scatterring: Application to the ultrasonic detection of a target embedded in a diffusive medium
Authors: Aubry Alexandre , Derode Arnaud
Published: May 2008

We present a new imaging technique dedicated to the detection of a target embedded in a strongly scattering medium. Classical imaging techniques such as echography fail in this kind of configuration because of multiply scattered echoes and aberration distorsions. The experimental set up consists in an array of programmable transducers placed in front of a collection of steel rods randomly distributed. The slab thickness is three times the mean free path. Behind this strongly diffusive slab, we set an echogene steel cylinder that we want to detect and localize. The impulse responses between each couple of transducers are measured and form the interelement matrix. Our technique separates the single-scattered echoes from the multiple scattering background. This is possible because of a deterministic coherence of single-scattering signals along the antidiagonals of the array response matrix, whatever the distribution of scatterers. Once this operation is performed, the detection of the target is achieved by applying the DORT method (French acronym for decomposition of the time reversal operator). The quality of detection is assessed theoretically with Random Matrix Theory and shown to be, by far, better than what is obtained with echography and the classical DORT method.
Time reversal processing to forward scattering waves of underwater targets
Authors: Tsurugaya Yoshiaki , Kikuchi Toshiaki , Mizutani Koichi
Published: May 2008

This paper presents the detection of underwater targets using a time reversal. When a target exists between a sound source and a time reversal array in shallow water, the time reversal array receive the sound waves from the sound source and the waves scattering by the target. If the time reversal processing to them is carried out and they are re-transmitted from the time reversal array, it will be thought that they are converged at the position of the sound source and the target. However, since the waves converging at the sound source have a high level, the waves converging at the target position are usually masked by the high level sounds. Then, we cannot observe the waves converged at the target. We eliminate only the high level sounds from the sound fields. In each array element, the signals in case of non target are subtracted from the signals including the target. As a result of subtraction, the components of the scattering wave by the target are left on the array elements. The time reversal fields of the scattering wave are constructed by radiating the components of the scattered waves from each element again.
Saclant ASW Research Centre contributions to underwater acoustics during the first sixteen years of its existence: Personal records
Authors: Laval Robert
Published: May 2008

In less than one year the NATO Undersea Research Centre, previously called Saclant ASW Research Centre, will celebrate its 50th anniversary. I had the great privilege to be one of the first scientists joining the Centre in 1959 and to work there until 1975. The main purpose of the research conducted in underwater acoustics was to reach a physical understanding of the different processes of sound propagation, including multipath effects, transformations introduced by a multilayered bottom and random scattering by the volume inhomogeneities and by the surface and bottom roughness. Experiments at sea first conducted from the research ships Aragonese and then Maria Paolina used explosives charges and active sonars FM pulses as sound sources. The digital analysis equipment, which had been designed to record, process and facilitate the interpretation of the received signals, had no equivalent at the time. The Centre has been a very active platform of intercommunications for most of the civilian and military organizations, which were involved in this discipline. The result of this continuous exchange of ideas has facilitated the harmonization of the research programs in the different NATO countries and has greatly contributed to the creation of a large international research community in underwater acoustics.
Shallow water environmental acoustics at NURC/SACLANTCEN
Authors: Kuperman William A.
Published: May 2008

The Centre has a long tradition of studying shallow water acoustics. The research has been experimental and theoretical, the latter effort often involving either using or developing models. Examples include innovations such as the concept of the optimum frequency of an acoustic channel, first measured and then precisely explained with modeling, to, for a recent example, the utilization of ocean ambient noise based on its special properties. Consequently, the Centre has continued to remain at the leading-edge for a half century. The common goals at the Centre that provided the impetus for the progression of research are presented. Then a brief historical review of a selection of the contributions the Centre has made in shallow water acoustics is given.
My personal perspective of sonar research at the NATO Undersea Research Centre
Authors: Wille Peter
Published: May 2008

As the first German Director of the NATO Undersea Research Centre I was contemporary to the fall of the Berlin Wall in November 1989 and the unification of Germany in October 1990, the breathtaking termination of the Cold War without bloodshed. NATO's Undersea Research Centre, thirty years of age when the Berlin Wall fell, has been a key player and focal point of a most important and demanding field of defense research. It has been exemplary in combining customer oriented applied research on sonar concepts up to system demonstrators with basic research to understand, to model and to predict the variability of the ocean environment, the most difficult challenge of successful surveillance and reconnaissance. Some unique capabilities enable the Centre to serve the nations and the NATO commands: Above all, the ability to attract first class scientists to work together at the Centre for a few years and then return to their home countries, forming an ever growing international network of intense scientific cooperation. The advanced sea trial related technology and data treatment realized by a capable and flexible technical staff is no less decisive. The many high-ranking visitors, scientific, military and political underline the international standing of the Centre.
19 years in the SCNR for NURC: A recollection
Authors: Bjo/rno/ Leif
Published: May 2008

I joined the SCNR for NURC by January 1st 1980 and left it again by the end of 1998. This period, on which I will concentrate my talk, includes the toughest period of the "Cold War" up to 1989 and the period after, when money to NURC became sparse and new ways of funding had to be found. This presentation will emphasize some of the highlights in NURC's contributions to Underwater Acoustics, ASW, MCM and to engineering aspects of underwater research. These highlights include for instance towed arrays, underwater acoustic modelling, MILOC, the MCM and the REA programmes, time-reverse underwater acoustics, operations research and engineering of underwater instruments. These important contributions over 19 years were supervised by 10 chairmen of the SCNR and by 6 directors, all putting their special fingerprint, closely related to their personality, on the course of NURC's development. A few humoristic features and experiences from the 19 years will briefly be referred to.