bluuebunny/crossref_metadata_2025 huggingface dataset:
Mucosal wave analysis of a human vocal fold in the hemilarynx experiment
Authors: Doellinger Michael , Berry David A. , Lohscheller Jörg
Published: May 2008

The mucosal wave propagation was investigated in laboratory experiments across a variety of phonatory conditions. The focus was on the medial and superior surface dynamics of the vocal fold, which quantify mucosal wave propagation, but have been relatively little studied. High-speed, digital imaging of the entire surface of the vocal fold was performed using an excised human hemilarynx setup. Surface dynamics were characterized and differentiated across a variety of phonatory conditions. During sustained, flow-induced oscillation, the local maxima of vocal fold mucosal displacements, velocities and acceleration and their particular phase delays in the glottal cycle were investigated. Statistical analysis was performed, examining the influence of applied stimulations. Increasing the airflow yielded higher values for lateral displacements as well as higher velocity/acceleration values. Elongating the vocal fold resulted in decreased lateral displacements. The mucosal wave propagation apparently increased for higher flow, elongated folds, and higher adduction forces. While an understanding of the correlation between vocal fold dynamics and phonatory physiology/pathology is still in its infancy, the data presented here help to establish such connections. The data are also useful for the development and evaluation of physical and numerical models of vocal fold vibration.
Discrimination of Mandarin tone 1 vs. tone 4 in disyllables by adult speakers of English
Authors: Berkowitz Shari
Published: May 2008

Previous research on cross-language perception of lexical tone has mainly used monosyllabic stimuli; however disyllables may be more difficult for non-natives to discriminate due to coarticulation and context effects (Berkowitz & Strange, 2007). Preliminary work suggests that there is an effect of context on discrimination of tone 1 vs. 4, despite the fact that this is usually considered an easy contrast when tested with monosyllabic stimuli. Disyllabic Mandarin nonsense words served as stimuli in a categorial same/different task. Pairings of tone 1 and tone 4 in initial and final position were tested in the context of all four tones. American English listeners with no background in tone languages completed the experiment without feedback. Data was scored with A' and was analyzed for effect of tone, initial vs. final position in the disyllable, and height and contour of the fundamental frequency. The results of this paradigm will be used to design a study of tone perception in preschool groups who speak Mandarin as their L1, English as their L1, and internationally adopted children who were previously exposed to Mandarin.
Articulatory features influencing regressive place assimilation across word-boundaries in German
Authors: Jaeger Marion , Hoole Phil
Published: May 2008

Within current phonological theories the greater tendency of C1 nasals vs. C1 plosives to undergo regressive place assimilation is often treated as the consequence of acoustic-perceptual properties of nasality (e.g. Steriade, 2001). Little is known about the articulatory patterns underlying this asymmetry. Our current EMA study aims to test and compare the effects of manner of articulation of C1 (alveolar nasal vs. alveolar plosive), place of articulation of C2 (labial vs. dorsal plosive), vowel context (palatal /i/ vs. non-palatal vowel /a/), and word frequency upon the intra- and intergestural timing and movement magnitude of various articulators in C1C2 sequences across word-boundaries in German subjects. Our analyses of non-palatal vowel contexts in three speakers showed a greater likelihood of reduction of the tongue tip both in words with a nasal C1 and in high frequency words. For those word pairs in which tongue tip displacement was measurable, tongue tip - tongue back overlap was significantly greater in word pairs with a nasal C1 and in word pairs with high frequency words. On the other hand, tongue tip - lower lip overlap was only significantly greater in word pairs with high frequency words.
Adaptive microphone array free of the desired speaker cancellation combined with postfilter
Authors: Jovicic Slobodan , Saric Zoran
Published: May 2008

The optimal microphone array includes two processing blocks - minimum variance distortionless response (MVDR) beamformer and the single-channel Wiener filter, which acts as post-filter. The main drawback of MVDR beamformer is the cancellation of the desired speech signal and its degradation in multi-path wave propagation environment. To make the adaptive algorithm robust against room reverberation and to prevent desired signal cancellation, an estimation of the unknown desired speaker's transfer function was proposed. The estimation is based on the imperfect signal and the interference covariance matrices estimated from available microphone signals during speaker activity and pause of speech respectively. As MVDR beamformer suppresses coherent interference, post-filter has to reduce diffuse acoustic noise. The post-filter proposed in this paper is developed under assumption that complex coherence function is unknown but time invariant. The additional improvement of the post-processing algorithm on low frequencies is obtained by combining a priori noise power attenuation factor for diffuse noise field with estimated one. The proposed algorithm is tested on simulated room with reverberation, and compared with some known post-processing algorithms with rather good results.
Model of the configurations of intonation movements in Danish - suggesting an 'intoneme'
Authors: Frovin Sophia
Published: May 2008

This study examines the inventory of intonation movements in Danish. The aim is to identify the micro intonation and to produce a model of the movement patterns in Danish, which would predict the intonation of synthetic speech in order to establish a close-to-natural intonation. The theoretical background is a modified version of the "close copy theory" ('t Hart et al.), in which movements are tendency lines in the intonation pattern. A new movement starts when the tendency line shifts direction. The acoustic analysis comprises measurements of time, semitones and slope (semitones/second = slope) and a description of the (parts of) phones in intonation movements of actual speech. The analysis suggests that the slope of the movement is one of the most important features of intonation, and that speakers are aiming to produce one of a set of 5 standard slopes, depending mainly on the phonetic segment, suggesting an underlying unit - an "intoneme"- displaying the same relationship as that of phoneme and phone.
Characterisation of the velocity distribution on rigid in-vitro upper airway replicas
Authors: Van Hirtum Annemie , Grandchamp Xavier , Marx David , Pelorson Xavier , Bailliet Helène
Published: May 2008

Qualitative and quantitative characterisation of the velocity distribution is an important aspect in order to study respiration related flow phenomena in the upper airways as well as for human speech production. Classical phonation models exploit one-dimensional flow descriptions in order to estimate the pressure forces exerted on the vocal folds tissues. In this case, the velocity distribution is expressed as an analytical relationship depending on volume velocity airflow, geometry and pressure distribution. Therefore simplified flow models can be validated by measuring the relevant quantities. However, more detailed and quantitative velocity predictions, aiming e.g. to improve phonation modelling or to study turbulent sound production, require increased precision for both the qualitative and quantitative characterisation of the velocity distribution. The current paper presents preliminary in-vitro measurements of the velocity distribution obtained by Particle Image Velocimetry combined with hot film anemometry. Several simplified rigid geometries are assessed in order to represent different portions of the upper airway. Besides qualitative results a first quantitative comparison between simulated and measured velocities is provided and discussed.
Experimental analysis of the relationship between the glottal flow and glottal area waveforms
Authors: Schwarz Raphael , Deliyski Dimitar D.
Published: May 2008

A very important but little studied aspect of human voice production is the relationship between the vocal fold vibration and the transglottal airflow. To analyze this relationship, in this study we combined high-speed videoendoscopy of the glottis for determining the glottal area waveform (GAW) with inverse filtering of the acoustic signal for estimating the glottal flow waveform (GFW). The high-speed camera system, recording at 20,000 pps, and the audio recording hardware were triggered by the same quartz oscillator to achieve synchronization with an unprecedented accuracy within 25 μs. We developed an image processing algorithm for automatic extraction of the GAW from the high-speed images. The high-speed video samples and the corresponding acoustic signals were obtained from 12 normophonic individuals (6 male, 6 female) for different voicing conditions: register (pulse, modal, falsetto); adductory adjustment (loose, normal, pressed); longitudinal tension within modal register (low, comfortable, high pitch); non-stationary phonation (variation in pitch and loudness). To compare the resulting GAWs and GFWs, the waveforms were parameterized concerning their temporal and spectral features. It is shown, that the revealed relationships between the vocal fold vibrations and the transglottal flow are comprehensible by accounting the different phonation conditions. [Work supported by NIH.]
Pitch production by bilingual Mandarin-Min speakers and monolingual Mandarin speakers in speaking Taiwan Mandarin and Min
Authors: Wu E-Chin , Fon Janice
Published: May 2008

In this study, Mandarin produced by bilingual Mandarin-Min speakers and monolingual Mandarin speakers were compared to see whether the presence of Min would pose influence on the pitch range utilized by bilingual Mandarin-Min speakers in speaking Mandarin. The results showed that bilingual Mandarin-Min speakers and monolingual Mandarin speakers did indeed differ in their pitch values when speaking Mandarin. Intriguingly, however, the difference did not go in the same direction for males and females. It is proposed here that the difference found in the male group was due to influence of Min, while the difference found in the female groups was due to social factors. The results of this study indicate that both language experiences and social constraints are important factors that influence a person's use of pitch.
Pitch Tracking using the Generalized Harmonicity Indicator
Authors: Haddad Darren , Noga Andrew , Sarkar Tappan
Published: May 2008

For many audio applications, a process is required to obtain an accurate estimate of the fundamental and harmonics of periodic sections of the audio signal. The Generalized Harmonicity Indicator (GHI) is to determine, assess and track the fundamental and harmonic frequencies of consecutive time segments of a speech signal using the Matrix Pencil (MP) technique[i]. Various methods of fundamental and harmonic frequency tracking have been proposed and developed, but most have been based on other low resolution techniques such as FFT and cepstral analyses. This is opposed to using a super-resolution frequency estimation technique as provided by the MP. The prior art in the area of super-resolution speech fundamental determination consists of the "super resolution pitch determinator" (SRPD)[ii] and the "enhanced SRPD" (eSRPD) methods. Because these prior methods do not explicitly process a spectral representation or decomposition of the input audio signal, they are not considered to be in the same class as the MP GHI, although they provide a baseline for comparing different estimation techniques. [i] Haddad, D.M., Sarkar, T.K., Noga, A.J.; "Speech Compression Using the Matrix Pencil Technique"; IEEE 12th DSP Workshop; Sept. 2006; Page(s):218-221. [ii] Y. Medan, E., Yair, D., Chazan, "Super Resolution Pitch Determination of Speech Signals," IEEE Trans. On Signal Processing, ASSP-39(1):40-48, 1991.
Mitigation of Nonlinear Distortion in Speech Signals Using Histogram Matching
Authors: Smolenski Brett
Published: May 2008

Nonlinear distortion is a common artifact in audio communication equipment. In addition, it is well know that the normalized amplitude distribution of speech signals converges to approximately a gamma distribution after a few seconds. Hence, the transfer function of any memoryless nonlinearity distorting the speech signal can easily be estimated, provided one has buffered enough data. This research shows how both parametric and nonparametric histogram matching algorithms can be employed to remove the effects of these types of distortions. Further, the improvement these algorithms have on speaker identification performance is also studied. This approach represents a radical departure from the traditional approach taken in speech enhancement. The traditional approach has been to first acquire a model or representation of the distortion, noise, or interference that is corrupting the signal, and then attempt to remove this from the signal, usually introducing other forms of distortion in the process. With this approach a model of clean undistorted speech is used that the distorted speech is then matched to.
"Polyaural" array processing for robust automatic speech recognition in noisy and reverberant environments
Authors: Stern Richard M. , Gouvea Evandro B. , Kumar Kshitiz
Published: May 2008

It is well known that human binaural processing is very useful for separating incoming sound sources as well as for improving the intelligibility of speech in reverberant environments. In this paper we present a new method of signal processing for robust speech recognition using multiple microphones. The method, loosely based on the human binaural hearing system, consists of passing the speech signals detected by multiple microphones through bandpass filtering and nonlinear halfwave rectification operations, and then cross-correlating the outputs from each channel within each frequency band. These operations provide rejection of off-axis interfering signals. These operations are repeated (in a non-physiological fashion) for the negative of the signal, and an estimate of the desired signal is obtained by combining the positive and negative outputs. We demonstrate that the use of this approach provides substantially better recognition accuracy than delay-and-sum beamforming using the same sensors for target signals in the presence of additive broadband and speech maskers, and it provides substantial improvements in specific reverberant environments as well. [Supported by NSF and DARPA]
Fractal analysis of signals of the seismic acoustic emission
Authors: Glushkov Alexander V. , Svinarenko Andrey A. , Lepikh Yaroslav I.
Published: May 2008

Paper is devoted to analysis fractal properties of signals of the seismic acoustic emission in periods between earthquakes. Earlier it has been carried out an analysis of natural data on distribution of the earthquake hypercenters (Mukhamedov, 1992; Salimi et al, 1993). To reveal the fractal properties for signals of the seismic acoustic emission the joint wavelet analysis is carried out by using the non-decimated wavelet transform (Glushkov et al, 2004, 2005). We present the fractal processing data for envelopes of signal of the seismic acoustic emission in different ranges of frequencies. The dependences of the Herst indicator and fractal dimension curve on the lengths of considered intervals are presented. It takes a place crossover upon a behaviour with the Herst indicator H=0,4-0,6. It is given the physical interpretation of the seismic acoustic emission. References: Mukhamedov V.A., Izv. Russian Acad.Sci. 3, 39 (1992). Salimi M., Robertson M., Sammis S., Phys.Rev.Lett. 70, 2186 (1993). Glushkov A.V. etal, Nonlinear Processes in Geophys. 11, 285 (2004). Glushkov A.V. et al, Atmospheric Res. (Elseiver). 77, 100 (2005).
A torpedo detection and 3-D tracking system
Authors: Roosnek Nico
Published: May 2008

Ship torpedo defence systems are ideally based on 3D information of incoming torpedoes. Therefore a Torpedo Detection and Location (TDL) sonar system has been designed based on a transmitting transducer, two receiving arrays and optimal signal processing with 3D-tracking capabilities with optimal estimation. The transmitted signal is a chirp. The signal processing with intrinsic beamforming is based on the phase of the signal and on optimal estimation (Kalman filtering). For testing the capabilities of such a design a test system on detection and tracking of a torpedo-like object in the atmosphere has been developed. Some figures about its capability, which can be extrapolated to underwater situations, will be given.
Source localisation on a single hydrophone
Authors: Le Touzé Grégoire , Nicolas Barbara , Mars Jérôme I.
Published: May 2008

The aim of the study is to localise an undewater Ultra Low Frequency (ULF) source in a shallow water environement. The acoustic signal is recorded on a single hydrophone and the source, which has to be short in time, is unknown. To perform the localisation, we have developed modal filters based on time-frequency techniques. Different localisation techniques are proposed: - Conventional Matched Mode Processing: results are good for range estimation but contain error on source depth estimation. We show that those errors are due to the ULF band. - Source depth estimation technique based on mode amplitude estimation: this estimation is precise but presents ambiguities. - Range estimation technique using mode phase estimation: we recently developed this method which also estimates mode signs. The source range is precisely estimated and mode sign estimation avoids ambiguity on source depth estimation combined with the previous method. These methods are validated on real data comming from the North Sea.
Aircrafts localisation and tracking with arrays of microphones
Authors: Caronna Gaetano , Testa Pierluigi
Published: May 2008

University of Roma "La Sapienza" will participate in a project, financed by the Italian Ministry of Research, aiming to detect and to track aircrafts near the airport using the acoustic emissions of the aircraft and the technique of acoustic beam-forming. Although the principle is well known and some applications are reported in literature and explored also by our group in a previous European FP6 project ("Safe Airport"), the final performance and the engineering value are not assessed. In this reseach effort the problem is again raised aiming to create a more realistic model including wind and temperature gradient that, in particular conditions, can drastically impair the validity of the results, estimating the related errors. A simulation software was written considering the steering drive on the microphones outputs, being the mechanical rotation of the array prohibited, due to the aerodynamic noises induced. At least two arrays, opportunely located, are requested in order to perform a triangulation and detect the source of the acoustic wave emitted by the aircraft. The precision in the localisation of the acoustic source was estimated in a simulation where real aircraft acoustic emissions and environmental noise were used as imput.
Performance of high-resolution sensor array processing algorithms in the localization of acoustic sources
Authors: Lardies Joseph , Ma Hua , Berthillier Marc , Foltete Emmanuel
Published: May 2008

The localization of noise sources from a specified direction may often be accomplished with an array of sensors. One commonly used processor consists of delay and add networks: a conventional beamformer, however its spectrum suffers from the Rayleigh resolution and its performance is highly degraded, specially in lower frequency range. In the communication, the performance of some typical high-resolution sensor array processing algorithms: Minimum Variance, MUSIC, Mini-Norm algorithms are investigated for wideband source location. Their performances are compared with a new source localization algorithm which is based on a sparse representation of sensor measurements with an overcomplete basis composed of samples from the array manifold. The key of the method is the use of the SVD for data reduction and the formulation of a joint multiple-sample sparse representation problem in the signal subspace domain. Increased resolution and improved robustness to noise is obtained with this algorithm applied to various numerical examples.
Objective quality measurement of the excitation of impact sounds in a source/filter model
Authors: Lagrange Mathieu , Giordano Bruno L. , Depalle Philippe , McAdams Stephen
Published: May 2008

For the modeling of percussive (non-sustained) sounds, the excitation signal can be estimated from an original sound in several ways, usually by a time-domain deconvolution process. The source signal obtained by such a process cannot be compared with the original excitation because it is usually unknown. Hence in most of the approaches available in the literature, the validation of the deconvolution process is quantifed in terms of spectral flatness, i.e. a source signal is considered as a good estimation of the excitation when most of the resonant content has been removed. However, the excitation signal is usually a percussive burst, the time domain properties of which are known to be very important, at least perceptually speaking. To evaluate the time domain properties of the estimated excitation, we propose in this paper to compare the estimated excitation to the recording of the acceleration of the hammer hitting a plate. In the recordings considered, the evolution of the acceleration of the hammer has a specifc pattern with several peaks due to the bouncing of the hammer on the suspended plate. This specifc pattern allows us to propose a metric that can be useful for objectively measuring the quality of the estimation process.
Estimation of reflection location by the correlation coefficient function
Authors: Shibayama Hideo , Araya Takeshi , Makabe Yoshiaki , Okarura Eiji
Published: May 2008

Gas is supplied to households through gas pipes that branch out from a main pipe. It is essential to carry out maintenance of these branch pipes to ensure the safe supply of gas. However, there are many cases in which the state of buried gas pipes and the connection condition of a large number of pipes are unknown because no detailed information is included in piping drawings or only plan views are available. If the piping arrangement can be estimated on the basis of response waves originating from an acoustic wave travelling in a gas pipe emitted by a loudspeaker, which is placed at one end of the pipe, this method will be important for the maintenance of branch pipes. To realize this, we report a method of estimating the length of gas pipes using correlation coefficient between a driving signal and measured waves. And, we show the experimental results with high accuracy.
Two Dimensional Wavelet Coefficient Statistics for Sea Bottom Classification
Authors: Koukos Ioannis , Mavroidis Theodoros , Vardoulias Georgios
Published: May 2008

In this paper we examine the classification of different seafloors based on the analysis of images obtained by side-scan sonar. For this purpose, we apply various two dimensional multilevel wavelet decomposition schemes on images obtained from three different seafloor types, i.e., sand ripples, rocks and sands, and then we examine the statistics of the corresponding wavelet coefficients. The observed Probability Density Functions (pdf) are modeled using various theoretical distributions such as the Alpha Stable, Sum of Gaussians, Log-normal. The parameters of the fit are subsequently used to classify the side scan sonar images according to well known cluster analysis techniques. The use of the energy of the wavelet coefficients as a tool for side scan sonar image classification is also evaluated. A new unsupervised classification scheme based on the pdf fitting parameters is proposed.
Adaptive coding/modulation for shallow-water UWA communications
Authors: Mani Sanjay , Duman Tolga M. , Hursky Paul
Published: May 2008

We consider adaptive modulation and coding techniques for Phase Shift Keying (PSK) transmission schemes over underwater acoustic (UWA) channels. Adaptive modulation and coding can be an effective means of obtaining higher data rates while retaining acceptable error levels by exploiting knowledge of the channel state. This is particularly significant in the UWA scenario, where attained spectral efficiencies are critical since usable bandwidth is a severe limitation. We examine the use of channel capacity and post-equalization signal to noise ratio (SNR) as adaptation metrics. We illustrate the ideas both through simulations and using the results of a recent experiment (AUVfest 2007). In the experiment, in order to evaluate effectiveness of adaptive coding/modulation for UWA communications, we have transmitted a bank of signals with varying spectral efficiencies (obtained by changing the number of transmit elements, modulation scheme, and code rate) back to back. Using the received signals, we illustrate the relevance of the channel condition metrics under consideration, that is, we evaluate the proposed metrics for various scenarios, and provide indications of the transmission rates that could have been achieved reliably.
Acoustic methods in extremely shallow water: validity of the far field approximation
Authors: Madricardo Fantina , Buogo Silvano , Calicchia Paola , Boccardi Emiliano
Published: May 2008

In this paper we present some experimental results concerning the possibility of using traditional echosounder in extremely shallow water environments. In the framework of the Echos Project, a collaboration between ISMAR and IA, a wide area of the Venice Lagoon has been explored with a traditional echosounder. Since the explored area was extremely shallow (up to 50 cm), it became necessary to check experimentally whether the far field approximation was still valid in such conditions. In this work the acoustic field of the echosounder ELAC LAZ72 used for the sub bottom investigation is experimentally characterized and the validity of the far field approximation is verified.
Acoustic measurements of bubbles in the wake of ship model in tank
Authors: Sutin Alexander , Benilov Alexander , Roh Heui-Seol , Nah Young In
Published: May 2008

The interest to bubble generation by moving ships is connected with large area of bubble wake reaching several kilometers that can be used for ship detection. We developed acoustic system for the measurements of bubble density and conducted measurements in 100 m long towing tank. The developed system measured attenuation of ultrasound in wide frequency band from 50 to 800 kHz between two acoustic sensors placed on the distance 20cm. The attenuation of sound produced by bubbles was observed during several minutes after model of ship passed the point of measurement. The attenuation was recalculated to the bubble size distribution for bubbles from 4 to 65 microns using theory of resonance bubble attenuation. The measured bubble size distribution can be interpolated by power dependence n(R) ̃ R (-3.5) that is typical for bubbles at sea subsurface layer. The dependencies of bubble concentration of model ship speed and type of propeller were investigated. The generation of bubbles was observed when the model speed exceeds definite threshold that can be connected with the cavitation threshold. The theory describing dynamics of wake turbulence based on the shear-free turbulent wake was developed. The measured bubble concentration was in good agreement with the developed theory.
On acoustic tomography method physical advantages in long range ocean inhomogeneities control
Authors: Semenov Andrew
Published: May 2008

An acoustic tomography method based on observation of signals transecting an inhomogeneity is discussed. While for various inhomogeneities (ships, underwater objects, sea mammals etc.) short range monitoring of back scattered sonar reflections could be used, their monitoring at longer ranges is still a problem. This is due to the fast decay of sonar signals with distance, especially in unfavorably absorbing shallow water regions where there is functional distance dependence of optimum frequency. New perspectives on ocean climate change measurement (ATOC) require monitoring of extra long (500 - 1000 km) tomography traces, where ocean vortices and icebergs could hamper the progress. The physical limits (optimum frequencies and efficiencies) for two tomographic methods for deep and, most unfavorable in the author's view, shallow ocean regions are compared. For several types of inhomogeneities tomorgaphic measurement optimization estimates are discussed. For ranges up to 500 km the acoustic tomography method has substantial advantages with respect to conventional sonar monitoring even for unfavorable shallow water regions at least up to 300 km distance and in all deep water regions. Tomographic measurements are also indispensable in long-range ocean vortices monitoring.
A hydrophone calibration centre for the Mediterranean area
Authors: Cannelli Giovanni Bosco , Buogo Silvano , Calicchia Paola
Published: May 2008

The Underwater Acoustics Laboratory of the Institute of Acoustics "O. M. Corbino" in Rome, Italy, has recently achieved accreditation for hydrophone calibration. It can offer hydrophone calibration services operating under a management system compliant with UNI EN ISO/IEC standard and traceability of measurements to NPL primary standards. Precise, traceable measurements are needed today in all applications of underwater acoustics, where a faithful response of measuring devices is essential for a correct evaluation of the system performance and for a rigorous assessment of the environmental impact of man-made activities at sea. To attain these requirements the Laboratory has a water tank 6 m long, 4 m wide and 5,5 m deep, equipped with a motorized two-trolley positioning system capable of handling loads of up to 100 kg. A variety of test and measurement equipment is available, controlled by a computer acoustic calibration system, and standalone instruments for signal generation, acquisition and analysis. Besides, two windows are provided on one of the tank walls for combined acoustical and optical measurements, which can be performed using the available Laser Doppler Vibrometer (up to 100 kHz) and the 16-mm film rotating prism high-speed camera (up to 10 000 frames per second).
Suppression of side lobe level on the cone characteristics of the directivity pattern of an antenna as an important factor in its directivity index and effective aperture
Authors: Milosic Zvonimir M.S.
Published: May 2008

This paper will present a universal procedure for precise de-embedding of the directivity index and effective aperture dependence of the suppressed side lobes on measured cone directivity patterns of sonar antennas. The procedure is derived on an idealized model of cone characteristics and directivity patterns of antennas. It is universally applicable to any contemporary sonar or hydroacoustic communication system with a cone directivity pattern. In accordance with the given expressions of directivity index and effective aperture of circle baffled pistons, this paper will present substitute analytical functions for hypothetical circle antennas with suppressed minor lobes. The presented graphs of functional behavior in a two-dimensional system, and in a three-dimensional coordinate system, are an excellent base for standardization, unification and quality evaluation of sonar or radar systems and also medical scanners. Key words: directivity index of sonar antenna, idealized model of cone directivity pattern, level of suppressed side lobes, effective aperture
Transmission of low frequency sounds from ships into water environment
Authors: Gloza Ignacy
Published: May 2008

The main goal is presentation the broad experimental research associated with the transmission of acoustic energy generated by moving objects into the water. I present the methodology of evoluation of the transmission of low frequency vibration from the mechanisms of ship into the water environment. For that purpose the research of vibration distribution over ship's hull were conducted along with the hydroacoustic field. The underwater sound measurements were performed both for sailing condition and during anchorage. The hull is source of complex surface shape inducing waves of different frequencies depending on the working parameters of ship's machinery. As a result of mechanical incitement of the hull's surface a complex distribution of vibration is created depending on vibration spectra of the primary sources as for the main propulsion units, generators or other mechanisms. This research allowed to determine the transmission coefficient for the mechanical vibration energy going through the ship's hull and the evaluation of influence on the hydroacoustic field of a ship. The values signaling the changes of the technical condition of machinery and propulsion system.
Detection of a resonant target buried in sediment using iterative time reversal: mid-frequency pond experiments
Authors: Dzikowicz Benjamin R. , Waters Zachary J. , Holt R. Glynn , Roy Ronald A.
Published: May 2008

Iterative time reversal techniques developed at smaller scales, [Waters et al., J. Acoust. Soc. Am. 122, 3023 (2007)], are applied to the detection of a 15 cm diameter stainless steel shell buried in sandy sediment at the acoustic test pond at the Naval Surface Warfare Center - Panama City Division. A mid-frequency, directional projector is located 1.5 m above the sediment and directed normally to it. A hydrophone is located midway between the sediment and the projector. This system gives a response between 20 kHz and 200 kHz. A calibration filter is designed using the direct path response between the projector and hydrophone. This filter is applied at each time reversal iteration to prevent the time reversal technique from converging to the transducer resonance. Application of iterative time reversal allows the detection of the target at greater depths than otherwise possible due to the resonance scattering of the target. Additional experiments explore the application of filters, and the effect of different window sizes. Also, comparisons with similar laboratory experiments and comments on real world applications are discussed. [Work supported by the Office of Naval Research.]
BIE and coupled BIE-FEM methods for scattering from underwater objects
Authors: Karasalo Ilkka
Published: May 2008

Two techniques are described for computation of acoustic scattering by a smooth elastic object in a fluid space or a fluid layer of a layered fluid-solid medium. Both techniques use a frequency-domain boundary integral equation (BIE) method for the exterior layered medium, with the Green's function computed numerically by adaptive high-order transform integration and exact finite elements. The surface of the object is parameterized by a smooth map of the unit sphere, and the BIE is discretized by B-splines, point collocation and numerical integration in polar-like coordinates centered at the collocation point. Two approaches for handling the interior dynamics of the object are described. In the first, applicable to objects structured into homogeneous layers separated by closed smooth interfaces, the interior dynamics is formulated as a system of BIEs with displacements and tractions at the interfaces as unknowns. In the second, the interior is discretized by a general-purpose finite element (FEM) method, and an impedance matrix connecting tractions and displacements on the exterior surface is computed by eliminating interior degrees of freedom by Gaussian elimination. Computational results on test cases of the 2006 target strength benchmark workshop at NURC are presented.
Calculation of the acoustic target strength of elastic objects based on BEM-BEM-coupling
Authors: Burgschweiger Ralf , Ochmann Martin , Nolte Bodo
Published: May 2008

Based on a BEM-BEM-coupling method, the scattered pressure from elastic objects placed in water and partially buried in the sediment is calculated. For this reason, a special variant of the boundary element method (BEM) is implemented. It contains a pre- and a postprocessor with 3D visualization, in order to define the geometry of the scattering objects in the interface layer between fluid and sediment and the parameters needed for characterizing the fluid and the elastic material. The solver is able to perform numerical calculations in a multiple parallel manner. For the solution of the underlying system of linear equations, we use different kinds of approximate and direct solution techniques. Simple acoustical exterior problems, for example, the sound scattering by elastic solid cylinders and spheres placed in a fluid are treated by the BEM-BEM-coupling method. The results will be compared with analytical solutions or solutions obtained from other numerical methods.
Modeling of scattering from targets in an oceanic waveguide using Kirchhoff/diffraction method
Authors: Lee Keunhwa , Seong Woojae , Joo Yongtaek
Published: May 2008

The target scattering model in an oceanic waveguide is presented. The target scattered pressure field is formulated using the generalized Green's function method [F. Ingenito, J. Acoust. Soc. Am. 82, 2051-2059 (1987)]. The concept of Kirchhoff/diffraction method is introduced in order to simplify the Fredholm integral equation. In numerical analysis, complex target is divided into numerous polygon facets, whose analytic solution for scattered field is derived based on the waveguide solution by the ray or normal mode theory. This solution is used in constructing the target scattered field for complex target. Comparison between ray and normal mode based target scattering model is shown. Finally discussion for conditions of the source/receiver and target which improves the numerical efficiency is given.
Comparison of two monostatic reverberation models based on ray theory
Authors: Chu Youngmin , Seong Woojae , Oh Won Tchon
Published: May 2008

Two monostatic reverberation models applicable for range dependent environment are presented. Considering the roughness of both the sea surface and bottom, we calculate the reverberation signal, based on the ray propagation model and scattering model using the composite-roughness theory and/or other empirical formulae. First model computes the reverberation time signal using two-way eigenray searching between source/receiver position along the discrete scattering area (dA). The other model uses one-way eigenray searching along the discrete ray angle (dθ) which generates a non-uniform scattering area providing less accurate solution at a reduced computational burden. The time series calculated from both models are shown and compared with the solutions presented in the Reverberation Modeling Workshop (Nov. 2006, Austin, TX).
On the analysis of the time spreading of sound diffusers
Authors: Redondo Javier , Picó Rubén , Avis Mark R.
Published: May 2008

Since the invention of sound diffusers three decades ago a substantial effort has been made to predict the acoustic behaviour of these structures. BEM methods are well established for this purpose after a systematic comparison between simulations and experimental data. Volumetric methods such as finite element methods (FEM) or the finite difference time domain method (FDTD) are not often used, due to their large computational cost. However, near to far field transformations (NFFT) can overcome that problem. Recently some of the authors have shown that the FDTD method is a useful technique to analyse the time domain signature of sound diffusers. In this paper a careful analysis of the performance of diffusers in the time domain (“time spreading”) are reported, opening a new field of research.
The intensity in a reverberant field as an acoustic energy-density gradient
Authors: Valeau Vincent , Boirlaud Matthieu , Picaut Judicaël , Hodgson Murray
Published: May 2008

The classical assumption for diffuse reverberant sound fields is that the acoustic energy flow at any location in a room -- i.e., the acoustic intensity -- is null. For rooms with disproportionate dimensions and/or uneven absorption distributions, this assumption no longer holds. Over the past few years, a room-acoustic diffusion model has been developed that can be seen as an extension of the statistical theory to nondiffuse sound fields. This work investigates the basic gradient-equation underlying this diffusion theory: it states that the energy-flow vector is proportional to both the gradient of the acoustic energy density, and to the room diffusion constant. The gradient equation for the acoustic intensity is confirmed numerically by comparison with calculations of the intensity vector using a particle-tracing model and a ray-tracing model. Examples are presented for a room with a diffuse sound field, and for elongated rooms where the sound field is strongly non-diffuse, in the case of both diffuse and specular reflections.
Prediction and tailoring of steady-state broadband sound fields in enclosures using absorption scaling and energy-intensity boundary elements
Authors: Bliss Donald B. , Michalis Krista A. , Franzoni Linda P.
Published: May 2008

Enclosures with diffuse reflection boundaries are modeled with an energy-intensity boundary element method using uncorrelated broadband directional sources. An absorption-based perturbation analysis shows the spatial variation of the acoustic field obeys certain scaling laws. A series expansion in terms of average absorption gives separate boundary integral problems at each order. The lowest-order solution has a uniform level proportional to the reciprocal of the average absorption. The next-order solution is independent of average absorption and primarily responsible for spatial variation of the acoustic field. This solution depends on the spatial distribution of absorption and input power sources, but not their overall level. For the primary spatial variation, the effects of the relative distributions of absorption and input power are linear and uncoupled. These distributions can be expressed in terms of constituent spatial modes corresponding to the ways absorption and input power can be distributed. Solved numerically once for each mode, the acoustic field can be expressed in terms of the modal amplitudes in closed form. These amplitudes can be adjusted to tailor the spatial variation. Examples include how to distribute absorption to minimize sound levels in one location, or how to achieve a uniform interior field. (Sponsor: NSF)
Defining perceptual requirements of dynamic (real-time) acoustic modeling and auralization
Authors: Gedemer Linda
Published: May 2008

Acoustic modeling is evolving from static sources and receivers to dynamic sources and receivers as computational speeds allow for faster renderings. Current research has focused on creating highly accurate dynamic models and auralizations but still struggle with the trade-off between accuracy and available processing speed. The question remains as to how accurate do these models need to be if the receiver (listener) is moving dynamically through the acoustic model. The focus of this paper is to address the accuracy required of dynamic models and/or auralizations from a perceptual standpoint. From this, the necessity for creating models that are perhaps more accurate than required will be considered.
Localization and mesoscopics in structures and rooms II
Authors: Weaver Richard
Published: May 2008

A new theory is developed which makes quantitative predictions for energy transport that are consistent with localization. The theory promises ultimately to apply to a wide variety of systems. It is based on an assumption of S-matrices that are incoherent - as conventional diffuse field theories would have it - but then corrected to assure exact energy conservation. It is shown that the resulting system responses (they are concatenations of S matrices) have coherence, are consistent with localization, and predict Greens function retrieval. Localization emerges naturally from competition between rates of transport and rate of eigen-mode resolution. Examples are presented corresponding to coupled rooms and to diffusion in a multiply scattering medium. Future generalizations are discussed and the mathematical challenges are outlined. This is the second talk in a pair begun in a different session. Work supported by NSF CMS 05-28096.
Monitoring field and laboratory performance for airborne and impact sound insulation for 110,000 homes per annum
Authors: Smith Sean , Baker Dave , Panter David
Published: May 2008

This paper provides an overview of a unique monitoring system used in England and Wales for tracking the sound insulation performance for 110 000 new build homes per annum. The regulatory approach, known as Robust Details was established within the building regulations and passed by parliament in 2004 after an 18 month initial research project. Annually over 5000 construction sites now use this system approach for separating walls and floors for attached houses and apartments. Acoustic site inspections and sound insulation field testing reports provide systematic feedback to the design and performance review. Since the RD systems inception on-site sound insulation compliance rates have risen from 40% to 97%. This paper will discuss the role of laboratory benchmark testing for products and also the role of acoustic "field" inspectors in tracking the airborne and impact performance. In addition the role of intervention and statistical analysis in upgrading or withdrawing specific wall and floor constructions will be presented.
Changed sound properties due to minor construction changes in a lightweight building
Authors: Ljunggren Fredrik
Published: May 2008

This paper relates to building acoustic measurement inside a two-story office house. The construction, which is known as lightweight, is prefabricated in volumes at a factory and is then transported to the building yard for assembling. It is build up of a wooden frame with particle boards and plaster boards attached. The building consists of a number of nominally, or almost nominally, identical rooms with assumed identical sound properties. In the projection stage, the construction was slightly modified in some of the rooms in order to see in what way the sound properties would be affected. In total eight impact sound pressure level measurements and eight sound reduction index measurements were performed and analysed for the different setups.
Acoustic and vibration characteristics of floated floors concrete structures
Authors: Yoo Seungyup Y. , Jeon Jin Yong
Published: May 2008

The floating floor with multilayers of resilient materials was investigated for the isolation of floor impact sound. The floor structures were evaluated in the reduction of the heavy-weight impact sound, which is mainly affected by the boundary conditions of the tested floors. The values of single-number ratings for floor impact sounds decrease dramatically for floated floors. Insertion loss is determined by the vibration characteristics of a bare slab, such as transmissibility. The logarithmic decrement (ξ) of the vibration responses was 4 times higher than that of the bare slab. Above 125 Hz, the vibration between the slab structure and floating floor is isolated effectively. It was revealed that the resonant frequency of the slab structure and the damping factor of the floating floor are important aspects for impact sound isolation.