bluuebunny/crossref_metadata_2025 huggingface dataset:
Optical detection of longitudinal and shear acoustic waves with laser picosecond acoustics
Authors: Matsuda Osamu , Wright Oliver B. , Hurley David H. , Gusev Vitali , Shimizu Ken'Ichi
Published: May 2008

The absorption of picosecond light pulses in a medium can generate sub-THz acoustic waves. These cause a transient optical reflectance change that can be monitored by delayed probe light pulses. This technique, termed laser picosecond acoustics, can be used for the nondestructive evaluation of the physical properties of thin films and substrates. This paper describes a general method for quantitative analysis of such reflectance changes. It is applicable to multiple anisotropic layers that may be opaque or transparent. Longitudinal or shear acoustic waves propagating along the stacking direction of the multilayers modulate the dielectric permittivity anisotropically and inhomogeneously through the photoelastic effect, through local rotation, or through the surface and interface displacements. We describe how the optical reflectance for obliquely incident probe light can be calculated for the modulated medium. We then demonstrate the method with reference to experimental results for a sample consisting of a silica film on a zinc substrate in which both longitudinal and shear acoustic waves are generated and detected. The analysis yields the film thickness, sound velocity, and photoelastic tensor components, for example. The method is also applicable to various light scattering problems involving the inhomogeneous modulation of optical properties such as in photothermal experiments.
Laser ultrasonic study of Lamb wave propagation in different film loading configurations
Authors: Xu Xiaodong , Shkerdin G. , Glorieux Christ
Published: May 2008

A study is performed concerning the propagation Lamb waves in technologically and scientifically interesting configurations. In a first configuration a thin film was loaded with different liquids on its two sides. Three experimental laser ultrasonic configurations, line excitation with scanning detection, grating excitation with single point detection, and grating excitation with scanning probe beam, were implemented. All experimental results were found to be consistent between experimental methods and with theoretical predictions. The configuration can be implemented in monitoring applications where elastic properties of liquids that are kept in containers can be determined via the outside container wall. In a second configuration laser excited and detected Lamb waves in a rubber-metal bi-layer were used to determine the highly damped elastic properties of rubber.
Binaural auralization based on spherical-harmonics beamforming
Authors: Song Wookeun , Ellermeier Wolfgang , Hald Jo/rgen
Published: May 2008

The binaural auralization of a 3D sound field using spherical-harmonics beamforming (SHB) techniques was investigated and compared with the traditional method using a dummy head. The new procedure was verified by comparing simulated room impulse responses with directly measured ones both monaurally and binaurally. The objective comparisons show that there is good agreement in the frequency range between 0.1 to 6.4 kHz. Psychoacoustic attributes of multi-channel reproduced sounds were measured in a listening experiment to validate the method subjectively. The results show that subjective ratings of the width, spaciousness and preference of different audio reproduction modes auralized based on SHB were not significantly different from those obtained for dummy head measurements. Thus binaural synthesis using SHB may be a useful tool to reproduce a 3D sound field binaurally while saving considerably on measurement time because head rotation can be simulated based on a single recording.
Comparison of subjective impression of copy machine noise between Japanese, American and German participants
Authors: Furukawa Tatsuya , Takehira Osamu , Nagamiya Masaki , Kuwano Sonoko , Namba Seiichoro , Fastl Hugo
Published: May 2008

Psychological evaluations of copy machine noise are in progress to lower acoustical noise and to avoid disturbing the office environment. Psychological experiments were made for Japanese, American and German participants to examine cultural differences in sensitivity to copy machine noise using the Semantic Differential. Seventeen copy machine and laser printer sounds were used for the experiments. The results were analyzed statistically. The experiments revealed that, just as with Japanese, Americans and Germans were mostly sensitive to "sound pressure levels." However, the second influential factor for Japanese participants was "impulsiveness" but "sharpness" for Americans and Germans. Evaluation formulas for "pleasing" noise using physical values were decided for each country. This suggests that copy machine noise will be evaluated without psychological experiments. The information gathered will possibly be used to adapt copy machine noise to levels that are comfortable to users in different countries.
Identification of transient events from a hard disk drive using non-stationary loudness
Authors: Ali Dave
Published: May 2008

The Hard Disk Drive (HDD) industry has been pushed so much to lower its limits of acoustic emissions by OEMs and customers that the levels of most HDDs flirt with the threshold of hearing and in some cases go below. With these much reduced levels come different problems and complaints from customers, namely transient events; latching, de-latching, spin up, spin down, etc. The difference in amplitude and duration of these events typically stand out to the human ear above the steady state nature of the HDD during idle and sometimes when it's active. This paper shows a method of characterizing these events with confidence to help in designing a better product for the industry.
The influence of speed bumps on perceived annoyance
Authors: Preis Anna , Kaczmarek Tomasz , Griefahn Barbara , Gjestland Truls
Published: May 2008

Recently, several attempts to use speed bumps as a noise reduction method have been made. Objective analyses of the effect of speed bumps on noise have been shown to result in a rather small reduction of noise. In the present paper the influence of speed bumps on perceived annoyance is investigated. The annoyance rating of a situation in which a passenger car approaches with constant velocity, then decelerates, crosses the bump, accelerates, and then recedes at a constant speed was compared with a car pass-by at a constant velocity without a bump. Three different velocities were analyzed: 40, 50 and 60 km/h, and two types of driving conditions: normal, and aggressive. Listeners judged their annoyance for all the investigated scenarios using the ICBEN scale (0-10) for annoyance assessment. Objective analyses showed a significant reduction of LAeqT in the bump situation for all tested velocities, and for both driving conditions. The results of this psychoacoustic experiment show no effect of the bump on annoyance rating for normal driving conditions. However, in aggressive driving conditions the bump resulted in a significant increase in annoyance. In the light of these results, speed bumps cannot be considered as a noise reduction method.
Fusion and masking threshold of a tone in narrowband noise
Authors: Hansen Hans , Weber Reinhard
Published: May 2008

When comparing different pitch phenomena, it is not clear whether the expression pitch strength points towards a unique perceptive phenomenon or object. As many studies concentrate on one special phenomenon, the question, what is actually judged, seems only implicitly answered. Kubovy and Van Valkenburg (2001) define a perceptual object as "that what is susceptible to figure-ground segregation". Pitch plays a major role in these grouping processes. In this context the judgment of pitch strength refers to two perceptual cases. The first one is the pitch strength of a tone-in-noise, i.e. the case where the pitch is linked to a separate contour within the narrowband noise (NBN), while the second is the pitch strength of tonal noise. Here, the noise evokes a pitch sensation that is not related to a separate object. In order to explore the transition from case /1 tone-in-noise to case /2 tonal noise, the identification threshold hearing a tone in NBN centered on the tone is determined in a lab experiment. This segregation or fusion threshold is compared to the measured masking threshold for center frequencies 250-4000 Hz octave-wise at 60 dB SPL noise level. The bandwith of the NBN is varied from 50-250 Hz accordingly.
Are absolute thresholds and loudness judgements influenced by different colours
Authors: Menzel Daniel , Faccinelli Elias , Fastl Hugo
Published: May 2008

Previous experiments showed that images of differently colored trains and sports cars can influence the loudness ratings given by subjects via free magnitude estimation. Red vehicles caused subjects to rate the loudness of simultaneously presented train or car sounds higher relative to green vehicles. To investigate whether these loudness differences correspond to shifts in absolute threshold, subject's threshold in quiet was measured via Békésy-tracking while viewing red, green, and neutral color patches. Also, the influence of color on the loudness of broadband noise was measured using a method of adjustment, in which subjects had to adjust the level of a test sound until it was perceived as loud as a reference sound. In both cases, no influence of color on either absolute threshold or loudness perception measured via adjustment could be found. These results support the hypothesis that, compared to previously used methods like magnitude estimation, methods that require subjects to concentrate on the auditory stimulus do not seem to be as applicable for measurements of audio-visual interactions.
Factor analyses of critical-band-filtered speech of British English and Japanese
Authors: Ueda Kazuo , Nakajima Yoshitaka
Published: May 2008

Two-hundred sentences of British English and Japanese, each uttered by 10 native speakers (5 females and 5 males) in each language, were analyzed through 20 bands of critical band filters. Smoothed power fluctuations derived from the filters were submitted to principal component analyses followed by varimax rotation. The first three factors explained 34-37% of variance. One of the factors exhibited two peaks along frequency axis in the standardized scores and two of the factors showed one peak for each. These three factors divided the whole frequency range of speech sound into four bands. The structure of the factors and frequency bands was essentially the same across the two languages. These frequency bands can be used for speech perception in general, because intelligible noise-vocoded speech sounds can be synthesized with the frequency bands.
How many psycho-acoustic attributes are needed
Authors: Pedersen Torben Holm , Zacharov Nick
Published: May 2008

Sounds may be characterized by objective perceptive attributes (for which there may exist physical metrics) or by subjective (affective or connotative) attributes. This paper will deal with the perceptive attributes. Within product sound quality the metrics for classical the psycho-acoustic attributes (loudness, sharpness, roughness and fluctuation strength) -maybe supplemented with tone and impulse prominence- are often used as the only attributes to characterize the sounds. But are these 4-6 attributes or dimensions sufficient to characterize a sound? Within room acoustics and reproduced sound many other attributes are used and in the language around 100 direct sound describing words may be found. This paper will give an overview over attributes used within different acoustic areas. The latter part of the paper will discuss the role of sensory evaluation methods as a means to systematically developing attributes for the objective qualification and quantification of sound characteristics.
Noise perception of wall-hung gas boilers
Authors: Chevret Patrick , Coulon Anne , Bessac François , Parizet Etienne
Published: May 2008

The aim of this study is to assess the quality image for domestic wall-hung gas-fired boilers, based on their noise. Nine boilers were recorded using an acoustic manikin in a hemi-anechoic room, for different operating conditions. Two of these operating conditions (maximum heat input, hot water tapping) were first studied. Five-second sequences were presented (through headphones) to sixty listeners using the mixed assessment method allowing for the comparison between signals. Analyses showed several assessment strategies: according to listeners, the relevant noise parameters could be the loudness, the sharpness or the presence of tonal components. A second experimental phase focused on tonal components by artificially modifying some sounds to offer a relevant perceptive indicator. The results of this second experiment will also be presented.
Just noticeable differences of loudness and sharpness for earth moving machines
Authors: Pedrielli Francesca , Carletti Eleonora , Casazza Camilla
Published: May 2008

This paper describes some results of a research concerning the improvement of the noise climate at the operator station of construction machines during real working conditions. Binaural noise signals were previously recorded and then used in subjective listening tests aimed at identifying the set of acoustic and psychoacoustic parameters which affect the auditory perception of these signals with respect to the annoyance sensation. Results showed that loudness and sharpness are the parameters best correlated to the annoyance. In order to verify the efficacy of some noise control solutions in improving the operator comfort conditions, the next necessary step is to build a specific metrics able to show the minimum differences in these parameters which are subjectively perceived. This paper describes the results of specific listening tests carried out in order to evaluate the differential thresholds of loudness and sharpness by the Method of Limits. The test was repeated at two different signal presentation levels. The loudness value of each original stimulus was varied by interval steps of 0,3 sone, while the Sharpness by interval steps of 0,02 acum. At this moment in time, the data analysis is still in progress and the ultimate results will be presented in the manuscript.
Advantages and disadvantages of fast and slow compression in hearing aids
Authors: Moore Brian
Published: May 2008

Compression is used in hearing aids to compensate for the effects of loudness recruitment. However, there is no consensus about the "best" compression speed. The theoretical advantages and disadvantages of slow and fast compression will be discussed. Studies comparing the relative merits of slow and fast compression have led to a great variety of outcomes. It is argued, following the work of Gatehouse and colleagues, that this is partly the result of a failure to consider individual differences and the auditory ecology of each individual. It is argued that listening in the dips of a fluctuating background sound, such as a competing talker, depends on the ability to process the temporal fine structure (TFS) of sounds, as represented in patterns of phase locking in auditory neurons. For people with a good ability to process TFS, fast compression can amplify sounds in the dips, increasing the effectiveness of dip listening. However, for people with a poor ability to process TFS, envelope cues may be critical for speech intelligibility and fast compression may disrupt such cues. It is proposed that a test of the ability to process TFS might be useful for selecting compression speed for an individual.
Dynamic compression in hearing aids based on an auditory model
Authors: Hohmann Volker
Published: May 2008

A multichannel dynamic compression algorithm is proposed that uses a novel nonlinear auditory filterbank which aims at effectively describing the basilar membrane (BM) response to arbitrary signals. It is based on a linear Gammatone filterbank, a subsequent instantaneous compression stage and a frequency-synthesis stage. In order to model the linear response to off-frequency tones and suppression effects from signals below the characteristic frequency (low-side suppression) the compression characteristics is controlled in each filter band by the deviation of the current sub-band instantaneous frequency from the band's center frequency. If the deviation is small, on-frequency components are prominent, and full gain and compression is applied. If the deviation is large, off-frequency components are prominent, and gain and compression is reduced. Simulations of responses to sinusoids as a function of frequency and level at a fixed BM place (i.e., within a fixed filter band) and for all filter bands (i.e., BM excitation patterns) show good correspondence with psychoacoustical excitation pattern models. Responses to two-tone stimuli quantitatively simulate psychoacoustical two-tone suppression. Likewise, nonlinear growth of simultaneous masking is quantitatively modeled. First results show that speech reception in modulated noise may be improved by the system for a subgroup of hearing-impaired subjects.
Acoustical frequency discrimination and pitch matching in bimodal and hybrid hearing
Authors: Baumann Uwe , Rader Tobias , Helbig Silke , Gstöttner Wolfgang
Published: May 2008

Frequency discrimination and pitch matching of implantees using combined electric and acoustic stimulation in either the same ear (EAS) or the opposite ear (bimodal condition) was assessed by means of adaptive procedures. EAS patients received either the MED-EL standard electrode or the recently introduced FLEX design with reduced diameter. Acoustic JNDF in EAS patients ranged from close to normal to grossly abnormal compared to a group of matched SNHL listeners. The median JNDF was 7.1% in the SNHL and 7.5% in the EAS group. There was no statistically significant difference in terms of JNDF between both groups of listeners. Frequency mapping was studied by means of an adjustment method where subjects were instructed to control the pitch of an acoustically presented sinusoid in reference to electrical stimulation. The findings demonstrate that the insertion of an intra-cochlear electrode does not significantly hamper the average frequency discrimination ability in EAS patients.
Speech-in-noise enhancement and sound localization with improved binaural hearing instruments
Authors: Wouters Jan , Doclo Simon , Moonen Marc , Van Den Bogaert Tim
Published: May 2008

Multi-microphone noise reduction schemes have become standard in commercial hearing aids and cochlear implants. Recent studies with bilateral hearing aids have shown that common adaptive directional microphone systems tend to distort localization cues, leading to inappropriate and reduced spatial awareness for bilateral hearing aid users. Here we show that binaural multi-microphone signal processing based on multi-channel Wiener filter (MWF) are capable of combining noise reduction with the preservation of directional hearing. Physical simulations and perceptual results from 10 listeners have been studied for different noise source scenarios, in different reverberant conditions, and for a number of signal processing schemes using up to 4 microphone inputs (2 each side). An overview is given of the localization performance and the speech reception benefits in these different listening conditions for the different noise reduction strategies. An adaptive directional microphone system (ADM) is used as a reference system. Signal processing based on MWF does, unlike ADM, provides a combination of noise reduction and preservation of spatial awareness. Moreover, in some conditions it even offers an improved spatial release from masking. The MWF outperforms the ADM in terms of localization and noise reduction if signals are not arriving from the most forward field of view.
Binaural hearing abilities of bilaterally fitted hearing aid users assessed using objective and subjective outcome measures
Authors: Behrens Thomas , Neher Tobias
Published: May 2008

Aided spatial hearing in the hearing impaired remains a rather sparsely explored topic. Therefore we do not know enough about the ability of the hearing impaired to exploit auditory cues for spatial hearing. In an attempt to advance our knowledge the following experiment was set up. A group of 21 experienced hearing aid users took part in a test using modern completely in the canal hearing aids in the field for at least 7 weeks. After acclimatization to the devices they were tested on a number of outcome measures. These included spatial unmasking, an interview administered selection of questions from the Speech Spatial and Qualities of hearing scale, and a baseline speech in noise measure. Spatial unmasking was assessed using three concurrent female talkers, with the target talker always presented directly ahead and the maskers presented either at +/- 50 degrees or both at 180 degrees. This presentation will provide results from the study described above along with possible relations to auditory and non-auditory predictors of spatial hearing performance.
Numerical modeling of Panphonics's G1 flat loudspeaker
Authors: Gulyás Krisztián , Augusztinovicz Fülöp
Published: May 2008

The basic idea behind AABC (Active Acoustic Barrier Control) is to reduce the sound radiation of a structure acoustically without influencing the vibration behavior of the structure. The large surface acoustic polymer material actuator/sensor systems are primarily meant to form the actuator/sensor system for the AABC concept. The main component of this package is a special active device: the rEMA - revised Elastic Mass Actuator based on the Panphonics' G1 panel loudspeaker element. This paper deals with the modeling and the low-frequency performance issues of the G1 flat loudspeaker. The aim was to create an accurate mathematical model to understand the operational principles of the loudspeaker, and then this model was used to optimize its low-frequency performance. Due to the special structure and the optimization tasks, a new numerical method was developed to model special multi-layer coupled vibro-acoustics systems based on the Finite Difference and Boundary Element Method. The investigation focuses on the mechanical behavior of the panel and describes the sound radiation properties also. The performed work was a part of the research of the project InMAR (Intelligent Materials for Active Noise Reduction) which was funded by the European Union.
Metrology and prediction for integrating a subsystem on a vehicle: Application to a fan system attached to the front end of a car
Authors: Moulet Marie-Hélène , Mapagha Saul , Martin Vincent
Published: May 2008

The car industry must satisfy physical acoustic and vibratory objectives in order to comply with safety and comfort norms. Among others, efforts entering the chassis should be limited. When a host structure is excited by a vibratory system, called subsystem, it also vibrates and may radiate an acoustic field. Usually, the subsystem is first tested on a bench and the question is then to deduce the efforts entering the chassis from those entering the test bench. Globalising notions of impedance already made evident via deformable structure configurations are used presently through measurements in a technological configuration, the complexity of which does not allow modelling. The case of a fan system attached to the front end of a car is under study. Some metrological conclusions, as well as more dedicated ones concerning the fan system itself, will be given. Finally, on the calculation and prediction sides, expected properties that are not satisfied (such as symmetry in measured matrices) are seen to be of no great consequence in the present case.
Numerical investigations into the squeal propensity of a railway disc brake
Authors: Chiello Olivier , Lorang Xavier
Published: May 2008

This paper comes within the scope of a research program concerning with the reduction of the squeal noise generated by high power railway disc brakes. It focuses on the numerical results provided by a finite element model of the brake including unilateral contact and Coulomb friction at the disc/pad interfaces. In particular, the dynamic stability of the sliding equilibrium is investigated by performing a complex eigenvalue analysis of the linearized equations verified by the structural displacement fields. Complex eigenvalues and complex modes are used to estimate the squeal propensity of the brake in a given frequency range. The effect of various mechanical and geometrical parameters is studied in order to better understand the mechanism leading to the system instability.
Sound propagation on a high pressure gas pipe
Authors: Schirmacher Rolf , Baars Robert
Published: May 2008

At a two stage high pressure gas compressor with intermediate cooler, the cooler radiate a tone at the compressor rotational frequency of approx. 160 Hz. For the design of noise reduction devices, the mechanism of sound propagation in between the compressor an the cooler on a DN 400 steel pipe with 24 mm wall thickness and 160 Bar internal gas pressure was to be determined. By non-invasive vibration measurements on the pipe, the dominance of the fluid borne sound (natural gas) over the structure borne sound was found. Later, pressure measurements in the pipe clearly approved this result. The concepts upon which the measurements and data evaluation were based (propagating modes of coupled system, wave separation, etc) as well as the results are presentend.
Analysis of the acoustic signals backscattered by a tube using the time-frequency representations
Authors: Laaboubi Mustapha , Aassif Elhoucien , Latif Rachid , Maze Gerard , Decultot Dominique , Moudden Ali , Dariouchy Abdelilah
Published: May 2008

The normal excitation of a tube immersed in water by the acoustic plane wave, circumferential waves are generated inside the shell. These circumferential waves, standing form stationary waves on the circumference of the tube for some frequencies. These stationary waves, constituting resonances of the tube which are perfectly visible on the backscattered spectrum. Moreover, the studies carried out on the diffusion of a plane acoustic wave by target were based primarily on the use of the monodimensional methods (Temporal domain and/or frequencial domain). To exceed the disadvantages of these methods, in this work, we used the time-frequency representations such as the Short-Term Fourier Transform (STFT), Wigner-Ville Distribution (WVD) and Wavelet Transform method. These representations are applied to a theoretical signal backscattered by a tube of aluminium, copper and steel with radii ratio b/a = 0.95 (a is the external radius, and b the internal radius). From the time-frequency images obtained we have visualized the dispersion of circumferential waves (S0, A1, S1,œ) and identified these different waves. This analysis permits to compare between these time-frequency representations. And also we have compared between the cut-off frequencies of circumferential waves obtained from these representations and those computed by the proper modes theory of the vibration.
Influence of material properties on sound radiation for flat loudspeakers
Authors: Ramis Soriano Jaime , Alba Fernandez Jesús , Frances Monllor Jorge , Arenas Bermudez Jorge P.
Published: May 2008

This work aims to analyze numerically, analytically and experimentally, the influence on sound power radiated, frequency response, and directionality of different materials in flat loudspeakers. Concerned experimental setup, the excitation force consists in a shaker which his mechanical characteristics are completely known. Finite Element Method has been used to simulate this configuration .The analytical model used couples the electrical, mechanical and acoustical system of the panel. The parameters of the panel material which we are focused are elastic modulus and loss factor. The analytical formalism serves as the basis to study the structural and vibrational behaviour of the system that also is analyzed with experimental results. The results obtained from different materials and different excitation position serves as basis for optimum configuration in this type of speakers.
Ground-borne vibration case studies in residential constructions
Authors: Perry James
Published: May 2008

Case studies are examined for several new high-rise residential and hotel buildings proposed for construction over or near underground and surface transit systems. Baseline vibration measurements were taken on grade and existing building structures and projected against standard functional vibration criteria (ANSI S3.29 / ISO 2631 2). Through post-construction testing and client feedback, the vibration and structure-borne noise impacts were field verified. Study results offer insight into the correlation between these vibration criteria and the subjective human perceptions of various sensitivities. Further items of discussion include the relative effects of various architectural constructions, building types, and applications as well as mitigating sound and vibration controls.
Characterisation of structure borne sound sources from measurement in-situ
Authors: Elliott Andrew , Moorhouse Andy T.
Published: May 2008

In-situ source characterisation methods are those which involve measurements made whilst source and receiver are coupled as they would be in a real installation. Potentially in-situ source characterisation may account for the physical reality lost in the "black box" approach. There are other potential benefits such as ease of measurement. In this work a structure borne sound source is characterised using in-situ measurements of blocked force and coupled mobility. Promising results from the method have been presented previously. Further to this, an extension of the method allowing the use remote measurement positions has been developed. Using reciprocity, the extended method will further ease measurement for situations where access poses a problem. The extended method is outlined and some preliminary validation results are presented.
Evaluation of acoustic environments using deteriorated speech sound
Authors: Nagatani Yoshiki , Sakaguchi Takefumi , Hosoi Hiroshi
Published: May 2008

Aged or hearing-impaired people require better acoustical environments for higher QOL. Many methods of evaluating the quality of speech in acoustical environments have been developed. However, since they mainly focus on quality of speech in bad conditions (e.g. environments with huge noise or long reverberation), they are not suited for evaluations in generic environments such as normal houses or public facilities for aged people. For instance, the scores of D value (deutlichkeit) or speech transmission index (STI) are too high and not sensitive enough to refer to such environments. The intelligibility test using normal speech sounds cannot clarify the differences between different room environments because the intelligibility scores reach almost 100 percent in ordinary room environments. Therefore, we proposed a new evaluation method using deteriorated speech sounds. In this method, signal-processed speech sounds are presented to trial subjects under target sound environments. In this study, Japanese monosyllabic speech sounds convoluted by the impulse responses of room reverberations were presented through a headphone in order to simulate the architectural acoustic environments. As a result, it was shown that this new method could detect the small difference of sound environments, which the conventional methods could hardly evaluate.
A system for automatic detection and correction of detuned singing
Authors: Lech Michał , Kostek Bozena
Published: May 2008

The aim of the paper is to show a system engineered for automatic detection and correction of detuned singing. For this purpose, existing methods of fundamental frequency detection and pitch correction are reviewed. In addition, main characteristics of some existing detuning systems are presented. As algorithms for fundamental frequencies detection and pitch correction, the fast autocorrelation and HPS (Harmonic Product Spectrum), and the modified phase vocoder and PSOLA (Pitch-Synchronous Overlap-Add) are chosen and examined. Four possible combinations of the algorithms are reviewed not only in the context of fundamental frequency detection and pitch shifting correctness but also with regard to the quality of the resulting singing signal. Experiments are performed on both male and female singing samples consisting of a variety of tones and various articulations. Basing on the obtained results, it is concluded that the HPS and PSOLA algorithms are the optimum choice as means to correct detuned singing. In addition, listening tests are performed in order to confirm objective measurements of pitch detection and correction. The system is implemented in JAVA. Conclusions are drawn and proposals of improvements are provided.
Spectrotemporal and f0 evidence for a theory of enhancement
Authors: Dutta Indranil
Published: May 2008

According to Keyser & Stevens, 2006, apart from universal features that are utilized to calculate motoric instructions, there is a second parallel and language-specific process called enhancement, where additional motoric instructions are processed for the enhancement of the saliency of features that are in danger of being obliterated. The "Standard View" on the distinction between voiced stops (VS) and voiced aspirated stops (VAS) in Hindi proposes that the breathy-murmured release following VAS is sufficient to make the contrast between the VS and VAS (Ladefoged & Maddieson, 1996). Hence, audible distinctions between VAS and VS during closure (CD/VLT) are not relevant in maintaining the contrast. Results from our studies show that CD is a relevant cue in making a distinction between aspirated and unaspirated stops. Spectral intensity measures show that speakers employ different glottal configurations to obtain the breathy/modal voicing contrast. VLT durations are inversely correlated with f0, such that longer VLT for VAS leads to further lowering of f0. This evidence suggests that f0 perturbations, differences in the durations of closure, and nature and extent of aspiration, all contribute in making the four-way stop contrast possible in Hindi, in support of a theory of enhancement.
Hearing the tongue and lips of vowel gestures: A new differential paradigm
Authors: Schwartz Jean-Luc , Vallée Nathalie , Kandel Sonia
Published: May 2008

It is an old question to know to what extent a listener can recover the articulatory dimensions of a speaker's gesture. In the case of vocalic configurations, a number of experiments have been done on expert phoneticians, showing that vowel height can be reasonably well estimated from the sound, but the front-back and lip rounding dimensions are much less well recovered. However, almost nothing has been done on naive listeners, due to the difficulty to perform absolute estimations in the lack of explicit phonetic knowledge. In the past years, we have developed an original paradigm, exploiting differential rather than absolute estimations. We show that French listeners, even naïve, are able to discriminate to a certain extent which vowel in a given pair has a higher vs. lower or more front vs. more back tongue position, or more or less rounded lips. From these data, we have elaborated an algorithm enabling to estimate what are the internal representations of vowel height, frontness and rounding, and correlated these representations with acoustic parameters, F1, F2-F1 and F'2 in Barks appearing to play a key role in the auditory recovery of these three motor dimensions.
Mathematical Evidence For Motor Theories of Speech Perception
Authors: Ramsay Gordon
Published: May 2008

Theories of speech perception have often proposed, controversially, that recovery of phonological information from the acoustic signal necessarily requires implicit knowledge of the physical processes underlying speech production. Most of the evidence supporting this hypothesis has been derived empirically from experiments in speech perception. In this paper, we show that many of the basic ideas underlying motor theories of speech perception can also be derived mathematically from first principles, using classic results in stochastic nonlinear filtering theory. Two key results, the Duncan-Mortenson-Zakai and Fujisaki-Kallianpur-Kunita theorems, show that the optimal state estimator for any partially-observed nonlinear stochastic dynamical system always takes the form of a "matched filter", which is itself a nonlinear stochastic dynamical system, the structure of which mimics, and resonates with, the structure of the original. Interpreting this in the context of speech perception, recovery of phonological information from sound produced by a human vocal tract necessarily involves construction of an internal model of the processes implicated in speech production. A key prediction, which we explore, is that any such model need not reproduce all the details of these processes, but is only required to predict the lawful conditional correlation between gesture and sound.
Speech recognition with body-conducted speech using differential acceleration
Authors: Nakayama Masashi , Ishimitsu Shunsuke , Nakagawa Seiji
Published: May 2008

Speech-recognition rates decrease in noisy environments. The body-conducted speech, conducted in solids such as body and skins, has a noise-robust characteristics and can be served for recognition systems even in 98 dBSPL-noise (-20 dBSNR) environments. However, the body-conduction could not capture high frequency sounds. Conventional methods for the improvement in sound quality of body-conducted speeches needs both speeches and body-conducted speeches. In this paper, a new body-conducted speech retrieval technique in sound quality without a speech signal itself is proposed. First, high-frequency components in the body-conducted speech were emphasized using differential acceleration. Second, conventional noise reduction method was adopted to make a clear body-conducted speech from a retrieval speech which contains constant noise. The recognition experiments using the proposed method showed that it improved recognition rate in all speakers.
Front-end processing of a distant-talking speech interface for control of an interactive TV system
Authors: Omologo Maurizio
Published: May 2008

This work addresses a research activity being conducted for the development of a user-friendly interface for the access to a virtual smart assistant enabling the interaction with TV-related digital devices and infotainment services. In the given scenario, the users can speak in a natural and comfortable way, not encumbered by any hand-held or head-mounted microphone. The environment is typically a living room, equipped with digital TV, Hi-Fi audio devices, etc., and populated by a group of people (e.g., family members). Among the most challenging issues involved in this scenario are a multi-microphone front-end for an effective processing of the given acoustic scene, an Acoustic Echo Cancellation (AEC) component to compensate the sound produced by loudspeakers, and a multi-modal distant-talking spoken dialogue system. As far as the front-end is concerned, multiple speaker localization, speech activity detection, speaker identification, and speech recognition will have to perform accurately even when AEC is applied to the given microphone array. The paper aims to present preliminary results of this research, which is being conducted under the European Project DICIT.
An efficient frame selection approach to variable frame rate analysis for noise robust speech recognition
Authors: Tan Zheng-Hua , Lindberg Bo/rge
Published: May 2008

This paper presents a low-complexity, effective variable frame rate (VFR) analysis method that conducts frame selection on the basis of a posteriori signal-to-noise ratio (SNR) weighted energy distance. It has two characteristics. First, energy distance (instead of cepstral distance) is used to make it computationally efficient and thus enable a finer granularity in search as compared with cepstral distance criterion. Secondly, SNR weighting is used to emphasize the reliable regions in noisy speech signals. In terms of frame selection, it is experimentally found that the method is able to assign a higher frame rate to fast changing events such as consonants, a lower frame rate to steady regions like vowels and no frames to silence, even for very low SNR signals. The VFR method is applied to speech recognition in noisy environments to improve noise robustness. Being a method that takes effect in the time-domain, it is moreover combined with spectral- and cepstral-domain techniques to gain further improvement. Experiments are conducted on the Aurora 2 database, which is the TI digits database artificially distorted by adding different noises, and very encouraging results are obtained.
Perceptual asymmetries induced by category learning
Authors: Bonnasse-Gahot Laurent
Published: May 2008

This paper takes an information-theoretic approach to study the perceptual consequences of the neural encoding of categories (e.g. vowels). We focus on two well-known psychophysical phenomena: categorical perception, characterized by greater cross-category than within-category discrimination, and perceptual magnet effect, stating that perceptual space is wrapped around prototypical instances of a given category, leading to a better ability to discriminate stimuli near non-prototypical exemplars of a category than near prototypical ones. Introducing a perceptual distance based on the Kullback-Leibler distance between the patterns of activity evoked by two stimuli, we show that both categorical perception and prototypical effects emerge from maximizing information contained in the neural representation about a set of categories. We discuss the relations between these two psychophysical phenomena and show that they go in hand with another kind of asymmetric effect. The discriminability between a category prototype and a non-prototype is not symmetric, depending on which stimuli serves as a referent. Non-prototypical stimuli are judged closer to prototypical exemplars than the reverse. Quantitative and qualitative comparisons with experimental data and previous theoretical work are presented and discussed.
Order effects and peripherality: a cross-linguistic perceptual study using an [i]-[e] articulatory continuum
Authors: Karypidis Charalampos , Colazo-Simon Antonia , Costagliola Angelica V. , Cecote Stein Cirineu , Guglielmi Gilles
Published: May 2008

This paper attempted to provide further evidence on whether peripherality is a universal bias triggering stimulus order effects. Participants from four linguistic systems - Brazilian Portuguese, Salentinian Italian, French and Spanish - participated in the study. In Experiment 1, listeners identified as /i/ or /e/ the ten stimuli of a continuum prepared with an articulatory model. Experiment 2 consisted in an AX 'same-different' discrimination task including pairs of stimuli differing in one or two steps along the continuum. Discrimination was better when the more peripheral stimulus was presented second in a pair. However, these order effects were found mainly within the /i/ category and only for Italian and French. Experiment 3 was a replication of Experiment 2, except that two modifications were introduced: a. identical pairs were added, and b. one- and two-step pairs were separately examined. Order effects were again found but, this time, they covered a wider area of the continuum. Peripherality alone was not able to account for order effects. The role of focalization was later discussed.
Perception of an infrequent assimilation: Labial-to-alveolar assimilation in German
Authors: Mitterer Holger
Published: May 2008

In German (and many languages) the alveolar nasal /n/ can assimilate to [m] if followed by a labial (e.g. 'in Berlin' ← "i[m] Berlin"). In a German spontaneous speech corpus (KielCorpus), one also finds a few cases in which an /m/ followed by an alveolar consonant surfaces as [n] (e.g., 'samstag' ← "sa[n]stag", Engl. Saturday). Four experiments investigated whether there is a similar pre-lexical compensation process for these labial-to-alveolar assimilations as previous research uncovered for alveolar-to-labial assimilations. This turns out to be the case: German and Dutch listeners-the latter potentially unfamiliar with this type of assimilation-tend to perceive the assimilated [n] in "sanstag" as /n/ if presented in isolation ("an"), but as /m/ if presented with minimal context ("ansta"). For German listeners, this context effect is larger if they hear the complete word "sanstag", introducing an additional lexical bias to perceive the [n] as /m/. Finally, phonetic detail such as transitions of the [s] fricative pole are also important: If the assimilated "an" is spliced into another /s/-initial syllable by the same speaker, the context effect is reduced. Perceiving infrequent assimilations seems therefore similar to perceiving frequent assimilations.
An overview of time-reversal acoustics
Authors: Fink Mathias
Published: May 2008

Time-reversal invariance is a very powerful concept in physics. In the field of acoustics where time reversal invariance occurs, time-reversal experiments may be achieved simply with arrays of transmit-receive transducers, allowing an incident wave field to be sampled, recorded, time-reversed and re-emitted. Time reversal mirrors (TRMs) may be used to study random media and chaotic reverberating structures. Common to these complex media is a remarkable robustness exemplified by observations that the more complex the medium between the probe source and the TRM, the sharper the focus. TRMs open the way to new signal processings that interest imaging, detection, telecommunications and therapy. Time reversal mirrors have plenty of applications including ultrasonic therapy and medical imaging, non destructive testing, telecommunications, underwater acoustics, seismology, sound control, home automation. An overview of these fields will be presented.
The detection and characterization of buried resonant targets by iterative, single-channel time reversal
Authors: Roy Ronald A. , Waters Zachary J. , Dzikowicz Benjamin R. , Holt R. Glynn
Published: May 2008

A technique for detecting buried resonant targets is described in which enhanced signal-to-noise ratio and convergence to a narrowband signal is achieved using the iterative time reversal of backscattered echo returns. [Waters et al., J. Acoust. Soc. Am. 122, 3023 (2007).] The center frequency of the converged-upon signal is typically a resonance frequency of the target and thus could be used for classification and identification purposes. This technique could offer a straightforward means for enhancing target return levels in a noisy or cluttered environment using existing SONAR systems. The procedure consists of exciting the transducer with a broadband pulse, digitizing the echo return windowed about the target, reversing the data stream in time, and using this signal as the source waveform for the next interrogation pulse. We report results derived from a multi-pronged investigation that includes numerical modeling, high-frequency (100 kHz - 2 MHz) scaled tank experiments employing both free-field and buried targets, and mid-frequency (20 kHz -200 kHz) buried target experiments run in a shallow pond. [Work supported by The Office of Naval Research and the Center for Subsurface Sensing and Imaging Systems (NSF ERC Award No. EEC-9986821).]