bluuebunny/crossref_metadata_2025 huggingface dataset:
On using ‘‘loudness-weighted’’ SEL, LEQ, and DNL to assess noise environments
Authors: Schomer Paul
Published: July 2002

LEQ and DNL are commonly used to assess the long-term noise environments around airports, roads, etc. The fundamental building block to these assessments is the A-weighted sound level or SEL. These are used to compute LEQ or DNL. But many problems have surfaced over the simple use of A-weighting. The A-weighting has been shown to be deficient in assessing sound with strong low-frequency content, and other characteristics, like impulsiveness, are not properly accounted for by the A-weighting. However, no other simple filter has been shown to be superior to the A-weighting filter. One can consider building a more complicated filter that dynamically changes with level and frequency to better reflect human response than does the A-weighting. The equal loudness contours expressed in phons offers a set of curves that can be used to design such a filter. Like A-weighted sound level or A-weighted SEL, signals processed with such a new ‘‘filter’’ would be termed ‘‘loudness-weighted’’ sound level or ‘‘loudness-weighted’’ SEL. This author has analyzed many sounds to illustrate, test, and evaluate the concept of loudness-weighted SEL. This paper discusses the concept of using ‘‘loudness-weighting’’ in place of A-weighting and the results indicated by the analysis of many common sounds.
Characteristics of amplitude modulation following responses in man
Authors: Mauer Günter , Doering Wolfgang H.
Published: July 2002

Amplitude modulation following responses (AMFRs) are a subset of envelope following responses (EFRs) and can be recorded from the human scalp. In AMFRs continuous amplitude-modulated (AM) stimuli with carrier frequency fc and modulation frequency fm are used. Responses arise at frequency fm and are largest for modulation frequencies in the range below 100 Hz depending on the subject’s age and state of arousal. Multi-channel recordings and brain electrical source analysis were carried out to get information about localization of AMFR generators and dependency on modulation frequency of the stimulus. Recording of AMFR seems to yield important information about frequency-specific hearing loss even in the low-frequency range. Responses are assumed to be evoked by neural excitation originating mainly from a basilar membrane region corresponding to the carrier frequency fc. However, in masking experiments, it could be shown that also regions of the basilar membrane related to frequencies up to several octaves above the carrier frequency contribute to the responses although stimulus bandwidth is very narrow. Models of cochlear mechanics were used to elucidate stimulus-response characteristics and effective stimulation bandwidth. Thresholds of AMFR in normal-hearing and hearing-impaired subjects are in close relation to psychoacoustically determined thresholds for the same stimuli.
Sensitivity of auditory units in the frog’s midbrain to amplitude decrements
Authors: Bibikov Nikolay G.
Published: July 2002

Responses to single and repetitive amplitude decrements in an otherwise steady best-frequency sinusoid were recorded in midbrain auditory units of the curarized grass frog. The modulators were either 1 or 12 full cycles of cosine function (frequencies: 10–100 Hz, maximal modulation depths: 6%–80%). All units were also classified according to their responses to pure tone and AM tone bursts. In some tonic units, which demonstrated enhancement of synchronization in the course of AM tone bursts, enhancement of the rate response to successive amplitude decrements was routinely observed. The response to the first decrement in succession could be omitted, but the probability of highly synchronized firing increased gradually to the end of the series of decrements. The effect could be explained by adaptation of the unit to a new average intensity level. Some onset units, which did not respond to any modulation periods of 100% AM tone bursts, demonstrated a prominent response to a single 10%–20% decrement in a pure tone. The dependence of the response upon duration of decrement (period of a cosine function) usually showed a prominent maximum in the range of 20–50 ms. The response latency increased with the decrement duration, showing that the firing was evoked by an amplitude recovery.
Speech synthesis of VCV sequence using a physiological articulatory model
Authors: Dang Jianwu , Honda Kiyoshi
Published: July 2002

A 3-D articulatory model has been constructed based on volumetric MR images for a Japanese male speaker, which is driven by a target-dependent control strategy to produce dynamic articulation for vowel sequences [Dang and Honda, ICSLP98]. This work describes a multi-point control strategy that is developed for generating vowel–consonant–vowel (VCV) sequence. Three control points are located on the mandible, tongue tip, and tongue dorsum, and they are operated in each muscle workspace. Contributions of the tongue and jaw muscles to each control point are evaluated by model simulation, and described as a weighted coefficient. Muscle activation signals are determined according to the articulatory vector from the current position to the target. The total activation pattern of the muscles is the sum of the activation signals calculated for the control points. Dynamic vocal tract shape is obtained by using the activation pattern to excite the model, and compared with MRI movies of VCV sequence for the same speaker [Masaki and Honda, J. Acoust. Soc. Am. 102, 3166(A) (1997)]. The VCV sequence is synthesized from a set of vocal tact area functions of the model, based on vocal tract widths in the midsagittal and parasagittal planes, and compared with real speech.
A preliminary study of speech transformation using empirically defined articulatory modes
Authors: Story Brad , Titze Ingo
Published: July 2002

In previous work it has been shown that a speaker-specific set of vocal tract shapes (acquired using MRI) corresponding to vowels can be decomposed into orthogonal components or ‘‘modes,’’ and effectively parameterized by the modal coefficients. Furthermore, a nearly one-to-one mapping was found to exist between the modal coefficients of the two most significant modes and the F1−F2 formant space. Thus an inverse mapping of time-varying formants extracted from recorded speech back to modal coefficients (and consequently to vocal tract area functions) was made possible; the derived sequence of area functions can be used in a simulation of the original speech utterance. In this study, it is first shown that similar orthogonal modes exist for three additional speakers. It will then be demonstrated that the time-varying modal coefficients determined for a recorded sentence via the inverse mapping of one speaker can also be used to create a sequence of area functions based on one of the other speaker’s vocal tracts. A subsequent simulation produces the original sentence but with the vocal tract characteristics of the new speaker. The results also suggest that the time-varying modal coefficients may define common gestures across speakers even though their acoustic characteristics can be different.
Perceptual training with native Japanese speakers on various L2 sounds
Authors: Pruitt Jesica C. , Akahane-Yamada Reiko , Strange Winifred , Kubo Rieko , Takada Tomoko
Published: July 2002

It has been found that adults can more robustly acquire segments in a second language (L2) with identification training which uses highly varied training stimuli, i.e., natural tokens from multiple talkers in multiple phonetic contexts. This approach has been applied systematically in cross-language training studies in which Japanese speakers were trained to acquire AE /r/ and /l/ [Lively et al., J. Acoust. Soc. Am. 96, 2076–2087 (1994)] as well as AE/v̥, æ, ʌ/ [Strange and Akahane-Yamada, J. Acoust. Soc. Am. 102, 3137(A) (1997)]. The current set of experiments expanded this line of research by testing whether the same procedure would be effective in training other consonants (/b/–/v/, /s/–/θ/ and /z/–/ð/) and vowels (11 AE vowels). With both consonants and vowels, the previous results were replicated; subjects showed significant improvement from pretest to posttest and showed generalization to an untrained talker. In the vowel training study, transfer of training across speaking style was further examined. Subjects were trained to distinguish vowels in CVC syllables in a carrier sentence context, and were then tested in both sentence and citation contexts. Interestingly, the improvements were significant for sentence contexts but not for citation. Taken together, these studies further our understanding of the complexities of L2 learning and training.
Recognition of Portuguese vowels by Galician-speaking and Spanish-speaking listeners
Authors: Feijóo Sergio , Fernández Santiago , Balsa Ramón
Published: July 2002

Galician, Portuguese, and Spanish are three romanic languages spoken in the Iberian peninsula. Galician and Portuguese are historically related and keep the same vocalic system of common latin: a, ε, e, i, ɔ, o, u. In Spanish the number of vowels was reduced to five: a, e, i, o, u. Nevertheless, Spanish has exerted considerable influence on Galician, contributing to weaken the vocalic contrasts ε–e and ɔ–o. The purpose of these experiments was to assess how Galician- and Spanish-speaking listeners perceived the vocalic system of Portuguese. Stimuli were pV syllables pronounced in a carrier phrase, where V was one of the seven vowels. Results show that Galician-speaking listeners have some difficulty in perceiving the contrasts ε–e and ɔ–o, probably due to the historical influence of Spanish, and despite the fact that F1 and F2 of the Portuguese vowels were similar to those reported for Galician vowels. Spanish-speaking listeners identified ε–e as a single vowel e, and ɔ–o as a single vowel o, associating the L2 vocalic inventory to their own vocalic system. Discussion is based on the acoustic characteristics of vowels in those languages.
Robust and adaptive spatialized audio for desktop conferencing
Authors: Ward Darren B. , Elko Gary W.
Published: July 2002

For multi-talker teleconferencing, providing multichannel audio to the listener has been advocated as an alternative to a conventional single audio channel. The use of multichannel audio provides a means of supplying the listener with spatialized audio cues. Specifically, it permits binaural signals to be delivered to the listener, which has distinct advantages for multi-talker teleconferencing. First it allows remote talkers to be virtually placed around the listener, providing the listener with the ability to easily discriminate between talkers based on their virtual location. Second, if a binaural microphone is used to acquire the talker’s speech signal, the listener can receive this binaural signal directly. Binaural signals can be delivered to a listener through headphones or using a loudspeaker array (which is more preferable for video conferencing). However, the use of loudspeaker arrays requires the listener’s head to be accurately positioned or else the spatialized audio effect will be lost. This paper contains discussion on the means of delivering binaural signals using loudspeaker arrays that allow the listener to move her or his head without losing the spatialized audio effect.
Hands-free teleconferencing unit using auxiliary loudspeaker and duo-filter echo canceller DSP
Authors: Shimauchi Suehiro , Haneda Yoichi , Kaneda Yutaka
Published: July 2002

In many telecommunication systems, hands-free operation is becoming popular. For this to be comfortable and easy, high echo cancellation performance and a simple setup are required. This paper presents a new hands-free teleconferencing unit that consists of a main loudspeaker, an auxiliary loudspeaker, a microphone, and an echo canceller DSP board. The auxiliary loudspeaker is used to reduce the acoustic coupling between the main loudspeaker and the microphone by reproducing the anti-phase sound of the main loudspeaker. Since the auxiliary loudspeaker is set very close to the microphone, its sound level is low enough for users not to hear. Experimental results showed a reduction in acoustic coupling of about 6 dB from 200 Hz to 1 kHz. This unit also incorporates a newly developed acoustic echo canceller (AEC) DSP. The AEC algorithm, which is based on a duo-filter control system [Y. Haneda, S. Makino, J. Kojima, and S. Shimauchi, ‘‘Implementation and Evaluation of an Acoustic Echo Canceller Using Duo-Filter Control System,’’ in Proc. EUSIPCO96 (1996), Vol. 2, pp. 1115–1118], helps to produce good and stable performance even during double-talk.
Interpretation of the different infrasounds detected in France during one year in a four-components prototype CTBT station
Authors: Blanc Elisabeth , Le Pichon Alexis , Cansi Yves
Published: July 2002

A four-components prototype CTBT station has been installed in France by the Department of Analysis and Monitoring of the Environment of the Atomic Energy Commission, for the development of instrument and methods for infrasound monitoring. The purpose of this paper is to present the characteristics of the main infrasound signals detected during a year. Infrasound from the Concorde supersonic airplane presents different phases corresponding to the different acoustic paths in the different layers of the atmosphere. The number of phases, their apparent velocity, and the signal shape are strongly dependent on the ray trajectory and then on the wind and temperature profiles. Characteristic seasonal variations have been determined. Microbarom signals have been used to follow the evolution of the ocean swell during periods of several days. Severe weather infrasound has been observed, related to lightning activity and convective motions of the atmosphere. Chemical industrial explosions provide interesting well-known sources used for testing the methods and for the development of propagation models. All these events are control events for the permanent evaluation of the detection and location capability of the station. A synthesis of one year’s observations is given.
One-year analysis of systematic detection and location of infrasounds in a protoype CTBT station
Authors: Le Pichon Alexis , Cansi Yves
Published: July 2002

Within the framework of the Comprehensive Test Ban Treaty, the Department of Analysis and Monitoring of the Environment (DASE) of the French Atomic Commission has developed an automatic processing software providing detection of coherent infrasound signals. The infrasound’s parameters are calculated by using the progressive multichannel correlation method as a real-time detector. Each week, an automatic bulletin summarizes all detections by giving the wave parameters (velocity and azimuth), the amplitude, the frequency range, and the correlation coefficient. The one-year bulletins enable the identification of most of the infrasounds (sonic boom of supersonic aircraft, mining explosions of known quarries, storms, ocean swell). Most of the man-made infrasounds are produced by quarry blasts and the Concorde. Other infrasounds are produced by severe weather (storms, hail). Excluding false detections, an average number of 40 detections is obtained each day. For some infrasounds, statistics show several phases related to the different reflections in the stratosphere and thermosphere with seasonal characteristics. In order to detect coherent signals with poor signal-to-noise ratio (detection in windy period, fine detection of phases with low amplitudes), the thresholds of detections are set at a low level. In the future, this study will help to determine operational thresholds.
Beamforming performance of a V-shaped array of sea-bed geophone sensors
Authors: Brind Richard J. , Goddard Nick J.
Published: July 2002

Geophone sensors on the sea-bed offer a mechanism for detecting and localizing acoustic sources in the ocean. Recent work is described in which a V-shaped array of sensor units was deployed on the sea-bed in a shallow water location. Each unit consisted of a three-component set of geophones and a hydrophone. Data were obtained on the response of the sensors to noise from ships in the vicinity, and to impacts on the sea-bed remote from the array. The positions of the units were determined by examining the phase of broadband energy at each pair of elements, and applying a least squares fit. The geophone data were then beamformed. Conventional and adaptive beamforming techniques have been applied, and the sensitivity of each to errors in the knowledge of the element positions assessed. The response of the array to noise from nearby ships has been compared to predictions. The beam pattern was found to be superior to that of an equivalent array of acoustic sensors. The ability of the V-shaped configuration in the experiment to resolve the wave number of the incident field has been investigated. The response of the array to energy arriving in the dispersive interface wave propagation path has been examined.
Evolution of vocal communication in the gray treefrog complex
Authors: Gerhardt H. Carl
Published: July 2002

The evolution of communication requires changes in both signals and the criteria used by receivers to select signals. The question arises: Do changes occur first in signals, first in receivers, or are tight co-evolutionary changes required in both signals and receivers? In the gray treefrog complex, spectral properties of calls are conserved: in both the diploid ancestral species (Hyla chrysoscelis) and in derived, tetraploid lineages (nominally, H. versicolor) there are two spectral peaks: a dominant band centered at about 2.4 kHz, and a secondary band (−8 to −10 dB) at about 1.2 kHz. Whereas females of the ancestral species have a weak preference for calls consisting only of the secondary band to calls having only the dominant band, females of the derived species strongly prefer calls with just the dominant band. Thus, an evolutionary change in preference has preceded any change in the spectral properties of the call. Temporal differences between the species include pulse rate, pulse duration, and pulse shape. Whereas females of the ancestral species use pulse rate alone for call discrimination, females of the derived species base preferences on all three properties, thus suggesting a co-evolutionary change. [Work supported by NSF and NIMH.]
Comparison of two in situ tuna target strength measurement methods
Authors: Bertrand Arnaud , Josse Erwan
Published: July 2002

Two approaches were developed for in situ tuna TS measurement in French Polynesia. One consists of coupling split beam TS measurement with sonic tracking of tagged fish. The other one consists of TS measurement on tuna aggregated close to a Fish Aggregating Device. The first method was used on well-identified fish, swimming freely in their environment and allowing observation of the effect of fish behavior on TS up to 500 m of depth. The second method was applied on juvenile fish located close to the surface. In this case, fish were not individually identified and a fish sample was caught by trolling line in order to determine the length and species composition of the aggregation. Both methods are complementary and results are similar according to body length.
Modeling considerations for acoustic scattering by fluidlike elongated zooplankton: Model intercomparisons and recommendations for euphausiids and copepods
Authors: Stanton Timothy K. , Chu Dezhang
Published: July 2002

Predictions of acoustic scattering by fluidlike elongated zooplankton are made using the deformed cylinder model and Distorted Wave Born Approximation over a wide range of animal size, shape, material property inhomogeneities, and orientation, as well as acoustic frequency. Each shape represents a digitized form of the shape of a euphausiid or copepod at low-, medium-, and high-resolution sampling of the shape (e.g., at low resolution, a straight smooth finite cylinder is used and at high resolution, a rough tapered bent finite cylinder is used). The results are analyzed in order to determine the conditions under which the various degrees of complexity in modeling the scattering are required. For example, the predictions illustrate that for modeling the scattering for animals near broadside incidence, relatively simple models can be used (smooth homogeneous bodies). However, for animals off broadside, which can be the case for downward-looking echosounders insonifying euphausiids and cope- pods, roughness and sometimes inhomogeneities must be used to model the scattering accurately (roughness effects dominate scattering well off broadside). Comparisons are made with laboratory data involving euphausiids, shrimp, and copepods. [Work supported by ONR.]
Meddies and front detection by acoustic tomography
Authors: Gaillard Fabienne
Published: July 2002

The CAMBIOS experiment was defined within the CANIGO project as an observing array for the Canary basin. A pair of 400-H acoustic tranceivers set up in the middle of the basin provided a 9-month reciprocal series of arrival amplitude versus time sequences, starting in July 1997 and ending in March 1998. In the Canary basin the temperature and current field are perturbed in the upper layers by the meandering and branching of the Azores front, and at mid-depth by the transit of warm and salty coherent structures called meddies. The first analysis of the acoustic measurements will be presented here, with particular attention dedicated to the signature of the front and meddies, both on direct and reciprocal travel times. [Work supported by the MAST III programme under Contract No. MAS3-CT96-0060.]
Hyperthermia therapy using phase conjugation
Authors: Roux Philippe , Porter Michael B. , Song Hee C. , Kuperman William A.
Published: July 2002

One of the advantages of ultrasound phased-array treatments is the noninvasive manner in which ultrasonic energy, and then heat, can be delivered to the treatment volume. However, the heat generated in the medium induces variations in the sound speed, which affects the acoustic focusing and leads to strong effects such as self-focusing or self-defocusing of the sound beam. In this work, a numerical simulation is used to study coupled acoustic-heat propagation in tissue. By simultaneously modeling the propagation of acoustic waves and the diffusion of heat, it was possible to simulate quantitatively acoustic refraction induced by heat in the treatment volume. Finally, using phase conjugation, one procedure to control temperature and maintain the acoustic focusing at the initially desired focal point was developed.
Sound scattering and local heating from pulse-driven microbubbles
Authors: Hilgenfeldt Sascha , Lohse Detlef , Zomack Michael
Published: July 2002

Using a bubble dynamical model based on the Rayleigh–Plesset equation, it is shown that the experimentally observed spectra of sound emitted from diagnostic microbubble contrast agents under the action of short, strong ultrasonic pulses can be understood from the interplay of enhanced sound emission due to nonlinear bubble oscillations and enhanced sound damping due to the occurrence of high frequencies in the emitted spectrum. The latter effect counteracts the pronounced increases in the scattering cross sections in the regime of nonlinear bubble response. Sound absorption furthermore leads to highly localized heating of the liquid (e.g., blood) over a distance of several μm near the collapsing bubble. This temperature rise is, however, too brief and the heated liquid volume too small to be hazardous to the patient’s health.
Effect of tone burst ultrasound in combination with PPAA (poly-propyl acrylic acid) on hemolysis of erythrocytes
Authors: Porter Tyrone M. , Mourad Pierre , Crum Larry , Murthy Niren , Hoffman Alan , Stayton Pat
Published: July 2002

In this study, high intensity focused ultrasound (HIFU) and PPAA (poly-propyl-acrylic acid) were combined at sublethal doses to achieve hemolysis of human erythrocytes. In a series of experiments, a suspension of erythrocytes (1A passive cavitation detection system) was utilized to monitor the acoustic emissions from the cell suspension during exposure to ultrasound. In the presence of PPAA, there is a significant increase in cavitation and hemolysis during ultrasound exposure at a pH of 6.1, but not at a pH of 7.4. PPAA is a pH-sensitive polymer, changing its conformation and its activity as the solution becomes more acidic, achieving its optimal active state at a pH of 6.1. The acoustic data and the activity of PPAA suggest that the polymer, in its active state, provides nucleation sites to the cell suspension and that cell/bubble interaction is the primary mechanism for cell damage.
Steered beams using a spatially phased array
Authors: Hughes W. Jack
Published: July 2002

A spatially phased transducer array has been developed and tested that can form a steered beam without the need for electronically applying phase shifts to an array of elements. In its simplest form, the transducer consists of two channels from spatially shaped cosine and sine apertures, where the sine channel is shifted 90 degrees and then combined with the cosine channel. The transducer produces a beam steered to a desired angle at a specific design frequency in either receive or transmit operation. Several have been fabricated and tested using low-lateral mode transducer materials such as polyvinylidene flouride (PVDF) and 1-3 PZT composite, and electroplated electrodes to form the cosine and sine apertures. The appearance of the electrode has led to calling this a ‘‘doily’’ transducer array. Beams were formed that were steered as far as 54 degrees with typical sidelobe levels of −20 dB. The apertures were also geometrically shaped to supress sidelobes along the steering axis. [This work was supported by Kam Ng of ONR.]
Ladle crane noise
Authors: Wiciak Jerzy W. , Iwaniec Marek , Panuszka Ryszard J.
Published: July 2002

The present paper considers the vibration analysis of the ladle crane (with hoisting capacity Q=160 tons) and noise caused by working units of the crane. The vibroacoustical investigations and theoretical analysis have been applied to the main bridge with box-trapezoid cross section. In the article there is shown comparison of the vibration acceleration and dynamic stress that have been measured in real structure with calculations obtained by means of the finite element method. The noise caused by working units of the crane have been presented for three operating conditions: when the tilting ladle was descending and was hoisting, and pass by load running of the teeming crane. [Work supported by TUMM DS 11.130.557, Cracow, Poland.]
The role of measurement standards in acoustics
Authors: Barham Richard G.
Published: July 2002

Recent times have seen a marked increase in public awareness of noise as both a cause of nuisance and a hazard to hearing. Consequently, demand for acoustical measurements has grown and is now an important issue across an ever broadening range of industrial sectors. If these measurements are to have any technical validity, it is essential that they be founded on some physical reference quantity or primary standard. In practice, this is achieved by means of a defined measurement infrastructure. In the UK, the National Measurement System serves this purpose and it is the role of the National Physical Laboratory (NPL) to establish the primary measurement standards to underpin the system. While different applications deal with a range of acoustical quantities, they all derive in some way from sound pressure. The primary standard for sound pressure is realized through the calibration of laboratory standard microphones, using the reciprocity method. NPL has for many years been at the forefront of research in this field and most recently has introduced a fully revised reciprocity calibration facility. The paper describes the facility and highlights the latest technical developments. It also discusses the value of third-part accreditation of such facilities.
The results of a unique Nordic HAKK interlaboratory REAT comparison
Authors: Poulsen Torben
Published: July 2002

The sound-field requirements in the hearing protector standard ISO 4869-1 are under discussion at present and are likely to be changed in a coming revision of the standard. The Nordic HAKK group comprises representatives from the hearing protector industry, labor protection institutes, audiological clinics, and universities. The test facilities at the laboratories of the members of the Nordic HAKK group cover various ways to fulfill the sound-field requirements of the ISO standard. Therefore, a round robin test was initiated with the aim of determining—and comparing between laboratories—the attenuation of two hearing protectors of the muff type and one of the plug type. The measurements were performed during the meetings of the group and with the group members (min. ten) as test subjects, thus permitting one group of subjects to be used by all laboratories. A statistical analysis of the results is in progress at the time of submission of this abstract. All factors (laboratory, protector-type, test-frequency, test-subject) are significant.
Alternative field methods of measuring hearing protector performance
Authors: Franks John R.
Published: July 2002

Real-world tests of hearing protector attenuation are scarce. This study evaluated data from three new field-test methods as compared to the subject-fit data from Method B of ANSI S12.6-1997 for the same protector. The new field-test methods were: fit-check headphone (FCH) method [K. L. Michael, Spectrum 15, Suppl. 1, 24 (1998)], modified fit-check in sound-field (MFCSF) method, and modified bone conduction loudness balance (MBCLB) method [T. W. Rimmer and M. J. Ellenbecker, Appl. Occup. Environ. Hyg. 12, 69–75 (1997)], all of which can be administered in small audiometric booths such as are commonly found in industry. Twenty audiometrically competent subjects naive to hearing protector use were tested in the laboratory and for the three field-test methods. The earplug tested was the EAR Express Pod Plug. The results showed that the FCH method and the ANSI A0S12.6 subject-fit method produced similar attenuation levels. The MFCSF method showed decreased attenuation levels at 250 and 500 Hz while the MBCLB attenuations were elevated at 250, 500, and 1000 Hz due to the occlusion effect of the HPD. The results suggest that the FCH method was the most feasible and reliable method for field testing, although it is limited to earplugs. [Portions of work supported by the U.S. EPA IA 75090527.]
The attenuation of combinations of ear muffs and ear plugs
Authors: Smoorenburg Guido F.
Published: July 2002

In a number of situations, a single hearing protector (earplug or earmuff) may not provide sufficient hearing protection. This may concern, for example, shooting noise in indoor firing ranges, shooting instructors also outdoors, and jet-engine testing sites. Combinations of earplugs and earmuffs may then be considered. The protection provided by these combinations cannot simply be obtained by adding the attenuation figures provided for the plug and muff individually. A principal limit is set by bone conduction of the sound through the skull to the inner ear. In addition, the total attenuation may be affected by acoustic coupling of the two protectors. It will be demonstrated how to estimate the total attenuation using attenuation measurements of combinations of three earmuffs (Bilsom Comfort, Clark 19A, Peltor H7A) and three earplugs (Bilsom Form, Bilsom Soft, Willson EP100) for 16 subjects and data on bone conduction. The experiment was carefully designed, aiming at a direct comparison of the attenuation for the protectors individually and together, within one subject and with a minimum of replacement of the protectors. The evaluation was based primarily on subjective measurements (REAT method) but the earmuffs have also been measured using miniature microphones in the outer ear canal.
Ultrasonic velocity and absorption in the binary mixtures of 1-butanol with isomeric butanediols at 298.15 K
Authors: Zorebski Edward , Zorebski Michal
Published: July 2002

From the early days of ultrasonic technology, ultrasonic velocity and ultrasonic absorption have been a rich source of information on the structure and condition of the materials trough in which the ultrasonic waves are propagated. Here an investigation into binary mixtures (in the whole concentration range) of 1-butanol with 1,2-butanediol and 2,3-butanediol at 298.15 K is reported. For the ultrasonic velocity measurements at 2.15 MHz, the pulse-echo-overlap method was used and the ultrasonic absorption in the frequency range 10–200 MHz was determined by the standard pulse technique, where the amplitude of the first transmitted pulse was registered as a function of distance. The results were compared with the previous data for 1-butanol with 1,3-butanediol and 1,4-butanediol and discussed in terms of molecular interactions in highly associated liquids.
Ultrasonic study of ionic association in aqueous solutions of lanthanide salts
Authors: Voleišienė Birutė , Miglinienė Gintara , Voleišis Algirdas
Published: July 2002

The association reactions between trivalent ions of lanthanum, cerium, neodymium, gadolinium, erbium, ytterbium, and nitrate or sulphate ligands has been investigated from ultrasound velocity dispersion measurement data. It is assumed that ultrasound velocity dispersion in aqueous solutions of lanthanide nitrates and sulphates is caused by an association process. The ultrasound velocity was measured by the ultrasonic laser interferometer within the frequency range of 3–200 MHz. The rates of formation of inner sphere complexes [LnNO3]2+aq and [LnSO4]+aq were calculated and the interaction between individual ions Ln(III) and water molecules is discussed across the series. A comparison of relaxation parameters for the nitrate solutions with corresponding sulphates was made. It has been shown that a magnitude of an ultrasound dispersion may be the quantity of complexation. Ultrasound velocity dispersion for nitrate solutions is much smaller than that of more diluted lanthanide sulphate solutions. The lanthanide nitrates form predominantly outer sphere complexes with some inner sphere substitution. In order to produce an appreciable amount of inner complexes in nitrate solutions for a similar dispersion and relaxation, a tenfold higher concentration of the nitrates must be used.
Temporal models of pitch processing
Authors: Yost William A.
Published: July 2002

Most models of pitch perception assume three basic stages of processing: cochlear filtering, haircell/neural transduction, and pitch extraction. In temporal models of pitch processing the pitch extraction stage is often based on autocorrelationlike processes. This talk will review our work using regular interval stimuli to study temporal models of pitch perception. Work with regular interval stimuli suggests that temporal models of pitch processing can account for the pitch and pitch strength of complex sounds with both resolved and unresolved harmonics. The pitch and pitch strength of regular interval stimuli appears to be based on the temporal properties of the waveform fine structure. Human and animal psychophysical results, physiological recordings from the cochlear nucleus, and model results will be presented. [Work supported by NIDCD and AFOSR.]
Internal representation of spectral information in dependence of signal type and auditory filter bandwidth
Authors: Laback Bernhard , Deutsch Werner A.
Published: July 2002

Previous studies have shown that enhancement of the spectral background (lower level spectral components) improves intelligibility as well as listening comfort of music signals in subjects with broader than normal auditory filters. This is in contrast to reported results on speech perception in noise, yielding reduced intelligibility as a result of spectral contrast reduction. A proposed model for the internal object representation by separation of ‘‘spectral layers’’ accounts for these opposing effects. ‘‘Spectral layers,’’ corresponding to different sound sources [in music: dominant/accompanying musical voices; in speech-in-quiet: formant peaks/modulation sidebands of the peaks (transitions); in speech-in-noise: speech foreground/speech background/noise components] are represented in coherent spectral amplitudes. The audibility of ‘‘spectral layers’’ is derived from individual masking properties. Pilot experiments with speech tokens and short music fragments agree relatively well with the model predictions: The effect of either increasing or reducing the spectral contrast on discrimination highly depends on the distribution of the perceptually relevant acoustical information in the ‘‘spectral layers’’ relative to their audibility. Performance varies for different signal types (simultaneous musical voices, vowel, consonant). [Work supported by the Austrian Academy of Sciences.]
An overview of reconstructing acoustic pressure fields using the HELS method
Authors: Wu Sean F.
Published: July 2002

The acoustic pressures radiated from complex vibrating structures are reconstructed by using the Helmholtz equation least-squares (HELS) method [Wang and Wu, J. Acoust. Soc. Am. 102, 2020–2032 (1997); Wu and Yu, ibid. 104, 2054–2060 (1998)]. Specific examples include an engine block and the interior space of a passenger vehicle. These structures are of arbitrary shapes and geometry, containing sharp edges and corners, and abrupt changes in surface contour. To test the robustness of the HELS method, measurements of field acoustic pressures are taken over a planar surface at a certain distance away from the structure. The reconstructed acoustic pressures, however, extend over the entire (nonplanar) surface of the structure. Note that the input data in these cases are not error free due either to measurement uncertainties or to the loss of the near-field effect. On the other hand, reconstructed acoustic pressures consist of predominantly the near-field effect. Hence the problem becomes mathematically ill-posed. To overcome this ill-posedness difficulty, an optimization scheme is developed which enables one to obtain satisfactory reconstruction results with a relatively few number of measurements in the field. The HELS method is shown to be effective in the low- to mid-frequency range, and can become a robust noise diagnostic tool for analyzing structure-borne sounds. [Work supported by NSF.]
Fast convergence algorithms for active noise control in vehicles
Authors: Vicente Luis , Masgrau Enrique
Published: July 2002

When the reference signal for the FXLMS algorithm is taken from an acoustic sensor, convergence can be very slow due to great eigenvalue spread. Using a nonacoustic sensor, such as a tachometer, cancellation of narrow-band noise in the sensed fundamental frequency and harmonically related ones can be achieved very fast, although other periodic noises and the underlying broadband noise will remain. Backward prediction errors resulting at the various stages of an adaptive lattice predictor (ALP) represent a time-domain orthogonalization of the input signal. An ALP structure, with the acoustic reference as input signal, before a FXLMS, makes up the FXGAL algorithm. Due to orthogonalization, FXGAL is significantly faster compared to FXLMS with reference from a microphone. When compared to FXLMS with a tachometer signal, it is not faster but it can cancel every periodic noise, independently of the harmonical relation between them, as well as the underlying broadband noise. Depending on the relative weights of the different components (periodic harmonically related, periodic nonharmonically related, and broadband noise) in the reference signal, the FXGAL algorithm can be an excellent alternative to FXLMS. Comparative results between FXLMS (with acoustic and nonacoustic reference) and FXGAL will be presented. [Work supported by CONSI+D-DGA.]
Noise reduction in acoustic signals using the perceptual coding and intelligent decision systems
Authors: Krolikowski Rafal , Czyzewski Andrzej
Published: July 2002

A new algorithm of broadband, nonstationary noise reduction was engineered employing the perceptual approach to the removal of noise from acoustic signals. It enables analysis and processing of sound according to characteristics of hearing sense, and employs a decision system to automatically adjust thresholds of masking. The presented algorithm provides an extension of perceptual coding applications, because it makes possible the reduction of noise included in source signals. Although noisy components of a signal may occur randomly, it is possible to estimate the noise distribution on the basis of signal analysis in silence passages of the transmission. Subsequently, the decision should be made as to the masking curves’ parameter settings which will make the noise inaudible. That is the reason why an intelligent approach is used in modeling some interrelations between available noise patterns and the noise affecting consecutive portions of useful signals. The decision system was implemented in various ways using fuzzy reasoning, rough set approach, and modified feedforward neural networks with functional links. The proposed methods of noise reduction and obtained results of noisy speech processing are presented in the paper. [Work supported by the Committee for Scientific Research, Poland, Grant No. 8 T11D 021 12.]
Issues in the design of an advanced unit selection method for natural sounding concatenative synthesis
Authors: Breen Andrew , Jackson Peter
Published: July 2002

This paper describes a method for selecting units from a database of recorded speech, for use in a concatenative speech synthesizer. The simplest approach is to store one example of every possible unit. A more powerful method is to have multiple examples of each unit. The challenge for such a method is to provide an efficient means of selecting units from a practical inventory, to give the best approximation to the desired sequence in some clearly specified way. The approach used in BT’s Laureate system uses mixed N-phone units. In theory, such units could be of arbitrary size, but in practice they are constrained to a maximum of three phones. This method dynamically generates the unit sequence based on a global cost. Units are selected using purely phonologically motivated criteria, without reference to acoustic features, either desired or available within the inventory. A sophisticated method of unit selection does not, however, guarantee natural sounding synthesis. The process of signal generation plays an equally important part. The style of speech generated by such an approach is highly influenced by the underlying nature of the original recorded speech data. The paper will discuss this and other limitations of purely concatenative synthesis.
On processing speech and non-speech signals in acoustics
Authors: Pazukhin Rostislav
Published: July 2002

In a nonspeech communication, the pitch, duration, formants arrangement, and other acoustic features are bearers of coded information. Certain meanings are assigned to them by convention. The processing of nonspeech signals, consists in distinguishing and, subsequently, interpreting them. Speech oral messages are not, by contrast, amenable to such one-average ‘‘technology.’’ Each oral utterance betrays a two-layer organization. Here, true information carriers are coded articulary changes within the speech channel. Such gestures may be successfully ‘‘read’’ directly in TADOMA communication (by the hand placed against the face of the speaker), or in the ‘‘inner speech’’ (through bioimpulses announcing rudimentary movements of our speech organs). Mostly, however, the invisible articulations need applying certain ‘‘echo-effects’’ to become perceivable. Usually, the ‘‘echoes’’ of the voice, noise, whistle, and light (cf. ‘‘lip-reading’’) are used. Accordingly, the acoustician’s mission in speech processing ought to be reduced to restoring—from the sound characteristics of utterances—the invisible changes within the throat of the speaker (through an improved ‘‘inverse mapping, the directional hearing methodology, etc.’’?). Starting from these data, physiologists and linguists could decipher messages using ‘‘lexicon of speech gestures.’’
Auditory displays using loudspeaker reproduction in a car cabin
Authors: Krebber Winfried , Gierlich Hans W.
Published: July 2002

Playback of binaural sounds generated by an auditory display is normally made by headphones. For automotive applications such as driving simulation or sound design, however, loudspeaker reproduction is often required. A setup using a cabin-specific four-channel loudspeaker arrangement is presented and discussed. In order to get a sufficient 3-D sound, the loudspeakers have to be equalized carefully. Best results could be achieved using individualized HRTF sets. The quality of loudspeaker reproduction in comparison to headphone reproduction was evaluated by several psychoacoustic experiments. Localization and distance perception experiments show that loudspeaker reproduction does not achieve the same quality as individualized headphone reproduction. However, for simulator applications the achieved quality is quite sufficient.
Communication and 3-D sound: Speech intelligibility and speaker recognition
Authors: Drullman Rob , Bronkhorst Adelbert W.
Published: July 2002

In a 3-D auditory display, sounds are presented over headphones in a way that they seem to originate from virtual sources in a space around the listener. The possible merits of such a display were investigated with respect to speech intelligibility and speaker recognition against a background of competing speech. Various conditions were investigated: speech material (words or sentences), presentation mode (monaural, binaural, or 3D), number of competing speakers (1–4), and virtual position of the speakers (in 45° steps around the frontal horizontal plane). Average results for 12 listeners show an increase of speech intelligibility for a 3-D presentation, with more than two competing speakers compared to conventional monaural or binaural presentation. The acuity to recognize a speaker is slightly better and the time required for recognition is significantly shorter for a 3-D presentation in the presence of two or three competing speakers. Although absolute localizability of a speaker is rather poor, spatial separation appears to have a significant effect on communication. For either speech intelligibility, speaker recognition, or localizability, no difference is found between the use of an individualized 3-D auditory display and a general display. [Work supported by the Royal Netherlands Navy.]
Simplified auralization of reflections in a virtual auditory environment
Authors: Dürrer Bernd
Published: July 2002

Based on the mirror-image method, a system for generating virtual auditory environments can simulate reflections using virtual sound sources. During the auralization process, the signal of each virtual sound source has to be convolved with the appropriate pair of HRTFs. In this investigation, a simplified model based only on interaural time and intensity differences was used for the auralization of higher-order reflections: This approach greatly reduces the necessary computational effort. Stimuli using the simplified model were compared in listening tests with stimuli using convolution with HRTFs. The tests were performed with artificial-head HRTFs (17 subjects) and individual HRTFs (9 subjects). In addition to the reduction of computational costs, significant perceptual effects on the extent of the auditory event were noted which are believed to be beneficial for certain applications (e.g., telecommunications). The usability of this model was evaluated in a speech intelligibility test in a concurrent speakers scenario. [Work supported by the EU project AUDIS (Esprit No. 22352).]