bluuebunny/crossref_metadata_2025 huggingface dataset:
Effects of reverberation and spatial diffuseness on the speech intelligibility of public address sounds in subway platform for young and aged people
Authors: Kim Yong Hee , Soeta Yoshiharu
Published: May 2013

This paper investigated how young and aged people response differently on the public address sounds in subway platform according to various reverberation and diffuseness. Both word intelligibility (WI) and listening difficulty (LD) tests were adopted as experimental method. Twelve simulated sound fields at the same position were prepared according to the combination of four different reverberation times (RT) and three different interaural cross-correlation coefficients (IACC). Auralized sounds with anechoic test words of actual station names were presented to young and aged subjects at the fixed sound pressure level. As results, LD results showed significant correlation with RT for both young and aged subjects, whereas significant correlation of WI results and RT was found only in aged subjects. Aged subjects showed worse speech intelligibility performances on public address sounds than young subjects due to their worse hearing level. RT was found as the most important factor to determine speech intelligibility for both young and aged subject, whereas aged subject showed better speech intelligibility performances with lower IACC. From the regression analysis, LD rating was estimated from the measured RT and IACC. Additionally, the effects of word familiarity, individual noise sensitivity and hearing level on the speech intelligibility were discussed.
Level-dependent effects on speech intelligibility
Authors: Pérez-González Patricia , Lopez-Poveda Enrique A.
Published: May 2013

The effect of noise on speech intelligibility is typically measured using fixed-level speech (or noise) and varying the speech-to-noise ratio (SNR). An assumption of this procedure is that intelligibility mostly depends on the SNR and barely depends on speech level. The effective SNR, however, (i.e., the SNR in the internal stimulus representation), possibly depends on peripheral compression. Indeed, compression could facilitate or hinder intelligibility for negative and positive SNRs, respectively. Insofar as compression varies with level, speech intelligibility might also vary with speech level. Here, we tested these hypotheses by measuring percent correct digit triplet identification as a function of speech level for fixed SNRs. Measurements were carried out for normal-hearing subjects and for hearing-impaired subjects with linear cochlear responses, as assessed using the temporal masking curve method. Results for both groups suggest that the detrimental effect of the noise on intelligibility is larger for speech levels near threshold, particularly for negative SNRs, a result that cannot easily be explained by compression. Alternative explanations for the result are discussed.
The Glasgow Monitoring of Uninterrupted Speech Task: A naturalistic measure of speech intelligibility in noise
Authors: MacPherson Alexandra , Akeroyd Michael A.
Published: May 2013

When listening to speech in noisy environments parts of the speech signal are often missed due to masking, degradation, or inattention. Sometimes the message can be reconstructed, but reallocating resources to recover the missed information can affect the efficiency and speed at which the message is understood. A slowing of processing will be particularly detrimental if there are few opportunities for understanding to catch up, e.g., when listening to the radio. It is likely then that the monitoring of uninterrupted flows of continuous speech will be especially sensitive to hearing or listening impairments. We developed a new task to test this. The Glasgow Monitoring of Uninterrupted Speech Task (GMUST) requires participants to listen to a 10-min segment of continuous speech while monitoring a scrolling transcript of the speech on the screen in front of them for any word substitutions. The proportion of word substitutions listeners are able to correctly identify is then used as a measure of speech intelligibility. When compared to speech reception thresholds (SRTs) given by a standard speech-in-noise test pilot results indicate higher SRTs for the GMUST. This suggests that the GMUST could be a more sensitive measure of deficits in speech-in-noise understanding than standard speech-in-noise tests.
Can envelope recovery account for speech recognition based on temporal fine structure?
Authors: Apoux Frederic , Youngdahl Carla L. , Yoho Sarah E. , Healy Eric W.
Published: May 2013

Over the past decade, several studies have demonstrated that normal-hearing listeners can achieve high levels of speech recognition when presented with only the temporal fine structure (TFS) of speech stimuli. Initial suggestions to explain these findings were that they were the result of the auditory system’s ability to recover envelope information from the TFS (envelope recovery; ER). A number of studies have since showed decreasing ER with increasing numbers of analysis filters (the filters used to decompose the signal) while intelligibility from speech-TFS remains almost unaffected. Accordingly, it is now assumed that speech information is present in the TFS. A recent psychophysical study, however, showed that envelope information remains in the TFS after decomposition, suggesting a possible role of ER in speech-TFS understanding. The present study investigated this potential role. In contrast to previous work, a clear influence of analysis filter bandwidth on speech-TFS understanding was established. In addition, it was shown that near perfect speech recognition from recovered envelopes can be achieved with as many as 15 analysis filters. Finally, the relationship between analysis and auditory filter bandwidths was explored in ER. Taken together, the present findings suggest that a role of ER in speech-TFS understanding cannot be excluded.
Intervention for restricted dynamic range and reduced sound tolerance: Clinical trial using a Tinnitus Retraining Therapy protocol for hyperacusis
Authors: Formby Craig , Hawley Monica , Sherlock LaGuinn P. , Gold Susan , Parton Jason , Brooks Rebecca , Payne JoAnne
Published: May 2013

Hyperacusis is the intolerance to sound levels that normally are judged acceptable to others. The presence of hyperacusis (diagnosed or undiagnosed) can be an important reason that some persons reject their hearing aids. Tinnitus Retraining Therapy (TRT), a treatment approach for debilitating tinnitus and hyperacusis, routinely gives rise to increased loudness discomfort levels (LDLs) and improved sound tolerance. TRT involves both counseling and the daily exposure to soft sound from bilateral noise generator devices (NGs). We implemented a randomized, double-blind, placebo-controlled clinical trial to assess the efficacy of TRT as an intervention for reduced sound tolerance in hearing-aid eligible persons with hyperacusis and/or restricted dynamic ranges. Subjects were assigned to one of four treatment groups (2x2): Devices: NGs or placebo NGs and Counseling: Yes or No. They were evaluated at least monthly on a variety of audiometric tests, including LDLs, the Contour Test for Loudness for tones and speech, and word recognition measured at each session's comfortable and loud levels. Eighty percentage of the participants who received full treatment benefited significantly; whereas the other treatment groups demonstrated ≤ 45% treatment efficacy. Treatment dynamics and examples of improved word recognition post-treatment will be described. [Work supported by NIH R01 DC04678.]
Evaluation of a binaurally synchronized dynamic-range compression algorithms for hearing aids
Authors: Ernst Stephan , Grimm Giso , Kollmeier Birger
Published: May 2013

Binaural cues such as interaural level differences (ILD) are used, among other cues, to organize auditory perception and to segregate sound sources in complex acoustical environments. Dynamic-range compression working independently at each ear in a bilateral hearing aid, however, can alter these ILDs, potentially affecting sound source segregation. Binaural synchronization of compression algorithms might thus be necessary to preserve potentially beneficial spatial cues. This study presents a binaurally linked model-based fast-acting dynamic compression algorithm designed to approximate the normal-hearing basilar membrane input-output function in hearing-impaired listeners. Aim of the evaluation was to assess the effect of binaural synchronization on speech intelligibility and listening effort in spatial masking conditions in comparison to bilateral fitting. Spatially symmetric and asymmetric masking conditions were used. A conventional multiband dynamic compression algorithm both implemented in a bilaterally independent and in a binaurally linked version, was tested as a reference. Hearing impaired listeners were aided individually with the algorithms for both experiments. Results indicate a small preference toward the model-based algorithm in challenging masking conditions. However, no benefit of binaural-synchronization could be found even for the fast-acting compressor, suggesting a dominant role of the better ear in all experimental conditions. [Work funded by BMBF 01EZ0741 and DFG FOR1732.]
Communication aid utilizing bone-conducted sound via teeth by means of mouthpiece form actuator
Authors: Muramatsu Mikio , Kurosawa Junko , Oikawa Yasuhiro , Yamasaki Yoshio
Published: May 2013

Since bone-conducted sound is conveyed to cochlea directly, without passing through eardrum, it is audible even for hard-of-hearing people whose inner ears are still normal. In this study, we utilize bone-conducted sound via teeth so as to support sound communication. We implement a bone-conducted actuator on a tooth, while actuators of prevalent hearing aids are attached to mastoid, forehead, or jaw in general. Teeth convey sound excitation more easily, because they are bare bones, not covered with skin. Our hearing aid is made in the form of mouthpiece, and it can be readily put on and taken off from a tooth. Plus, we carry out experiments regarding sound localization and thresholds of bone-conducted sound via teeth, using this actuator. The results offer hearing characteristic of bone-conducted sound via teeth and show that examinees can perceive right and left using bone-conducted sound via teeth. In addition, we also attempt to record vibrations of teeth through a microphone, which is embedded on the mouthpiece form actuator. The aim of this study is to realize a hybrid actuator that enables both hearing and recording simultaneously and to suggest a new communication aid system not only for hard-of-hearing people but also for the robust.
Three curious results of radiation loading calculations
Authors: Garrelick Joel M.
Published: May 2013

One of the many joys in reading Miguel C. Junger’s contributions to the journal and elsewhere was the physical interpretations that accompanied theoretical results, which were at times unanticipated but always enlightening. In three instances however, he found the task of offering such an explanation particularly challenging and was beguiled by the fact. This is documented in his paper “Three apparently paradoxical results of sound radiation theory” [J. Acoust. Soc. Am. 106(3, Pt. 1), 1589–1590 (1999)]. The specific results are as follows: (1) At frequencies above coincidence, the power radiated by a point driven infinite thin plate exposed to a low impedance fluid is equal to the power radiated by that plate when submerged in a high impedance fluid. (2) At low frequencies, the entrained mass that is associated with a translating circular piston fully submerged in an acoustic medium is equal to that acting on the piston when it is baffled and exposed to the medium on one side only. (3) The low frequency admittance of a fluid loaded infinite thin plate driven by a point force exhibits a spring-like reactance and yields a phase angle magnitude that is equal to that of the plate when driven by a line force. These three topics are revisited in this paper with an attempt to distinguish between paradox and coincidence.
Sound radiation by parallel coated plates separated by a fluid layer: Now and Then
Authors: Stokes Ann
Published: May 2013

In 1987, I had the opportunity to work with Miguel Junger on a paper entitled “Sound Radiation by parallel coated plates separated by a fluid layer.” A critical piece of the analysis involved analytically inverting a 5x5 matrix, which resulted in an extremely complex algebraic expression for the far-field of a point-excited 5-layer configuration. In order to understand the physical mechanisms, and validate the results, Dr. Junger developed asymptotic forms of the solution for familiar configurations and made use of spring-mass models to understand resonance enhancement effects. This talk introduces our current semi-analytical waveguide analysis of sound radiation by parallel, multi-layer plates. An advantage of the waveguide approach is that finite-length structures can be modeled, and axial discontinuities, including ribs and wavebearing bulkheads, can also be included very efficiently. Physical mechanisms are interpreted in terms of the propagating and evanescent waves of the structures. A focus of this talk is the extension of the perfectly matched layer (PML) model of the exterior fluid to waveguide models.
Intelligibility of interrupted speech in listeners of different age and hearing status
Authors: Shafiro Valeriy , Sheft Stanley , Risley Robert , Gygi Brian
Published: May 2013

In most real-world communicative environments, speech signals are fragmented and incomplete due to masking by other concurrent sounds. Experiments in the perception of gated and time-compressed speech provide a useful approach for systematically investigating factors involved in the perception of temporally fragmented speech. In the current work, the intelligibility of spoken sentences was measured following periodic signal interruption at various square-wave gating rates with time compression applied by either omitting or doubling silent intervals during gated-off times. Across interruption rates (0.5–16 Hz), speech perception of younger and older normal-hearing and older hearing-impaired adults was similar for slow interruption rates, but differed substantially for the faster rates. This rate-dependent variation was consistently found for different interruption methods and conditions. Importantly, speech perception at fast interruption rates correlated strongly with pure-tone thresholds and performance on speech-in-noise tests, while it did not correlate with results from tests of working memory or spectro-temporal pattern discrimination. These findings suggest that different perceptual processes are involved in the perception of interrupted speech at slow and at faster interruption rates, and have implications for developing diagnostic tests applicable to real-world listening environments. [Work supported by NIH.]
Impaired speech recognition under a cognitive load: Where is the locus?
Authors: Mattys Sven
Published: May 2013

Improving the validity of speech-recognition models requires an understanding of the conditions in which speech is experienced in everyday life. Listening conditions leading to a degradation of the signal—noise, competing talkers, disordered speech—have received most of the attention in that literature. But what about adverse conditions that do not alter the integrity of the signal, such as listening to speech under a non-auditory cognitive load (CL)? Drawing upon a variety of behavioral methods, this presentation investigates the effects of a concurrent attentional or mnemonic task on the relative reliance on acoustic cues and lexical knowledge during speech-perception tasks. The results show that listeners under CL downplay the contribution of acoustic detail and increase their reliance on lexical-semantic knowledge. However, greater reliance on lexical-semantic knowledge under CL is a cascaded effect of impoverished phonetic processing, not a direct consequence of CL. Ways of integrating CL into the functional architecture of existing speech-recognition models are discussed.
Quality of voices processed by hearing aids: Intra-talker differences
Authors: Muralimanohar Ramesh Kumar , Kronen Caleb , Arehart Kathryn , Kates James , Pichora-Fuller Kathleen
Published: May 2013

The effects of hearing aid signal processing depend on the voice characteristics of a talker. For example, we have found that the perceptual and acoustic consequences of hearing aid signal processing vary across talkers and that these effects can be explained, in part, by the acoustic differences between the different voices. However, we find that different utterances spoken by the same talker are also differentially affected by the hearing aid signal processing. In this study, we quantified the acoustic and perceptual consequences of hearing aid signal processing on several utterances spoken by the same talker. The voices were presented in quiet and in the presence of babble noise (10 dB SNR). The voices were processed with varying amounts of frequency compression, wide dynamic range compression and noise suppression (spectral subtraction). We considered intra-talker variability using an objective quality metric (Hearing Aid Sound Quality Index) and perceptual ratings of quality. We analyzed sentences from several different speech corpora, including the Hearing-in-Noise Test (HINT), IEEE, and TIMIT. The results showed interactions between the effects of signal processing and the acoustic characteristics of specific utterances spoken by an individual. [Work supported in part by GN ReSound and NSERC.]
Perception of Thai distinctive vowel length in noise
Authors: Onsuwan Chutamanee , Tantibundhit Charturong , Saimai Nantaporn , Saimai Tanawan , Chootrakool Patcharika , Thatphithakkul Sumonmas
Published: May 2013

A forced choice identification perception experiment using 150 monosyllabic rhyming-word stimulus pairs (with identical consonants and tone) in four conditions of white Gaussian noise was conducted to explore vowel confusions in Thai, a language with nine monophthongs and length (short-long) contrast for all vowels (e.g., /i/-/i:/ and /o/-/o:/). Each stimulus containing speech and noise portions is equal in length. Perceptual results of 18 vowels from 36 Thai listeners at a noise level (SNR) of -24 dB, where the percent intelligibility is the most interpretable, showed that stimuli with short vowels are more accurately perceived than those with long vowels (93.46 vs. 85.64%) with /o:/ and /e:/ as the most confusable. Interestingly, asymmetrical confusions are observed with very few short vowels being misperceived as long vowels, but a larger number of long vowels misperceived as short. Consistent with previous studies of perception of English vowels in white noise [e.g., Benki (2003)], the findings confirm perceptual robustness of vowel height (correlating with F1) over vowel front/backness (correlating with F2). Lastly, an analysis for listeners’ misidentified responses shows that the listeners generally favor short over long vowels.
Perception of English vowels and use of visual cues by learners of English and English native speakers
Authors: Pereira Yasna I.
Published: May 2013

English vowels may be difficult to discriminate for many learners of English (L2 learners). Research in L2 speech perception has shown that the use of visual cues improves speech perception, at least for visually salient contrasts. This study investigated the use of visual cues in the perception of English vowels by L2 Advanced learners (Spanish native speakers) and English native speakers (ENS). Thirty-seven L2 learners and 20 ENS were given a vowel test that presented real CVC words in audio (A), audiovisual (AV), and video-alone (V) mode. The A and AV conditions were presented in noise (-10 dB SNR) to ENS and in quiet to L2 learners. For ENS, identification rates were significantly higher in AV than in A condition, suggesting there were visual cues to vowel identity. For L2 learners, A scores were significantly lower than for ENS, and AV scores did not differ significantly from results in A mode. This suggests low sensitivity to visual cues to vowel identification, though L2 learners achieved better than chance scores when forced to attend to visual information in the V mode. These results support previous findings of relatively poor sensitivity to visual cues to phoneme identity in L2 learners.
The development of clear speech strategies in 9–14 year olds
Authors: Pettinato Michèle , Hazan Valerie
Published: May 2013

This study investigated the development of global clear speech strategies of child talkers. Two groups of 20 talkers aged 9–10 (children) and 13–14 (teens) were recorded in pairs while they carried out spot the difference picture tasks (diapix), either hearing each other normally (NB condition) or with one talker hearing the other via a three-channel noise vocoder (VOC condition). Acoustic-phonetic analyses focused on the talker having to overcome the communication barrier. Data were compared to those for twenty of the adults in Hazan and Baker [J. Acoust. Soc. Am. 130, 2139–2152 (2011)]. The three age groups did not differ in task transaction time for NB, but children took significantly longer to complete the task in VOC than teens or adults who took equally long. Children spoke at a slower speech rate overall than teens, while teens and adults did not differ; all groups significantly reduced their speech rate in VOC relative to NB. Adults hyperarticulated vowels in VOC, but children and teens showed only minor adaptations. These results suggest that although 9–10 year olds use some strategies to clarify their speech in difficult conditions, other strategies continue to develop into late adolescence.
The role of high-frequency envelope fluctuations for speech masking release
Authors: Jo/rgensen So/ren , Dau Torsten
Published: May 2013

The speech-based envelope power spectrum model [sEPSM; Jo/rgensen and Dau (2011), Jo/rgensen et al. (2013)] was shown to successfully predict speech intelligibility in conditions with stationary and fluctuating interferers, reverberation, and spectral subtraction. The key element in the model was the multi-resolution estimation of the signal-to-noise ratio in the envelope domain (SNRenv) at the output of a modulation filterbank. The simulations suggested that mainly modulation filters centered in the range from 1 to 8 Hz contribute to speech intelligibility in the case of stationary maskers whereas modulation filters tuned to frequencies above 16 Hz might be important in the case of fluctuating maskers. In the present study, the role of high-frequency envelope fluctuations for speech masking release was further investigated in conditions of speech-on-speech masking. Simulations were compared to various measured data from normal-hearing and hearing-impaired listeners [Festen and Plomp (1990), Christiansen et al. (2013)]. The results support the hypothesis that high-frequency envelope fluctuations (>30 Hz) are essential for speech intelligibility in conditions with speech interferers. While the sEPSM reflects effects of energetic and modulation masking in speech intelligibility, the remaining unexplored effect in some conditions may be attributed to, and defined as, “information masking.”
Using landmark detection to measure effective clear speech
Authors: Boyce Suzanne E. , Hamilton Sarah , MacAuslan Joel , Krause Jean , Smiljanic Rajka , Bradlow Ann
Published: May 2013

While the relationship of speaking style to intelligibility under challenging conditions has been established, it is a common observation that some speakers seem to be more intelligible than others for most listeners. In previous work, we have reported that automatic measures based on the technique of Landmark Detection appear to track differences between Clear and Conversational speaking style. One question that remains is whether Landmark measures can be used to predict which speakers are most likely to produce highly intelligible speech. In this study, we took advantage of a set of previously acquired databases covering a total of 31 American English speakers who produced Clear and Conversational Speech to examine correlations between our Landmark-based measures and the Clear Speech productions of highly intelligible speech. Across these databases, we had data on intelligibility for 13 speakers. Results showed that speakers with high overall intelligibility in Clear Speech showed significantly different patterns on Landmark-based automatic measures, compared to speakers with more moderate performance on intelligibility measures. Applications of these results to problems in speech technology, linguistic education, and clinical practice will be discussed.
Feedback active noise control in a crew rest compartment
Authors: Sachau Delf , Pabst Oliver
Published: May 2013

Active systems for noise cancelation are typically used to reduce low frequency noise (below 500 Hz). In this study, a method to reduce the delay in the control loop of feedback noise control systems is discussed. There are certain applications in which we may choose to control a specific frequency range; hence, a selection of the control-band must be performed. This selection is commonly done with analog filters introducing delay in the signal path. However, in feedback systems, the reference signal is calculated from the error signal which makes their performance more sensitive to these delays. In this contribution, the filtered-error feedback filtered-reference LMS algorithm (FE-FBFxLMS) is analyzed. Consequently, control-band selection is performed in an auxiliary loop removing the additional delay required in the signal path. This method of band selection also allows a reduction of the delay due to the anti-aliasing filters by using faster low-order filters in terms of group delay. With this delay reduction, higher noise attenuation is expected at the cost of a slower convergence rate. This method is applied to control broadband noise in a single channel feedback system in simulation and experiment.
An investigation into the relationship between sound quality and demodulation ratio for parametric loudspeakers
Authors: Ikefuji Daisuke , Nakayama Masato , Nishiura Takanobu , Yamashita Yoichi
Published: May 2013

Recently, parametric loudspeakers with a powerful ultrasound have been utilized for reproduction of sound to particular area. Amplitude of the ultrasound is modulated with a target audible sound. The emitted amplitude modulated wave is demodulated into the target audible sound in the air. Demodulation ratio affects sound quality of the reproduced sound with parametric loudspeakers. Moreover, demodulation ratio depends on the distance between the parametric loudspeaker and the listener. Therefore, the listener should utilize parametric loudspeakers at the suitable distance, which is required for sufficient demodulation of the amplitude modulated wave. Thus, we have proposed the criterion for measuring the demodulation ratio. However, it had not been conducted to investigate a relationship between subjective sound quality and demodulation ratio. In this paper, we investigate the relationship for estimation of the suitable distance. We carried out subjective evaluation experiment for confirming sound quality at each distance. As a result, we confirmed the consistency about fluctuation tendency of the subjective sound quality and demodulation ratio at each distance.
Monterey Bay ambient noise profiles using underwater gliders
Authors: Chandrayadula Tarun K. , Miller Chris W. , Joseph John E.
Published: May 2013

In 2012, during two separate week-long deployments, underwater gliders outfitted with external hydrophones profiled the upper 100 m of Monterey Bay. The environment contains various noises made by marine mammals, ships, winds, and earthquakes. Unlike hydrophone receivers moored to a fixed location, moving gliders measure noise variability across a wide terrain. However, underwater mobile systems have limitations such as instrument and flow noise, that are undesired. In order to estimate the system noise level, the hydrophones on the gliders had different gain settings on each deployment. The first deployment used a 0 dB gain during which the ambient noise recordings were dominated by the glider. The second used two hydrophones, one with a 0 dB gain and the other with 20 dB. Apart from system sounds, the higher-gain hydrophone also recorded far-away sources such as whales and ships. The noise recordings are used to estimate the spectrograms across depth and record time. The spectrograms are integrated with the glider engineering data to estimate histograms of noise power as a function of depth and glider velocity. The statistics from the two different deployments are compared to discuss the value of gliders with external hydrophones in ambient noise studies.
Measuring the spatial characteristics of the ambient noise field from an autonomous underwater vehicle
Authors: Fried Stephanie , Schmidt Henrik
Published: May 2013

For autonomous underwater vehicles (AUVs), the primary method of sensing the local environment is through acoustics. The local noise field contains a wealth of information the AUV uses—from target tracking to communication to general understanding of the environment. A measure of the spatial composition of the ambient noise field can provide details about the physical environment as well as information for the AUV to incorporate into its control decisions. The challenge is in accurately measuring the directionality of the noise field from a single line array and continuously updating this measure to reflect changes in the environment and additional information as the AUV moves. Here we present a method for continuously assessing the spatial characteristics of an ocean ambient noise field measured by an AUV with a towed hydrophone array.
Effective ice model for under-ice propagation using the fluid-fluid parabolic equation
Authors: Heaney Kevin D. , Campbell Richard L. , Freitag Lee
Published: May 2013

An approach is presented to permit efficient computation of the waterborne acoustic field in the presence of sea ice. The range-dependent wide-angle parabolic equation (PE) is used to model the acoustic field with a hard, lower-density layer (ice) placed above the ocean and seafloor. The ice layer is characterized by its thickness, compressional speed, density, and attenuation. Acoustic loss due to sea ice is primarily driven by conversion to shear waves, and in this model the effect will be approximated by attenuation within the ice layer. Rough interface scattering at the air-ice and ice-water interfaces will be handled by generating range-dependent realizations from a data-derived ice thickness model. An inversion for ice parameters is conducted using the work of Jin et al. [J. Acoust. Soc. Am. 96(5)] and by matching the transmission loss of Diachok [J. Acoust. Soc. Am. 59(5)]. Model-data comparisons between this three-layer PE and measurements the taken in the Fram Strait in 2010 and in the Canada Basin in 2011 will be presented.
Optimizing acoustics for spoken word using active acoustics
Authors: Ellison Steve , Germain Pierre
Published: May 2013

Teleconferencing, classrooms, lectures, drama, and worship all rely on spoken word to convey a message. The successful receipt of the message is largely dependent on the acoustic of the room, the vessel for the message, both in amplified and unamplified situations. A room that supports teleconferencing well will have minimal early reflections and reverberation, yet the same room may be used in a classroom environment that would benefit from early reflections. Active acoustic systems can be used to provide this acoustic energy. Early Reflection Benefit (ERB) will be revisited, and active acoustic systems utilized for speech in various contexts will be described.
Subjective experiment on suitable speech-rate of public address announcement in public spaces
Authors: Yokoyama Sakae , Tachibana Hideki
Published: May 2013

In such public spaces as railway stations, airport terminal buildings, shopping arcades, etc., it is often the case that information provided through public address (P.A.) system is deteriorated by the influences of reverberation and background noise. This problem is serious especially for the case of providing various information and emergency evacuation alarms in the case of such disasters as earthquake and fire. Regarding this acoustic problem in public spaces, the authors performed field surveys in various public spaces, in which actual P.A. announcements were recorded and reproduced in an anechoic room by applying the 6-channel recording/reproduction technique to realize 3D aural impression. As a result of the subjective experiments on speech intelligibility (ease in listening) performed in the simulated sound field, it has been found that the speed of speech is an essential condition as well as reverberation and noise level. To investigate the way of improving speech intelligibility in such spaces, subjective experiment was performed by changing the reverberation time, background noise level, and the speech rate in Japanese announcements at several steps. In the experiment, a Text-To-Speech (TTS) software was used to change the speech rate. As the test subjects, students from abroad participated in the experiment and the difference of the effects of the adverse conditions between natives and non-natives was also investigated.
Design of the new public address system for the cathedral of Münster, Germany
Authors: Behler Gottfried , Vorländer Michael
Published: May 2013

One of the most renowned cathedrals in Germany, the Dom St. Paul in M̈unster was completely closed for renovation for almost one year. During this time, the entire electro acoustical sound reinforcement system has been renewed. As for most buildings of this type, the acoustical situation is far away from optimal. This mainly is due to a huge reverberation time, which makes the understanding of spoken words almost impossible. Moreover, the old concept of sound reinforcement by using distributed loudspeaker systems all over the church is not satisfying anymore with respect to nowadays demands for quality and speech intelligibility. Due to the situation that the number of people in the cathedral during service times is varying from only some hundreds to over 2500 a more exible PA system is required, that takes into account that only occupied areas inside of the cathedral should be supplied with amplified sound. To achieve the target, an entirely new concept for the sound reinforcement based on digital signal distribution and modern digitally operated array loudspeakers was developed. The requirement for the speech intelligibility was to reach at least an STI of more than 0.5. The system will be discussed and results will be shown.
Acoustics evolution of the Moscow Conservatory Great Hall after renovation in 2011
Authors: Kanev Nikolay , Livshits Anatoly
Published: May 2013

The Great Hall of the Moscow P.I. Tchaikovsky Conservatory is one of the best concert halls in Russia. Its acoustics is appreciated very much by both musicians and audience. In 2011, the Great Hall was renovated, after renovation its acoustics conditions remained at very high level that was confirmed by means of objective impulse response measurements and subjective estimations. In order to control how acoustics conditions are changing with time the observation of main acoustics parameters is carrying out with half year gaps. In this work we present the results of four measurements fulfilled in June 2011 (just after the renovation), December 2011, June 2012, and December 2012. The reverberation time decreases at low frequencies, whereas it is stable at middle frequencies. Periodic variations of the reverberation time take place at high frequencies. These variations are probably connected with seasonal changes of temperature and humidity conditions in the hall. Changes of other acoustics parameters correlate to reverberation time.
Auditory object formation in Cope’s gray treefrogs (Hyla chrysoscelis)
Authors: Bee Mark A. , Schrode Katrina M.
Published: May 2013

Hearing and acoustic communication in “real world,” multi-source environments require animals to group sound elements produced by the same source into perceptually coherent “auditory objects.” However, research on nonhuman animal communication rarely investigates perceptual processes involved in forming auditory objects of communication sounds. We tested the hypotheses that spectral and spatial proximity promote the sequential integration of temporally separated sounds produced by the same source into coherent auditory objects of acoustic signals. Male gray treefrogs produce a pulsatile advertisement call; females prefer longer calls (= more pulses) to shorter calls and discriminate against calls missing pulses. We gave females a choice between a short but spectrally and spatially coherent call (25 pulses) and a longer call (35 pulses) in which alternating groups of 5 pulses had different frequencies (ΔF, 0–12 semitones) and came from different locations (Δθ, 0° or 90°). Females generally preferred the longer call at smaller values of ΔF and Δθ, indicating a role for spectral and spatial proximity in sequential integration. Under some conditions, however, subjects showed a surprising willingness to integrate pulses despite large ΔFs. Together, these data shed light on the perceptual cues that receivers exploit to form coherent auditory objects of communication sounds.
Acoustical tomography in the shallow water ocean: Dream or reality?
Authors: Roux Philippe
Published: May 2013

Acoustical tomography in shallow waters relies on the identification and tracking of stable ray arrivals. The variation of the arrival time of these rays is used to solve the inverse problem and to estimate the physical properties as sound speed variations associated to internal waves or currents. In practice, however, technical difficulties appear (1) when the number of resolved eigenrays in this multipath environment is too small using a set of point-to-point recordings or (2) when the travel-time fluctuations are dominated by the fast-evolving surface that blur the slower internal-wave driven perturbations around the thermoclyne. Recent experimental studies based on laboratory-scaled demonstrators showed the efficiency of array processing algorithms in combination with source-receiver arrays to tackle issue (1) above. Depending on the waveguide geometry, the acquisition procedure can also be adjusted to provide a separation between the slow and fast travel-time fluctuations. Inversion results based on laboratory experiments are presented that focus on the dynamics of (1) a gravity wave at the air-water interface and (2) a thermal plume in the water column. Finally, these results are discussed in the framework of ocean acoustic research.
Effective ultrasound-targeted microbubble destruction-mediated gene transfer into the livers of small and large animals
Authors: Miao Carol H. , Noble Misty L. , Song Shuxian , Sun Ryan R. , Kuhr Christian S. , Graves Scott S. , Keilman George W. , Morrison Kyle P. , Loeb Keith R. , Brayman Andrew A. , Paun Marla , Storb Rainer F. , Sun Samuel S.
Published: May 2013

Ultrasound (US)-targeted microbubble (MB) destruction (UTMD) can significantly enhance gene delivery in mouse livers when pDNA/MBs were injected into the portal vein (PV) with simultaneous US exposure using a focused transducer. However, this transducer was ineffective in enhancing gene transfer into rats. A 13-mm diameter unfocused transducer was designed and the delivery route of pDNA/MBs was modified into a specific liver lobe, resulting in >100-fold increase in luciferase expression in rats. To facilitate the translation into human application, many technical issues were explored in large animal models. We applied 1.1 MHz US to a targeted canine liver lobe with simultaneous injection of pDNA/MBs into a PV segmental branch and occlusion of the inferior vena cava. For more effective treatment of large tissue volumes, a 52-mm planar unfocused transducer was specifically constructed. Its apodized dual element configuration greatly reduced the near field transaxial pressure variations, producing a uniform field of US exposure for the treated tissues. Together with a 15 kW-capacity US amplifier, a 692-fold increase of gene expression in canines was achieved at 2.7-MPa. Transaminase levels and histology analysis indicated minimal tissue damage. These results demonstrated that UTMD is highly promising for safe and efficient gene delivery into the liver.
Small interfering ribonucleic acid delivery with phase-shift nanoemulsions
Authors: Burgess Mark T. , Porter Tyrone M.
Published: May 2013

Acoustic cavitation offers a unique approach to small interfering RNA (siRNA) delivery compared to current methods. Typically, preformed microbubbles are used as cavitation nuclei to permeabilize cells and facilitate siRNA entry into the cytoplasm. However, microbubbles are restricted to the vasculature space and suffer from stability issues that limit their applicability. Alternatively, phase-shift nanoemulsions (PSNE) possess the long circulation and extravasation properties of nanoparticles, while also serving as cavitation nuclei in tissue upon acoustic droplet vaporization. Here we report the use of PSNE for delivery of siRNA engineered to knockdown green fluorescent protein (GFP) expression. A cell suspension (5 x 106 cells/mL) of GFP expressing breast adenocarcinoma cells was exposed to 5 MHz pulsed ultrasound (4 MPa peak negative pressure, 3 cycles, 250 Hz, 100 s exposure duration) in the presence of PSNE (~109/mL) and free siRNA (1.8 µM). Flow cytometry was used to quantify GFP expression and cell viability. There was 20% (p < 0.05, n = 6) reduction in GFP fluorescence for cells treated with GFP siRNA and 80% (+/− 6.4%) cell survival. This work highlights the potential for PSNE to serve as interstitial cavitation nuclei for siRNA delivery in tissue. [Work supported in part by NIH grants R25CA153955 and R03EB015089.]
Non-invasive toroidal high intensity focused ultrasound transducer for increasing the coagulated volume in depth
Authors: Vincenot Jeremy , Melodelima David , Chavrier Francoise , Chapelon Jean-Yves
Published: May 2013

A device composed of 32 elements (78 mm2 each) arranged on a toroidal transducer (operating frequency: 2.5 MHz) was developed to increase the coagulated volume. To date, our previous work on toroidal transducers used the outer envelope of a torus as a reference surface. Here, the transducer geometry was based on the interior part of a torus. This produces a focus that is ring-shaped but the ultrasound beams also intersect between the principal focal ring and the transducer surface to form a secondary focal zone, which contributes to increase the size of the lesion. The radius of curvature was 70 mm with a diameter of 67 mm. A 7.5 MHz ultrasound imaging probe was placed in the center of the device. Twenty ablations were produced in vitro by using electronic beam steering, each ablation was created in 55 s. The average depth of intervening tissues (skin-fat-muscle) was 11±1 mm and the average depth of liver tissues that have been spared was 21±4 mm. No significant temperature rise in intervening tissues was measured (maximal temperature: 41°C). The dimensions of these ablations were an average diameter of 10±1 mm and an average depth of 27±3 mm.
Simulative measures for structure borne sound radiation of composites
Authors: Klaerner Matthias , Marburg Steffen , Kroll Lothar
Published: May 2013

Due to the high stiffness-to-weight ratio, composite structures tend to be acoustic sensitive. The sound emission of such radiating components is commonly measured by the sound power requiring the determination of the sound intensity in normal direction and—in numerical simulations—the sound pressure on the radiating surface. Assuming a unit radiation efficiency all-over the surface and neglecting local effects, the equivalent radiated power (ERP) is a common approach for an upper bound of structure borne noise. Therein, the sound power finally results from the squared velocity integrated over the radiating surface and the fluid impedance. As ERP usually requires extra post processing to consider the velocity in normal surface direction, the kinetic energy is essential in common FEA results including all velocity components apart from the normal direction, too. Moreover, ERP necessitates the knowledge of the radiating surfaces increasing the effort especially for complex geometries. Thus, the possibilities and limits of estimating the emitted sound power by the kinetic energy as well as using the ERP method will be shown. Test cases are a rectangular plate and a thin-walled bonded part with linear anisotropic material properties.
Effect of speaking rate variation on the perception of singleton and geminate consonants in Japanese by native and Korean listeners
Authors: Sonu Mee , Arai Takayuki , Kato Hiroaki , Tajima Keiichi
Published: May 2013

Perception of phonemic length contrasts in Japanese is difficult for non-native listeners. To better understand the source of this difficulty, the present study investigated native Korean listeners’ perception of consonant length contrasts at different speaking rates. Stimuli were created by modifying the duration of the second consonant of a non-word /ereC:e/ along a continuum to /ereCe/, where C was /k/ or /s/. The base words were spoken by a professionally trained native Japanese speaker with a carrier sentence at three rates, fast, normal, slow. Twenty-seven native Korean and eleven native Japanese listeners participated in a perception test. They listened to one of the created stimuli and identified whether the second consonant was singleton or geminate. Results show that even though Korean listeners’ perceptual boundary location between singleton and geminate consonants shifted according to speaking rate in a similar manner as the natives, their boundary location was more variable than native listeners at all speaking rates. Korean listeners also showed greater perceptual boundary width than Japanese listeners. These results suggest that Korean listeners have ambiguous criteria for phonemic length contrasts. Results are discussed in terms of the perceptual similarity between Korean and Japanese consonants. [Work supported by JSPS.]
Synchronous visualization of multimodal measurements on lips and glottis: Comparison between brass instruments and the human voice production system
Authors: Hézard Thomas , Fréour Vincent , Caussé René , Hélie Thomas , Scavone Gary P.
Published: May 2013

Brass instruments and the human voice production system are both composed of a vibrating “human valve” (constriction in a pipe) coupled to an acoustic resonator: lips coupled to the brass instrument or vocal folds coupled to the vocal tract. In both cases, the aeroacoustic coupling is responsible for the self-oscillations and the large variety of regimes. Additionally, brass instruments and voice share difficulties for the in-vivo measurement of the exciter activity. Hence, the development of a common tool is relevant. It is also relevant to explore the effect of some known differences between these systems, namely, the strength of the coupling and the physiological characteristics. This paper introduces components for the development of such a tool. First, two corpuses of multimodal measurements are presented: one for a singer's larynx during sustained vowels, one for a musician's lips during sustained notes. They include high-speed video (HSV) recordings, electrical impedance (EI) measurements (electrolabiograph/electroglottograph), and audio recordings (AR). Then, we introduce two estimation algorithms: (AW) one of the opening area waveforms from videos, (LF) one of the LF-model parameters on these waveforms. Moreover, we build a video tool displaying, synchronously, the signals (HSV), (AW), (EI), and (AR) in time and frequency domains. Finally, this tool is exploited to exhibit common behaviors and relevant differences between brass instruments and human voice.