bluuebunny/crossref_metadata_2025 huggingface dataset:
Influence of duration on the perception of consonants /x/ and /j/ in Chinese
Authors: Feng Li , Yoshitaka Nakajima , Kazuo Ueda
Published: May 2013

In the current study, we investigated whether Mandarin Chinese native speakers' perception of /x/ and /j/ was affected by the duration of consonant parts. Two perceptual experiments, in which the method of constant stimuli was employed, were conducted. Six normal-hearing adults (four females) took part in the experiment. Their average age was 24 yr. All subjects were native speakers of Mandarin Chinese. In the first experiment, Chinese syllables that begin with /x/ or /j/ were extracted from a speech database, and the consonant parts were manipulated in terms of duration. As the duration of /x/ was decreased or the duration of /j/ was increased, to a certain extent, the consonant which had been originally /x/ was perceived as /j/, and vice versa. Synthesized noises instead of recorded consonants were utilized in the second experiment, similar effects of the consonant duration appeared.
Quality of older voices processed by hearing aids: Acoustic factors explaining inter-talker differences
Authors: Goy Huiwen , Pichora-Fuller Margaret K. , van Lieshout Pascal , Arehart Kathryn H.
Published: May 2013

Hearing aid signal processing algorithms are often evaluated with professional recordings of voices. However, hearing aid users often listen to older talkers who may have poorer voices than younger talkers. The purpose of this study was to quantify the extent to which the acoustic and perceptual consequences of hearing aid digital signal processing algorithms differ for talkers that vary in their vocal characteristics. There were six older talkers (3 males and 3 females) selected from a larger database of 79 talkers; their voices had good, moderate, or poor quality based on perceptual data from younger and older listeners. 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). There were interactions between signal processing and voice characteristics as measured using the Hearing Aid Speech Quality Index (HASQI) and listener ratings of perceived sound quality of the processed speech. In this paper, we examine the possible acoustic sources that may explain these interactions, including inter-talker differences in formant space and pitch variation. [Research supported by NIH, GN Resound, NSERC, and the Canada Research Chairs program.]
The effect of background noise on the ability to perceive and remember unrelated words in nonnative listeners
Authors: Avivi-Reich Meital , Leung Caterina Y. , Schneider Bruce A.
Published: May 2013

Nonnative listeners find it more difficult to meet the challenges presented by additional background noise than do native listeners [Ezzatian et al., Speech Commun. 52, 919–929 (2010)], but it is not known whether it is more difficult for them to remember what was said in noisy situations than it is for native listeners. Previous studies have acknowledged that the effect of background noise on the ability to perceive and remember unrelated words is greater in older adults than younger adults. The present study investigates auditory memory performance in nonnative younger adults, using a paired-associate paradigm in three conditions: quiet, continuous babble and babble during word presentation only. Noise levels were adjusted to equate for individual differences in the ability to identify single words in noise. The initial results suggest that nonnative listeners perform similarly to native young adults in the quiet and continuous conditions but worse in the babble during the word-presentation-only condition. These results suggest that stream segregation may be slower in nonnative listeners when the masker and the target words start at the same time.
Advantage of talker differences and spatial separation for speech-on-speech listening in younger and older adults with good audiograms
Authors: Besser Jana , Pichora-Fuller Kathleen , Festen Joost M.
Published: May 2013

Older adults have more difficulty than younger adults understanding speech when there is competing speech, even if they have good audiograms. Age-related differences in listening may be due to declines in auditory temporal processing and/or cognition. We administered the LiSN-S [Cameron et al. (2011)] to measure speech reception thresholds (SRTs) in younger and older adults with good audiograms. There were four test conditions, in which the target and competing speech were presented with the same or different voices at the same or different locations. Compared to younger listeners, older listeners obtained worse SRTs in all test conditions and they realized less advantage from talker differences and spatial separation between the target and competing speech. For both groups, the results obtained in the four test conditions were strongly associated with each other. We also assessed cognitive abilities and auditory temporal processing in the older adults. LiSN-S results in this group were strongly associated with measures of cognition, measures of temporal processing (tapping the use of fine structure and gap cues), as well as pure-tone averages (PTA) for 9 and 10 kHz, but not PTAs for frequencies in the standard audiometric range.
Ideal signals and processing for continuous active sonar
Authors: DeFerrari Harry A.
Published: May 2013

The ideal signal for continuous active sonar would be linear in both time and Doppler and have no time or Doppler leakage. Here two signals/signal processing methods are presented that have half the requisite condition—zero time leakage. But we show a way to use the time property to eliminate Doppler interference (clutter) from zero Doppler reverberation (bottomed) sources and from direct arrivals. Here we develop an m-sequence processed with a matched filter approach and a special class of inverted binary sequences processed with a matched-inverse filter. In both cases, there is a perfect pulse correlation property; that is, a large signal when they line up in time and hard zero when they do not. This property is used to develop a Complete Ortho-Normal (CON) data sets in both waveform and pulse response space. The CON pulse response allows elimination of the easily identified zero Doppler reverberation arrivals. Then, reprocessing for Doppler, the zero Doppler signals are cleanly removed as is all of their Doppler leakage. In effect one has the ideal clutter free signal with leakage from direct arrivals and bottom reverberation removed and a target signal displayed in a time-Doppler plane against a noise only background.
Toward blind reverberation time estimation for non-speech signals
Authors: Santos João F. , Peters Nils , Falk Tiago H.
Published: May 2013

Reverberation time (RT) is an important parameter for room acoustics characterization, intelligibility and quality assessment of reverberant speech, and for dereverberation. Commonly, RT is estimated from the room impulse response (RIR). In practice, however, RIRs are often unavailable or continuously changing. As such, blind estimation of RT based only on the recorded reverberant signals is of great interest. To date, blind RT estimation has focused on reverberant speech signals. Here, we propose to blindly estimate RT from non-speech signals, such as solo instrument recordings and music ensembles. To estimate the RT of non-speech signals, we propose a blind estimator based on an auditory-inspired modulation spectrum signal representation, which measures the modulation frequency of temporal envelopes computed from a 23-channel gammatone filterbank. We show that the higher modulation frequency bands are more sensitive to reverberation than the modulation bands below 20 Hz. When tested on a database of non-speech sounds under 23 different reverberation conditions with early decay time (EDT) ranging from 0.26 to 7.6 s, a blind estimator based on the ratio of high-to-low modulation frequencies outperformed two state-of-the-art methods and achieved correlations with EDT as high as 0.80 for solo instruments and 0.75 for ensembles.
Efficient speech encryption using chaotic cat map for code-excited linear prediction based coders in packet networks
Authors: Merazka Fatiha
Published: May 2013

The increasing importance of multimedia applications is placing a great insist on content protection and customer privacy. Communications can be intercepted, especially over wireless links. Since encryption can effectively prevent eavesdropping, its use is widely advocated. The codec G. 729 based CS-ACELP algorithm is standardized as voice codec by ITU-T for multimedia and Voice over Internet Protocol (VoIP) applications. In this paper, we introduce a speech encryption method based chaotic cat map algorithm. Cat map extended to two-dimensional NxN matrix. It takes concepts from linear algebra and uses them to change the positions of the values of the matrix. The result after applying the Cat Map will be shuffled signals that contain the same values of the original signals. We applied our encryption scheme to the standard ITU-T G.729 standard speech coder to evaluate its performance. Simulation results show that G.729 based cat map encryption is very efficient since the encrypted speech is similar to a white noise. The perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD) tests for speech extracted from TIMIT database confirm the efficiency of our proposed scheme.
Dereverberation of a closed test section of a wind tunnel with a mult imicrophones cesptral method
Authors: Blacodon Daniel , Bulté Jean
Published: May 2013

Today, the manufacturers of aircraft want to perform at the same time aeroacoustic and classical aerodynamic tests in the wind tunnels with closed test sections in order to reduce test duration and costs involved in new aircraft developments. However, this kind of wind tunnels is not optimal for acoustics testing because the measurements are often masked by background noise (i.e., noise generated by the wind tunnel flow mechanism) and reverberations on the walls without acoustic liner within the test section. This is a serious problem because these unwanted phenomena limit the accuracy of identification, and quantification of acoustic sources. The reduction of background noise can be obtained with subspace techniques or noise subtraction based on a noise reference measurement. Concerning the reduction of the contamination of the echos, it can be achieved with an inverse filtering technique or with a single microphone cepstral method. A new approach is proposed in this study to remove the spurious effects of the echos. It is based on multi microphones cesptral technique. This new technique has been successfully applied on numerical simulations, and experimental data. The theoretical developments of this method and the obtained results will be presented in the full paper.
On the origins of ambient biological sounds in shallow water tropical ecosystems
Authors: Freeman Simon E. , Rohwer Forest L. , Gregg Allison , Coleman Laura , Buckingham Michael J.
Published: May 2013

Although discovered more than 60 years ago, the origins of much ambient underwater biological noise remain unclear. Snapping shrimp sounds dominate some environments but elsewhere the shallow-water biological sound field is often heterogeneous. Here we show that dominant components of underwater ambient noise recorded on coral reefs around five islands in the central Pacific may be caused by the interaction of hard-shelled benthic macro-organisms with the substrate. Recordings show a consistent, nightly 4.7 to 6.9 dB increase in estimated pressure spectral density level in the 11 to 17 kHz band with a spectral peak centered between 14 to 15 kHz. Underwater time-lapse photography reveals a marked night-time increase in benthic invertebrate activity at most locations, temporally consistent with the increase in pressure spectral density level. Intensity-filtered recordings of an example species, the hermit crab Clibanarius diugeti, in quiet aquarium conditions reveal that transient sounds produced by interaction between the crustacean’s carapace, shell, and coral substrate are spectrally consistent with the central Pacific recordings. Passive acoustic monitoring of such ambient noise may be useful as a complementary ecological survey technique to SCUBA-based visual observations, which are typically poor in estimating the abundance and diversity of cryptobenthic organisms.
Prediction of underwater noise and far field propagation due to pile driving for offshore wind farms
Authors: Lippert Stephan , Lippert Tristan , Heitmann Kristof , von Estorff Otto
Published: May 2013

Wind energy plays a key role toward a greener and more sustainable energy generation. Due to limited onshore areas and possible negative effects on human living space, offshore wind parks become increasingly popular. During construction by pile driving, however, high levels of underwater sound emission are observed. To avoid negative effects on marine mammals and other sea life, different approaches, like, e.g., bubble curtains or cofferdams, are currently investigated to cut down the sound pressure levels. In order to predict the expected underwater noise, both with and without sound damping measures, numerical simulation models are needed to avoid complex and costly offshore tests. Within this contribution, possible modeling strategies for the prediction of underwater noise due to pile driving are discussed. Different approaches are shown for the direct adjacencies of the pile and for the far field sound propagation. The effectivity of potential noise mitigation measures is investigated using a detailed finite element model of the surroundings of the pile. The far field propagation in the kilohertz range at distances of several kilometer from the pile, on the other hand, is computed by a model based on wavenumber integration. Finally, the model validation with corresponding offshore tests is addressed.
A quasi-analytic model of the underwater sound signal from impact driving of an offshore semi-infinite pile
Authors: Hall Marshall V.
Published: May 2013

A quasi-analytic model is derived for the underwater sound signal radiated when an offshore pile is struck on its face by a hammer. The pile is modeled as a semi-infinite cylindrical shell of an elastic solid. The impact generates a pulse of vibration that travels down the pile at the longitudinal sound-speed. At a given distance below the pile face, the axial displacement after the peak has arrived decreases exponentially with time. There are two coupled equations of motion for the axial and radial displacements. A closed form expression is derived for the radiated sound pressure (which is proportional to the radial acceleration) in terms of the Poisson ratio and Young’s Modulus of the pile material, the hammer velocity, contact area between hammer and pile, pile radius, hammer mass, the pile’s longitudinal sound-speed, and the sound-speed and density of the external medium. This model is applied to a published scenario for which the radiated sound pressure had been computed using a Finite Element Model, but is found to produce a different result. Some assumptions used in the model are identified that may explain the difference.
Under-ice shallow-water sound propagation and communication in the Baltic Sea
Authors: Sangfelt Erland , Ivansson Sven M. , Karasalo Ilkka
Published: May 2013

Sound propagation under ice in the Baltic is of interest for military as well as civilian purposes. Important questions are, for example, how sonar systems are affected by ice in shallow waters and how marine mammal life under ice is affected by ship traffic. Changing climate and increasing ship traffic are factors of great concern also in the Arctic region. Modeling results for sound propagation under ice in the Baltic Sea are presented for low as well as high frequencies. The sound propagation is influenced by bottom as well as ice-cap interaction in these shallow waters. The low- and high-frequency modeling is performed with wavenumber and ray models, respectively. Both types of models are amended to include a solid ice layer on top of the water column. In addition, the modeling efforts are extended to study the performance of an underwater communication system developed at FOI. The modeling results are evaluated using data from sea trials in the Gulf of Bothnia under ice-covered and ice-free conditions.
Arctic ambient noise measurements in support of the Northern Watch Project
Authors: Heard Garry J. , Pelavas Nicos , Pecknold Sean , Lucas Carmen E. , Martin Bruce
Published: May 2013

During August 2012, acoustic recording systems were deployed in Barrow Strait as part of the Defence Research and Development Canada (DRDC) Northern Watch Technology Demonstration Project. Two Starfish Sensor Cubes each with a 1-m cube of seven hydrophones operating in the frequency range of 5–750 Hz, and two single-hydrophone, Autonomous Multichannel Acoustic Recorders (AMAR) providing a 30-kHz signal bandwidth were deployed. The Starfish were deployed for two one-week intervals. One AMAR was deployed for two weeks partially overlapping the Starfish deployment. The second AMAR was deployed for a period of one year with recovery planned for August 2013. The observed underwater noise picture is one of high variability ranging from an extremely quiet to a noisy environment. Noise sources included: A 500-m long iceberg grounded within 500 m of one of the Starfish; a large ice island (4–5 sq-km) that passed within 4 km of the sensors; a small number of motoring vessels; significant wind events that caused rapid and strong variations in the noise field; and a small number of marine mammal detections. After our departure, a large number of Beluga whales were observed visually. The remaining AMAR may detect these late summer visitors.
Source bearing and range estimation using an ice-mounted tri-axial geophone
Authors: Dosso Stan E.
Published: May 2013

This paper presents results from Arctic field trials to estimate the bearing and range of an acoustic source in the water column using seismic particle motion measured at a tri-axial geophone on the sea ice surface. Measurements were carried out on smooth, rough, and ridged annual ice, and on a multi-year ice floe. At each site, a hammer seismic study was carried out to selectively excite various ice seismic waves and investigate their propagation properties. Impulsive acoustic sources were deployed in the water at a variety of bearings and ranges from 0.2 to 50 km. Source bearings are estimated by applying polarization filters to suppress shear waves with transverse particle motion and computing the incident power as a function of radial look angle; the inherent 180-degree ambiguity is resolved by requiring prograde particle motion in the vertical-radial plane. Results indicate good bearing estimation (<10-degree average errors) at all ranges with little dependence on ice type. Source range is estimated from the time difference between the water-borne arrival and the critically-refracted longitudinal plate (Lp) wave. Results are limited due to strong attenuation of the Lp wave, with good range estimation to <1 km for smooth annual ice and <0.5 km for other sites.
Objective evaluation of musical instrument quality: A grand challenge in musical acoustics
Authors: Campbell D. Murray
Published: May 2013

Over the last few decades, increasingly sophisticated experimental and computational studies have clarified the processes involved in sound production in musical instruments. One of the principal goals of this research effort has, however, remained tantalizingly elusive: the establishment of clear and unambiguous relationships between objectively measured properties of an instrument and judgments of its musical qualities by an experienced player. This is partly because player evaluation is a subtle and highly subjective process in which many different aspects of the instrument's performance may be tested. Early studies concentrated on the steady state spectra of sound recorded in the far field of the instrument. More recently, it has been recognized that transient aspects of an instrument's performance are important in judgments of quality made by performers. These aspects include the ease with which a stable regime of oscillation can be initiated, and the flexibility with which pitch, amplitude, and timbre can be modified during performance. Attempts to define “playability” of an instrument in scientific terms, and to relate these scientific metrics to the vocabulary used by performers in judgments of playability, have been partially successful, but many questions remain unanswered.
Modeling and simulation of Gamelan Bali concert hall based on objective acoustic parameters
Authors: Nitidara Ni Putu Amanda , Merthayasa I G. Nyoman , Sarwono Joko
Published: May 2013

Gamelan Bali music performances require special place to highlight the quality of the acoustic performances. A concert hall dedicated for Gamelan Bali was proposed to perform a better acoustic quality. Studies on Gamelan Bali has been done to achieved an optimum value of sound fields in Gamelan Bali concert hall. This paper shows a geometrical model designed to fulfill the suitable sound fields of Gamelan Bali Concert Hall. The model was modified from shoebox shaped with rear and side balconies. The acoustic performances of the model at 1 kHz summarized as follows: Tsub = 1.41 s, LL = 80.2 dBA, Δt1 = 39.43 ms, and IACC = 0.33. Those values meet the optimum values from the result of previous studies. Auralization of sound in the room was also done for the purpose of subjective judgment.
Assessing the acoustics of offices: A case of standardization
Authors: Asselineau Marc
Published: May 2013

Standards are supposed to help promote better understanding between the various interested parties. As such they can help an end user understand what the acoustic performances of his future premises will be… as long as such a standard exists. Over the years, various standards have been developed by several countries to characterize the acoustics of non partitioned offices. The tactics developed by ISO follow the Scandinavian standards trend with emphasis on the spatial sound level decay in the furnished space while the north American standards are more concerned with speech discretion. Are those tactics really opposed, or are they complementary? This paper intends to submit a comparison of a few standards (ISO, NF, CAN) applied on a couple of representative cases. It turns out that such standards represent the habits and practice of those who wrote it.
Acoustical design of Turkish Religious Affairs Mosque
Authors: Sü Gül Zühre , Çalışkan Mehmet
Published: May 2013

The new Turkish Religious Affairs (DIB) Mosque with its prayer capacity and outstanding volume is the largest classical-contemporary mosque project of the past decade built in Ankara, Turkey. The Mosque is also one of unique examples of its scale for which the room acoustic design is applied in its design phase. Acoustical design of DIB Mosque is critical considering speech and music related activity patterns held in such religious spaces. Interior surface forms and materials of walls and pendentives, floor finishing alternatives, geometry, and finishing of the dome are studied simultaneously with the architectural design as early as in the concept phase. Impedance tube is used for testing alternative materials for specifying sound absorption characteristics of reliefs and perforations. Computer simulation is applied as an acoustical design tool and estimations are held by ODEON v.11.23. Objective acoustical parameters including reverberation time, speech transmission index, and A-weighted sound levels are assessed with and without sound reinforcement systems for fully and partially occupied mosque conditions. Auralizations are held for imam and muezzin in different forms of religious call out to prayers. Evaluation of the space indicates that the optimized acoustical field is proper for intended functions of use in a mosque and satisfies desired tranquil environment.
Recognizing objects from multiple orientations using dolphin echoes
Authors: DeLong Caroline M. , Heberle Amanda , Mata Kayla , Harley Heidi E. , Au Whitlow W.
Published: May 2013

Object constancy, the ability to recognize objects despite changes in orientation, has not been well studied in the auditory modality. Dolphins use echolocation for object recognition, and objects ensonified by dolphins produce echoes that can vary significantly as a function of orientation. In four experiments, human listeners had to classify echoes from objects ensonified with dolphin signals. Participants were trained to discriminate among the objects using an 18-echo stimulus from a 10 degree range of aspect angles, then tested with novel aspect angles across a 60 degree range. In the first two experiments, the three objects varied in material, size, and shape. Participants were typically successful recognizing the objects at all angles (M = 78%). In experiment 3, the three objects had the same material but different shapes. Participants were often unsuccessful recognizing the objects at all angles (M = 46%). In experiment 4, participants had to classify echoes from four fish species across a wider range of angles (330 degrees). Preliminary results show overall poor performance (M = 45%). These results suggest that object characteristics play a role in whether performance is more view-dependent or view-invariant. These studies can provide insight into the process dolphins use to identify objects.
Dolphin strategies for long-range object detection and change detection
Authors: Finneran James J.
Published: May 2013

Dolphin strategies for detecting objects and changes in objects were investigated by having three trained bottlenose dolphins perform long-range echolocation tasks. The tasks featured the use of “phantom” echoes produced by capturing the dolphin’s outgoing echolocation clicks, convolving the clicks with the impulse response of a physical target to create an echo waveform, then broadcasting the delayed, scaled echo waveform back to the dolphin. Dolphins were trained to report the presence of phantom echoes or a change in phantom echoes. Target simulated ranges varied from 25 to 800 m. At ranges below 75 m, all dolphins followed a single click-echo paradigm, where inter-click intervals exceeded the two-transit time (i.e., the dolphins waited to receive the echo from a click before emitting the next click). As the range increased beyond 75 m, two of the three dolphins increasingly produced bursts, or “packets,” of several clicks, then waited for the packet of echoes to return before emitting another packet of clicks. The third dolphin instead utilized very high click repetition rates. The use of click packets may be a response to a limitation in the dolphin’s ability to employ multi-echo processing with large inter-echo delays.
Ultrasound-mediated gene delivery – Cardiovascular applications for Chronic ischemia, heart failure, and ischemia-reperfusion injury
Authors: Leong-Poi Howard
Published: May 2013

Ultrasound-mediated gene delivery (UMGD) is a non-invasive gene transfer technique, utilizing high power ultrasound and DNA-bearing microbubbles. Despite modest transfection efficiency, its high organ, tissue specificity, and repeatability make it an attractive therapeutic option. UMGD has been used in a variety of in vivo applications, including cardiac and skeletal muscle, kidney, liver, cerebral, and even lung, and have been studied using many gene vectors, including plasmid, viral, and small interfering RNA. This presentation will focus specifically on cardiac applications using plasmid DNA, including (1) introduction of UMGD in the heart, including optimization of parameters and protocols, (2) UMGD for therapeutic angiogenesis in chronic ischemia, including multi pro-angiogenic gene therapy and combination gene- and progenitor cell-based therapies for chronic hindlimb ischemia, and (3) applications for anti-apoptotic therapy in heart failure and ischemia-reperfusion injury.
Transducer modeling for optimal in-situ performance in a hearing instrument context
Authors: Kuster Martin
Published: May 2013

In a hearing instrument context, the transducers design has to be optimized for optimal performance in the installed configuration. This is typically behind or inside the outer ear of the instrument's wearer. Several examples will be shown where the in-situ acoustics is taken into account in the design or selection process of transducers. In this process, acoustic modeling is used extensively. Transducers are modeled by network analogs and they are coupled under defined impedance conditions to 3D numerical acoustics models, which incorporate the in-situ influence of, e.g., the wearer's ear anatomy. Moreover, in some cases, the modeling also includes part of the signal processing present in the hearing instrument.
Design of resonant frequencies of the piezoelectric actuator with integrated components
Authors: Kuroda Jun , Onishi Yasuharu , Komoda Motoyoshi , Oikawa Yasuhiro , Yamasaki Yoshio
Published: May 2013

The piezoelectric actuators have been applied to various electrical devices such as piezoelectric speakers, buzzers, haptics, and ultrasonic transducers. The improvement of the electromechanical conversion efficiency is the most important issue of piezoelectric actuator systems in mobile devices. The electrical power consumption of actuators must be suppressed as possible, due to mobile devices having small batteries. The frequency response around the mechanical resonance should be carefully designed for low power consumption driving. The resonant frequencies of the piezoelectric actuators consist of integrated components, for example, metal corns of ultrasonic speakers, are decided by the energy dispersion of the total system. Therefore, design factors such as sizes and physical properties of each component, are necessary to optimize the resonant frequencies for practical applications. The total energy of the piezoelectric system is described by Lagrange-Maxwell equation. Even though it is not easy to solve the differential equations written in the Lagrangian coordinate system by the exact calculation, useful information for design of the system can be derived by the approximate calculation. In this paper, we will indicate the design guideline to optimize resonant frequencies of the piezoelectric actuators with integrated components, based on the analysis in the Lagrangian coordinate system.
Technical ear training: Tools and practical methods
Authors: Corey Jason
Published: May 2013

Broadly defined, technical ear training seeks to make associations between aural impressions of sound quality and quantifiable characteristics of audio signal processing and acoustical measurements. Technical ear training typically focuses on attributes of sound such as spectral balance (e.g., filtering and parametric equalization); dynamic range of musical signals (including artifacts produced by dynamics processing); reverberation, delay, and early reflections (from real acoustic spaces or generated artificially); and spatial extent (width and depth). These elements of recorded sound can be broken down into graduated levels of audibility for the development of critical listening skills. With repeated and regular practice of carefully chosen exercises, listeners can gain increased sensitivity to subtle details of sound, as well as efficiency and accuracy in identifying specific parameters of signal processing by ear. With applications primarily in sound recording and production, technical ear training is also highly relevant to the evaluation of acoustical spaces as a complement to objective measurements. This presentation will review a selection of software modules developed by the author to teach critical listening skills to undergraduate students. The author will also discuss some practical methods and exercises used for teaching technical ear training and critical listening.
Effectiveness of technical listening training in Department of Acoustic Design of Kyushu University
Authors: Kawahara Kazuhiko , Takada Masayuki , Iwamiya Shin-ichiro
Published: May 2013

What is the professional listening? The listening ability of Sound/Acoustic Professionals listening is categorized into three phases: the ability to discriminate between different sounds, the ability to correlate the auditory difference with the physical properties of sounds, and the ability to imagine the proper sounds when given the acoustic properties of the sounds. These kinds of ability can be trained through listening training. Furthermore, through trainings, trainees can share same auditory experiences. The shared experiences improve ability of trainees to express their auditory impression with appropriate words and this ability contributes to smooth communication on auditory imagery in their group. In this paper, as a listening training, the overview and effectiveness of Technical Listening Training in Kyushu University is described. To evaluate the effectiveness of training, the average correct answer ratios of trainees were examined through the training. The improvement of correct answer ratios was observed. We could show the effectiveness of Technical Listening Training.
Synergistic interactions underlying the production of voice
Authors: Kaburagi Tokihiko
Published: May 2013

This paper presents recent progress in research examining the mechanisms of voice generation and a method for physiologically based speech synthesis. The overarching goal of this research is the precise modeling of interactions among physical systems involved in the processes underlying voice generation. In the basic voice generation system, a flow-structure interaction between glottal flow and the vocal folds causes self-oscillations of the folds, where flow separation, a nonlinear aerodynamic phenomenon, plays an important role. The fluid dynamic theory implies that a thin boundary layer formed near the glottal wall characterizes the flow behavior, including flow separation, jet formation, and pressure loss across the channel. We therefore use the interactive boundary layer method to analyze glottal flow and show how the flow-structure interaction is effective in maintaining vocal fold oscillations. In addition, the interaction between the voice generation system and the vocal-tract filter, i.e., the source-filter coupling, has been found to involve nonlinear factors in speech, such as skewing of glottal flow, unsteadiness in vocal fold oscillations, and transitions in voice register.
Theoretical and experimental study of glottal geometry in phonation
Authors: Pelorson Xavier , Van Hirtum Annemie , Wu Bo , Silva Fabrice
Published: May 2013

Most existing theoretical models of phonation assume that the vocal folds are parallel and that the glottis forms a two-dimensional channel for the flow. However, during phonation the vocal folds can be very accurately abducted or adducted using intrinsic muscles acting on the arytenoid cartilages. The resulting shape of the glottis can then vary between an almost uniform slit (when the vocal folds are parallel) to a V shape (with an angle between the vocal folds up to 20 °). Further, the vocal folds surface can present some irregularity which can be severe, in some pathological cases such as cysts or nodules. In this paper, we present a theoretical and experimental study performed in order to evaluate and to predict these geometrical effects. Several theoretical models to predict the pressure losses in a non-uniform glottis will be presented and tested against an in-vitro experimental set-up using a self-oscillating latex replica of the vocal folds. The angle between the artificial folds could be controlled using micrometers while surface irregularities could be simulated by inserting small spheres with various masses and diameters.
Influence of epithelium and fiber locations on glottal closure and sound production at soft-phonation conditions
Authors: Zhang Zhaoyan , Xuan Yue
Published: May 2013

Previous studies showed that isotropic vocal fold models often vibrated with incomplete glottal closure at onset despite the vocal folds were in contact at rest. This contrasts with human phonation in which complete glottal closure is observed even during soft phonation with minimal or low laryngeal muscle contraction. Based on previous experimental studies, we hypothesize that this difference in glottal closure patterns is due to the relatively large stiffness in the anterior-posterior direction or the presence of the epithelium layer. These hypotheses were tested in self-oscillating physical vocal-fold models, with anisotropic stiffness conditions simulated by fibers loosely imbedded at different locations in otherwise isotropic vocal folds. The results showed that, compared to isotropic one-layer models, the presence of a stiff epithelium layer led to complete glottal closure along the anterior-posterior direction, increased maximum glottal opening, strong excitation of high-order harmonics in the resulting voice spectra and reduced noise production. Similar improvement in glottal closure and high-order harmonics excitation was observed with fibers in the cover layer, but to a less degree. Presence of fibers in the body-layer led to reduced maximum glottal opening but did not yield noticeable improvement in glottal closure and harmonic excitation. [Work supported by NIH.]
Development of a real time compliance system for wind farms regulated by ambient-relative noise standards
Authors: Hankard Michael
Published: May 2013

Some noise level regulations in the United States require wind turbine farms to not exceed the ambient sound levels at nearby residences by more than a fixed amount. For the project discussed herein, compliance with such a regulation requires curtailment of turbine operations to some degree at times when turbine operations are at or near maximum, atmospheric conditions are conducive to sound propagation, and sound from other sources including vegetation rustle are at a minimum. Based on the analysis of months of time-synchronized sound, meteorological, and operations data, a system was developed to assess compliance on a real-time, ongoing basis. The primary element in the determination of compliance is the shape of the one-third octave band spectrum at the residence, augmented by ground and hub-height meteorological conditions, and wind farm operations information. Without the spectral filter, curtailment would need to take place under a broader array of meteorological conditions to ensure compliance, which would result in loss of power generation revenue. This paper will describe the data collection and analysis methods, the development of the spectral filter, and the results of field testing including both the partial and entire shut down of the wind farm.
Prevalence of complaints related to wind turbine noise in northern New England
Authors: Kaliski Kenneth
Published: May 2013

As of September 2012, there were a dozen large operating wind projects with a total capacity of approximately 565 MW in northern New England, with more coming online by the end of the year. This paper evaluates the prevalence of noise complaints to regulatory authorities from those wind projects. Where possible, the exposure of residences to wind turbine sound is calculated. Exposure is estimated through standard ISO 9613-2 modeling procedures. A comparison of the exposure of complainants and non-complainants is made with the goal of assessing the prevalence of complaints at various modeled sound levels.
Generation of wind turbine noise signature for use in lab environment
Authors: Zosuls Aleks , Kelley R. Morgan , Mountain David , Grace Sheryl
Published: May 2013

The fact that wind turbines produce infrasound continues to draw attention and discussion. Some argue that while the infrasound level produced by wind turbines is quite low, it still may be affecting the vestibular system or the hearing system, particularly via activation of the outer hair cells. Others hypothesize that the infrasound may be inducing whole body, chest cavity, or other human organ resonance. In order to study these hypotheses, it is first necessary to be able to recreate the turbine noise signature in a lab environment. Thus, the goal of this work is to create an acoustic system that can produce low-level infrasound. The system requirements include low cost, high fidelity, and imperceptible structural coupling to the lab. In addition, the system must be able to produce a broadband spectrum as well as a single tone. Progress toward the design of this audio system is discussed in this paper.
Wind turbine sound prediction–The consequence of getting it wrong
Authors: Palmer William K.
Published: May 2013

The application to permit a wind turbine power development usually involves submission of a prediction for the sound level that will occur at residences, schools, places of worship, and elsewhere people gather for restorative rest. This paper uses the example of a wind power development, and follows iterations taken to finalize the sound level prediction. The paper provides quantitative information collected since the start up of the wind power development on measured sound levels and octave band distribution; and qualitative observations on the special characteristics of the sound. Actual observations are compared to the predictions. More importantly, the paper reviews the consequences self-reported in qualitative interviews by citizens living with the changed environment after four years of operation of the wind power development. Reported impacts included difficulty sleeping, loss of jobs, and changes to social relationships, caregiving, pursuit of hobbies, leisure, learning, and overall health. Changes in measured health outcomes are identified. Both the quantitative and qualitative findings justify revision of the permitting process.
Comparison of supersonic full-scale and laboratory-scale jet data and the similarity spectra for turbulent mixing noise
Authors: Neilsen Tracianne B. , Gee Kent L. , Wall Alan T. , James Michael M. , Atchley Anthony A.
Published: May 2013

The broadband, partially correlated noise radiated from supersonic jets has characteristics that scale with nozzle size and flow properties. In particular, the spectral content of jet noise and the variation with angle in many cases agree with empirically derived similarity spectra for large and fine-scale components of turbulent mixing noise [Tam et al., AIAA Paper 96-1716]. In previous studies, measurements made near the F-22 Raptor agreed remarkably well with the similarity spectra, with two exceptions. First, the high-frequency slopes seen in the data were shallower than the similarity spectra at many angles. Second, the data exhibit a double frequency peak, which is absent from the similarity spectra [Neilsen et al., J. Acoust. Soc. Am. 132, 1993 (2012)]. These observations are explored further by examining the spectral characteristics of noise from a different military jet and a laboratory-scale, unheated jet. In both cases, there is evidence that for supersonic cases the measured spectra are shallower than the similarity spectra due to nonlinear propagation effects. In addition, the military data support the observation that the double spectral peak is a feature of full-scale jet noise. Recommendations are made for applying the similarity spectra to predict spectral levels for full-scale jets. [Work supported by ONR.]
Complexity penalized hydraulic fracture localization and moment tensor estimation under limited model information
Authors: Ely Gregory , Aeron Shuchin
Published: May 2013

In this paper, we present a novel technique for micro-seismic localization using a group sparse penalization that is robust to the focal mechanism of the source and requires only a velocity model of the stratigraphy rather than a full Green's function model of the earth's response. In this technique, we construct a set of perfect delta detector responses, one for each detector in the array, to a seismic event at a given location and impose a group sparsity across the array. This scheme is independent of the moment tensor and exploits the time compactness of the incident seismic signal. Furthermore, we present a method for improving the inversion of the moment tensor and Green's function when the geometry of seismic array is limited. In particular, we demonstrate that both Tikhonov regularization and truncated SVD can improve the recovery of the moment tensor and be robust to noise. We evaluate our algorithm on synthetic data and present error bounds for both estimation of the moment tensor as well as localization. Furthermore we discuss the estimated moment tensor accuracy as a function of both array geometry and fault orientation.
Comparison of dispersive relation of the oil well with that of the fluid cylinder and the hollow in solid
Authors: Zhang Hailan , Cui Hanyin , Lin Weijun , Wang Xiuming
Published: May 2013

Acoustical well logging is a widely used technique in oil fields to investigate the formation outside wells and the quality of wells. The knowledge of the acoustical behavior of the wells is important to the technique and has been widely studied. The wells are usually modeled as a fluid filled cylindrical borehole in an infinite solid medium. This structure with an infinite section, sometimes called as open waveguide, is more difficult to study than the typical acoustical waveguide with finite section and free or rigid boundaries. In this presentation, the well is studied as a coupled vibration system consisted of the fluid cylinder inside the well and the solid formation outside the well. The dispersive relations of three systems, i.e., the circular fluid cylinder with rigid boundary, the circular hollow in solid medium, and the fluid filled borehole in solid, are numerically calculated and compared. The properties of the dispersive relations, such as the cutoff frequencies, normal and abnormal dispersion, are compared and discussed. [Work supported by National Natural Sciences of China, Grant No. 11134011.]
Space-time methods for robust slowness estimation for monopole logging while drilling
Authors: Aeron Shuchin , Bose Sandip , Valero Henri-Pierre
Published: May 2013

In this paper, we present methods for interference cancelation for robust slowness estimation from noisy monopole logging while drilling (LWD) data. The main contributions are two fold. First, we show via tests on real data sets presence of systematic propagative interferences in monopole LWD data, which is the primary reason for loss of compression and shear semblance in the slowness time coherence (STC) processing of the LWD data. This interference in turn is mostly dominated by Stoneley type propagative component, which, unlike the main Stoneley mode, is time persistent over the entire acquisition interval. In addition, we also show that in fast formations the shear wave can significantly interfere with the compressional wave making the compressional slowness estimates quite bad. Second, based on these observations we propose a Successive Interference Cancellation (SIC) algorithm to estimate and cancel these interferences leading to STC enhancement and improved slowness estimation of the head waves. The algorithm exploits a novel representation of borehole acoustic signals using a dictionary of space-time propagators and is more robust compared to traditional slowness filtering methods, especially for the low aperture borehole acoustic array. We show the superior performance of the proposed algorithm on synthetic and real data sets.