bluuebunny/crossref_metadata_2025 huggingface dataset:
Current developments at the National Institute for Standards and Technology in pressure calibration of laboratory standard microphones
Authors: Nedzelnitsky Victor , Wagner Randall P. , Fick Steven E.
Published: October 2013

Current research effort aims at improving the apparatus and methods for determining the pressure sensitivities of IEC types LS1Pn and LS2aP laboratory standard microphones. Among the improvements that are being systematically incorporated in an evolving test bed is the capability to operate at adjustable power line frequencies other than the usual 60 Hz. Suitable choices of line frequency relative to frequencies of calibration and adjustable bandpass filter characteristics can be used to improve the signal-to-noise ratios of measurements performed near the usual line frequency and its first few harmonics. This can enable the use of relatively large volume couplers for which uncertainties in microphone front cavity volume and equivalent volume, capillary tube effects, and heat conduction corrections have a lesser influence than they have for small-volume couplers. Another improvement aims to control and to stabilize the ambient static pressure during microphone calibrations, to reduce or eliminate the effects of barometric pressure fluctuations on these calibrations.
Free-field reciprocity calibration of laboratory standard (LS) microphones using a time selective technique
Authors: Rasmussen Knud , Barrera-Figueroa Salvador
Published: October 2013

Although the basic principle of reciprocity calibration of microphones in a free field is simple, the practical problems are complicated due to the low signal-to-noise ratio and the influence of cross talk and reflections from the surroundings. The influence of uncorrelated noise can be reduced by conventional narrow-band filtering and time averaging, while correlated signals like cross talk and reflections can be eliminated by using time-selective postprocessing techniques. The technique used at DPLA overcomes both these problems using a B&K Pulse analyzer in the SSR mode (steady state response) and an FFT-based time-selective technique. The complex electrical transfer impedance is measured in linear frequency steps from a few kHz to about three times the resonance frequency of the microphones. The missing values at low frequencies are estimated from a detailed knowledge of the pressure sensitivities. Next an inverse FFT is applied and a time window around the main signal is used to eliminate cross talk and reflections. Finally, the signal is transformed back to the frequency domain and the free field sensitivities calculated. The standard procedure at DPLA involves measurements at four distances and the repeatability of the calibrations over time is within ±0.03 dB up to about 1.5 times the resonance frequency of the microphones.
Microphone calibration by comparison
Authors: Wong George S. K.
Published: October 2013

The absolute method of microphone calibration by the reciprocity method (IEC 61094-2 1992) provides the highest accuracy of approximately 0.04 to 0.05 dB, and the procedure requires three changes of microphone in the ‘‘driver-receiver combination’’ that needs approximately 1 to 2 days. The system capital cost is relatively high. The NRC interchange microphone method for microphone calibration by comparison has been adopted internationally by the International Electrotechnical Commission as IEC 61094-5 (2001-10). With this method, the test microphone is compared with a reference microphone calibrated by the reciprocity method and the procedure requires approximately 3 h. The uncertainty of the comparison method is between 0.08 to 0.09 dB, which satisfies most industrial needs.
Development of a laser-pistonphone for an infrasonic measurement standard
Authors: Horiuchi Ryuzo , Fujimori Takeshi , Sato Sojun
Published: October 2013

Acoustical standards for audio frequencies are based on pressure sensitivities of laboratory standard microphones calibrated using a coupler reciprocity technique. There is a growing need to extend the frequency range downward for reliable infrasonic measurement. The reciprocity technique, however, has limitations on low-frequency calibration (1–20 Hz) because signal-to-noise ratio deteriorates and a sound leak occurs from capillary tubes that equalize the static pressure inside and outside of the coupler. These factors rapidly increase the measurement uncertainty as the frequency is lowered. NMIJ has therefore recently developed a laser-pistonphone prototype, which enables precise calibration of microphones at low frequencies. Compared with the reciprocity technique, the laser-pistonphone produces a higher sound pressure within a cavity by the sinusoidal motion of a piston and has a significantly improved signal-to-noise ratio. Sound pressure is calculated from the piston displacement, which is determined via a Michelson interferometer. A test microphone is inserted into the cavity, exposed to the sound pressure, and its open-circuit voltage is measured. Static pressure equalization is realized through the gap between the piston and its guide. Careful design of the dimensions and relative position of the cavity and piston minimizes sound leakage and friction between them.
Sensitivity change with practical use of electret condenser microphone
Authors: Yasuno Yoshinobu , Miura Kenzo
Published: October 2013

Dr. Sessler and Dr. West invented the electret condenser microphone (ECM) in 1966. It has since been applied in various ways as a sound input device. The ECM has become an important component as a microphone for communications because of its stable sensitivity frequency characteristic. Materials and production methods have been improved continually up to the present. In particular, the ECM reliability is based on the electret’s stability. For that reason, the electret surface charge decay is the main factor in ECM sensitivity degradation. This study analyzed the changes of an ECM preserved for 28 years in the laboratory and actually used for an outdoor interphone unit for 29 years. The change of diaphragm stiffness and electret surface voltage were compared with the evaluation result of a heat-acceleration test and verified. A degradation estimate of sensitivity change of ECM was performed. Regarding the life of the electret predicted in the report of former study [K. Miura and Y. Yasuno, J. Acoust. Soc. Jpn. (E) 18(1), 29–35 (1997)], the validity was verified using actual data from this long-term observation.
Development of a small size narrow directivity microphone
Authors: Iwaki Masakazu , Ono Kazuho , Sugimoto Takehiro , Ishii Takeshi , Inamaga Keishi
Published: October 2013

We developed a new microphone that has very sharp directivity even in the low-frequency band. In an ordinary environment of sound pick-up, the energy of background noise is distributed mainly in frequencies lower than 1000 Hz. In such frequencies, a typical cardioid microphone has the directivity pattern close to that of omni-cardiod. Consequently, it is difficult to pick up the objective sounds clearly from background noises. To suppress the noises with very small level, the directivity pattern should be also sharpened in the low-frequency band. In this report, we describe a new method to sharpen directivity for the low-frequency band. The method requires three microphone capsules. One capsule is the main microphone with a very short acoustic pipe. The others compose a second-order gradient microphone to cancel the signal that comes from behind the main microphone. A special feature of this microphone is to control a dip frequency of behind sensitivity without changing the frequency response of the front sensitivity.
Infrasound calibration of measurement microphones
Authors: Frederiksen Erling
Published: October 2013

Increasing interest for traceable infrasound measurements has caused the Consultative Committee for Acoustics, Ultrasound and Vibration (CCAUV) of BIPM to initiate a key comparison calibration project (CCAUV.A-K2) on pressure reciprocity calibration down to 2 Hz. Ten national metrology institutes, including the Danish Primary Laboratory of Acoustics (DPLA), take part in this project. In addition DPLA has started its own infrasound calibration project, which is described in this paper. The purposes of this project are verification of the CCAUV results and development of methods for calibration of general types of measurement microphone between 0.1 and 250 Hz. The project includes the design of an active comparison coupler, an experimental low-frequency reference microphone, and new methods for its frequency response calibration. One method applies an electrostatic actuator and requires a low-pressure measurement tank, while the other requires an additional microphone, whose design is closely related to that of the reference microphone that is to be calibrated. The overall calibration uncertainty (k=2) for ordinary measurement microphones is estimated to less than 0.05 dB down to 1 Hz and less than 0.1 dB down to 0.1 Hz, if the reference is calibrated in the latter mentioned way, i.e., by the related microphones method.
Various acoustical aspects of an Asian (South) Indian classical music concert
Authors: Prasad M. G. , Raman V. K. , Jagadishan Rama
Published: October 2013

An Asian (South) Indian classical music concert is an integrated acoustical experience for both the audience and the player(s). A typical concert team, either vocal or instrumental, consists of a main vocalist (or an instrumentalist) accompanied by a violinist, up to three percussion instrument players, and a reference drone. The concert is comprised of many songs. Each song has two main parts, namely Alapana and Kriti. The Alapana is an elaboration of a raga (tune) and the Kriti refers to the lyrics of the song. The violinist actively follows and supports the main musician during the concert. The percussion player(s) are provided an opportunity to present a solo of their rhythmic skills. The players and the audience communicate emotionally and intellectually with each other. Elements such as aesthetics, rhythm, skill, and emotional aspects of the players are evaluated and appreciated by the audience. This talk will present various aspects of a concert that brings about an integrated and holistic experience for both the audience and the player(s). Some samples from live vocal and instrumental music concerts will be presented.
Artificial neural network models of sound signals in urban open spaces
Authors: Yu Lei , Kang Jian
Published: October 2013

Sound signals, known as foreground sounds, are important components of soundscape in urban open spaces. Previous studies in this area have shown that sound preferences are different according to social and demographic factors of individual users [W. Yang and J. Kang, J. Urban Des., 10, 69–88 (2005)]. This study develops artificial neural network (ANN) models of sound signals for architects at design stage, simulating subjective evaluation of sound signals. A database for ANN modeling has been established based on large-scale social surveys in European and Chinese cities. The ANN models have consequently been built, where individual’s social and demographic factors, activities, and acoustic features of the space and sounds are used as input variables while the sound preference is defined as the output. Through the process of training and testing the ANN models, considerable convergences have been achieved, which means that the models can be applied as practical tools for architects to design sound signals in urban open spaces, taking the characteristics of potential users into account. Currently ANN models combining foreground and background sounds are being developed.
Describing soundscape and its effects on people where soundscape is understood as an expansion of the concept of noise engineering
Authors: Kawai Keiji
Published: October 2013

This study discusses how to describe sound environment and people in terms of ‘‘soundscape’’ as an expansion of the ‘‘noise engineering.’’ In the framework of the conventional study field of noise evaluation, typically, sound environments are represented by loudness-based indices such as A-weighted sound pressure levels, and the impact of sound environments on people is represented by annoyance response or some physiological metrics. In the case of soundscape studies, however, the description should be expanded beyond what has been used in noise engineering. This matter has already been frequently discussed, but it doesn’t seem that much consensus has been achieved concerning it yet. With respect to the effects of sound environment on people, since the concept of soundscape focuses on personal and social meanings of environmental sounds including the historical or aesthetic contexts, the effects are considered to be represented not by a singular concept such as comfortableness or quietness, but by multiple dimensions of emotional and aesthetic concepts. Also, descriptions of sound environment should include some qualitative aspects, such as what types of sounds can be heard at what extent. In this paper, the methodology to describe human-soundscape relationships is discussed through a review of related studies.
The interaction between ultrasound and human cancellous bone
Authors: Wear Keith
Published: October 2013

Attenuation is much greater in cancellous bone than in soft tissues, and varies approximately linearly with frequency between 300 kHz and 1.7 MHz. At diagnostic frequencies (300 to 700 kHz), sound speed is slightly faster in cancellous bone than in soft tissues. A linear-systems model can account for errors in through-transmission-based measurements of group velocity due to frequency-dependent attenuation and dispersion. The dependence of phase velocity on porosity may be predicted from theory of propagation in fluid-filled porous solids. The dependence of phase velocity on frequency (negative dispersion) can be explained using a stratified two-component model. At diagnostic frequencies, scattering varies as frequency to the nth power where 3>n>3.5. This may be explained by a model that represents trabeculae as finite-length cylindrical scatterers.
Dependence of phase velocity on porosity in cancellous bone: Application of the modified Biot-Attenborough model
Authors: Yoon Suk Wang , Lee Kang Il
Published: October 2013

This study aims to apply the modified Biot-Attenborough (MBA) model to predict the dependence of phase velocity on porosity in cancellous bone. The MBA model predicted that the phase velocity decreases nonlinearly with porosity. The optimum values for input parameters of the MBA model, such as compressional speed cm of solid bone and phase velocity parameter s2, were determined by comparing the prediction with the previously published measurements in human calcaneus and bovine cancellous bones. The value of the phase velocity parameter s2=1.23 was obtained by curve fitting to the experimental data only for 53 human calcaneus samples with a compressional speed cm=2500 m/s of solid bone. The root-mean-square error (rmse) of the curve fit was 15.3 m/s. The optimized value of s2 for all 75 cancellous bone samples (53 human and 22 bovine samples) was 1.42 with the rmse of 55 m/s. The latter fit was obtained by using cm=3200 m/s. Although the MBA model relies on empirical parameters determined from the experimental data, it is expected that the model can be usefully employed as a practical tool in the field of clinical ultrasonic bone assessment.
Simulation of fast and slow wave propagations through cancellous bone using three-dimensional elastic and Biot’s trabecular models
Authors: Hosokawa Atsushi
Published: October 2013

The propagation of ultrasonic pulse waves in cancellous (trabecular) bone was numerically simulated by using three-dimensional finite-difference time-domain (FDTD) methods. In previous research [A. Hosokawa, J. Acoust. Soc. Am. 118, 1782–1789 (2005)], two two-dimensional FDTD models, the commonly used elastic FDTD model and an FDTD model based on Biot’s theory for elastic wave propagation in an isotropic fluid-saturated porous medium, were used to simulate the fast and slow longitudinal waves propagating through cancellous bone in the direction parallel to the main trabecular orientation. In the present study, the extended three-dimensional viscoelastic and Biot’s anisotropic models were developed to investigate the effect of trabecular structure on the fast and slow wave propagations. Using the viscoelastic model of the trabecular frame comprised of numerous pore spaces in the solid bone, the effect of the trabecular irregularity, that is the scattering effect, on both the fast and slow waves could be investigated. The effect of the anisotropic viscous resistance of the fluid in the trabecular pore spaces on the slow wave could be considered using Biot’s anisotropic model.
3-D numerical simulation of wave propagation in porous media: Influence of the microstructure and of the material properties of trabecular bone
Authors: Haiat Guillaume , Padilla Frederic , Laugier Pascal
Published: October 2013

Finite-difference time domain simulations coupled to 3-D microstructural models of 30 trabecular bones reconstructed from synchrotron radiation microtomography were employed herein to compare and quantify the effects of bone volume fraction, microstructure, and material properties of trabecular bone on QUS parameters. Scenarios of trabecular thinning and thickening using an iterative dedicated algorithm allowed the estimation of the sensitivity of QUS parameters to bone volume fraction. The sensitivity to bone material properties was assessed by considering independent variations of density and stiffness. The effect of microstructure was qualitatively assessed by producing virtual bone specimens of identical bone volume fraction (13%). Both BUA and SOS show a strong correlation with BV/TV (r2=0.94 p10−4) and vary quasi-linearly with BV/TV at an approximate rate of 2 dB/cm/MHz and 11 m/s per % increase of BV/TV, respectively. Bone alterations caused by variation in BV/TV (BUA: 40 dB/cm.MHz; SOS: 200 m/s) is much more detrimental to QUS variables than that caused by alterations of material properties or diversity in microarchitecture (BUA: 7.8 dB/cm.MHz; SOS: 36 m/s). QUS variables are changed more dramatically by BV/TV than by changes in material properties or microstructural diversity. However, material properties and structure also appear to play a role.
Singular value decomposition-based wave extraction algorithm for ultrasonic characterization of cortical bone in vivo
Authors: Sasso Magali , Haiat Guillaume , Talmant Maryline , Laugier Pascal , Naili Salah
Published: October 2013

In the context of bone status assessment, the axial transmission technique allows ultrasonic evaluation of cortical bone using a multielement probe. The current processing uses the first arriving signal to evaluate the velocity while later contributions are potentially valuable and are not yet analyzed. However, all those contributions interfere, which disrupts the signal analysis. A novel ultrasonic wave extraction algorithm using a singular value decomposition method is proposed. This algorithm aims at characterizing a given energetic low-frequency (ELF) contribution observed in vivo at around 1 MHz. To evaluate the performances of the proposed algorithm, a simulated data set was constructed taking into account the influence of noise and of random interfering wavefront. The velocity of the ELF contribution is estimated on simulated datasets and compared to the input velocity. For a signal-to-noise ratio of 10 dB, the mean error associated with this method is 5.2%, to be compared with 34% with a classical signal analysis. The algorithm was also tested on real in vivo measurements. Results show the ability to accurately identify and possibly remove this wavefront contribution. Results are promising for multiple ultrasonic parameters evaluation from different wavefront contributions in our configuration.
Direct evaluation of cancellous bone porosity using ultrasound
Authors: Liu Peiying , Lewis Matthew , Antich Peter
Published: October 2013

Quantitative measurements of trabecular bone porosity would be a great advantage in diagnosis and prognosis of osteoporosis. This study focused on evaluation of the relationship between ultrasonic reflection and the density and porosity of cancellous bone. Theoretical simulation using matlab and Field II ultrasound simulator predicted a linear model of a material’s porosity and parameters of the reflected ultrasonic signals. Experimentally, four plastic phantoms fabricated with different porosities were tested by a 5-MHz ultrasound transducer and the results agreed with simulations. Twelve specimens of bovine cancellous bone were measured. The porosities of these specimens were estimated by calculating the ratio of the mass in air to the wetted mass when they were immersed in water and all the air was removed from the pores. Among all the parameters, the peak value of the reflected ultrasound signal demonstrated highly significant linear correlations with porosity (R2=0.911) and density (R2=0.866). This encouraging result shows that this technique has the potential to be used to monitor porosity changes noninvasively for clinical purpose such as noninvasive assessment of osteoporotic fracture risk. [Work supported by Pak Foundation.]
Do we have better hearing sensitivity than people in the past?
Authors: Kurakata Kenji , Mizunami Tazu
Published: October 2013

Our hearing sensitivity to tones of various frequencies declines progressively as we become older. ISO 7029 describes a method for calculating expected hearing threshold values as a function of age. However, more than 30 years have passed since the ISO standard source data were published. An earlier paper of the present authors [K. Kurakata and T. Mizunami, Acoust. Sci. Technol. 26, 381–383 (2005)] compared hearing threshold data of Japanese people in recent years to the ISO standard values. The results of that comparison showed that the age-related sensitivity decrease of Japanese people was smaller, on average, than those described in the standard. A large discrepancy was apparent at 4000 and 8000 Hz: more than 10 dB for older males. In response to that inference, the ISO/TC43/WG1 ‘‘threshold of hearing’’ initiated a project in 2005 to explore the possibility of updating ISO 7029. This paper presents a summary of those comparison results of audiometric data and the work in WG1 for revising the standard.
Use of perceptual weights to test a model of loudness summation
Authors: Leibold Lori J. , Jesteadt Walt
Published: October 2013

We recently examined the contribution of individual components of a multitone complex to judgments of overall loudness by computing the perceptual weight listeners assign to each component in a loudness-matching task [Leibold et al., J. Acoust. Soc. Am. 117, 2597 (2005)]. Stimuli were five-tone complexes centered on 1000 Hz, with six different logarithmic frequency spacings, ranging from 1.012 to 1.586. When all components fell within the same critical band, weights varied little across components. In contrast, the range of weights increased with increasing frequency separation, with greater weight given to the lowest and highest frequency components. Perceptual weights were largely in agreement with the Moore et al. loudness model [J. Audio Eng. Soc. 45, 224–237 (1997)], except at the widest bandwidth. In the current study we further examined predictions of the loudness model, focusing on the widest frequency-spacing condition. Masked thresholds and jnds for intensity discrimination were measured for each component and were compared to weights. The model predicts more interaction in the widely spaced conditions than simple critical band models, but underestimates the true interactions in conditions where components are widely spaced. Central factors appear to influence loudness, masked thresholds, and intensity discrimination in these conditions. [Work supported by NIH/NIDCD.]
Increased loudness effect at the absolute threshold of hearing
Authors: Yoshida Junji , Kasuga Masao , Hasegawa Hiroshi
Published: October 2013

This study investigated the effects of a previous sound on loudness at the absolute threshold of hearing. Change of the absolute threshold of hearing was measured when a pure tone preceded the test tone in the measurement of the threshold. The previous sound at 60 dB SPL was presented first in one ear, followed by the presentation of the test sound in either the contralateral or ipsilateral ear at an interval of 0.5 s. Both the previous and test sounds had the same frequency of 500 Hz, and the same duration of 3 s. The change of the threshold was obtained from the difference between the thresholds with and without the previous sound. The threshold was found to be decreased significantly by approximately 2 dB when the previous sound was presented in the contralateral ear. On the other hand, the threshold was only slightly changed when the previous sound was presented in the ipsilateral ear. These experiment results suggest that loudness was increased by perceiving the previous sound in the contralateral ear.
On the perception of epenthetic stops in American English
Authors: Arvaniti Amalia , Shosted Ryan , Kilpatrick Cynthia
Published: October 2013

This study examines the perception of epenthetic stops in American English. Stimuli were created from prince, prints, mince, mints, quince, and quints by removing all traces of /t/ and splicing in 0–72 ms of silence, in 12-ms steps, with or without a following burst. Subjects saw the minimal pairs on screen and selected the word they heard. It was hypothesized that stimuli with bursts and longer closure would result in more t responses (prince identified as prints) and that frequent words (prince/prints) would be more difficult to distinguish than infrequent words (quince/quints), as our production results suggest that frequent pairs are more likely to be produced similarly. Results from 19 subjects show shorter response times with longer silence intervals, but no effect of burst or stimulus identity. Similarly, stimuli with bursts were not identified as nts words more frequently than those without. Generally, stimuli originating from nts words were more likely to be identified as such if they contained a burst, while the opposite was true for stimuli from nce words. Finally, frequent words were less likely to be correctly identified than infrequent words, suggesting that /t/ epenthesis is not as widespread throughout the lexicon as generally believed.
New anthropomorphic talking robot—investigation of the three-dimensional articulation mechanism and improvement of the pitch range
Authors: Fukui Kotaro , Ishikawa Yuma , Sawa Takashi , Shintaku Eiji , Honda Masaaki , Takanishi Atsuo
Published: October 2013

We developed a new three-dimensional talking robot WT-6 (Waseda Talker-No. 6), which generates speech sounds by mechanically simulating articulatory motions as well as aeroacoustic phenomenon in the vocal tract. WT-6 consists of a 17-DOF mechanism (1-DOF lungs, 5-DOF vocal cords, 1-DOF jaws, 5-DOF tongue, and 4-DOF lips). It has 3-D lips, tongue, jaw, and velum, which form the 3-D vocal tract structure. It also has an independent jaw opening/closing mechanism, which controls the relative tongue position in the vocal tract as well as the oral opening. The previous robot, which had a 2-D tongue [J. Acoust. Soc. Am. 117, 2541 (2005)], was not enough to realize precise closure to produce humanlike consonants such as /s/ or /r/. The new tongue, which could be controlled to form the 3-D shape, makes it possible to produce more realistic vocal tract shape. The vocal cord model was also improved by adding a new pitch control mechanism pushing from the side of the vocal cords. The pitch range becomes broader than the previous robot, which is enough to reproduce normal human speech. Preliminary experimental results showed that synthesized speech quality improves for vowels /a/, /u/ and /o/. Some experimental results and video demonstration of the talking robot will be presented.
Acoustic analysis and perceptual evaluation of nasalized /g/ consonant in continuous Japanese
Authors: Kuwabara Hisao
Published: October 2013

It is well known that the /g/ consonant, a velar voiced plosive, in Japanese continuous speech is often nasalized unless it appears at the word-initial position. The nasalized /g/ consonant takes place in dialects mainly spoken in northern districts including the Tokyo area where the standard Japanese is spoken. However, the number of nasalized /g/ consonants is said to be decreasing year by year according to a survey. This paper deals with acoustic and perceptual analysis of this phenomenon. Test materials used in this experiment are read version of Japanese short sentences by NHK’s (Japan Broadcasting Corporation) professional announcers. Each sentence includes at least one /g/ consonant that would likely be pronounced as nasalized. An evaluation test reveals that less than 60% of nasalization has been found to occur for /g/ consonants for which 100% nasalization had been observed decades ago. Acoustic analysis for nasalized and non-nasalized /g/ sounds has been performed mainly through waveform parameters. It has been found that the power ratio between consonant and vowel is the most effective parameter for distinguishing nasals from non-nasals, but it is highly speaker dependent.
Production and perception of place of articulation errors
Authors: Stearns Adrienne M. , Frisch Stefan A.
Published: October 2013

Using ultrasound to examine speech production is gaining popularity because of its portability and noninvasiveness. This study examines ultrasound recordings of speech errors. In experiment 1, ultrasound images of participants’ tongues were recorded while they read tongue twisters. Onset stop closures were measured using the angle of the tongue blade and elevation of the tongue dorsum. Measurements of tongue twisters were compared to baseline production measures to examine the ways in which erroneous productions differ from normal productions. It was found that an error could create normal productions of the other category (categorical errors) or abnormal productions that fell outside the normal categories (gradient errors). Consonant productions extracted from experiment 1 were presented auditory-only to naive listeners in experiment 2 for identification of the onset consonant. Overwhelmingly, the participants heard normal productions and gradient error productions as the intended sound. Categorical error productions were judged to be different from the intended sound. The only effect of erroneous production on perception appears to be a slight increase in reaction time, which may suggest that error tokens are abnormal in some way not measured in this study.
Role of linguistic experience on audio-visual perception of English fricatives in quiet and noise backgrounds
Authors: Wang Yue , Jiang Haisheng , Danyluck Chad , Behne Dawn
Published: October 2013

Previous research shows that for native perceivers, visual information enhances speech perception, especially when auditory distinctiveness decreases. This study examines how linguistic experience affects audio-visual (AV) perception of non-native (L2) speech. Native Canadian English perceivers and Mandarin perceivers with two levels of English exposure (early and late arrival in Canada) were presented with English fricative-initial syllables in a quiet and a caf-noise background in four ways: audio-only (A), visual-only (V), congruent AV, and incongruent AV. Identification results show that for all groups, performance was better in the congruent AV than A or V condition, and better in quiet than in caf-noise background. However, whereas Mandarin early arrivals approximate the native English patterns, the late arrivals showed poorer identification, more reliance on visual information, and greater audio-visual integration with the incongruent AV materials. These findings indicate that although non-natives were more attentive to visual information, they failed to use the linguistically significant L2 visual cues, suggesting language-specific AV processing. Nonetheless, these cues were adopted by the early arrivals who had more L2 exposure. Moreover, similarities across groups indicate possible perceptual universals involved. Together they point to an integrated network in speech processing across modalities and linguistic backgrounds. [Work supported by SSHRC.]
The influence of noise and reverberation on vowel recognition: Response time
Authors: Blaszak Magdalena , Rutkowski Leon
Published: October 2013

This study examines the perceptual effect of two types of noise and reverberation on vowel recognition. Multitalker babble and traffic noise (European Standard EN 1793–3) were generated simultaneously with Polish vowels /a, e, i, o, u, y/ in two different sound fields and an anechoic chamber. The experiment was performed under various conditions of signal-to-noise ratio (−9, −6, −3, 0, +3, no noise). A new procedure for listeners’ selection based on the Bourdon’s psychometrics test was proposed. The effects of noise and reverberation were quantified in terms of (a) vowel recognition scores for young normal-hearing listeners (YNH) and (b) ease of listening based on the time of response and subjective estimation of difficulty. Results of the experiment have shown that (a) the response time can be a good measure of the effect of noise and reverberation on the speech intelligibility is rooms, and (b) in this type of experiment, of great significance is the choice of the subjects based on the psychometric tests.
Quantifying the benefits of sentence repetition on the intelligibility of speech in continuous and fluctuating noises
Authors: Mercille Isabelle , Larose Roxanne , Giguère Christian , Laroche Chantal
Published: October 2013

Good verbal communication is essential to ensure safety in the workplace and social participation during daily activities. In many situations, speech comprehension is difficult due to hearing problems, the presence of noise, or other factors. As a result, listeners must often ask the speaker to repeat what was said in order to understand the complete message. However, there has been little research describing the exact benefits of this commonly used strategy. This study reports original data quantifying the effect of sentence repetition on speech intelligibility as a function of signal-to-noise ratio and noise type. Speech intelligibility data were collected using 18 normal-hearing individuals. The speech material consisted of the sentences from the Hearing In Noise Test (HINT) presented in modulated and unmodulated noises. Results show that repeating a sentence decreases the speech reception threshold (SRT), as expected, but also increases the slope of the intelligibility function. Repetition was also found to be more beneficial in modulated noises (decrease in SRT by 3.2 to 5.4 dB) than in the unmodulated noise (decrease in SRT by 2.0 dB). The findings of this study could be useful in a wider context to develop predictive tools to assess speech comprehension under various conditions.
Selective attention and perceptual learning of speech
Authors: Francis Alexander L. , Kaganovich Natalya , Driscoll Courtney
Published: October 2013

Phonetic experience can change the perceptual distance between speech sounds, increasing both within-category similarity and between-category distinctiveness. Such warping of perceptual space is frequently characterized in terms of changes in selective attention: Listeners are assumed to attend more strongly to category-differentiating features while ignoring less relevant ones. However, the link between the distribution of selective attention and categorization-related differences in perceptual distance has not been empirically demonstrated. To explore this relationship, listeners were given 6 h of training to categorize sounds according to one of two acoustic features while ignoring the other. The features were voice onset time and onset f0, which are normally correlated and can both serve as a cue to consonant voicing. Before and after training, listener’s performance on a Garner selective attention task was compared with assessment of the perceptual distance between tokens. Results suggest that training can induce both warping of perceptual space and changes in the distribution of selective attention, but the two phenomena are not necessarily related. Results are consistent with a two-stage model of perceptual learning, involving both preattentive adjustment of acoustic cue weighting and higher-level changes in the distribution of selective attention between acoustic cues. [Work supported by NIH-NIDCD 1R03DC006811.]
Investigation of interaction between speech perception and production using auditory feedback
Authors: Akagi Masato , Dang Jianwu , Lu Xugang , Uchiyamada Taichi
Published: October 2013

This study employed an auditory feedback paradigm with perturbed fed-back speech to investigate interaction between speech perception and production by measuring simultaneous fluctuations of speech production organs using the electromyographic (EMG) signals, articulatory movements, as well as spectral analyses, where the articulatory data were obtained by the electromagnetic articulographic (EMA) system. Chinese vowels pair [i]-[y] and Japanese vowels pairs [e]-[a], [e]-[i] and [e]-[u] were chosen as the experimental objects. When the speaker is maintaining the first vowel, the feedback sound is randomly changed from the first vowel to the second one in each pair by manipulating the first three formants. Spectral analysis showed that a clear compensation was seen in the first and second formants of the vowels. Analyses of EMG and EMA signals also showed muscle reactivation and tongue movements to compensate for the perturbations. Latency of the compensating response is about 150 ms to start and about 290 ms for maximum compensation from the onset of the perturbation. According to the measurements, it seems that in most cases the speaker attempts to compensate for the ‘‘error’’ caused by the auditory perturbation by a real-time monitoring, and the auditory feedback takes place simultaneously often during speech production.
Influence of the prosody of spoken language on recognition and memory for vocal quality
Authors: Shigeno Sumi
Published: October 2013

This study examined whether recognition and memory for vocal quality of a speaker who speaks either a native language or non-native language should be influenced by the prosody of the language that the speaker utters. Voices of 12 speakers were recorded. They were six Japanese people and six Americans and Britons. All speakers uttered short sentences in their respective native languages (i.e., Japanese for Japanese speakers and English for Americans and Britons) and in a non-native language (i.e., English for Japanese speakers and Japanese for Americans and Britons). Ten Japanese participants rated the vocal quality of speakers in the first session. After 1 week the same experiment was again conducted in the second session. Results showed that the performance of identification of speakers as Japanese or as non-Japanese was comparatively accurate even though the ratings on the speakers’ voices were varied as the language spoken by the speakers. Ratings of the voices were compared further between two sessions and little difference was found, irrespective of a 1-week blank. Results suggest that the memory for vocal quality is robust, but that the recognition of vocal quality is dependent on the prosody of the language spoken by speakers.
Brain activity during auditory processing affected by expectation of speech versus nonspeech
Authors: Nota Yukiko
Published: October 2013

fMRI was used to clarify whether there is any differential brain activity invoked by expectation for speech versus nonspeech sounds. Auditory stimuli were created by acoustically morphing between either sustained vowels or tones, respectively, and a buzz sound. The two sets of interpolation were performed in nine nonlinear steps; the stimuli retained for perceptual experiments were only the three most vowel-like, the three most tone-like, and the three most buzz-like tokens morphed from the vowels. In the ‘‘speech expectation’’ session, subjects were instructed to discriminate the vowel-like and buzz-like stimuli; in the ‘‘nonspeech expectation’’ session, subjects were instructed to discriminate the tone-like and buzz-like stimuli without knowing that the buzz stimuli had been morphed from the vowels. Thus the buzz-like stimuli in both experiments were the same, but the subjects’ expectations were different because they were told to expect either speech (vowel-like) or nonspeech (tone-like) stimuli. Comparison of brain activation during processing of the buzz-like stimuli under these two conditions revealed that BA40 and thalamus were more activated in speech expectation, while right BA20 was more activated in nonspeech expectation. These results suggest that subjects’ speech/nonspeech expectation for sound stimuli influences brain activity for actual auditory processing. [Work supported by MEXT.]
Changes in formant frequencies associated with postural change in adult male speakers over 50 years old
Authors: Hashi Michiko , Nanto Tomoki , Ohta Natsuki
Published: October 2013

It is possible that changes of direction of gravity relative to the vocal tract associated with changes in posture influence acoustic characteristics of speech including vowel formant frequencies. Studies examining such effects had produced mixed results and demonstrated the possibility of substantive interspeaker variability in the effect of postural changes on vowel formant frequencies. Recent work by Takakura et al. [‘‘Changes in formant frequencies associated with postural change,’’ paper presented at the Fall meeting of Acoustical Society of Japan (2006)], using young adult male speakers, revealed a small number of speakers demonstrating changes in vowel formant frequencies and suggested effect of age. The present study attempts to examine changes of vowel formant frequencies in upright and supine position among older male speakers. Attempts will be made to eliminate the effect of differences in neck position between the postures through the use of a power-bead-based neck stabilizer. The results will be compared with data from young normal speakers in the previous study and inferences will be made relative to speech production models.
Towards estimation of Japanese intelligibility scores using objective voice quality assessment measures
Authors: Kaga Rui , Kondo Kazuhiro , Nakagawa Kiyoshi , Fuzimori Masaya
Published: October 2013

We investigated the use of objective quality measures to estimate the intelligibility of Japanese speech. We initially focused on PESQ (perceptual evaluation of speech quality), which is the state-of-the-art objective assessment method, and can estimate the mean opinion scores (MOS) at an extremely high accuracy. Since we can assume that speech quality is correlated with intelligibility, it should be possible to estimate the intelligibility from the estimated opinion scores or some of its derivatives. We tried to estimate the intelligibility of the Japanese Rhyme Test (DRT). The DRT uses minimal word pairs whose initial phone differs by one phonetic feature. We estimated the MOS of the DRT word samples mixed with noise and tried to correlate this with the measured intelligibility. However, the estimated MOS showed no difference between phonetic features. However, the difference in the estimated MOS between the word pairs seems to differ by a phonetic feature for SNR above 10 dB, which suggests that the estimation of intelligibility by a phonetic feature may be possible. We also plan to selectively use the internal data used to calculate the MOS estimate for better intelligibility estimation.
Two-microphone system using linear prediction and noise reconstruction
Authors: Nakano Hirofumi , Fujii Kensaku , Amitani Tomohiro , Miyata Satoshi , Itoh Yoshio
Published: October 2013

This study proposes a new adaptive microphone system that is characterized by a linear prediction circuit inserted previous to a noise reconstruction filter corresponding to an adaptive delay used in conventional systems. This insertion provides various advantages to the adaptive microphone system. One is that steering a null for a noise source becomes possible irrespective of the incidence of speech signal. Another is that the new microphone system works as an omnidirectional microphone for the speech signal. Consequently, setting the microphones at arbitrary intervals is possible. For example, building one microphone in a handset and another into a telephone baseset becomes possible, which provides higher noise reduction effect. In practical use, microphone systems must function in reflective surroundings. In this study, such performance of the proposed system is verified first using computer simulations and then using an experimental system put in an ordinary room. This study also presents experimental results verifying that the proposed system can successfully reduce noise incident from the same direction as a speech signal, as well as crowd noise recorded in an airport.
Adaptive matched field processing enhancements to forward sector beamforming
Authors: Dunne Jeffrey A.
Published: October 2013

A study was undertaken to examine the potential benefit of adaptive matched field processing (AMFP) to the forward sector capability of single-line, twin-line, and volumetric arrays. Comparisons are made with conventional MFP (CMFP) and adapative and conventional plane-wave beamforming (APWB and CPWB) in order to assess the degree of ownship noise reduction obtainable and any corresponding improvement to the signal-to-noise ratio (SNR). A minimum variance distortionless response beamformer using dominant mode rejection was implemented, applied to both uniform and distorted array shapes. Significant improvement over CMFP and CPWB in tracking and SNR was seen for modeled data in both cases, with the distorted array showing, not surprisingly, better left-right rejection capability. [Work was undertaken with support from the Defense Advanced Research Projects Agency (DARPA) Advanced Technology Office (ATO).]
Adaptive filtering using harmonic structure of voiced speech for reducing nonstationary known noise
Authors: Ota Kenko , Yanagida Masuzo , Yamazaki Tatsuya
Published: October 2013

Proposed is an effective method for reducing nonstationary known noise. The objective of this research is to develop a scheme of preprocessing for speech recognition that keeps the same speech recognition rate even in a worse acoustic environment and to realize a TV control system using speech recognition. The basic idea of the proposed method is to estimate the frequency spectrum of noise including sounds from the TV itself and to remove noise components from the frequency spectrum of the received signal. A transfer function from the TV set to the microphone is calculated in an iterative manner estimating the noise signal at the microphone. Traditional ANC techniques do not directly use spectral features such as harmonic structure or fundamental frequency, for example. As the proposed method uses harmonic structure of vocalic segments in command speech, it is expected to have an advantage compared with the traditional adaptive methods. Results of evaluation using speech recognition show that the proposed method significantly improves speech recognition rate compared with conventional spectral subtraction. [Work supported by Knowledge Cluster Project, MEXT, and by Academic Frontier Project Doshisha University.]