bluuebunny/crossref_metadata_2025 huggingface dataset:
Spatially continuous model of the broadband time-of-arrival in the head-related transfer functions
Authors: Majdak Piotr , Ziegelwanger Harald
Published: May 2013

Head-related transfer functions (HRTFs) describe the filtering of the incoming sound by the human anatomy. They contain the so-called broadband time-of-arrivals (TOAs), which interaural differences yield the well-known interaural time differences used to estimate the lateral position of sound sources by the human auditory system. The TOAs are essential for a time-synchronous binaural rendering of multiple virtual sound sources or for interpolation of the timing information in an existing HRTF set. Estimation of the TOA is usually done separately for each spatial direction and is thus prone to errors and directional outliers. A method for a robust estimation of spatially continuous TOA function from a set of listener-specific HRTFs is presented. The method relies on a geometric model of the HRTF-measurement setup represented by parameters like head position, radius, and ear position. The model parameters were fit to HRTFs of a sphere numerically calculated under various conditions, and to measured HRTFs of 160 listeners. The resulting model parameters and TOA functions corresponded well with the measurement geometry and manually derived TOAs, respectively. The model parameters were further compared to those resulting from a simplified model which assumes the listener being in the center of the HRTF-measurement setup, demonstrating the impact of the usually neglected aspect of listener position on the HRTF timing quality.
Measuring pressure and particle velocity along the human ear canal
Authors: Hiipakka Marko , Pulkki Ville
Published: May 2013

A non-invasive method of measuring or estimating accurately the head-related transfer functions (HRTFs) and headphone transfer functions (HpTFs), i.e., the pressure at the eardrum rather than at the blocked ear canal entrance is called for. In this work, a miniature-sized acoustic pressure–velocity sensor is used to measure both pressure and velocity along the ear canals of human test subjects. The measurements are used to study the applicability of a recently proposed method of estimating the pressure at the eardrum from pressure–velocity measurements made at the ear canal entrance. The measurement results are compared to results from computational modeling with human ear canal parameters. In addition, the effect of the PU-sensor itself on the pressure at the eardrum is studied. It is shown that the estimation method is reliable and accurate for most human subjects. The diameter and the shape of the ear canal affect the results in such a way that the best results are obtained with wide and straight ear canals whereas less accurate results are obtained with narrow and curved ear canals. It is concluded that the estimation method facilitates the obtaining of individual HRTFs and HpTFs at the eardrum using non-invasive measurements.
Effects of compression on the use of onset time differences to detect one tone in the presence of another
Authors: Madsen Sara M. , Moore Brian
Published: May 2013

It is easier to hear one of two notionally “simultaneous” tone complexes if the onset of the masker complex is delayed relative to that of the signal. However, the ability to use onset asynchrony as a cue may be reduced when using amplitude compression, due to distortion of the onset of sounds (overshoot effects). We assessed how fast- and slow-acting five-channel compression affects the ability to use onset asynchrony to detect one (signal) complex tone when another (masking) complex tone is played almost simultaneously. A 2:1 compression ratio was used with normal-hearing subjects and individual compression ratios and gains recommended by the CAM2 hearing aid fitting method were used for hearing-impaired subjects. For the normal-hearing subjects, performance improved with increasing onset asynchrony in all conditions. The improvement was greatest with fast compression and least with no compression. Preliminary results for the hearing-impaired subjects indicate smaller but similar effects of onset asynchrony and a greater benefit of compression. The benefit of compression probably occurs because compression increases the level of the part of the signal that occurs before the masker relative to the masker. [Work supported by Starkey (U.S.A.) and the MRC (UK).]
Study on effects of presence of cue-tone on psychophysical tuning curves
Authors: Kidani Shunsuke , Miyauchi Ryota , Unoki Masashi
Published: May 2013

Our previous study indicated that tunings of the auditory filter were sharpened by the presence of a cue-tone [Kidani et al. (2012), ISH2012]. It is unclear, however, whether the variation of the auditory filter due to the cue-tone is caused by excitation or suppression, because tip of the filter is normalized at 0 dB. Psychophysical tuning curves (PTCs) can show that the detection threshold is decreased at the probe frequency or increased around the probe by the presence of cue-tone, indicating excitation and suppression respectively. PTCs, because, are measured as masked threshold of probe by narrow-band noise. This study aims to consider the effect of the presence of cue-tone by measuring of PTCs. In present study, PTCs were measured in simultaneous masking in the absence and presence of cue-tone for four probe frequencies. The probes were presented at 10 dB above each absolute threshold. The frequency and level of the cue-tone were same as the probe. The result revealed that filter-Q, regarded as the sharpness of tuning, was increased by the presence of cue-tone when the probe frequencies were 1.0 and 2.0 kHz, while the filter-Q was not changed when the probe frequencies were 0.5 and 4.0 kHz.
Reliability of procedures used for scaling loudness
Authors: Jesteadt Walt , Joshi Suyash N.
Published: May 2013

In this study, 16 normally hearing listeners judged the loudness of 1000-Hz sinusoids using magnitude estimation (ME), magnitude production (MP), and categorical loudness scaling (CLS). Listeners in each of four groups completed the loudness scaling tasks in a different sequence on the first visit (ME, MP, CLS; MP, ME, CLS; CLS, ME, MP; CLS, MP, ME), and the order was reversed on the second visit. This design made it possible to compare the reliability of estimates of the slope of the loudness function across procedures in the same listeners. The ME data were well fitted by an inflected exponential (INEX) function, but a modified power law was used to obtain slope estimates for both ME and MP. ME and CLS were more reliable than MP. CLS results were consistent across groups, but ME and MP results differed across groups in a way that suggested influence of experience with CLS. Although CLS results were the most reproducible, they do not provide direct information about the slope of the loudness function because the numbers assigned to CLS categories are arbitrary. This problem can be corrected by using data from the other procedures to assign numbers that are proportional to loudness. [Work supported by NIH.]
Multicomponent signal detection: Tones in noise and amplitude modulation detection
Authors: Thompson Eric R. , Simpson Brian D. , Iyer Nandini
Published: May 2013

In order to predict the detectability of broadband acoustic signals, a model must include a means of integrating information across frequencies. There have been several previous studies measuring the detectability of multicomponent signals, but it is still not clear what the best model is when signal components are not equally detectable. Some researchers have proposed that thresholds are driven by the most detectable component (max-d’ model), while others have found that the best model for their data is a statistical summation model, where component sensitivities are combined using a Pythagorean sum. In the present study, detection thresholds were collected in broadband noise for single tones at three frequencies and three signal-to-noise ratios (SNRs) and for all combinations of SNRs for the three tones presented together. Also, amplitude modulation (AM) detection thresholds were measured for a 16-Hz AM signal imposed on 300-Hz-wide noise bands centered at three frequencies at three modulation depths for each band individually, for combinations of two bands and for all three bands presented together. While both models (max-d’ and Pythagorean sum) can predict the general trend of the multicomponent data from the single component data, neither model fits the data very well.
An automated procedure for detecting human frequency-following responses to voice pitch
Authors: Jeng Fuh-Cherng , Hu Jiong
Published: May 2013

The frequency-following response (FFR) to voice pitch has been widely examined in research laboratories and has demonstrated its potential to be transformed into a useful tool for patients with hearing, speech, and language disorders in the clinic. During the past decade, many aspects of the FFR have been reported. The presence of such a response, however, still relies on subjective interpretation of the observer. Aside from a recent study reporting two algorithms for detecting such a response, there has been limited number of studies reporting the development of an automated procedure for FFR. The purpose of this study is (1) to develop an automated procedure that utilizes the statistical properties of the temporal and spectral energy distributions in the recorded waveforms and (2) to explore the effectiveness, accuracy, and efficiency of the automated procedure and compare them with those obtained from conventional algorithms and human judgments.
Sensory consonance of two simultaneous sine-tones
Authors: Frosch Reinhart
Published: May 2013

In Chapter 4 of my book “Musical Consonance and Cochlear Mechanics” (vdf, Zurich, 2012), four psychoacoustic experiments on the sensory consonance of two simultaneous sine-tones are described. In each of those experiments, the deeper-tone frequency fd was kept fixed, at fd = 132, 264, 528, or 1056 Hz. Each experiment was done twice, at sound-pressure levels of 50 and 70 dB (SPL). The resulting consonance curves (sensory consonance versus higher-tone frequency fh) exhibit consonance minima at beat-rates bm.d. = fh - fd (where “m.d.” = “most dissonant”) ranging from 13 Hz (at fd = 132 Hz) to 39 Hz (at fd = 1056 Hz). In Section 15.1 of the above-mentioned book, these most dissonant beat rates are shown to agree well with the following empirical law: bm.d. = (1.07s-0.5) * sqrt(favg), where favg = fd + bm.d./2. The present study was prompted by the comments of a reader; the just described empirical law is unsatisfactory because in the underlying experiments the deeper-tone frequency fd [rather than the average frequency (fd + fh)/2] was kept constant. It was found that the data agree equally well with the following modified empirical law: bm.d. = (1.09s-0.5) * sqrt(fd). This modification does not affect the validity of the complex-tone consonance theories described in Chapters 15 and 16 of the mentioned book.
Ventriloquism effect and aftereffect in the distance dimension
Authors: Hládek Ľuboš , Le Dantec Christophe C. , Kopčo Norbert , Seitz Aaron R.
Published: May 2013

When an auditory target is presented simultaneously with a spatially displaced visual target, the perceived auditory target location shifts toward the visual target. This effect, known as the ventriloquism effect or visual capture, has been extensively studied in the horizontal dimension, but not in distance. Here, we measured distance localization performance in a reverberant room. Stimuli were either audio-visual (AV) 300-ms broadband noise bursts presented synchronously with spatially congruent or incongurent visual stimuli/LEDs, or auditory-only (A-only) noise bursts. One of eight speakers (distance 70 cm to 203 cm directly ahead of the listener) presented a stimulus on each trial. During adaptation runs, the AV stimuli were presented with the V-component closer or further by 30% than the A-component (displacement direction fixed within session). The ventriloquism effect was observed for both V-closer and V-further AV stimuli, with slightly stronger shifts induced by the V-closer stimuli. Ventriloquism aftereffect, assessed by presenting A-only trials interleaved with the adaptation-AV trials, was also observed, but was weaker than the ventriloquism effect. The results suggest that visual targets do capture auditory targets in the distance dimension, but visual modulation might be asymmetrical with respect to distance. [Work supported by EU FP7-247543, VEGA-1/0492/12, NSF (BCS-1057625).]
An assessment of virtual auditory distance judgments among blind and sighted listeners
Authors: Kolarik Andrew J. , Cirstea Silvia , Pardhan Shahina , Moore Brian
Published: May 2013

Auditory distance perception is a crucial component of blind listeners’ spatial awareness. Many studies have reported supra-normal spatial auditory abilities among blind individuals, such as enhanced azimuthal localization [Voss et al. (2004)] and distance discrimination [Kolarik et al. (in press)]. However, it is not known whether blind listeners are better able to use acoustic information to enhance judgments of distance to single sound sources, or whether lack of visual spatial cues prevents calibration of auditory distance information, leading to worse performance than for sighted listeners. Blind and sighted listeners were presented with single, stationary virtual sound sources between 1.22 and 13.79 m away in a virtual anechoic environment simulated using an image-source model. Stimuli were spoken sentences. Sighted listeners systematically underestimated distance to remote virtual sources, while blind listeners overestimated the distance to nearby virtual sources and underestimated it for remote virtual sources. The findings suggest that blind listeners are less accurate at judging absolute distance, and experience a compression of the auditory world, relative to sighted listeners. The results support a perceptual deficiency hypothesis for absolute distance judgments, suggesting that compensatory processes for audition do not develop among blind listeners when estimating the distance to single, stationary sound sources.
Noise control using lightweight acoustic metamaterials
Authors: Naify Christina J.
Published: May 2013

Metamaterials have emerged as promising solutions for manipulation of sound waves in a variety of applications. Negative dynamic mass has been explored in metamaterial applications to improve sound insulation in both three-dimensional (ball-in-rubber), and two-dimensional (membrane-type) approaches. Noise control utilizing locally resonant acoustic materials (LRAM) resulted in improved sound insulation by 500% over acoustic mass law predictions at peak transmission loss (TL) frequencies. The LRAM contribute minimal added mass, making them appealing for weight-critical applications such as aerospace structures. In this study, an overview of LRAM for noise control applications will be presented, including potential issues associated with scale-up of the structure. TL of single-celled and multi-celled LRAM was measured using an impedance tube setup with systematic variation in geometric parameters to understand the effects of each parameter on acoustic response. Finite element analysis (FEA) was also performed to predict TL as a function of frequency for structures with varying complexity, including stacked structures and multi-celled arrays. [Work supported by the Office of Naval Research.]
Importance of container geometry of an elastomer particulate damper called Enidamp™ into the damping of vibrations
Authors: Bustamante Marcelo , Gerges Samir N. , Cordioli Júlio A. , Martin Ovdalio P. , Weisbeck Jeffrey N. , Ott Mark
Published: May 2013

Research testing has been conducted to development of an innovative damping treatment, called Enidamp™. This treatment can add considerable damping to a structure by leading a vibration through a rigid connection, to a set of elastomer particles, which behaves as a damper. It is of interest to study the parameters involved in this mechanism of energy dissipation and to achieve optimal performance of the Enidamp™ system. Particularly, this paper experimentally analyzes the importance of the container geometry which houses the elastomer particles. For this purpose, the fluidization point at which the elastomeric particles become optimally excited to maximize damping is found for different depths and widths of the Enidamp™ container keeping the volume constant. Important conclusions from this experiment guide future studies for prototype improvements.
Experimentally elicited productions: Differences and similarities between mixed effects and ANOVA analyses
Authors: Goldrick Matthew
Published: May 2013

Currently, many experimental studies of speech production use fully counterbalanced designs to examine variation in categorical (e.g., correct/incorrect) or relatively continuous measures (e.g., reaction times, voice onset times). These data present several challenges to ANOVA analyses. Some of these issues are well known to be the speech community; for example, the non-normality of dependent variables such as proportion correct. Others have been less extensively addressed; for example, many speech studies account for participant- but not item-specific contributions to variance. I’ll discuss the opportunities and challenges in using linear mixed effects models to address these issues. I’ll review some of the common issues that arise in using such models and discuss how to interpret and report their results.
Autocorrelation function mechanism for pitch salience and cross-correlation function mechanism for sound localization revealed by magnetoencephalography
Authors: Soeta Yoshiharu , Shimokura Ryota , Nakagawa Seiji
Published: May 2013

A model of primary sensations and spatial sensations is proposed by Ando (2001). The model of the auditory-brain system includes the autocorrelation function (ACF) and the interaural cross-correlation function (IACF) mechanisms. At present, environmental noises are evaluated by sound level such as equivalent continuous A-weighted sound pressure level (LAeq). However, we sometimes feel annoyed with sound with low sound level because of the quality. Sound quality can be characterized by factors obtained from ACF and IACF of sound. For example, pitch and pitch strength can be characterized by delay time and amplitude of the maximum peak of the ACF. Directional sensation can be characterized by delay time and amplitude of the maximum peak of the IACF. To verify the model, we investigated how ACF and IACF factors are coded in our human brain. The results indicated that delay time and amplitude of the maximum peak of the ACF and IACF are coded by the latency and strength of brain activity. In addition, we applied the model to analyze a Buddhist sutra chanted in temples. The results indicated that some characteristics of the sutra could be characterized by the ACF and IACF factors.
Autocorrelation-based features for speech representation
Authors: ando Yoichi
Published: May 2013

This study investigates autocorrelation-based features as a potential basis for phonetic and syllabic distinctions. The work comes out of a theory of auditory signal processing based on central monaural autocorrelation and binaural crosscorrelation representations. Correlation-based features are used to predict monaural and binaural perceptual attributes that are important for the architectural acoustic design of concert halls: pitch, timbre, loudness, duration, reverberation-related coloration, sound direction, apparent source width, and envelopment (Ando, 1985, 1998; Ando and Cariani, 2009). The current study investigates the use of features of monaural autocorrelation functions (ACFs) for representing phonetic elements (vowels), syllables (CV pairs), and phrases using a small set of temporal factors extracted from the short-term running ACF. These factors include listening level (loudness), zero-lag ACF peak width (spectral tilt), τ1 (voice pitch period), φ1 (voice pitch strength), τe (effective duration of the ACF envelope, temporal repetitive continuity/contrast), segment duration, and Δφ1/Δt (the rate of pitch strength change, related to voice pitch attack-decay dynamics). Times at which ACF effective duration τe is minimal reflect rapid signal pattern changes that usefully demarcate segmental boundaries. Results suggest that vowels, CV syllables, and phrases can be distinguished on the basis of this ACF-derived feature set.
The role of normalization in phoneme recognition and speaker definition
Authors: Patterson Roy D. , Irino Toshio
Published: May 2013

There is size information in speech sounds because the vocal tract and the vocal cords both grow as a child develops into an adult. Specifically, average pitch and mean formant frequency decrease as speaker size increases. Nevertheless, human speech recognition is effectively size invariant across the full range of sizes in the normal population of speakers and well beyond. It is also the case that listeners can discriminate speaker size with great accuracy; indeed, with greater accurately than they can discriminate the loudness of sound or the brightness of light. The first part of this talk describes how the peripheral auditory system normalizes speech sounds automatically to produce a size invariant representation for speech recognition. The second part presents a model of how the central auditory system transforms information in the cochlea into our perception of who is speaking and what they are saying. The model suggests that the system combines information about vocal resonator size with a small amount of contextual information to determine what the person is saying (at the phonological level), and then it adds voice pitch information to determine who is speaking (in the sense of the sex and size of the speaker).
Establishing typicality: A closer look at individual formants
Authors: Hughes Vincent
Published: May 2013

Research into the forensic performance of individual formants has offered considerable evidence to support the traditional acoustic-phonetic view that whilst F1 and F2 encode broad phonetic contrast, higher formants may offer greater speaker-discriminatory potential (Peterson 1959, Ladefoged 2006, Clermont and Mokhtari 1998, Rose 2002). However, the comparative performance of formants has largely been assessed using posterior assessments of speaker-specificity (McDougall 2004, 2006; Clermont et al. 2008). Using quadratic polynomial fittings of F1 to F3 from spontaneous tokens of /ai/ extracted from all 100 speakers in the DyVis database (Nolan et al. 2009), this paper discusses issues relating to p(H|E)-based voice comparison analysis (particularly the use of discriminant analysis, DA). Further, DA performance is compared with an analysis based on likelihood ratios (LRs). LRs based on F3 are found to only marginally outperform F1 and F2 with regard to the magnitude of same-speaker and different-speaker strength of evidence, as well system performance metrics (EER and Cllr). The poorer than expected F3 LRs are assessed with regard to the distributions of values within- and between-speakers for the best performing F3 coefficient: the constant. The data go some way to establishing F3 population statistics, which may potentially be applied to voice comparison casework.
An automated framework for the extraction of ultrasonic echoes embedded in noise
Authors: Pedrycz Adam , Valero Henri-Pierre , Hori Hiroshi , Nishimiya Kojiro , Sugiyama Hitoshi , Sakata Yoshino
Published: May 2013

Proposed is an automated framework for the extraction and characterization of the arriving echo in ultrasonic signals embedded in high noise. Commonly, in order to correctly characterize the first echo hidden within a noise-ridden signal, multiple traces are stacked in a gather to improve the SNR, hence facilitating easier extraction and characterization of the recorded echo. Such first order statistical methods require multiple traces and usually fall short in the accuracy of the echo estimate when the variance of the noise does not belong to a known distribution. To mitigate this problem, a framework has been developed comprised of a multi-step procedure, i.e., pre-processing, localization, gating, and finally parameterization of the given echo. This automatic framework operates on single traces and does not require the setting of processing parameters. By means of this method, the true echo can be extracted in one-shot from other overlapping noise components. Furthermore, because the method operates on a trace-by-trace basis, it is insensitive to large non-stationarities in the baseline. Experiments conducted using synthetic as well as aluminum reflector pulse-echo lab data demonstrate the effective extraction of the true echo under the presence of noise and ringing at varying levels of severity.
Extract voice information using high-speed camera
Authors: Akutsu Mariko , Oikawa Yasuhiro , Yamasaki Yoshio
Published: May 2013

Conversation is one of the most important channels for human beings. To help communications, speech recognition technologies have been developed. Above all, in a conversation, not only contents of utterances but also intonations and tones include important information regarding a speaker’s intention. To study the sphere of human speech, microphones are typically used to record voices. However, since microphones have to be set around a space, their existences affect a physical behavior of the sound field. To challenge this problem, we have suggested a recording method using a high-speed camera. By using a high-speed camera for recording sound vibrations, it can record two or more points within the range of the camera at the same time and can record from a distance, without interfering with the sound fields. In this study, we extract voice information using high-speed videos, which capture both a face and a cervical part of the subject. This method allows recording skin vibrations, which contain voices with individuality and extrapolating sound waves by using an image processing method. The result of the experiment shows that a high-speed camera is capable of recording voice information.
Multi-stage identification for abnormal/warning sounds detection based on maximum likelihood classification
Authors: Hayashida Kohei , Ogawa Junpei , Nakayama Masato , Nishiura Takanobu , Yamashita Yoichi
Published: May 2013

In recent years, the methods utilizing environmental sounds have been increasingly employed for monitoring the safety of the elder who lives in distant place. Environmental sounds should consist of various sounds in daily life, and identified ones enables to detect abnormity. To detect abnormity, it is therefore required that abnormal/warning sounds are accurately identified among environmental sounds. In the past, environmental sound identification methods have generally utilized acoustic models constructed by each sound source for all environmental sounds. In our former research, we proposed multi-stage identification for detecting abnormal/warning sounds accurately. However, these methods design individual acoustic models from similar sounds. Therefore, the sound identification performance is degraded. To overcome this problem, in this study, we proposed environmental sound classification based on acoustic features for model construction. The proposed method classifies environmental sounds based on the difference of acoustic likelihood and designs acoustic models are constructed by classification results. Moreover, the proposed method detects the abnormal/warning sounds more accurately by combining it with the multi-stage identification. We carried out the evaluation experiment with environmental sound database. Experimental results of this experiment demonstrate that the identification performance for the proposed method is higher than that for the conventional methods.
Backward waves and quasi-resonance of shells and invariants of the time reversal operator
Authors: Franck Philippe D. , Dominique Clorennec , Cès Maximin , Anankine Romain , Prada Claire
Published: May 2013

Backward waves propagating on shell are guided modes with opposite phase and group velocities. For a shell in vacuum, backward modes are linked to zero group velocity modes and resonances, which have been the object of recent studies. For a shell embedded in water, the group velocity does not vanish because of the leakage into the fluid. However, the group velocity of the backward mode has a minimum associated to a quasi-resonance. These phenomena are studied on air filled steel and zircaloy hollow cylinders, using a 3 MHz linear array in pulse echo mode. The circumferential guided modes are generated and their radiation into water detected by the array. The modes are separated using the decomposition of the time reversal operator (TRO), each pair of counter-propagating modes being associated to two invariants of the TRO [Prada et al. J. Acoust. Soc. Am. (1998)]. Two resonances are revealed by the eigenvalues of the TRO, one is associated with the first longitudinal thickness resonance and the other, very high, occurring at a slightly lower frequency, corresponds to the minimum of the group velocity of the backward mode. The back-propagations of the eigenvectors of the TRO provide the phase velocities of these modes.
Including frequency-dependent attenuation for the deconvolution of ultrasonic signals
Authors: Carcreff Ewen , Bourguignon Sébastien , Idier Jérôme , Simon Laurent
Published: May 2013

Ultrasonic non-destructive testing (NDT) is a standard process for detecting flaws or discontinuities in industrial parts. A pulse is emitted by an ultrasonic transducer through a material, and a reflected wave is produced at each impedance change. In many cases, echoes can overlap in the received signal and deconvolution can be applied to perform echo separation and to enhance the resolution. Common deconvolution techniques assume that the shape of the echoes is invariant to the propagation distance. This can cause poor performances with materials such as plastics or composites, in particular because acoustic propagation suffers from frequency-dependent attenuation. In geophysics, biomedical imaging or NDT, various frequency-dependent attenuation models have been proposed under different formulations. This communication compares the related possible constructions in order to account for attenuation in deconvolution methods. Especially, we introduce a discrete model for the data, that includes an attenuation matrix in the standard convolution model. Experimental data acquired from Plexiglas plates show that, for this material, attenuation varies roughly linearly with frequency, leading to a unique parameter identification. Finally, we show that such an advanced model manages a better fitting of the data, and promises improvement for the deconvolution of complex ultrasonic data.
Low latency audio coder design for high quality audio service on server-client environment
Authors: Moon Han-gil , Lee Nam-suk , Kim Hyun-wook
Published: May 2013

Low latency audio coding attracts increasing attention among high quality communication applications such as video conferencing system and server-client media applications such as cloud computing based interactive A/V service. This paper presents a low latency audio coding scheme which can achieve both low delay and high subjective audio quality at the same time. In order to guarantee low delay, the proposed coding system incorporates low overlap window while preserving the window size as same as that of conventional (AAC) long window. The supplementary signal processing tool is incorporated to enhance the audio quality. The proposed coding scheme achieves the delay of 24 ms in 48 kHz and the MUSHRA score, which is comparable to that of commercialized AAC.
Optimized hermetic transform beam-forming of acoustic arrays via cascaded spatial filter arrangements derived using a chimerical evolutionary genetic algorithm
Authors: Woodsum Harvey C. , Woodsum Christopher M.
Published: May 2013

Hermetic transforms are complex matrices, having particular mathematical properties, that have recently been introduced to the field of acoustic array signal processing. Cascade sequences of Hermetic transform matrices have been shown to have direct utility in accomplishing spatial filtering and beam-forming of data from oversampled arrays. The present work details the adaptation of techniques previously shown to be successful in the processing of radio-wave phased-array antenna systems [Woodsum et al., 16th International Conference on Cognitive and Neural Systems (2012)] to the processing of sampled digital data from acoustic arrays. As in our earlier work, the use of a Chimerical, Evolutionary, Genetic Algorithm, having a “feature seeking” fitness function, is retained, for deriving optimal multiplicative arrangements of non-commuting elemental transform matrices. Each elemental matrix represents a spatial “pole” or “zero,” and cascaded arrangements of these are utilized to create a desired spatial pattern response for the array. In terms of acoustic reception, the technique is especially successful in dealing with null placement in order to mitigate large numbers of interfering signals, and in achieving super-resolution beams for arrays that are “acoustically small.” Experimental results are compared to theoretical predictions of performance.
Adaptive active control of free space acoustic noise
Authors: Tabatabaei Ardekani Iman , Abdulla Waleed H.
Published: May 2013

This paper concerns adaptive active control of acoustic noise in free space. Traditional adaptive active noise control algorithms are efficient in acoustic ducts; however, they are very unstable and sensitive when being used in free space. An efficient adaptive active noise control algorithm for free space noise is derived based on a root locus analysis on the adaptation process performed in adaptive active noise control. The traditional algorithm and the proposed algorithm are fully implemented by using a high performance embedded controller. The controller is then used for active control of acoustic noise in a duct and, also, in free space. Different experiments show that the traditional active noise control algorithm is not stable when the setup is used in free space. However, the proposed algorithm is stable and converges at a high convergence rate until reaching steady state conditions.
Parametric loudspeaker for speech signal based on the combination of amplitude and frequency modulations
Authors: Iwasaki Toru , Ikefuji Daisuke , Nakayama Masato , Nishiura Takanobu
Published: May 2013

A parametric loudspeaker has been used for audio guidance to a specific area because it has a sharper directivity compared with the conventional electrodynamic loudspeakers. The parametric loudspeaker emits an ultrasound as a carrier wave, which is modulated with an audio signal and has large-amplitude. An audible sound is reproduced by the modulated ultrasound with large-amplitude distorted by the nonlinearity on the air. The conventional modulations have been proposed as the amplitude modulation and the frequency modulation. In the sound quality, the amplitude modulation is superior to the frequency modulation. However, in the sound pressure level, the frequency modulation is superior to the amplitude modulation. In the present paper, we especially focus on that the parametric loudspeaker will emit the speech signals for the audio guidance. Therefore, we propose new modulation method based on the combination of amplitude and frequency modulations, which are specialized for speech signals. More specifically, we apply the amplitude and frequency modulations to the divided frequency bands of speech signals, respectively. In order to confirm the effectiveness of the proposed method, we carried out evaluation experiments. As a result of evaluation experiments, we confirmed the effectiveness of the proposed method compared with conventional methods.
A design of audio spot based on separating emission of the carrier and sideband waves
Authors: Matsui Tadashi , Ikefuji Daisuke , Nakayama Masato , Nishiura Takanobu
Published: May 2013

Parametric loudspeaker, which utilizes an ultrasonic of non-linear interaction, is developed to achieve audio spot. The parametric loudspeaker has sharper directivity, but reflections and intercepts by emitted sounds become severe problems. This is because reflections and intercepts lead to an invasion of privacy, and become noise to other listeners except a target listener. Principle of the parametric loudspeaker can formulate as non-linear interaction of carrier and sideband waves in emitted ultrasonic sounds on air. This suggests that we can design audio spot by individually emitting the carrier and sideband waves. In the present paper, therefore, we propose the design method of audio spot with the separating emission of the carrier and sideband waves. More specifically, the audible sound is demodulated at an area where the carrier and sideband waves individually emitted from each parametric loudspeaker are overlapped. We carried out evaluation experiments to measure sound pressure level (SPL) of demodulated audible sound. In addition, we evaluated the speech articulation of the demodulated audible sound with the proposed method. As a result of evaluation experiments, we confirmed the effectiveness of the proposed method.
The impact of finite ensonified area on the scattering cross section
Authors: Olson Derek R. , Lyons Anthony P.
Published: May 2013

The definition of the scattering cross section for the pressure field scattered by a rough interface uses the underlying assumption that the ensonified area does not affect its shape. This assumption holds so long as the incident field is a good approximation of a plane wave, which situation occurs when the ensonified area is large compared to a wavelength. In the opposite situation, when the ensonified dimensions approaches a wavelength, the incident plane wave assumption does not hold and the cross section can depart from modeled behavior. This research uses the perturbation approximation to derive a model for the low grazing-angle behavior of the scattering cross section. The primary results are (1) the appearance of an additive Lambertian term and (2) a separation of scales imposed by the acoustic resolution. This separation can then be used as a criterion for the application of the composite roughness model. Model results are checked against direct numerical solution of the governing integral equations. Implications for inversion of seafloor parameters based on acoustic scattering experiments is discussed.
Geo-acoustic parameter estimation using a multistep inversion technique based on normal mode method
Authors: Wan Lin , Badiey Mohsen , Knobles David
Published: May 2013

The geo-acoustic parameters are of great importance in determining how the sea bottom affects sound propagation in the ocean. Many inversion techniques have been developed to estimate geo-acoustic parameters. One-step inversion algorithms using a cost function defined only by energy loss may not result in a unique solution of geo-acoustic inversion problem because of the correlation between seabed sound speed and attenuation. The present paper utilizes different characteristics of normal modes, including modal dispersion, modal attenuation, and modal based spatial coherence, to define appropriate cost functions in a multistep inversion algorithm for geo-acoustic parameter estimation. This inversion scheme is applied to the long-range broadband acoustic data obtained from L-shaped arrays in the Shallow Water 2006 experiment. The seabed sound speed and attenuation are estimated by minimizing the cost function at each step. The results show a nonlinear frequency dependence of attenuation, which is similar to the seabed attenuation derived from measured time series and transmission loss data at the same experimental site [Knobles et al., J. Acoust. Soc. Am. 124, 2008]. The uncertainties caused by the range dependent water column variability and bathymetry are discussed. [Work supported by ONR322OA.]
Basics and applications of psychoacoustics
Authors: Fastl Hugo
Published: May 2013

The field of psychoacoustics studies relations between acoustic stimuli, defined in the physical domain, and the hearing sensations elicited by these stimuli. The plenary lecture will address questions of stimulus generation and presentation. Both traditional methods and more recent methods like wave field synthesis will be touched. Concerning hearing sensations, basic magnitudes as for example absolute thresholds or loudness, but also advanced topics like pitch strength will be covered. Applications of psychoacoustics will include rather different fields such as music, noise evaluation or audiology, and address also cognitive effects as well as audio-visual interactions.
Analysis of sound absorption behavior of polyester fiber material faced with microperforated panels
Authors: Borelli Davide , Schenone Corrado , Pittaluga Ilaria
Published: May 2013

Perforated facings used in lined ducts or sound absorbing panels can have various purposes: protecting the porous sound absorbing material from dust or grazing flow, acting as a rigid support for the porous material, or also affecting the behavior of the “backing” material, modifying this way the acoustical performance of the porous layer. This paper describes the effect of perforated facings on sound absorption characteristics of samples made by polyester fiber, experimentally investigated in accordance with ASTM C384 04 standard by means of two Kundt’s tubes with different diameters. The polyester (PET) fiber material had bulk density of 30 kg/m3 and melting point at 260°C. The analysis was performed for a sample thickness equal to 100 mm. The samples were faced by means of different metal plates perforated with circular holes. The holes diameter was equal to 2 mm for all facings, while the percent open area was varied from 4.9% to 30%. The microperforated panels (MPPs) were positioned in adherence of the PET fiber material or at a distance of 2, 4, and 6 mm. The different behaviors due to the multiple combinations of percent open area and distance from the sample have been then analyzed and discussed.
Odontocete click train deinterleaving using a single hydrophone and rhythm analysis
Authors: Le Bot Olivier , Bonnel Julien , Mars Jérôme I. , Gervaise Cédric
Published: May 2013

Most odontocetes live in pods of several individuals, resulting in an overlapping of click trains recorded by passive acoustic monitoring systems. Localization algorithms and click classifiers are usually used for train separation. However, their performances fall down if individuals are too close to each other or if acoustical parameters vary greatly from click to click, respectively. Assuming odontocete clicks follow rhythmic patterns, we propose to use a rhythm analysis to separate mixed click trains from a single hydrophone. The proposed algorithm is based only on inter-click-intervals (ICI) to cluster clicks into trains. It uses information given by complex-valued autocorrelation to compute a histogram, which will exhibit peaks at ICIs corresponding to interleaved trains. By this technique, subharmonics corresponding to multiples of ICIs are automatically suppressed. The algorithm is then extended by a time-period analysis leading to a time-varying ICI spectrum. A threshold can be applied on this spectrum to detect the different interleaved trains. The final result is a binary time-ICI map on which trains can be fully and easily distinguished and extracted. We validate it on simulated and experimental data, and we show that the algorithm is particularly suitable as a preprocessing tool prior to localization and classification schemes.
Exploring the shelf-slope dynamics in the Adriatic Sea using numerical models and seismic oceanography
Authors: Bergamasco Andrea , Falcieri Francesco , Book Jeff W. , Carniel Sandro , Wood Warren W. , Sclavo Mauro , Hobbs Richard W.
Published: May 2013

Dense shelf waters are formed and spread in the Adriatic Sea during winter periods, which dynamics are usually investigated by means of sea truth campaigns and modeling efforts. The former are either based on observational approaches (moored instruments, CTD, current meters, etc.) or on more innovative techniques, e.g., employing Seismic Oceanography (SO). Recent studies have shown that SO techniques can produce maps of vertical transects along the survey lines with horizontal and vertical resolution of, respectively, 10 and 100 m, suitable to explore the horizontal structures of BBL dynamics. Elaborating on these considerations, a novel approach combining the SO dataset collected during the ADRIASEISMIC cruise and high-resolution numerical model (ROMS) results was performed in two restricted areas of the Adriatic Sea: off the Gargano promontory and off the Bari shelf break. We present the first steps along the definition of a novel methodology. On one hand, SO can help to image the existing dynamical structures and their spatial/temporal resolution; on the other, the numerical model can quantify these acoustic snapshots in terms of temperature, salinity, and density, integrating the XBTs that are acquired during SO lines, and help identifying the nature of other processes (e.g., turbulence, internal waves, etc.).
Mapping turbidity layers using a combination of high resolution seismic oceanographic and physical oceanographic data
Authors: Vsemirnova Ekaterina A. , Hobbs Richard W.
Published: May 2013

Synchronized seismic and oceanographic data were acquired during the Geophysical Oceanography (GO) project cruise in the Gulf of Cadiz in April–May 2007. The small volume (117 cu-in.) mini GI-gun seismic source used during the GO calibration experiment provided high resolution seismic data, which unveiled new features of the internal structure of the ocean. The seismic acquisition design gave a usable bandwidth of 50–250 Hz with a vertical resolution of 1.25 m, which is similar to that achieved by co-located CTD casts. We focus on the reflections observed on seismic data covering the moorings area. To test the hypothesis that measurable reflections can be generated by suspended sediment, we perform forward modeling of seismic response based on the temperature, salinity, and light attenuation measurements, available from CTD casts. Forward modeling based solely on temperature and salinity profiles show that thermohaline structure does not always explain reflections in water column, but they are consistent with light attenuation measurements.
Characterization of thermohaline staircases in the Tyrrhenian sea using stochastic heterogeneity mapping
Authors: Buffett Grant G. , Hobbs Richard W. , Vsemirnova Ekaterina , Klaeschen Dirk , Hurich Charles , Ranero C. , Sallares V.
Published: May 2013

We apply Stochastic Heterogeneity Mapping based on the band-limited von Kármán power law function to stacked migrated seismic data of thermohaline staircases in the Tyrrhenian Sea. This process allows the estimation of stochastic parameters such as the Hurst number (a measure of surface roughness) and scale length. Thermohaline staircases are regular, well-defined step-like variations in vertical profiles of temperature and salinity. In the ocean, they are thought to arise from double diffusion processes driven by the large difference in the molecular diffusivities of heat and salt. They are thought to have an anomalously weak internal wave-induced turbulence, making them suitable for the estimation of a lower detection limit of turbulent dissipation. The Tyrrhenian Sea is a natural laboratory for the study of such staircases because, due to the internal basin’s dynamic stability, steps as small as 10’s of meters can be seen. Lower Hurst numbers represent a richer range of higher wavenumbers corresponding to a broader range of heterogeneity in reflection events. We interpret a broader range of heterogeneity as indicative of a greater degree of turbulence.