bluuebunny/crossref_metadata_2025 huggingface dataset:
Robust F0 estimation based on complex speech analysis
Authors: Kinjo Tatsuhiko , Funaki Keiichi
Published: October 2013

This paper proposes a novel robust fundamental frequency (F0) estimation algorithm based on complex-valued speech analysis for an analytic speech signal. Since the analytic signal provides spectra only over positive frequencies, spectra can be estimated accurately at low frequencies. Consequently, it is considered that F0 estimation using the residual signal extracted by complex-valued speech analysis can perform better for F0 estimation than that for the residual signal extracted by conventional real-valued LPC analysis. In this paper, the autocorrelation function (AUTOC) weighted by a reciprocal of the AMDF is also adopted for the F0 estimation criterion. The proposed F0 estimation algorithm is evaluated using three criteria, AUTOC, AMDF, and the weighted AUTOC, with complex-valued residual extracted by complex-valued speech analysis. We also compared the proposed method with those for three signals: the speech signal, analytic speech signal, and LPC residual. Speech signals used in the experiments were corrupted by addition of white Gaussian noise whose noise levels are 10, 5, 0, and −5 [dB]. The performance was evaluated using the gross pitch error (GPE) and fine pitch error (FPE). The results demonstrate that this algorithm based on complex speech analysis can perform better than conventional methods in an extremely noisy environment.
Cross-linguistic examination of listeners frequency thresholds for musical tones and vowels
Authors: Pape Daniel
Published: October 2013

There is clear evidence that musical education increases the discriminability of the frequency threshold (dl). If musical education increases sensitivity, the question arises if listeners of a language that uses F0 for prosody are more sensitive to musical tones and vowels’ dl thresholds. A perception experiment was conducted examining the dl of musical tones and vowels for both professional musicians and nonmusicians. The experiment was carried out cross linguistically: The languages include German and English (Germanic languages) and Italian, Portuguese, and Spanish (Romance languages). Results indicate that dl is significantly smaller for musicians compared to nonmusicians for all examined languages. The native vowel dl is significantly smaller for the Germanic languages. This is attributed to the more frequent use of F0 for prosody, thus claiming linguistic reasons. Surprisingly the dl for the musical tone in all languages is smaller for the Germanic musicians compared to the Romance musicians, although musical education and the stimulus were identical for all musically educated musicians. These results indicate an influence of the native language on the pitch accuracy of the listeners. Thus we give evidence that interference of speech pitch and musical pitch processing occurs, which supports the ‘‘common processing theory.’’
Lexical stress without postlexical head-marking: Evidence from Tagalog
Authors: Anderson Victoria B.
Published: October 2013

Jun (2005) compares prosodic prominence types in 21 languages representing diverse families (including Australian, Chinese, Germanic, Muskogean, Romance, and Semitic, but none from Austronesian). Jun observes an implicational relationship between the prominence type a language uses at the lexical level (i.e., lexical tone, stress or pitch accent) and the prominence type it uses postlexically (i.e., head-marking or edge-marking). Fourteen languages in the sample employ lexical stress; all use postlexical head-marking (i.e., postlexical pitch accent on a phrases most prominent syllable), and four additionally mark a phrases’ edge(s). No stress language in the sample employs postlexical edge-marking alone. However, Tagalog may be such a language. Tagalog uses lexical stress (LAyas ‘‘leave;’’ laYAS ‘‘carefree’’), but arguably does not use head-marking. This study examined map-task data from four native speakers of Tagalog. Syllables bearing main stress in a phrase, and occurring between zero and three syllables from the end of the phrase, were analyzed for pitch, intensity, and duration. Results show that although syllables with primary stress have increased intensity, and duration, pitch prominences do not reliably align with these syllables. Instead, low pitch reliably appears on the first syllable in an accentual phrase, while high pitch reliably appears on the last syllable.
Prosodic correlates to meaning in infant-directed speech
Authors: Herold Debora S. , Namy Laura L. , Nygaard Lynne C.
Published: October 2013

The present investigation examined the extent to which speakers produce reliable prosodic correlates to meaning across a range of semantic domains and whether parents use these kinds of cues to teach word meanings to children. In study 1, acoustic properties including fundamental frequency (F0), F0 variation, duration, and amplitude of infant-directed utterances were examined. Speakers were asked to produce question phrases in which novel words were used to convey one of two meanings from a set of antonym pairs (e.g., big/small, yummy/yucky). Herold (2006) has shown that adult listeners can identify word meanings from the prosodic information contained in these utterances. Acoustic analyses revealed that each word meaning had a unique acoustic signature, and that related word meanings such as big and tall and small and short had similar acoustic profiles. Study 2 explored whether naturalistic parental speech to young children recruits similar acoustic properties in the service of providing prosodic cues to meaning. Results demonstrated that mothers systematically varied prosodic cues as a function of word meaning when producing speech to their children. These findings suggest that speech directed to infants contains reliable prosodic markers to word meaning and that mothers may use prosodic cues to differentiate word meanings in word learning contexts.
Influence of voice prosody on preferential behaviors
Authors: Hasui Naoki , Takeuchi Yugo
Published: October 2013

In mother-infant interactions, the mother talks in a special voice that is characterized by elevated fundamental frequency (pitch) and exaggerated intonation contours, and the infant turns a face toward the voice and smiles naturally. This case suggests that humans react automatically to the special prosody without understanding of the lexical meaning in the word. We propose a hypothesis that adults react automatically as in the case of the infant by not only a difference of prosody but also the situation that one is in. This study aims to figure out the function of prosody on voice cognition and utilize these findings for constructing the voice user interface systems with a low cognitive load for users. In this paper, using the virtual maze on a display, we gave the participants a task that they should escape as quickly as possible from the maze. In the situation where the participants choose the route on T-shaped intersection, we gave the participants two leading voices of different prosody, and observed the preferential behavior for the two voices. The experimental results indicate that the voice of the voice user interface systems synthesizing the high pitch or the high rate of speech is naturally accepted and can direct the users.
Utilization of acoustic cues in identifying speech intonation contrasts in cochlear implant recipients and listeners with normal hearing
Authors: Peng Shu-Chen
Published: October 2013

This study investigated the utilization of acoustic cues in identifying speech intonation contrasts in adult cochlear implant (CI) recipients and normal-hearing (NH) individuals listening to spectrally and temporally degraded stimuli. Fundamental frequency, intensity, and duration patterns of a disyllabic word, popcorn, were manipulated orthogonally, resulting in 360 resynthesized stimuli. In a two-alternative forced-choice task, CI and NH participants identified whether each stimulus was question-like or statement-like. Each NH listener also identified stimuli that were noise vocoded. Preliminary results from seven CI and four NH listeners indicated that (a) weighting of fundamental frequency in the identification of CI listeners was less pronounced than that of NH listeners, (b) unlike NH listeners who showed little reliance upon intensity or duration patterns in identifying unprocessed stimuli, CI users demonstrated systematic dependence upon intensity and duration cues in identifying the same set of stimuli, and (c) with noise-vocoded (spectrally and temporally degraded) stimuli, NH listeners’ identification exhibited weighting patterns for intensity and duration cues in a way similar to that of CI listeners. Implications for the processing of multidimensional acoustic cues for speech intonation in normal and electrical hearing will be discussed. [Work supported by NIDCD-R01DC04786.]
Median plane localization using a parametric model of the head-related transfer function based on spectral cues
Authors: Iida Kazuhiro , Itoh Motokuni , Itagaki Atsue , Morimoto Masayuki
Published: October 2013

The present study has two purposes. One is to examine a simulation method for localizing a sound image. The authors propose a parametric HRTF model to simulate vertical sound localization. The parametric HRTF is recomposed only of the spectral peaks and notches extracted from the measured HRTF, and the spectral peaks and notches are expressed parametrically with center frequency, level, and sharpness. The other purpose is to clarify the contribution of each spectral peak and notch as a spectral cue. Localization tests were carried out in the upper median plane using the subjects’ own measured HRTFs and the parametric HRTFs with various combinations of spectral peaks and notches. The results show that the parametric HRTF recomposed using the first and second notches (N1 and N2) and the first peak (P1) provides almost the same localization accuracy as the measured HRTFs. Observations of the spectral peaks and notches indicate that N1 and N2 change remarkably as the source elevation changes, whereas P1 does not depend on the source elevation. In conclusion, N1 and N2 can be regarded as spectral cues, and the hearing system could utilize P1 as the reference information to analyze N1 and N2 in ear-input signals.
Evaluation of head-related transfer function on the opposite side to a sound source
Authors: Nishino Takanori , Oda Takayuki , Takeda Kazuya
Published: October 2013

A head-related transfer function (HRTF) is a complex acoustic transfer function because it includes the effect of reflection and refraction due to the head and ears. The HRTFs on the opposite side to a sound source have more complex characteristics and lower power than HRTFs that are on the same side as a sound source. Moreover, many previous studies about monaural sound localization showed that performances of sound localization on the opposite side were not precise. From these results, it can be assumed that the HRTFs on the opposite side are not necessary to use listener’s own. In our experiments, several stimuli that convolved with the HRTFs were presented to evaluate individuality. The left ear HRTF was the subject’s own, and the right ear HRTF was the subject’s own and others. The differences between the subject’s HRTF and others’ HRTFs were evaluated using the spectral distortion score. The sound source azimuths were from 0 to 180 deg at intervals of 30 deg; the front was 0 deg, the left side was 90 deg, and the back was 180 deg. The results shows that the right ear HRTFs between 30 and 150 deg azimuth have low individualities.
Classification head-related transfer functions for an individualization procedure by a tournament-style listening-test
Authors: Iwaya Yukio , Suzuki Yôiti
Published: October 2013

Individualization of head-related transfer functions (HRTFs) is necessary for high-accuracy sound image localization in three-dimensional spatial rendering by virtual auditory displays (VADs). However, measurement of HRTFs for each listener for all sound-source directions requires a special measuring apparatus and a long measurement time with a listener’s physical load. The authors have therefore proposed an individualization method of HRTFs called Determination method of OptimuM Impulse-response by Sound Orientation (DOMISO), in which the best set of HRTFs is selected using a tournament-style listening test without any measurement. In DOMISO, our HRTF corpus, including more than 120 sets of measured HRTFs, is used. In this study, a classification method of these HRIRs is examined to select good seeds for the tournament. First, relationships among various parameters of HRTFs and individualization results by DOMISO are analyzed using multiple-correlation analysis. Second, using promising parameters, all sets of HRTFs in the corpus are classified as a structured tree. Third, using results of classification, listening tests by DOMISO procedure are performed using these HTRFs. Finally, a listening test of sound image localization is conducted with the individualized HRTFs to examine the effectiveness of the classification.
Capture and rendering of spatial sound over headphones
Authors: Duraiswami Ramani , Zotkin Dmitry N. , O’Donovan Adam
Published: October 2013

A theory for capturing an audio scene and then rendering it remotely over headphones is developed. The method relies on capture of a sound field up to a certain order in terms of spherical wave functions. Then, the captured sound field is convolved with the head-related transfer function and rendered to provide presence in the auditory scene. The sound-field representation is then transmitted to a remote location for immediate rendering or stored for later use. A system that implements the capture using a spherical array is developed and tested. Head-related transfer functions are measured using the system described in [D.N. Zotkin et al., J. Acoustic. Soc. Am. (to appear)]. The sound renderer, coupled with the head tracker, reconstructs the acoustic field using individualized head-related transfer functions to preserve the perceptual spatial structure of the audio scene. [Work partially supported by VA.]
Acoustic rendering of a virtual environment based on virtual microphone control and binaural room scanning
Authors: Braasch Jonas , Martens William L. , Woszczyk Wieslaw
Published: October 2013

Binaural room scanning (BRS) is a convolution technique that utilizes a set of spatially indexed impulse responses that are selected in response to listener head movements during virtual acoustic rendering, such that the direction of sonic elements is automatically updated according to the angle determined by a head tracker. Since the room impulse responses have to be measured for only a few loudspeaker positions, a very successful application of BRS is the simulation of a control room. In order to create a flexible headphone-based virtual environment, it is proposed to simulate the input signals for the BRS system using virtual microphone control (ViMiC). ViMiC renders a virtual environment by simulating the output signals of virtual microphones that can be used to address a surround loudspeaker system. The advantages of this approach are twofold. First, the measured impulse responses of the BRS system ensure high spatial density in overall reflection patterns, avoiding the typical gaps in between the early reflections that occur when using solely mirror images or ray tracing. Second, the resulting imaging is closer to common audio engineering practice, where a pleasant but plausible environment is favored over a more purely realistic rendering.
A comparison of listener loudspeaker preference ratings based on in situ versus auralized presentations of the loudspeakers
Authors: Olive Sean , Welti Todd , Martens William
Published: October 2013

Auralization methods have several practical and methodological advantages for studying the perception of loudspeaker reproduction in rooms. For subjective measurements of loudspeakers, the auralization should be capable of eliciting the same sound quality ratings as those measured using an original acoustic presentation of a loudspeaker. An experiment was designed to test whether this is possible. Listeners gave preference ratings for both acoustic (in situ) and auralized double-blind presentations of four different loudspeakers, and the results were compared. The auralized presentations were generated from binaural room-scanned impulse responses of the loudspeakers convolved with the music test signals and presented over headphones equipped with a head-tracking device. For both in situ and auralized methods nine trained listeners gave preference ratings for four different loudspeakers using four different programs with one repeat (eight trials in total). The results show that the auralized and in situ presentations generally produced similar loudspeaker preference ratings.
Equalization of a one-dimensional sound field with a movable source
Authors: Chen Xi , Leishman Timothy W. , Sommerfeldt Scott D. , Faber Benjamin M.
Published: October 2013

An experimental plane-wave tube with a movable side-branch loudspeaker was used to study simultaneous source and sound field equalization. The effects of equalization and inverse filtering were evaluated for different measured field quantities. Parametric equalization filters were designed based on acoustic pressure and total energy density at a point within the tube. An adaptive filter based on total energy density and the filtered-x algorithm was also implemented. The results of the equalization methods are compared and their advantages and disadvantages are explained.
Numerical calculation method of binaural signals suitable for auralization of a reflective sound field
Authors: Otani Makoto , Ise Shiro
Published: October 2013

Based on the Kirchhoff-Helmholtz integral equation, a numerical calculation method, which enables a wave-theory-based auralization of a sound field inside room, is proposed. Binaural signals are synthesized from pressures and particle velocities on the boundary surface and corresponding head-related transfer functions (HRTF). The proposed method can shorten the extra computation time that is required because of changes in numerical conditions caused by head movement and rotation. The boundary element method (BEM) was used to calculate surface values on sound-reflective objects. In addition, HRTFs for various source positions were calculated using a BEM-based HRTF calculation system [M. Otani and S. Ise, J. Acoust. Soc. Am. 119, 2589–2598 (2006)]. Results obtained using the proposed method are compared with those of the conventional BEM, whose boundary consists of a head and a wall, and an image source method, which does not incorporate sound wave motion. Those comparisons reveal that the proposed method can accurately simulate binaural signals, including indirect sound from reflective objects. In addition, interactions between the head and wall are discussed, which are omitted in any auralization method using HRTFs to synthesize binaural signals and have not been investigated.
A new digital module for variable acoustics and wave field synthesis
Authors: de Vries Diemer , van Dorp Schuitman Jasper , van den Heuvel At
Published: October 2013

In the 1980s, wave field synthesis (WFS) was proposed by Berkhout [J. Audio Eng. Soc., 36, pp. 977–995 (1988)] as a new and powerful concept for spatial reproduction of sound fields without sweet-spot-limitations. Based on this concept, with implementation of simplifications not deteriorating the perceptual quality, a system for variable acoustics was developed: acoustic control system (ACS). The system has been widely applied as a solution for multipurpose halls. Until now, ACS processors have been built using analog electronic technology. Recently, however, a digital module has been designed that not only enables us to generate the ACS reflection patterns, but also to perform true WFS, e.g., for direct sound reinforcement with preservation of natural spatial properties. In the paper, design and specifications of the module will be discussed. Also, the potentials of integrated application of WFS and variable acoustics will be explained.
Audio noise reduction utilizing inharmonic-frequency analysis of generalized harmonic analysis (GHA)
Authors: Muraoka Teruo , Ifukube Thoru
Published: October 2013

Audio noise reduction is a desired technology for better telecommunications. Currently, the technology is based mostly upon spectral subtraction methods. However, its fundamental problem of musical noise generation remains unsolved. Musical noise is produced by mismatching of instantaneous phases of presampled noise and target noise to be canceled, which causes noise-breathing effects synchronous with the object’s intensity variation. Additionally, unnecessary frequency components inherent to Fourier analysis to resynthesis process are mixed in the processed output. For this reason, spectral subtraction is inapplicable to high-fidelity audio. The authors have devoted attention to inharmonic frequency analysis of generalized harmonic analysis (GHA), which was originally proposed by Wiener. The analyses were intended to examine stochastic signals utilizing harmonic frequency analysis. Results clarified that any signal can be represented as an almost periodic function whose frequency components have an inharmonic relationship. In 1993, Dr. Hirata proposed inharmonic frequency analysis based upon successive matching with frequency component templates, which is applicable to high-fidelity audio signal processing. This method came to be called GHA, following Wiener’s theory. The authors mostly attempted noise reduction in damaged SP records of historical value, retrieving almost noise-free musical sounds.
Observations of seafloor ripples and a time-dependent ripple geometry model
Authors: Traykovski Peter
Published: October 2013

Recent seafloor acoustic scattering experiments and theoretical developments have shown that the presence of ripples on the seafloor can increase the amount of subcritical angle acoustic penetration into the seafloor. A ripple observation and acoustic scattering experiment (SAX04) was recently conducted near Fort Walton Beach, Florida. During this experiment a sidescan sonar on an REMUS AUV was used to observe ripple geometry from the nearshore (5-m depth) to the shelf break (50-m depth). The ripples were found to increase in wavelength from 20 cm in the nearshore to 1.2 m at the shelf break. The offshore ripples were a product of Hurricane Ivan, which had passed the site 2 weeks prior. Time series of ripple geometry, measured with a tripod-mounted rotary sidescan sonar and a two-axis rotary pencil beam sonar, from a coarse sand site at the Martha’s Vineyard Coastal Observatory, also revealed long-wavelength (0.6 to 1.2 m) relict ripples after wave events. Ripple geometry models that assume the ripples are in equilibrium with hydrodynamic forcing underpredict the wavelength of the relict ripples at both sites; thus a new model that allows for a time delay between ripple adjustment and forcing was developed. [Work supported by ONR.]
Biodegradation of wave-formed sand ripples during SAX04 (Sediment Acoustics Experiment 2004)
Authors: Hay Alex E.
Published: October 2013

The gradual decay of 50- to 70-cm-wavelength sand ripples over a 20-day period of continuous in situ observations is presented. The ripples formed midway through the SAX04 experiment during Tropical Storm Matthew, and the observation period began immediately following this event. Bed elevation profiles and planform images of the ripple field were obtained at centimeter-scale resolution using rotary sonars, and at millimeter-scale resolution using a laser-video bed profiling system. Low-frequency currents, wave orbital velocities, and near-bed turbulence were monitored simultaneously with colocated instruments. The observed rates of change of the principal components of the bed elevation spectrum and of the spatially integrated backscatter intensity differences are presented and compared to predictions made with a diffusivity-based model of the ripple decay process. The model-data comparisons yield values for the horizontal sediment diffusivity which are consistent with previous measurements using particle-counting techniques.
Velocity dispersion in water-saturated granular marine sediments
Authors: Kimura Masao
Published: October 2013

The velocity dispersion in water-saturated granular marine sediments is an important subject that has remained unresolved in the community of sediment acoustics. Reported velocity dispersion results cannot be explained using the Biot model with the constant frame bulk modulus. The author showed that the values of the frame bulk modulus of water-saturated glass beads and beach sands are approximately 109 Pa at a frequency of 500 kHz and about ten times greater than those used typically in the community. Moreover, the author proposed a modified gap stiffness model incorporated into the Biot model (BIMGS model). Using this model, the author demonstrated that the frame bulk modulus has frequency dependence [M. Kimura, J. Acoust. Soc. Am. 1205 (2006)]. In this study, the BIMGS model is established using the measured results of the longitudinal wave velocities in glass beads saturated with alcohol-water mixed liquids. Then the velocity dispersion in glass beads and beach sands with different grain sizes are analyzed using the measured results of the longitudinal wave velocity at a much higher frequency and in the region of relaxation frequency. Finally, it is shown that the reported velocity dispersion can be explained using the BIMGS model.
Sound waves and shear waves in saturated unconsolidated sediments
Authors: Buckingham Michael J.
Published: October 2013

A recent, continuum-mechanics theory of compressional and shear wave propagation in saturated unconsolidated sediments relies on grain-to-grain interactions as the sole source of internal stress in the porous medium. On the basis of a statistical argument, nonlinear microscopic processes occurring at grain boundaries are incorporated into the intrinsically linear grain-shearing theory. The result is two sets of algebraic dispersion relationships, one of which represents a compressional wave and the other a shear wave. The shear wave is predicted to exist even in the absence of an elastic ‘‘frame.’’ For both types of wave, the theory predicts logarithmic dispersion and a near-linear dependence of attenuation on frequency. The grain-shearing dispersion relations, which are functions not only of frequency but also of the geophysical parameters of the sediment (porosity, bulk density, grain size, and overburden pressure), show excellent agreement with multiple data sets that have appeared in the literature over the past half century. Since the physical mechanisms behind the grain-shearing theory and Biot’s classical theory of wave propagation in porous media are quite different, some comments will be offered in an attempt to clarify the circumstances in which the respective theoretical approaches might be valid. [Research supported by ONR.]
A comparison of two observations of compressional wave velocity dispersion in marine sediments
Authors: Holland Charles W. , Turgut Altan , Dettmer Jan , Dosso Stan
Published: October 2013

Compressional velocity dispersion exists in all marine sediments. If the dispersion is great enough, it may play a significant role in acoustic interaction with the seabed. On the other hand, if dispersion is weak, seabed models and databases can be substantially simpler. The ocean acoustics community is divided on this issue, in part because of the lack of observations. Here we compare two recent methods for measuring dispersion. One technique measures the broadband reflection coefficient (frequency dependence of the critical angle) and thus infers the velocity dispersion. The other technique measures time of flight in an in situ probe over a broadband of frequencies. Measurements on New Jersey Shelf (STRATAFORM area) are compared and implications for the frequency dependence of the compressional wave attenuation are discussed. [Work supported by the Office of Naval Research and NATO Undersea Research Centre.]
Low-frequency laboratory measurements of sound speed in water-saturated granular sediments
Authors: Argo Theodore F. , Wilson Preston S.
Published: October 2013

Accurate knowledge of the acoustic behavior of water-saturated granular sediments is required for effective sonar operation in certain environments. Acoustic dispersion at frequencies below about 20 kHz has been the topic of much recent study and debate. While the understanding of real ocean-bottom sediments in their full complexity is the ultimate goal, laboratory measurements are significantly less difficult and subject to less uncertainty. This plays an important role in sediment acoustics since the difference between the predictions of competing propagation models is only a few percent. A new laboratory impedance tube apparatus, which operates in the 0.5–10-kHz frequency range, has been developed to address this issue. The device can be operated as a traditional acoustical impedance tube, in which water-borne sound waves interact with the water-saturated sediment. The device can also be operated as a driving-point impedance instrument, in which the mechanical input impedance of a column of water-saturated sediment is investigated. The effective-fluid sound speed is then inferred from the impedance measurements. Preliminary measurements of the sound speed in reconstituted natural sand sediments and artificial granular sediments obtained with the new apparatus will be presented and compared to existing theory. [Work supported by ONR.]
In situ acoustic scattering, attenuation, and dispersion measurements in marine sediments
Authors: Turgut Altan , Gauss Roger , Holand Charles , Zurk Lisa
Published: October 2013

The degree of acoustic velocity dispersion and nonlinear frequency dependency of attenuation in various types of sediments is investigated. A down-looking chirp sonar and a wideband acoustic probe system are used to measure compressional wave speed and attenuation within 2–150-kHz frequency band. Measurements indicate slight velocity dispersion and nonlinear frequency dependency of attenuation in silty and sandy sediments that can be effectively modeled by an extended Biot theory (Yamamoto and Turgut, J. Acoust. Soc. Am. 83, 1744–1751 (1988)]. The degree of acoustic scattering due to volume inhomogeneities and surface roughness is also estimated from the incoherent components of the measured signals. Acoustic scattering results are favorably compared with those predicted by pseudospectral FDTD simulations that used colocated vibracore data as input. Robustness of the attenuation and dispersion measurements by using chirp sonar and acoustic probe systems is also verified by the FDTD simulations [Work supported by the ONR.]
Weak consolidation model for wave propagation in ocean sediments
Authors: Berryman James
Published: October 2013

Within the context of Biot’s theory of poroelasticity, both effective density approximations and grain-to-grain elastic and inelastic effects are incoporated into a complete theory of acoustic propagation through ocean-bottom sands. The sedimentary material is assumed to be at best weakly consolidated. A variety of possible attenuation mechanisms are considered in order to explain ultrasonic data for both laboratory studies on glass beads saturated with water and silicon oil, as well as in situ data from the SAX99 experiment. An effective medium model is introduced to treat the weakly consolidated frame constants. Wave speeds and attenuations are explained to first order by the effective density approximation alone, but the remaining small discrepancies in velocity can be modeled successfully using the consolidation constants. Remaining attenuation discrepancies can then be fit using several methods including (a) increase of viscosity in fluid trapped between grains, (b) increase of viscosity due to high frequencies alone, and also (c) a new approach that specifically takes dynamic—but small amplitude—grain rearrangements into account. None of these mechanisms was taken into account in Biot’s original low-frequency theory.
Dispersion and attenuation due to scattering from heterogeneities in the porosity of sand sediments
Authors: Hefner Brian T. , Jackson Darrell R. , Calantoni Joseph , Reed Allen H.
Published: October 2013

Close packed granular materials are inherently heterogeneous at small length scales due to the random packing of the grains. These heterogenieties should manifest themselves in the frame moduli and the porosity of the medium. For sound propagation through fluid-saturated granular materials, the small scale of the heterogenieties may lead to scattering from the fast compressional wave into the slow compressional wave, increasing the attenuation of sound in the medium. This mechanism may account for the increase in high-frequency attenuation relative to expectations based upon Biot theory observed in recent measurements in sand and glass-bead sediments. Previously, we presented the results of a perturbation theory of sound propagation that accounted for losses due to scattering from heterogenieties in the bulk frame modulus [B. T. Hefner et al., J. Acoust. Soc. Am. 119, 3447 (2006)] and showed that the losses due to this mechanism were not enough to account for the measured attenuations. The theory has since been expanded to account for heterogenieties in the porosity. This theory relies on knowledge of the spatial statistics of the porosity and efforts are underway to measure these spatial variations in both simulations of glass beads and through CT scans of sand sediments. [Work supported by ONR.]
Measurements of the frequency dependence of sound speed and attenuation in a sandy sediment
Authors: Hefner Brian T. , Jackson Darrell R. , Williams Kevin L. , Thorsos Eric I.
Published: October 2013

Among the systems deployed during the sediment acoustics experiments, SAX99 and SAX04, were a fixed buried hydrophone array and a diver-deployed mobile array. The buried hydrophone array was used to measure the sound penetration into the sediment from a mobile source tower over a frequency range of 11–50 kHz during SAX99. For very steep incident grazing angles well above the critical angle, the penetrating field is only weakly influenced by the surface roughness and can be used to determine the sound speed and attenuation in the sediment. For SAX04, the frequency range was extended such that the new range covered 2–50 kHz. The diver-deployed array consisted of two sources and two receivers, which could be inserted into the sediment to a depth of 10 cm to measure sound speed and attenuation from 100–260 kHz during SAX99. For SAX04, the frequency range was increased to 40–260 kHz such that the measurements overlapped those made using the buried array. The sound speeds measured during both experiments using these systems are consistent with the predictions of Biot theory while the attenuation follows a linear frequency dependence at high frequencies. [Work supported by ONR.]
Modification of time and frequency dimensions of speech
Authors: Flanagan James L.
Published: October 2013

Vocal and auditory mechanisms impose constraints on speech signals. These constraints are central to techniques for coding and bandwidth compression, and hence to efficient transmission and storage. A key representation is the short-time Fourier transform, describing the signal in terms of two variables—time and frequency. This separation affords opportunity, within limits, to scale these information-descriptive parameters. This report recollects efforts (incentivized by bandwidth conservation) to modify speech in time and frequency. The ‘‘Analytic Rooter’’ [Schroeder et al., Proc. IEEE 55, 396–401 (1967)] divides bandwidth by a factor of 2, keeping temporal characateristics intact. The ‘‘Phase Vocoder’’ [J. Flanagan and R. Golden, Bell Syst. Tech. J. 45, 1493–1509 (1966)] allows nonrational manipulations in both time and frequency. And, ‘‘Silence Detection,’’ based upon automatic step-size adjustment in adaptive differential PCM coding [Flanagan et al., IEEE Trans. Commun. COM-30, 336–345 (1982)], permits speed variations with constant bandwidth. Recordings illustrate the effects.
Early linear predictive speech coding at Bell Telephone Laboratories and Electrical Communication Laboratory
Authors: Itakura Fumitada
Published: October 2013

At the 6th ICA in Tokyo (1968), two papers on the linear predictive coding of speech were presented by Manfred Schroeder and by me in the same session. The first paper was on the adaptive linear predictive waveform coding for medium band coding (time domain modeling) and the latter was on the LPC vocoder based on the autoregressive spectral estimation for narrow-band coding (frequency domain modeling). Although their approaches were very different in both concept and scope, there were dramatic common points: the covariance and autocorrelation methods were used to solve normal equation to determine the LPC parameters and also the pitch period was extracted to reproduce speech periodicity in both systems. Subsequently the multipulse (1982), stochastic (1984), and code (1985) excitation LP were proposed at AT&T BTL, whereas the transformation of LPC parameters to PARCOR (1969) and LSP (1975) and their efficient quantization were intensively studied at ECL of NTT. These efforts were eventually combined for practical implementations in the secure, cellular, and IP telephone in the 1980s. The brilliant ideas by Manfred Schroeder and his associates at BTL, in particular Bishnu Atal, have always been a helpful lighthouse to navigate the world-wide speech coding research in the right direction since 1980s.
The roles of models in speech communication research
Authors: Fujisaki Hiroya
Published: October 2013

The use of models is a key to development of science and technology. Only through modeling can one test various ideas, hypotheses, and theories against what one observes in nature. This paper first gives a systematic review of models that have been developed in the past on various aspects of speech communication, both between humans and between human and machine. These include speech production models, speech acoustics models, speech perception models, as well as language models and dialogue models. It will then describe briefly some of the models developed by the current author. Finally, it will discuss those aspects that have not been modeled so far, and present the author’s prospects on how to model them. This paper is dedicated to Manfred Schroeder, whose distinguished contributions in many areas of acoustics were enlightening to a large number of fellow researchers.
New directions in room acoustics: Probabilities, computer simulation, number theory
Authors: Kuttruff Heinrich
Published: October 2013

This paper highlights just a few of M. R. Schroeder’s major contributions to room acoustics. In his first investigations Schroeder presented a new approach to the frequency response of rooms by recognizing its random character, at least in the range above a critical frequency limit (today known as the Schroeder frequency). Soon afterwards he began employing the then-new digital computer as a research tool. Together with B. S. Atal he simulated the sound transmission in rooms, and in 1961 both authors demonstrated what nowadays is known as auralization and cancellation of acoustical cross-talk between two loudspeakers. In 1967 they published probably the first paper on ray tracing. Furthermore, Schroeder contributed to measuring techniques, for instance by introducing the integrated impulse response method of recording sound decay. Another astounding achievement is Schroeder’s invention of pseudorandom surface structures based upon number-theoretical schemes (Schroeder diffusers). Such devices may be used to scatter sound waves into all directions and hence to improve the acoustics of concert halls by creating lateral sound reflections. By his unconventional ideas and numerous innovations Manfred R. Schroeder has changed our understanding of sound transmission in rooms and initiated many new research activities.
Acoustic contribution to travel comfort in airplanes— insight delivered from room acoustics
Authors: Mellert Volker
Published: October 2013

Travel comfort in airplanes is affected by numerous parameters from which the acoustic environment provides a major contribution, as revealed in measurements of various symptoms of passenger and crew in long-haul flights. Tools known from room acoustics are useful to calculate and estimate an optimal acoustic design of the cabin interior, e.g., background noise levels must match the demand for privacy, or an appropriate STI ensures good public address and communication between the passenger and crew member. Many measures assume a statistical description of the acoustic field. But the strong frequency-dependant absorption leads to different Schroeder frequencies fs, which indicate the transition from well-separated resonances to many overlapping normal modes. In consideration of the extended noise sources as boundary conditions it is useful to determine with fs the regimes of analytical solutions of the wave field within the cabin from a more costly numerical (FEM/BEM) approach.
Recent discoveries and continuing mysteries in the auditory systems of bony fishes
Authors: Edds Walton Peggy L.
Published: October 2013

Recent research has revealed aspects of the auditory system that reflect the diversity of bony fishes as a group. This presentation will review some of the more exciting finds. For example, the long-standing classification of the hair-cell orientation patterns in saccules may require revision given the diversity of hair-cell orientations documented in closely related fishes recently. Hearing studies have shown that, although auditory specialist fishes (otophysines, with auditory accessories for detection of the pressure component of a sound wave) hear higher frequencies than auditory generalists (those that respond primarily to the particle motion component of a sound wave), work on the toadfish has shown that sensitivity at their best frequencies is not as different as once believed. However, a new kind of auditory specialist has emerged within the herrings. Some herring species respond to ultrasonic frequencies (\?20 kHz), apparently via a specialized region of the utricle. The utricle may be the rising star of the fish auditory periphery as physiological and anatomical-projection studies in nonherring species indicate that the endorgan may serve both the vestibular and the auditory systems. Emerging insights into auditory processing among fishes will be discussed. [Research supported by the NIH.]
Seasonal variation in the frequency response of the ear: Implications for communication
Authors: Sisneros Joseph
Published: October 2013

Seasonal variation in vocal-acoustic behavior is well known among nonmammalian vertebrates; however, studies of the neural mechanisms underlying such seasonal variation have focused mainly on the anatomical rather than the neurophysiological correlates of seasonal reproductive periodicity. In this talk, I present the neurophysiological evidence for seasonal variation in the frequency response of the ear in the plainfin midshipman fish (Porichthys notatus). Vocal communication is essential to the reproductive success of the seasonally breeding midshipman fish, which migrates during the late spring and summer from deep ocean sites into the shallow intertidal zone to breed. Female midshipman fish use the auditory sense to detect and locate males that are singing from their nests. The seasonal onset of male mate calling in the midshipman fish coincides with a dramatic increase in the range of frequency sensitivity of the female’s saccule, the midshipman’s main organ of hearing, thus leading to increased encoding of the male’s mate call. I will present evidence for a steroid-dependent mechanism that leads to this form of auditory plasticity and discuss why this plasticity of peripheral auditory frequency sensitivity may represent an adaptation to enhance the acquisition of auditory information needed for mate identification and localization during the breeding season.
Sound production and hearing ability in the Hawaiian sergeant fish
Authors: Maruska Karen P. , Boyle Kelly S. , Dewan Laura R. , Tricas Timothy C.
Published: October 2013

Sounds provide important signals for inter- and intraspecific communication in fishes, but few studies examine behavioral contexts of sound production and hearing ability within a single species. This study characterized the sounds and their behavioral contexts in a wild population of Hawaiian sergeant damselfishes (Abudefduf abdominalis), and compared their features to hearing sensitivity measured by both auditory-evoked potentials (AEP) and single-cell recordings in the auditory midbrain. The sergeant fish produces low intensity (∼85–115 dB rms re: 1 μPa), low-frequency (∼100–400 Hz) sounds during nest preparation, courtship an agonistic interactions. Both AEP and single-cell recordings show that fish are most sensitive to these frequencies (80–300 Hz) with best frequency at 100 Hz. However, auditory thresholds to tonal stimuli were approximately 20 dB lower for single-cell recordings than those measured by AEP experiments. Further, midbrain auditory neurons were on average more sensitive to playbacks of conspecific sounds than to pure tones. These experiments show that the hearing ability of this damselfish closely matches the intensity and spectral characteristics of sounds produced in the wild. Multidimensional studies that incorporate analyses of and responses to biologically relevant natural sounds are necessary to understand acoustic processing and the evolution of acoustic communication in fishes.
Acoustico-lateralis communication in coral reef butterflyfishes
Authors: Tricas Timothy C. , Boyle Kelly S.
Published: October 2013

Butterflyfishes are conspicuous members of coral reefs and well known for their pairing behavior and visual displays during social interactions. Members of the genus Chaetodon have a unique arrangement of the anterior swim bladder horns that connect with the lateral line (the laterophysic connection) and also project towards the inner ear, but functions for this putative acoustico-lateralis adaptation were not known. Field experiments demonstrate that the monogamous multiband butterflyfish, C. multicinctus, produces hydrodynamic pulses and sounds during territorial defense, and pulse trains that may be an alert call to the mate. In field choice tests, individuals could distinguish mates from nonmates and produced different calls to each, thus demonstrating the existence of mate recognition and context-specific sound production. Laboratory experiments show that sound production differs among sexes, hydrodynamic pulses are of low frequency (<100 Hz), and sounds have peak energy from 100–500 Hz. Auditory evoked potential experiments demonstrate that the auditory system has best sensitivity from 200–600 Hz and that the swim bladder horns provide a 10–20-dB gain within this band. These social sounds produced by Chaetodon can provide relevant stimuli to the auditory, vestibular, lateral line and possibly the laterophysic systems.