bluuebunny/crossref_metadata_2025 huggingface dataset:
Automatic detection of vowel nasalization using knowledge-based acoustic parameters
Authors: Pruthi Tarun , Espy-Wilson Carol Y.
Published: October 2013

Researchers in the past have suggested several acoustic correlates of nasalization including extra pole-zero pairs near the first formant (F1), a reduction in F1 amplitude, and an increase in F1 bandwidth. Even though these correlates have been known for a long time, considerable work is still needed to automate the extraction of acoustic parameters (APs) for nasalization. This work looked at 37 different APs which were pared down to 8 APs based on F statistic obtained by ANOVA. In preliminary experiments, an accuracy of 69.79% has been obtained for the task of discriminating between oral and nasalized vowels on the TIMIT database using a support vector Machine (SVM)-based classifier. The classification was done on a frame basis, and a segment was declared nasalized if more than 30% of the frames were found to be nasalized. Note that all vowels adjacent to nasal consonants were assumed to be nasalized. Thus, the accuracy may actually be higher since some vowels before nasal consonants may not be nasalized. Further, these results were obtained by using a linear kernel in SVMs. We hope the results would improve when a radial basis function kernel is used. [Work supported by Honda and NSF Grant BCS0236707.]
A novel method for the measurement of vowel formant frequencies and its application to the analysis of Japanese vowels over a wide range of ages
Authors: Kasuya Hideki , Ohta Kanae , Mori Hiroki , Deguchi Toshisada , Ohyama Ghen , Muraishi Yukimasa
Published: October 2013

The first three formant frequencies of the Japanese vowels, measured over a wide range of ages of talkers by one of the authors in 1968, have widely been cited in many speech textbooks and research papers. We decided to update those old data, measured by sound spectrograms, by using a more advanced method. We developed an iterative method for measuring vowel formant frequencies based on an autoregressive with exogenous input (ARX) speech production model that incorporates both voice source and vocal-tract models [J. Acoust. Soc. Jpn. 58, 386–397(2002)]. In using this method to measure vowel formant frequencies of high-pitched voice, the procedure for determining the initial value of the open quotient (OQ) of the source model was carefully investigated, since it affects significantly the estimation of the lower first formant frequency. Estimated OQs of the vowel /a/ of a talker were found to be stable against various initial values and so were used for analyzing the other vowels of the same talker. Perceptual confirmation of phonetic quality of resynthesized vowels with the fundamental frequency perturbed was also of great importance for accurate measurement. This procedure was successfully applied to 320 male and female talkers, aged 6–24 years.
An online customizable music retrieval system with a spoken dialogue interface
Authors: Hara Sunao , Miyajima Chiyomi , Itou Katsunobu , Takeda Kazuya
Published: October 2013

In this paper, we introduce a spoken language interface for music information retrieval. In response to voice commands, the system searches for a song through an internet music shop or a ‘‘playlist’’ stored in the local PC; the system then plays it. To cope with the almost unlimited size of the vocabulary, a remote server program with which a user can customize their recognition grammar and dictionary is implemented. When a user selects favorite artists, the server program automatically generates a minimal set of recognition grammars and a dictionary. The system then sends them to the interface program. Therefore, on average, the vocabulary is less than 1000 words for each user. To perform a field test of the system, we implemented a speech collection capability, whereby speech utterances are compressed in free lossless audio codec (FLAC) format and are sent back to the server program with dialogue logs. Currently, the system is available to the public for experimental use. More than 100 users are involved in field testing. In our presentation, we will report details of the system and the results of field tests, which include motorcycle environments. a)Currently at Graduate School of Computer and Information Sciences, Hosei University, Tokyo 184-8584, Japan.
A speech communication environment using open source software library for active sound image control
Authors: Kitashima Yuichiro , Kondo Kazuhiro , Nakagawa Kiyoshi
Published: October 2013

This paper describes a three-dimensional (3-D) conference system using an open source software library on conventional PCs. We will attempt to use both 3-D graphics and audio to construct a virtual conference environment for effective communication between remote parties. A rough prototype system was developed using OpenGL and OpenAL. The system uses local files for voice output, whose image location is rendered according to the user input. We initially evaluated the perceived sound image location accuracy of the rendered sound image using the prototype. Users were asked to identify the location of the rendered sound image from among four choices: front, back, left, and right. The users were able to identify the left and right images correctly at virtually 100%, but the front and back identification were lower than 10% for some sounds, particularly male speech. We plan to implement audio-streaming functions to achieve real-time audio conferencing and evaluate the benefits of 3-D audio for conferences. We would also like to implement HRTF (head-related transfer function) and RTF (room transfer function) for improved audio image localization, especially to achieve accurate elevation perception.
Finite-difference modeling of scattering from shallow buried objects in sea floor
Authors: Moen Inger Helene , Dong Hefeng
Published: October 2013

An object buried in the sea floor positioned close to the water/sediment surface can, when exposed to an incident wave, be a source of interface wave generation. The amplitude of a surface wave tends to dominate at greater distances, and surface waves can thus be a viable tool for detecting shallow buried objects. In this paper, the 2-D velocity-stress finite-difference formulation of Virieux [Geophysics 51, 889–901 (1986)] is adopted to model the wave propagation from a point source located above the sea floor in an open ocean environment. The wave responses from an object situated below the surface are computed and analyzed, with respect to various object depths, properties, and sizes. A second consideration of the paper is the analysis of the artificial reflections from the model boundaries. The perfectly matched layer absorbing boundaries is a widely used technique that has been shown to be very efficient and has therefore been implemented in order to reduce the grid reflections. The numerical results show that the amplitude of the surface wave decreases with the depth of the object, and that interface waves can be of great importance for detection and identification of shallow buried objects in the sea floor.
Time-frequency analysis of backscattered echoes from absorbing spherical shells
Authors: Ou Hui , Wang Xudong , Syrmos Vassilis L. , Allen John S.
Published: October 2013

The backscattering of tone bursts by spherical shells immersed in water has been investigated by several researchers. For a very thin shell, the backscattering specular echo is followed by enhanced echoes. The mechanism of these enhanced echoes has been attributed to a subsonic surface guided wave from recent ray synthesis analysis [P. L. Marston and N. H. Sun, J. Acoust. Soc. Am. 96, 1862–1874 (1992)]. In this study, time-frequency analysis is performed on the scattering signal. A discrete Wigner distribution function is employed to express the information content in the scattered sound in relation to the size and material of the shell [G. C. Gaunaurd and H. C. Strifors, IEEE 84, 1231–1248 (1996)]. The simulation results support that the time delay between the echoes is determined by the outer radius of the shell as well as the shell material (with or without absorption), or the Rayleigh velocity of the shell material. Furthermore, the backscattered enhancement occurs at different ranges in frequency domain due to various shell materials, which provide a potential method to determine the shell material properties. The relationship of this work to target discrimination methods for sonar applications is highlighted. [Work supported by ONR.]
Mechanism for sea bottom reverberation in long range sonar in a range-dependent continental shelf environment
Authors: Galinde Ameya , Ratilal Purnima
Published: October 2013

The statistically stationary background reverberation in long-range sonar data acquired in the New Jersey Continental shelf during the ONR 2003 Main Acoustic Clutter Experiment is mainly due to seafloor scattering. We compare three mechanisms or models for scattering from the sea bottom in a range-dependent waveguide with the experimental data. They are the Rayleigh-Born volume scattering model, a rough surface scattering model, and an empirical Lambertian model. Each of these models has a different decay rate as a function of range and different frequency dependence. These models are compared to the measured background reverberation as a function of range from the sonar system and at various frequency bandwidths from 300 to 1500 Hz to determine which mechanism or model for seafloor scattering best describes the data.
‘‘Through-the-sensor’’ monostatic bottom loss and bottom backscatter measurements at mid-frequencies
Authors: Barlett Martin , Brown Walter
Published: October 2013

Determinations of bottom backscatter and bottom loss parameters for the seafloor at mid-frequencies are of interest for understanding the performance of modern tactical sonar systems. A variety of techniques exist for measuring bottom scattering properties, but many of these methods require significant resources and/or time to perform. Over the last several years, a through-the-sensor approach for making bottom backscatter measurements using standard tactical sonars has been developed. Recently, these techniques have been extended to monostatic bottom loss estimates. In this presentation, a description of the overall approach to these measurements, advantages and limitations of these techniques, and examples of through-the-sensor bottom backscatter and bottom loss measurements made from data collected through a tactical sonar system will be given.
Ocean bottom scattering: characterization with chirp sonar
Authors: Quijano Jorge , Zurk Lisa , Turgut Altan , Tang D. J.
Published: October 2013

Shallow water acoustic propagation in the mid-frequency range (∼1–10 kHz) is strongly affected by scattering from the ocean bottom. In general, this scattering is a combination of rough surface scattering from the water-sediment interface and sediment layers, as well as volume scattering from embedded inhomogeneities. It is necessary to develop an understanding of the spatial and temporal characteristics of the acoustic field to optimize Navy sonar performance. Although there has been a growth of research activity in this area, the validity of different modeling approaches still remains an open research question. One promising tool for bottom sediment characterization is a chirp sonar. These sonars can produce high-quality data in which time of arrival is often associated with location of subbottom sediment layers to produce sediment profiles. However, features of the acoustic scattering such as multiple scattering or particle resonances can have complicated signatures that are not straightforward to interpret. In this work, the effect of these signatures on the chirp sonar data is studied through application of a numerical FDTD model applied to a variety of ocean bottom geometries. Results obtained from the chirp sonar system are presented and compared with ocean bottom scattering models.
Using the ultrasound standing wave in vibroacoustography for surface roughness imaging applications
Authors: Mitri Farid G. , Fatemi Mostafa
Published: October 2013

The surface roughness (of about 20 microns) of an acrylic block immersed in water was measured using the acoustic emission pressure field (in the kHz range) induced by the dynamic radiation force of ultrasound (2 MHz). The dynamic radiation force is produced by mixing two beams of different frequencies to excite the probed object. The acoustic emission is a result of object vibration at the difference frequency of the incident ultrasound beams and is detected by a low-frequency hydrophone placed nearby the block. Due to the presence of the block, an ultrasound standing wave field is generated as a result of multiple reflections between the transducer and the block’s surface. The acoustic emission by the block’s surface varies according to its relative position to the transducer. This variation is used to form an image of the surface roughness at two difference frequencies Δf=9.9 and 22.4 kHz. To have a one-to-one map between the surface roughness and the image, the surface variation has to lie within the distance from a maximum to its nearest minimum in the standing wave. Images obtained here demonstrate that vibroacoustography may be used as a powerful tool for the nondestructive inspection and imaging of surface roughness [Mitri et al., Appl. Phys. Lett. 88, 234105 (2006)]. [Work supported by NIH.]
Tower-based breaking wave noise measurements
Authors: Means Steven L. , Schindall Jeffrey A.
Published: October 2013

A tower-based sea surface noise experiment began collecting data January 2006 in the shallow waters (25 m) approximately 75 km off the coast of Savannah, Georgia. A 32-phone, three nested-aperture, vertical hydrophone array was deployed nominally 100 m from a Navy tower that stands 50 m above the water surface. A high-resolution video camera was mounted near the top of the tower along with a dual-polarized marine-band radar to record the location, size, and lifetime of the surface expressions of the breaking waves above of the array. The array is cabled back to the tower for power and signal collection. The tower is microwave-linked to shore for internet-based control and data retrieval. To date, measurements have been made in wind speeds ranging from 5–21 m/s and wave heights of 1–3.4 m over a 6-month period. Empirical relationships between the time-frequency structures of the generated noise, obtained on the endfire beam of the array, and the size of the surface expressions of the breaking waves will be given for a range of environmental conditions. [Work supported by ONR base funding at NRL.]
Numerical simulation of noise generated by wave breaking in a tank
Authors: Chen Xuemei , Means Steven L. , Szymczak Bill G. , Rogers Joel C. W.
Published: October 2013

A coupled hydro-acoustic source model in relating the physical parameters of wave breaking to the source quantities has been developed. The physical processes of wave formation and breaking are modeled using a generalized hydrodynamics formulation with the initial wave profile calculated by boundary integral method. Hydrodynamic parameters, such as pressure variations and air cavity shapes, etc., will be provided through the simulation. In the acoustic simulation, an algorithm has been developed in handling wave propagation in irregular regions, such as the bubbly liquid generated by wave breaking. To validate the modeling, an experiment on wave formation, propagation, and breaking is carried out in a wave tank. The waves are generated mechanically with a computer controlled vertically oscillating wedge. Favorable agreement is found upon comparing prediction and measurement. [Work supported by ONR.]
Long-term trends in ocean noise
Authors: Reeder D. Benjamin , Pucan Rommel , Collins Curtis A.
Published: October 2013

The effect of ocean noise, particularly from anthropogenic sources such as Navy sonar systems, on marine mammal hearing, behavior, and reproduction is of great concern. However, relatively few studies exist to make a definitive conclusion regarding rising ambient noise levels, due largely to the dearth of information over sufficiently long time periods for a sufficient number of geographical locations. A few studies have estimated that ocean noise is rising at a rate of approximately 3–5 dB per decade at some frequencies in some locations. This study is based upon ambient noise measurements from U.S. Navy sonobuoys deployed by operational units between 1993 and 2004. Data and analysis are presented for three areas of the ocean—the Northwestern Atlantic Ocean, the Greenland-Iceland-United Kingdom (GIUK) gap, and the Mediterranean Sea. Analysis of this data demonstrates a less significant increase, and in some cases, a decrease, in ocean noise for these three ocean areas at frequencies between 50 and 2000 Hz. Ocean noise levels in this study are comparable to measured ocean noise levels reported in the literature during the 1960’s.
Vessel noise measurements underwater in the Haro Strait, WA
Authors: Veirs Scott , Veirs Val
Published: October 2013

In the recent listing of southern resident orcas as endangered under the Endangered Species Act, disturbance by vessels is cited as a risk factor for the population and a factor in the designation of critical habitat. To help determine whether vessel noise underwater is disturbing orcas as they use sound to navigate, forage, or communicate, the source level of common large vessels transiting Haro Strait was measured. Since large vessels are the dominant source of underwater sound for the southern residents (based on water volume ensonified and duration of exposure at a given receive level), it is convenient that all commercial vessels longer than 65 ft are now required to broadcast their location, speed, and other data in real-time. The automatic identification system (AIS) broadcasts data every 2 s via VHF that can be received and decoded with specialized radio receivers that are now coming on the market. AIS was used to log range, speed, and relative bearing while simultaneous recordings were made of the received sound pressure levels using calibrated hydrophones and photographs were taken. From these observations, the broadband and spectrum source level of each vessel was calculated as a function of bearing relative to vessel axis.
A transparency model and its applications for simulation of reflector arrays and sound transmission
Authors: Christensen Claus Lynge , Rindel Jens Holger
Published: October 2013

The paper describes a new method for simulating the frequency-dependent reflection and transmission of reflector arrays, and the frequency-dependent airborne sound insulation between rooms by means of a room acoustic computer model. The method makes use of a transparency method in the ray-tracing process. In the first step of the calculation the rays hitting the relevant surfaces may either be reflected or transmitted, using a probability of 50%. In the next step the impulse responses in the receiver positions are calculated using a frequency-dependent correction to account for the reflected or transmitted energy. The method applied for the reflector array is based on a theoretical model that takes into account the dimensions of the reflecting surface, path lengths, and angle of incidence. The transmission calculation is based on the users’ data for the frequency-dependent transmission loss of the partition, and this is useful for the auralization of sound transmission through different building constructions. The acoustic properties like volume, reverberation time, and the area of the transmitting surfaces are included in the simulation.
Automatic vocal individual recognition of acorn woodpecker (Melanerpes formicivorus) based on hidden Markov models
Authors: Yao Yuan , Lin Ying , Ali Andreas , Taylor Charles
Published: October 2013

The acorn woodpecker (Melanerpes formicivorus) is a highly social cooperatively breeding species. A variety of conspecific interactions inside and between family groups is mediated by vocalizations. In the current study, acoustic sensor networks are used to monitor vocal behaviors of acorn woodpecker. In order to identify the callers in our behavioral study, we present a method for the automatic vocal individual recognition using hidden Markov models (HMMs). Field recordings from eight woodpeckers in two family groups were made in northern California. Individual identification accuracy is 92.65% for isolated waka calls, 84.31% for isolated syllable wa, 73.04% for isolated syllable ka, using 7-state HMMs with MFCC—E—D parametrization. For the continuous recordings, the recognition rate is 66.67%–100.00%. We studied the influence of two major factors in the performance of the HMMs: (i) different structures of HMMs and parametrizations of data, (ii) different qualities of signals, including degraded calls after propagation and calls from two simultaneous sources split by using beamforming of sensor networks. These results suggest that HMMs, combined with sensor networks, are a promising tool for data collection in vocal behavioral studies of animals.
Effects of scattering layer composition, animal size, and numerical density on the frequency dependence of volume backscatter
Authors: Benoit-Bird Kelly
Published: October 2013

The mesopelagic boundary community around the Hawaiian Islands is a land-associated, sound scattering layer that undergoes diel migrations with both a vertical and horizontal component. A video camera system was developed to quantitatively examine the numerical density, size, and taxonomic composition of micronekton. The camera system was combined with a four-frequency vessel-mounted echosounder system (38, 70, 120, and 200 kHz) to document the full migration range of micronekton and describe the changes in composition and density throughout their diel migration. Migrating animals split into multiple, distinct layers at night with differences in micronekton density, composition, and size. These differences were correlated with differences in the frequency response of volume backscatter. The relationship between these variables and backscatter intensity relationship between frequencies is complex. The results suggest strong partitioning of habitat by these animals in space and time, which, along with the high densities of animals, indicate that competition is important in driving the behavior and structuring the community.
Probing the deep: Acoustic characterization of the Mid-Atlantic Ridge ecosystem
Authors: Horne John K. , Anderson Cairistiona I. H. , Godoe Olav Rune , Patel Ruben
Published: October 2013

Implementation of the ecosystem approach to resource management potentially increases the use of acoustics during assessment surveys. A Census of Marine Life survey on the mid-Atlantic ridge during summer 2004 was an opportunity to develop a practical approach. The Norwegian vessel, G. O. Sars, is an acoustically-quieted platform equipped with a five-frequency (18, 38, 70, 120, and 200 kHz) echosounder, acoustic Doppler current profilers (ADCPs), multibeam sonars, and a deep-towbody. Paucity of information on species composition limited the ability to conduct a traditional acoustic survey. Acoustic structure independent of biological sampling was quantified and then integrated acoustic density and target strength observations with biological community composition and length frequency data. Species were not assigned backscatter thresholds or water column regions at the onset of analysis. Near-real-time products monitored biomass distributions: daily echograms, target strengths, and frequency-differenced echograms. Persistent biological layers at different depths occurred in the water column. Layers or components migrated toward the surface during dark hours. Modes of target strength frequency distributions and trawl catch compositions differed among layers. The use of acoustic and net technologies can be integrated with near-real-time analytic results to quantitatively characterize pelagic ecosystems. [Work supported by ONR, IMR, and NOAA Fisheries.]
In situ microbubble generation from liquid precursor for diagnostic and therapeutic application
Authors: Kawabata Ken-ichi , Yoshizawa Akiko , Azuma Takashi , Yoshikawa Hideki , Watanabe Hiroko , Sasaki Kazuaki , Hirata Koji , Umemura Shin-ichiro
Published: October 2013

Stabilized microbubbles show high echogenicity and characteristic nonlinear acoustic responses and also work as ‘‘sensitizers’’ for HIFU therapy. Accordingly, if microbbles were selectively placed in targets inside the body, targeted diagnosis and therapy would be possible. However, unfortunately, microbubbles are too large to apply tissues other than in vessels. For tumor detection and therapy, we propose the use of a nano-sized liquid precursor of microbubbles, which is small enough to accumulate in tumor tissues and generate microbubbles with ultrasound pulses. We have developed emulsion-based precursors containing perfluoropentane and found that microbubbles can be produced with a 10 μ pulse of 2-MHz ultrasound at intensities of more than about 150 W/cm2 in water. The intensity threshold was controllable by the added nonvolatile fluorocarbons. The microbubble generation was also observed in mice tumors at the same ultrasound intensity range. It was also found that when microbubbles were generated in tumor tissues, CW ultrasound exposure resulted in tumor damage. Our results indicate the potential usefulness of our precursor for targeted ultrasound diagnosis and therapy. [Part of this work was supported by the Japanese Ministry of Education, Culture, Sports, Science and Technology through a grant-in-aid for the creation of innovations through business-academic public sector cooperation.]
Ultrasound-mediated nanoparticle drug delivery in an in vivo tumor model
Authors: Raju Balasundar I. , Hall Christopher S. , Hughes Miachael S. , Wickline Samuel A. , Lanza Gregory M.
Published: October 2013

One of the exciting possible future clinical uses of ultrasound is site-targeted drug delivery using focused acoustic energy. In an earlier study, it was shown that ultrasound exposure promoted lipid delivery from nanoparticles into melanoma cells in culture [Crowder et al., UMB 2005]. In order to extend the study to in vivo situations, a novel, combined ultrasound therapeutic and imaging system was developed. The system consisted of a therapeutic transducer (−6-dB BW 1.1 to 2.3 MHz; natural focus 8 cm; f-number 1) with a centrally located and coregistered imaging transducer connected to a commercial ultrasound imaging system. The system provided capability for arbitrary acoustic excitation, electronic axial steering, and mechanical translation of the therapy transducer as well as real-time acquisition of images. The device was tested on bilateral murine tumor models (MDA435). Tumor images provided real-time feedback for treatment planning. The animals were injected with perfluorooctyl bromide nanoparticles targeted to αvβ3 integrin and with a radioactive complex. Initial results showed that application of therapeutic ultrasound affected the delivery of the radioactive complex as measured by radioactivity 24 h postinjection. Experimental results and experiences will be presented detailing the importance of imaging for in vivo treatment guidance and planning.
Ultrasound-assisted penetration of rt-PA into clots to enhance thrombolysis
Authors: Datta Saurabh , Vaidya Sampada , Beiler Shauna , De Courten-Myer Gabrielle , Holland Christy K.
Published: October 2013

Enhanced rt-PA thrombolysis has been shown in the presence of pulsed ultrasound. Porcine clots were exposed to ultrasound (120 kHz, 80% duty cycle, 1667-Hz PRF, 30 min) and rt-PA at exposure levels exhibiting no cavitation (0.15 MPa), stable cavitation (SC) alone (0.24 MPa), and inertial and stable (IC+SC) combined (0.37 MPa). A separate group of clots was also exposed to rt-PA alone (sham). Percent clot mass loss after each treatment was used to determine thrombolysis efficacy. Cavitation activity was monitored throughout the exposure. After the treatment protocol, penetration of rt-PA and plasminogen was evaluated by immunofluorescene imaging. The highest percent mass loss was observed for SC exposure (26.8%) followed by the IC+SC exposure (19.8%) and the lowest for no cavitation exposure (14.7%). The penetration of rt-PA was distinctly higher for clots exposed to SC compared to unexposed clots. Enhancement of thrombolysis is related to presence of cavitation and SC exposure appears to enhance thrombolysis by facilitating more rt-PA penetration into the clot.
An analysis of note deviation manner in piano music
Authors: Toda Yukiko , Nakayama Yosuke , Okura Takuya , Yasuda Naoto , Okawa Shigeki
Published: October 2013

Human music performance usually contains ‘‘out-of-score’’ features such as players’ physical characteristics and emotional expressions besides the on-score information (notes, rests, and various marks). Although those features are not written on scores, they have a very important role to add an artistic sense to the music. As one of the ‘‘out-of-score’’ features, the deviation manner of each note between the score and the human performance is focused on. In this study, among various genres and forms of music, F. Chopin’s mazurkas (a form of Polish folk dance), which have triple time and the accent on the second or third beat, are used. By limiting the music genres, it becomes simplified to compare the tendency of players. For the experimental dataset (three pianists, 51 mazurkas), after extracting the fluctuations (time differences) from the regular position of each note, the Fourier transform is applied to the sequence. As a result, a significant difference among the players and the playing modes is obtained, which show the temporal and the rhythm patterns. As a future work, by using the patterns, it is possible to create a virtual performer who has/imitates a certain player or a certain playing mode.
A technical look at swing rhythm in music
Authors: Lindsay Kenneth A. , Nordquist Peter R.
Published: October 2013

We investigate swing rhythm in music using computer analysis techniques. Swing is not a genre of music, rather a style of performance. The same musical piece (data) can be played in swing or straight time. Musical notes and structure are identical in both performances, but the notes temporal patterns have slight, significant differences between straight and swing performances. We demonstrate a technical approach analyzing these differences, and show examples of several styles of swing: American Swing, Brasilian Samba, Jamaican Reggae. Prior research into swing rhythm discovered swing ratio representing the temporal variation of the music as performed compared to the music as written. Our work shows that in some cases swing ratio is inadequate to properly model the real patterns of temporal variation. We have extracted details showing swing exists hierarchically, i.e., the patterns of temporal variation played at one time scale may differ from the types of patterns of temporal variation played at a slower or faster time scale. We also show examples of different instruments playing swing differently from each other, locking together at certain canonical time locations that represent standard counting and subdivision of the musical meter. This latter phenomenon has been previously investigated as ensemble swing.
Hidden Markov model-based speech synthesis as a tool for constructing comunicative spoken dialog systems
Authors: Tokuda Keiichi
Published: October 2013

For constructing spoken dialog systems which can realize natural human-computer interaction, speech synthesis systems are required to have an ability to generate speech with arbitrary speaker’s voice and various speaking styles and/or emotional expressions. Although state-of-the-art speech synthesis systems based on unit selection and concatenation can generate natural-sounding speech, it is still difficult to synthesize various voices flexibly because they need a large-scale speech corpus for each voice. In recent years, a kind of corpus-based speech synthesis system based on hidden Markov models (HMMs) has been developed, which has the following features: (1) original speaker’s characteristics can easily be reproduced because all speech features, not only spectral parameters but also fundamental frequencies and durations, are modeled in a unified framework of HMM, and then generated from the trained HMMs themselves; (2) using a very small amount of adaptation speech data, voice characteristics can easily be modified by transforming HMM parameters by a speaker adaptation technique used in speech recognition systems. From these features, the HMM-based speech synthesis approach is expected to be used as a tool for constructing communicative spoken dialog systems: keeping such a viewpoint in mind, basic algorithms and techniques for HMM-based speech synthesis are reviewed.
Expressive speech synthesis using multilevel unit selection
Authors: Klabbers Esther , van Santen Jan
Published: October 2013

Generating natural sounding and meaningful prosody is a central challenge in text-to-speech synthesis, especially when generating expressive speech. Recently, we proposed a multilevel unit sequence synthesis approach, based on the general superpositional model of intonation, which describes a pitch contour as the sum of component curves that are each associated with different phonological levels, specifically the phoneme, foot, and phrase. During synthesis, segmental perturbation curves, accent curves, and phrase curves are extracted from the acoustic signal and are combined into target pitch curves; these target curves are then imposed on the acoustic unit sequences using standard pitch modification methods. This approach represents an attempt to combine the strengths of the two dominant approaches to speech synthesis: unit selection synthesis, which preserves all details of natural speech but struggles with coverage of the very large combinatorial space of phoneme sequences and prosodic contexts, and diphone synthesis, which addresses coverage by generating rule-based synthetic target prosody and imposing it on acoustic units using signal modification methods. This approach minimizes prosodic modification artifacts, optimizes the naturalness of the target pitch contour by using quasi-natural contours, yet avoids the combinatorial explosion of standard unit selection synthesis.
The focused acoustic fields time reversal experiment series
Authors: Stevenson Mark , Guerrini Piero , Akal Tuncay , Song Heechun , Sabra Karim , Roux Philippe , Hodgkiss William S. , Kuperman William A.
Published: October 2013

We review a progression of at-sea experiments that brought the concept of underwater, acoustic time-reversal (TR) focusing in the sea from an unverified hypothesis to reality. When the experiments began, it was completely unknown whether the time-varying nature of the oceanic environment would support sufficient temporal stability for the method to succeed, even at long (λ>3 m) acoustic wavelengths. A 10-year progression of experiments showed that not only was this possible, but that various applications could be implemented using higher frequency sound (up to 18 kHz), including phase coherent digital communications. Our most recent experiment, FAF-06, investigated multiuser applications for passive TR communications using small transmit and receive arrays. We demonstrated throughput of 10 kbps at several km range in a shallow-water propagation channel where the transfer function had an ISI span of several hundred symbols. [Work supported by NATO and ONR.]
Experiment of time-reversal communication in deep ocean
Authors: Shimura Takuya , Watanabe Yoshitaka , Ochi Hiroshi
Published: October 2013

The basic study on the long horizontal acoustic communication using time-reversal waves have been carried out by the authors, and the method of combining time reversal and adaptive filter was proposed to enhance the communication accuracy and to achieve the communication with a moving source-receiver. This time-reversal technique will be applied to the communication in the deep ocean with an autonomous underwater vehicle, ‘‘Urashima,’’ which is being developed in JAMSTEC. The experiment was executed to demonstrate the effectiveness of the proposed method in the area at the depth of 1000 m in Suruga-bay. In the experiment, the 500-Hz projectors were used, and the range between the focus and the time-reversal array (TRA) was 10 km. The measurement system of the TRA side was moored with the underwater winch to compose the virtual array by changing its depth. Unfortunately, the projector of the TRA side had a breakdown, so it was abandoned to execute active time-reversal communication. However, experiments of passive time-reversal communication could be achieved. It was verified that the proposed method makes it possible to assure the communication channel with the TRA that consists of 20 elements. [This work was supported by MEXT KAKENHI 16686050.]
Auralization of an orchestra using multichannel and multisource technique
Authors: Vigeant Michelle C. , Wang Lily M. , Rindel Jens Holger
Published: October 2013

Previous work has shown the importance of including source directivity in computer modeling for auralizations. A newer method to capture source directivity in auralizations is the multichannel technique, which uses multichannel anechoic recordings. In this study, five-channel anechoic recordings were obtained for every orchestral part of two symphonies at the Technical University of Denmark. Five-channel auralizations were then created for each instrument, located at its typical position on-stage in a concert hall, by convolving five impulse responses from sources that each represent approximately one-fifth of a sphere with the corresponding anechoic recording channel and mixing these together. The multichannel auralizations from each instrument were subsequently combined to produce a full multichannel auralization of the entire orchestra. The results from listening tests will be presented, illustrating the perceived changes in realism, source width and depth, and source separation between the output of this full orchestral mix of multichannel auralizations and other auralization methods, including the following: (i) a mix of single-channel auralizations from each instrument, and auralizations made with a single channel orchestral anechoic recording using (ii) a surface source and (iii) a single omni-directional source. [Work supported by the National Science Foundation.]
Prediction of scattering coefficients for use in room-acoustic simulation
Authors: Ahnert Wolfgang , Feistel Stefan , Bock Steffen
Published: October 2013

While absorption coefficients have become easily available and published worldwide, computer modeling of room acoustics suffers until today under the lack of data for scattering surfaces. As a result, in most cases rules of thumb have to be used. In this work a new approach of predicting scattering coefficients is introduced. Based on the geometrical parameters of the surface, the complex reflection properties are calculated utilizing an elementary-wave method. In this respect, computed data are compared afterwards with measurement data and its applicability for use in acoustic simulation is investigated.
Optimal dispersion/dissipation-error finite-difference method for room impulse response simulation
Authors: Naka Yusuke , Oberai Assad A. , Shinn-Cunningham Barbara G.
Published: October 2013

The finite difference method is suitable for the calculation of acoustic impulse responses because of its computational simplicity and efficiency. However, for room acoustics applications, where reverberation makes it necessary to calculate the room impulse response and acoustic energy over a long duration, most standard finite-difference formulations are not practical because of the accumulation of dispersion and dissipation errors. In order to calculate very long impulse responses, a new, efficient, high-order finite-difference method is developed in this study. The new method is optimized to minimize the total dispersion and dissipation errors and hence the error accumulated in each time step resulting from discretization in both space and time. The error of the new method is significantly reduced so that the impulse response can be calculated over the whole reverberation time (about 1 to 3 s), using a relatively coarse spatial (6 grid points per wavelength) and a moderate temporal (CFL=1.0) resolutions. [Work supported by AFOSR.]
Temporal decay of broadband sound fields in enclosures using a boundary element method
Authors: Bliss Donald , Manning Ben , Rouse Jerry
Published: October 2013

A method to predict time-dependent behavior of sound fields in enclosures with diffuse reflection boundaries is described. A boundary element formulation utilizes broadband energy-intensity boundary sources, including propagation time delays. Temporal behavior is expressed in terms of a higher-dimensional eigenvalue/eigenmode problem, with source strength distributions expressed as eigenmodes. A temporal interpolation scheme is used to reexpress the actual delays as a set of integer-multiple delays. The interpolation scheme produces a discrete set of primary eigenvalues and eigenvectors, the total number of which is an integer multiple M of the number of panels N. The value of M is set by the number of interpolation subdivisions of the enclosure wave transit time. For decay from steady-state conditions, the initial value problem involves specifying the initial panel strengths, and M−1 time derivatives of each panel strength set to zero. In temporal decay from steady state, solutions exhibit rapid short-time spatial redistribution of energy, followed by long-time decay of a predominant spatial eigenmode. Long-time decay depends on the behavior of the most slowly decaying eigenmode and the relative source-panel strengths do not depend on initial conditions. Short-time adjustment and decay depends on initial source characteristics and the distribution of wall absorption.
Boundary element prediction of steady-state broadband interior sound fields using absorption-based scaling
Authors: Bliss Donald , Rouse Jerry
Published: October 2013

Energy-intensity boundary elements are used in combination with the method of absorption-based scaling to predict steady-state broadband sound fields in enclosures with diffuse reflection boundaries. The wall absorption is expressed in terms of an overall absorption parameter—the spatially averaged value—and spatial variations around this mean. Boundary element strengths are expressed in a power series of the overall absorption, treated as a small parameter, thereby giving a separate problem at each order. The first problem has a uniform mean-square pressure proportional to the reciprocal of the absorption parameter. The second problem gives mean-square pressure and intensity distributions that are independent of the absorption parameter and are primarily responsible for the spatial variation of the reverberant field. This problem depends on the location of sources and the spatial distribution of absorption, but not overall absorption. Higher order problems proceed in powers of the absorption parameter, as corrections to the primary spatial variation. Boundary element solutions are obtained for each order, and the simplest model for spatial variation gives fairly accurate predictions. The method gives physical understanding of the causes of the spatial variation of enclosure sound fields and provides specific insights into the behavior and design of acoustic spaces.
Comparison of sound fields in rooms computed by finite-element method and by cone beam method
Authors: Okamoto Noriko , Otsuru Toru , Tomiku Reiji , Okuzono Takeshi , Isobe Kouta , Furuya Hiroshi
Published: October 2013

Sound fields in realistic rooms were simulated by different kinds of computational methods, and comparisons were conducted. One is based on the wave acoustics and the other is based on the geometrical acoustics, i.e., the finite-element method (FEM) and the conical beam method (CBM). As for the software applied here, the authors have developed their own FEM code, namely LsFE-SFA, to conduct FEM computations, and they employed commercial software, i.e. raynoise, to conduct CBM computations. First, theoretical basis of LsFE-SFA was given briefly. Next, sound-pressure distributions in a reverberation room with the volume of 165 cubic meters were computed by both methods to compare the agreement with measured sound-pressure levels. The boundary conditions due to the sound absorption were changed to examine the relation between the absorbent condition and the agreements. The agreement between FEM and measurement were found to be better throughout the examination. Next, both FEM and CBM were applied onto the analysis of large-scale sound fields in a multipurpose hall with the volume of 37000 cubic meters to discuss the agreement and difference in detail.
Numerical analysis for attenuation due to sound absorption in small and large closed spaces using finite-element method
Authors: Yamaguchi Takao , Kanazawa Junichi , Akamatsu Katsuji
Published: October 2013

A finite element for media having sound absorption is derived and a code is developed. For this element, an unknown parameter is pressure, and both effective density and bulk modulus of the media have complex quantity to represent damped sound fields. This model for the media having sound absorption was adopted by H. Utsuno’s research group for their boundary element formulation [Utsuno et al., AIAA J. 28(11), 1870–1875 (1990)]. Firstly, a three-dimensional sound field in a small closed space (306×255×204 mm3), containing a sound-absorbing material, is calculated using the finite element method. Validity of the calculated response is verified by comparison with the corresponding experimental result carried by H. Utsuno. Secondly, a two-dimensional sound field in relatively large closed space (about 7×3 m2) is calculated varying damping of its medium. The shape of the large space is rectangle, and the rectangular space has a roof. Distribution of sound pressure is calculated when the velocity of the roof is given as white noise. Under the low damping condition for the medium, distribution of sound pressure is almost uniform like the diffused sound field. Under the high damping condition for the medium, sound pressure become larger when the observation point is closer to the roof.
Parallel computing on the sound fields by bread slicing method
Authors: Tokita Yasuhiro , Oikawa Yasuhiro , Yamasaki Yoshio
Published: October 2013

Numerical analyses by the explicit scheme of the finite difference method greatly reduce the number of known variables that are needed to solve an unknown variable in comparison with the implicit methods. This fact means that the amount of communications decreases among each node of the computer cluster on parallel computing. Generally, on the parallel computing of the distributed memory system, the communicating time occupies most of the total calculating time. Therefore, the reduction to the amount of the communication directly causes the reduction of the calculating time. The parallel computing by the explicit scheme of the finite difference method can enable the building up of high-performance algorithms. In order to reduce the amount of communication among each node, one has proposed a method to divide the field of numerical analyses. It is called the ‘‘bread slicing method.’’ With this method, the fields must be divided into pieces parallel to the section that has the least area. The results showed that the method in this study enabled the total amount of communication in the computer cluster to be reduced and the geometries of communication to be made simpler.