bluuebunny/crossref_metadata_2025 huggingface dataset:
Audio and acoustic design of the University of Sydney’s Indoor Environmental Quality Laboratory
Authors: Cabrera Densil , Crow Robert , Miranda Luis , de Dear Richard
Published: May 2013

The quality of indoor environments such as commercial offices is affected by many factors, including temperature, humidity, air movement, illumination, ambient sound, and room acoustics. In 2012, a new laboratory was established at the University of Sydney to examine how such factors affect human occupants. In terms of sound, the design of the laboratory has three components: the acoustic design of the testing rooms; the audio system design (for introducing artificial soundscapes); and the design of generic soundscapes to support experimental work in the laboratory. Acoustic design considerations of the laboratory allow for the testing rooms to be configured as high grade office environments. The laboratory has a 24-channel audio system for introducing realistic and potentially complex sound fields in to the testing rooms, both from within and outside the rooms. Parametrically controlled soundscapes have been developed for interior sources (such as building services noise) and exterior sources (such as transport noise). This paper describes how the combination of the laboratory’s acoustics, audio systems, and soundscapes can be used for scientific studies of indoor environmental quality.
Acoustics and estuarine ecology: Using active and passive methods to survey the physical environment, vegetation, and animals in North Carolina’s coastal estuaries
Authors: Luczkovich Joseph J. , Sprague Mark W. , Krahforst Cecilia S. , Walsh John P. , Pleva Audrey J. , Carpenter Dean E.
Published: May 2013

Estuarine systems have complex interactions of physical and biological processes. Regular observations are needed in order to understand their dynamics. Acoustic observation systems (echosounders, acoustic Doppler current profilers (ADCPs), and passive acoustic dataloggers) can provide observations on a wide spectrum of processes in estuaries. We have used echosounders to monitor changes in bathymetry, submerged aquatic vegetation, fishes, and invertebrates over time. In addition, sediment changes, resupsension events, turbidity, and waves are monitored using ADCPs. The higher trophic level species of fishes and marine mammals that are soniferous have been monitored by passive acoustic methods. We provide examples of each acoustic method used to study the dynamics of seagrasses, fishes, and the physical environment of the Albemarle, Pamlico, Currituck, and Core Sounds in North Carolina. While it is possible to combine methods to use acoustics to measure the dynamics of estuarine systems (estuarine observing systems), the challenge we face is to ground-truth these acoustic metrics using traditional sampling methods (e.g., quadrats for plants, trawls for fishes, and water samples for sediments) and integrate each of these measures. We could then examine the effect of storms, waves, and resuspension events on estuarine plant and animal distributions and abundances using acoustics metrics.
Quantitative ultrasound tomography
Authors: Dongen Koen W.A. v. , Ozmen-Eryilmaz Neslihan
Published: May 2013

Whole breast ultrasound is gaining interest as a possible alternative to mammography, as it is cost effective, patient friendly, and avoids the usage of ionizing radiation. Due to the similarity in both measurement setups, scientists have investigated if tomographic reconstruction algorithms originally developed for x-ray tomography, such as inverse radon transforms, are also applicable to ultrasound tomography. However, the multiple scattering of an acoustic wave field inside the breast as well as diffraction and refraction effects results in a severe blurring of the obtained images. In order to overcome these limitations, people are developing imaging algorithms, which are based on the acoustic wave equations. To show the limitations and potentials of the various imaging algorithms, we computed synthetic data for a cancerous breast model using a full wave solution method. Next, we tested and compared different imaging algorithms varying from a ray based inverse radon transform up to a full-wave nonlinear inversion technique. The latter one has the advantage that, as we will show, it allows for accurate speed of sound profile reconstructions at the cost of a severe computational load.
Flawed transducer detection using random sample consensus for ultrasound tomography
Authors: Wang Tianren , Jing Yun
Published: May 2013

In this paper, we present a random sample consensus (RANSAC) based ultrasound travel-time tomography method. Conventionally, all the time-of-flight (TOF) data between each two transducers are used to estimate the sound speed distribution. However, failing to identify the inaccurate TOF data (outliers) due to flawed transducers would reduce the accuracy of the estimated sound speed distribution. In our proposed approach, a small subset of TOF data were first randomly selected from the original TOF data, and then applied to the tomography algorithm to estimate a rough sound speed distribution. The rest of the TOFs data was applied to the rough distribution, and the goodness of fit was calculated. If most of the data fitted well in the estimated distribution, then all the well-fitted data (including the subset) was used to estimate a final sound speed distribution. Otherwise, there were outliers expected in the subset, and a new subset of the TOFs data would be randomly selected again. This repeated until most of the data fitted well in the estimated distribution. Simulation results showed that our method could effectively detect and eliminate outliers and increase the accuracy of estimating sound speed distribution.
Calculation of thermoacoustic functions with computational fluid dynamics
Authors: Bühler Simon , Wilcox Douglas , Oosterhuis Joris P. , van der Meer Theo H.
Published: May 2013

Thermoacoustic functions are important parameters of one-dimensional codes used for the design of thermoacoustic engines. The thermal and viscous thermoacoustic functions allow the inclusion of three dimensional effects in one-dimensional codes. These functions are especially important in the regenerator of a thermoacoustic engine, where the thermoacoustic heat pumping occurs. Even though analytical solutions were derived for uniform pores, the thermoacoustic functions for complex geometries such as stacked screen or random fiber regenerators cannot be calculated analytically. In order to gain more insight into the geometry induced complex flow fields, the procedure of Udea et al. (2009) to estimate the thermoacoustic functions was applied in computational fluid-dynamic simulations. By using two measurement locations outside of the regenerator and modeling the regenerator as an array of uniform pores, it is possible to estimate the thermoacoustic functions for complex geometries. Furthermore, a correction method is proposed to quantify the entrance effects at the beginning and end of a regular pore. The simulations are first validated for a uniform cylindrical pore with the help of the analytical solution. Then the correction method is successfully applied to a cylindrical pore with the results closely matching the analytical solution. Finally, the method is applied to the model of a staggered, stacked screen regenerator.
Study on thermoacoustic system to drive by low temperature—Effects of loop-tube thermoacoustic system connected with parallel double stacks on the onset temperature ratio
Authors: Nakano Yosuke , Sakamoto Shin-ichi , Watanabe Yoshiaki
Published: May 2013

As the temperature ratio of the both ends of the stack increases gradually and reaches a critical value, sound waves begin to oscillate in the thermoacoustic system. This temperature ratio is called the onset temperature ratio. It is necessary to decrease the onset temperature ratio for practical use of the thermoacoustic system: the use of the factory exhaust heat and the solar heat. In a previous study, thermoacoustic system with series connected a number of prime movers was designed. This system can drive by lower onset temperature ratio than thermoacoustic system with a prime mover. However, it is considered that the heat loss increase when the heat is carried to a number of high heat exchangers. Therefore, loop-tube thermoacoustic system connected with parallel double stacks (parallel loop system) was proposed. This system can drive two prime movers by a heat input part because it is connected prime movers in parallel. In this report, the onset temperature ratio of this system was compared with that of normal loop-tube system with a prime mover. As a result, we confirmed that parallel loop system can drive by lower onset temperature ratio than normal loop-tube system.
The effect of resonance mode control by expanding of cross-section area on cooling capacity in a loop-tube type thermoacoustic cooling system
Authors: Inoue Manabu , Sakamoto Shin-ichi , Nakano Yosuke , Watanabe Yoshiaki
Published: May 2013

One of assignments for practical realization of a loop-tube type thermoacoustic system is improvement of cooling capacity. It is known that cooling capacity of the system is improved by setting phase adjuster (PA), which reduces cross-section of the system locally. This is considered that because setting PA enables to control resonance mode and setting PA controls to one wavelength. However, there is much dissipation of energy in PA because of reducing cross-section. More cooling capacity of the system is expected by control resonance mode at less dissipation of energy. Therefore, expanding phase adjuster (EPA) as a devise, which expands cross-section locally, is proposed to reduce dissipation of energy. If setting EPA enables to control resonance mode, cooling capacity is expected to improve more. At first, we make sure that setting EPA enables to control resonance mode. Next, we make a comparison between the system set EPA and PA. As a result, cooling capacity of the system with PA is higher than with EPA. However, more cooling capacity is expected by shift the condition of EPA because energy used on cooling in the system with EPA is more than with PA.
On discontinuity waves and vibrations in thermo-piezoelectric bodies
Authors: Montanaro Adriano
Published: May 2013

With regard to a body composed of a linear thermo-piezoelectric medium, referred to a natural configuration, we consider processes for it constituted by small displacements, thermal deviations, and small electric fields superposed to the natural state. We show that any discontinuity surface of order r greater than 1 for the above processes is characteristic for the linear thermo-piezoelectric partial differential equations. We show that discontinuity surfaces of order 0 generally are not characteristic; hence, the conditions are written, which characterize the discontinuity surfaces of order 0 that are characteristic. We find the ordinary differential equations of propagation for plane progressive waves and standing waves. Then we characterize the ones whose wavefronts are characteristic.
Eigenvalue shapes compared to forced oscillation patterns of guitars
Authors: Muenster Malte , Richter Jan , Bader Rolf
Published: May 2013

Thirty-two guitars are measured geometrically and acoustically. The geometries of the top plate with its bracing as well as its thickness, the back plate with all bracing, the ribs, and rims are transferred to a CAD model. The top plate and the back plate of these guitars are measured using a 121-microphone array, back-propagating the sound field onto the top and back plates. Therefore, the guitars are once driven by impulses at the guitar bridge, once by plucking all notes on all strings up to the 12th fret to reconstruct the forced-oscillation patterns. Large differences are found with respect to the basic modes between the different guitars in terms of frequency and shape of their eigenmodes. Comparing the measured and calculated eigenvalues with the forced-oscillation modes driven by the strings, it appears that the eigenmode shapes often differ from the forced-oscillation patterns considerably.
Acoustic imaging of string instrument soundboxes
Authors: Waltham Chris , Koster Evert , Smit-Anseeuw Nils , Zimmer Aaron
Published: May 2013

A circular 30-microphone array of 90 cm radius has been used to produce acoustic images of several string instruments. The soundboxes were excited by an automated impact hammer, and the instruments were suspended inside the array in such a manner that both the soundbox and hammer mechanism could be rotated in the horizontal plane. By normalizing all microphone signals to the hammer signal, data could be assembled as if from an array of many times 30 microphones. Images were formed using the inverse frequency response function method. The array and data analysis code were tested with a rectangular plate set in a large plywood baffle, a system that was straightforward to simulate numerically. Due to the limited spatial resolution set by the array geometry and frequency range—typically 200–1000 Hz—instruments with long soundboxes were chosen for initial testing: a gothic harp, a guzheng, and a guqin.
Attenuation characteristics of fit-compromised earmuffs and various non-standard hearing protectors
Authors: Wells Laurie , Berger Elliott H. , Keiper Ron W.
Published: May 2013

Excessive noise exposure can be successfully mitigated by proper use of legitimate hearing protection devices. However, real-life circumstances sometimes drive people to use compromised or alternative means of protection. This paper reports attenuation data measured in the 3M E•A•RCAL facility over several years, in conformance with ANSI real-ear attenuation at threshold test standards (S3.19-1974, S12.6-1984, and S12.6-2008 Method A) and also provides, for comparison, one dataset from the open literature (fingers/palms). The loss of attenuation was measured for various earmuffs worn in less than ideal conditions, including earmuffs worn in conjunction with various safety glasses, hairnets, head covers, hoods, earmuff cushion covers, and ball caps. Data were also obtained for non-standard means of blocking sound, including long hair, cotton balls, and even use of palms and/or fingers to block the ears. Results demonstrated that the effects on earmuff attenuation varied from none at all (suitable cushion cover) to as much as 12 dB (hooded sweatshirt). Realizing that people adapt hearing protectors to meet their needs is one step toward optimizing hearing protection selection and use; knowing the significance of these adaptations is the next step. [The authors are employees of 3M and the research was funded by 3M.]
Influence of the external ear tissue domains on the sound attenuation of an earplug predicted by a finite element model
Authors: Viallet Guilhem , Sgard Franck , Laville Frédéric
Published: May 2013

Earplugs are a widespread solution to prevent the problem of hearing loss in the workplace environment, but they do not always perform as desired. Using a model of the ear canal occluded by an earplug could be helpful to perform sensitivity analyses (geometry and materials of the earplug) and to better understand the role of the earplug. The human external ear is a complex system made up of different tissues with a 3D geometry. In practice, it is reduced to a 2D cylindrical geometry for the acoustical tests fixtures. The purpose of this study is to compare the insertion loss predicted by a 3D complex finite element model of the ear canal surrounded by different tissue domains (skin, soft tissue, and bone) and occluded by a silicon earplug versus a 2D axisymmetric model of the same system. In both models, some investigations are made in order to verify if the models could be simplified by replacing the tissue domains by mechanical impedances. These investigations are made to reduce the complexity of the models and to discuss the relevance of whether or not including external ear tissue domains in a sound attenuation model of an earplug.
Large volume flow rate acoustophoretic phase separator for oil water emulsion splitting
Authors: Dionne Jason P. , McCarthy Brian , Ross-Johnsrud Ben , Masi Louis , Lipkens Bart
Published: May 2013

Efficient separation technologies for multi-component liquid streams that eliminate waste and reduce energy consumption are needed. Current technologies suffer from high cost of energy, use of consumables, fouling, and limited separation efficiency of micron-sized particles. We propose a novel platform technology consisting of a large volume flow rate acoustophoretic phase separator based on ultrasonic standing waves. The acoustic resonator is designed to create a high intensity three dimensional ultrasonic standing wave resulting in an acoustic radiation force that is larger than the combined effects of fluid drag and buoyancy, and is therefore able to trap, i.e., hold stationary, the suspended phase. The action of the acoustic forces on the trapped particles results in concentration, agglomeration, and/or coalescence of particles and droplets. Heavier than water particles are separated through enhanced gravitational settling, and lighter particles through enhanced buoyancy. A first prototype consists of a 2 in. by 1 in. flow chamber driven by a single 1 in. by 1 in. transducer at 2 MHz, with flow rates of 30 L/h, and measured oil separation efficiencies in excess of 95%. A second prototype is designed to further scale the system to flow rates of 150 L/h. [Work supported by NSF SBIR 1215021 and NSF PFI:BIC 1237723.]
Direct numerical simulations of acoustic streaming in standing wave tubes using the Lattice Boltzmann Method
Authors: Rafat Yasser , Habibi Kaveh , Mongeau Luc
Published: May 2013

One important factor in the efficiency of thermoacoustic engines is acoustic streaming, which causes convective heat transfer between high and low temperature reservoirs. Most experimental and numerical studies performed so far have focused on Rayleigh streaming. Less work has been done on acoustic streaming due to the stack. Most numerical studies of Rayleigh streaming were performed using Navier-Stokes based numerical methods. In this study, large eddy numerical simulations were performed using schemes based on the lattice Boltzmann method (LBM). The model considered a simplified thermoacoustic refrigerator made of a rectangular standing wave resonator with a flat plate spoiler. Low-amplitude results obtained for Rayleigh streaming velocity magnitudes were compared with linear acoustic theory for verification. High amplitude recirculated streaming flow structures around the edges of the flat plate spoiler were identified. These are likely to contribute to heat transfer much more than Rayleigh streaming. Parametric studies were performed to investigate the effects of Strouhal number and spoiler edge shape. The results confirm that vertical edge streaming flows play a significant role in thermoacoustic heat transport.
An overview of Bertram Scharf’s research in France on loudness adaptation
Authors: Meunier Sabine
Published: May 2013

Since 1978, Professor Bertram Scharf divided his time between the United States and France. He was a Visiting Scientist at the Laboratoire de Mécanique et d'Acoustique in Marseille until the mid-1990s and collaborated with the University of Marseille (Faculté de Médecine) until his death. One of Bertram Scharf’s major contributions to the field of psychoacoustics is in the area of loudness. He first studied spectral loudness summation, when he started working at Harvard University. In France, his work on loudness focused mainly on loudness adaptation. He wrote, “Loudness resembles pain in that it decreases as a function of time only under special stimulus conditions.” Bertram Scharf's work with his French colleagues defined aspects of loudness adaptation in its direct (simple loudness adaptation) and indirect (induced loudness adaptation) forms. They studied how the auditory system recovers from loudness adaptation and examined a possible physiological basis for loudness adaptation.
Spectral loudness summation: From the 1960s to the present
Authors: Verhey Jesko L. , Hots Jan
Published: May 2013

In general, the loudness level of a broader sound is higher than that of a narrower sound centered at the same frequency, an effect commonly referred to as spectral loudness summation. From the late 1950s onwards, Scharf published several articles on this topic investigating how stimulus parameters such as the level, number, and spectral separation of components of a complex tone and the spectrum shape affects the magnitude of spectral loudness summation, and how spectral loudness summation is altered in hearing-impaired listeners and under masking. In a contribution to the proceedings of the first international symposium on hearing, Scharf also provided important information of the effect of duration on spectral loudness summation, stimulating our own research in this field. This talk will provide an overview of Scharf’s work on spectral loudness summation and how the view on this topic has changed over time. It will be shown that even today there are aspects of this effect that are not completely understood in the light of current loudness models.
Context effects in loudness
Authors: Epstein Michael , Florentine Mary
Published: May 2013

Bertram Scharf made contributions to numerous topics in the loudness literature. In particular, he brought a great deal of insight into the current understanding of contextual effects in loudness. Some of the contextual effects that he studied include: (1) loudness adaptation, the decline in loudness of the latter portion of a continuous sound, (2) induced loudness reduction, the phenomenon by which a preceding stronger tone reduces the loudness of a weaker tone, (3) temporary loudness shift, a decline in the loudness of weaker sounds due to a physical fatigue of the cochlear amplifier, and (4) loudness enhancement, in which a brief sound is made louder when it follows a stronger sound within a short duration. Context effects serve as complex reminders of the necessity of careful design of any psychoacoustical experiment in which level varies. These effects also result in the breakdown of all loudness models, as virtually all calculations of loudness are performed for sounds without regard for previous stimuli. [Work supported by NIH-NIDCD.]
On internal mean flow in porous absorbers and its effect on attenuation properties
Authors: Boij Susann , Färm Anna , Göransson Peter , Glav Ragnar
Published: May 2013

In vehicle applications, absorbing materials are often used to attenuate sound. In, for example, exhaust systems and on noise encapsulations, the absorber is exposed to flow. This creates a boundary layer above the absorber, which affects the impedance of the surface, and hence alters the absorption properties. In addition to this effect, the flow itself may enter the absorbent material due to high pressure and forced flow paths. An investigation of the effects that internal flow in the absorber imposes on the acoustic properties is presented. One way to describe the effect is by a change in flow resistivity. The effect is investigated for typical absorbers used in noise encapsulation for trucks. The Transfer Matrix Method is applied to calculate the resulting absorption and reflection coefficient for absorbers with changed flow resistivity in layers at the surface. The possibility to model the changed properties of the absorber with internal mean flow by means of Biot theory is also explored, together with a discussion on suitable experimental methods to verify and further investigate the effects.
Acoustic characterization of graded porous materials under the rigid frame approximation
Authors: Groby Jean-Philippe , Dazel Olivier , De Ryck Laurent , Khan Amir , Horoshenkov Kirill
Published: May 2013

Graded porous materials are of growing interest because of their ability to improve the impedance matching between air and material itself. Theoretical models have been developed to predict the acoustical properties of these media. Traditionally, graded materials have been manufactured by stacking a discrete number of homogeneous porous layers with different pore microstructure. More recently, a novel foaming process for the manufacturing of porous materials with continuous pore stratification has been developed. This paper reports on the application of the numerical procedure proposed by De Ryck to invert the parameters of the pore size distribution from the impedance tube measurements for materials with continuously stratified pore microstructure. Specifically, this reconstruction procedure has been successfully applied to retrieve the flow resistivity and tortuosity profiles of graded porous materials manufactured with the method proposed by Mahasaranon et al. In this work, the porosity and standard deviation in pore size are assumed constant and measured using methods, which are applied routinely for homogeneous materials characterization. The numerical method is based on the wave splitting together with the transmission Green's functions approach, yielding an analytical expression of the objective function in the least-square sense. The objective function is constructed to minimize the discrepancy between the predicted and measured reflection coefficient spectra.
Analysis criteria for forensic musicology
Authors: Begault Durand R. , Heise Heather D. , Peltier Christopher A.
Published: May 2013

Expert testimony for forensic musicology addresses a broad spectrum of legal issues, including the authentication and differentiation of published compositions and musical recordings, performance rights, and legal determinations regarding copyright infringement. While legal cases involving music and performance infringement date back as far as the 19th century, the field of forensic musicology has no stated methodology by which an objective forensic determination can be made. Expert opinions based merely on subjective impression or resulting from the “golden ear” syndrome are pseudo-scientific and not objectively based. This paper proposes scientific methods and recommendations for analysis based on stated criteria, with the goal of controlling examiner bias. Considerations include analyses of composition, performance, and acoustical features, and factors such as melody, harmony, rhythm, and orchestration; pitch, tone, vibrato, and embellishment; metadata analysis; recording technologies; and digital signal processing, including “effects.” By engaging in a series of structured categorizations, the forensic expert can establish a consistent, replicable, and objectively verifiable means of determining whether or not a recorded piece of music has been misappropriated.
Using a computational model for the auditory midbrain to explore the neural representation of vowels
Authors: Carney Laurel H. , Li Jiashu , Li Tianhao , McDonough Joyce M.
Published: May 2013

A formant-based approach to representing vowel quality is anchored in acoustic theory and is well documented in perception studies and in auditory modeling. This ongoing study investigates the representation of vowels in the responses of auditory models at the level of the midbrain (inferior colliculus). Previous modeling and physiological results have shown that formant structure is conveyed by changes in neural rates of midbrain cells that are tuned to amplitude modulations near voice pitch frequency. The current study examined model population responses to 20 speakers (10 males, 10 females) reciting 12 English vowel contrasts from the Hillenbrand et al. database [J. Acoust. Sci. Am. 97, 3099 (1995)]. Pairwise correlations across model population responses for each vowel were used to evaluate variability in the neural representations. Results show that the acoustical variability associated with the vowel contrasts is maintained in these neural representations. Thus, variability in the acoustic vowel space is maintained after the nonlinear responses of realistic auditory-nerve models and midbrain models for amplitude modulation tuning. Our goal is to extend our knowledge of the neural representation of the vowel space using a computational model for the responses of auditory neurons to ensembles of speech tokens.
Generative approach for robust acoustic model training for blindly separated speech recognition
Authors: Kitaoka Norihide , Dekiura Yuto , Takeda Kazuya
Published: May 2013

We propose a generative acoustic model training method for robust speech recognition with blind sound source separation as a front-end. Multiple microphone systems are often used for the separation. In such situation, separated speech is severely distorted and thus the recognition rate significantly drops. If we can measure transmission characteristics from the sound sources with various directions to the microphones, we can simulate to receive various mixed speech made by multiple speakers speaking with overlaps to each other. Then we separate the simulated overlapped speech using a blind source separation method such as frequency domain independent component analysis (FDICA) and use the separated speech to train HMM acoustic models to recognize such separated speech. Our method can generate such distorted speech enormously without recording the real speech spoken to the microphone system. We evaluate the models in the continuous Japanese speech recognition and show the effectiveness.
Suppression of clipping noise in observed speech based on spectral compensation with Gaussian mixture models and reference of clean speech
Authors: Hayakawa Makoto , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: May 2013

In recent years, the development of communication system allows people to easily record and distribute their speech. However, in the speech recording, clipping noise degrades sound quality when the level of input signal is excessive for the maximum range of amplifier. In this case, it is necessary to suppress clipping noise in the observed speech for improving its sound quality. Although a linear prediction method has been conventionally proposed for suppressing clipping noise, it has a problem with degradation of the restoration performance by cumulating error when the speech includes a large amount of clipping noise. This paper describes a method for suppression of clipping noise in observed speech based on spectral compensation. In this method, the power spectral envelope of speech on each frame in the lower frequency band is noise suppressed to by using Gaussian Mixture Models (GMM), and the one in the higher frequency band is restored by referring to the clean speech. We carried out evaluation experiments with a speech quality, and confirmed the effectiveness of the proposed method toward the speech, which includes a large amount of clipping noise.
Detection for Lombard speech with second-order mel-frequency cepstral coefficient and spectral envelope in beginning of talking-speech
Authors: Furoh Takayuki , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: May 2013

In noisy environments, the recorded speech is distorted by the additional noise and the Lombard effect. Thus, the automatic speech recognition (ASR) performance is degraded in noisy environments. To solve this problem, noise reduction methods have been proposed as the conventional study. However, in the conventional study, the improvement of ASR performance for the Lombard effect was not discussed well enough. In the present paper, we focus on the robustly detection for Lombard effect speech (Lombard speech). This is because the ASR system can employ a suitable acoustic model by detecting the Lombard speech. We previously proposed the detection for Lombard speech based on second-order MFCC and fundamental frequency. The previously proposed method however requires longer utterances to detect Lombard speech. We therefore newly propose the detection method for Lombard speech with second-order MFCC and spectral envelope in beginning of talking-speech. To detect the Lombard speech at a short time, the proposed method employs variable weights corresponding to elapsed time for second-order MFCC and spectral envelope. As a result of evaluation experiments, we confirmed that the detection time was reduced from the conventional method.
A Hidden Markov Model based speaker identification system using mobile phone database of North Atlantic Treaty Organization words
Authors: Agrawal Shyam S. , Bansal Shweta , Pandey Dipti
Published: May 2013

This paper describes results of an experiment to conduct text independent speaker identification of large number of speakers (about 100) using a standard vocabulary of about 23 NATO words—such as Alfa, Bravo, etc. These words in isolation were spoken in a sound treated room by Hindi natives having very good education in English (both males and females) and recorded by a three channel data recording system—the cardioid microphone, electret condenser microphone, and a NOKIA mobile telephone. The pre-processed digitized database of isolated words was further processed to determine 39 MFCC's and their derivatives and used to build an HMM model for each speaker based on all the words. The HMM model was trained using an HTK tool kit to generate the model parameters and tested using Viterbi algorithm. The identification of speakers was done in a closed set manner, based on comparison of each NATO word in the model. In addition to correct identification, false acceptance and false rejection scores were also found. The results show varying performance due to variations in channels, male/female speakers. The overall identification scores vary between 60% and 70%. The paper gives detailed analysis of results.
An investigation of vowel substitution rules in the automatic evaluation system of English pronunciation
Authors: Sato Kei , Kato Masaharu , Kosaka Tetsuo
Published: May 2013

We investigate the performance improvement of an automatic evaluation system of the English pronunciation of Japanese learners. In this system, Japanese and English acoustic models are used to detect mispronunciation at a phoneme level. Hidden Markov models (HMMs) are used as acoustic models. Mispronunciation is detected by comparing the output likelihoods of the two models. In order to improve the performance of this system, we investigate certain mispronunciation rules, which represent common mispronunciations among Japanese learners. We use four mispronunciation rules: vowel insertion (at the end of a word), vowel substitution, vowel insertion (between consonants), and consonant substitution. In this system, the accuracy of the mispronunciation rules is particularly important. The rules are determined on the basis of the knowledge of phonetics in our previous system. However, the effectiveness of the rules has not been analyzed quantitatively, and we do so in this work. A knockout procedure is used to select effective rules. By selecting effective rules, we found that the correlation coefficient between the subjective evaluation value and the system performance improved from 0.757 to 0.858.
A method to estimate a temporally stable spectral envelope for periodic signals
Authors: Morise Masanori , Yamashita Yoichi
Published: May 2013

Vocoder-based speech synthesis system requires fundamental frequency (F0) and spectral envelope. Since the sound quality of synthesized speech depends on the estimation performance, methods that can accurately estimate two parameters are crucial to synthesize natural speech. In particular, spectral envelope estimation is more difficult than F0 estimation. One of the problems in spectral envelope estimation is that the result depends on the windowing time, type of window function, and its length. Conventional methods such as LPC or Cepstrum cannot remove the temporal variation. TANDEM-STRAIGHT can estimate a temporally stable spectral envelope, whereas the processing that consists of averaging two power spectra, smoothing and post-processing is complex. To simplify the processing, we propose a method based on pitch synchronous analysis and spectral smoothing. The proposed method can estimate a temporally stable spectral envelope from only one power spectrum processed by a specialized window function. The window function and its length are determined to remove the temporal variation. The objective evaluation was conducted to verify the temporal variance and the estimation performance. The result suggested that the proposed method can estimate the temporally stable power spectrum as well as TANDEM-STRAIGHT.
Processing time improvement for speech enhancement based local projection using dynamic parameters
Authors: Phienphanich Phongphan , Tantibundhit Charturong , Onsuwan Chutamanee
Published: May 2013

Local projection (LP) has been widely used to enhance speech by transforming noisy speech into two orthogonal subspaces: noise (S1) and signal plus small amount of noise (S2) subspace. S1 is removed and S2 is transformed into time domain resulting in the enhanced speech. Satisfactory results with significantly improved speech quality have been reported by several works although the processing time was not taken into account. Four parameters to be considered are embedding dimension (d), time delay (τ), numbers of iteration, and minimal embedding dimension. Speech quality is increased by the increase of d-parameter, resulting in decrease of τ-parameter value (d×τ kept constant) and the dramatic increase of the processing time. The goal is to come up with the best d×τ parameter for each iteration, while speech quality remains almost unaffected. Rather than using a fixed d×τ parameter, a dynamic approach is taken. Specifically, τ is initially set to 1 and incremented by 1 for next iteration. The experimental results tested on Thai initial rhyming words corrupted by three noise types (white, car, and street) each at SNR levels of 10, 5, 0, −5 dB showed that the proposed method significantly reduced the processing time for white noise by 35% (p < 0.01).
Acoustic-to-articulatory inversion by analysis-by-synthesis using cepstral coefficients
Authors: Busset Julie , Laprie Yves
Published: May 2013

This paper deals with acoustic to articulatory inversion of speech by using an analysis by synthesis approach. We used old X-ray films of one speaker to (i) the develop a linear articulatory model presenting a small geometric mismatch with the subject’s vocal tract mid sagittal images (ii) and design an adaptation procedure of cepstral vectors used as input data. The adaptation exploits the bilinear transform to warp the frequency scale in order to compensate for deviation between synthetic and natural speech. This enables the comparison of natural speech against synthetic speech without using cepstral liftering. A codebook is used to represent the forward articulatory to acoustic mapping, and we designed a loose matching algorithm using spectral peaks to access it. This algorithm, based on dynamic programming, allows some peaks in either synthetic spectra (stored in the codebook) or natural spectra (to be inverted) to be omitted. Quadratic programming is used to improve the acoustic proximity near each good candidate found during codebook exploration. The inversion has been tested on speech signals corresponding to the X-ray films. It achieves a very good geometric precision of 1.5 mm over the whole tongue shape unlike similar works evaluating the error at 3 or 4 points corresponding to sensors located at the front of the tongue.
Hybrid immersive three-dimensional sound reproduction system with steerable parametric loudspeakers
Authors: Shi Chuang , Tan Ee-Leng , Gan Woon-Seng
Published: May 2013

A loudspeaker must be both dispersive and directive to accurately reproduce spatial audio from digital media. To address this problem, an audio system that has a unique combination of conventional and parametric loudspeakers has previously been proposed and proved to be effective to reproduce an immersive 3D soundscape. However, this system has two drawbacks: (1) There is only one fixed “sweet spot,” and (2) only one listener within the “sweet spot” can enjoy the complete experience. Therefore, a hybrid 3D sound reproduction system combining conventional loudspeakers with a pair of steerable parametric loudspeakers is proposed in this paper. By using this new combination of conventional and steerable parametric loudspeakers, the “sweet spot” can be steered toward the listener's head position. Thus, the listener no longer needs to keep his head stationary while watching movies or playing games, which resulting in a more relaxing and pleasant experience. Furthermore, a dual-beamsteering method is proposed for the parametric loudspeaker, which provides a flexible software-control solution to allow the 3D sound experience to be enjoyed by two listeners simultaneously. This paper provides the system overview and highlights the key processing techniques in rendering a “steerable” immersive 3D soundscape.
Extension of perceived source width using loudspeaker array
Authors: Choi Jung-Woo , Kim Yang-Hann
Published: May 2013

In this paper, a method to extend the perceived spatial width of a virtual sound source using multiple loudspeakers is proposed. Control of perceived source width or apparent source width (ASW) has been attempted by decreasing the inter-aural correlation. For this purpose, numerous decorrelators were proposed for stereo loudspeakers or headphones. However, these techniques are inadequate for sound field reproduction system incorporating multiple loudspeakers. For sound field reproduction, extension of source width has to be realized with three requirements. First, extension should not deteriorate the localization cue, provided by the reproduction system. Second, the coloration artifact, which induces by extra wavefronts other than the direct wave, should be minimized. Most importantly, the effect of source width extension has to be maintained over a large listening area. To design a spatial decorrelator that can meet these requirements, we design a proper target sound field with reduced inter-aural correlation over a zone of interest. The target sound field is reproduced by loudspeakers driven from a multipole expansion technique. The performance of the proposed method is verified by examining the inter-aural correlation coefficient (IACC) of the reproduced sound field over a wide area, as well as the ITD and ILD distributions.
The small-slope approximation for layered seabeds
Authors: Jackson Darrell
Published: May 2013

The small-slope approximation has found application to unlayered seabeds and is generally regarded as an improvement over methods that employ either small-roughness perturbation theory or the Kirchhoff approximation. Unfortunately, the usual small-slope ansatz fails when applied to layered seabeds, as it is inconsistent with perturbation theory. This ansatz is replaced by an alternative, which is found to satisfy the criteria of reciprocity and consistency with the perturbation and Kirchhoff approximations. This approach will be illustrated by computation of the coherent reflection coefficient and scattering strength for a seabed consisting of a single rough fluid layer over a semi-infinite, elastic basement with flat upper boundary. Computation time is significantly longer than for the unlayered case, increasing as the desired accuracy increases. The results will be contrasted with those obtained using a variety of existing approximations.
Energy exchange and scattering loss within a two-way coupled-mode formulation
Authors: Stotts Steven A. , Koch Robert A.
Published: May 2013

The loss of energy due to propagation in an ocean environment arises from several possible processes. In a two-way coupled-mode description scattering effects from a rough bottom Pekeris waveguide can be isolated by excluding bulk attenuation, and intermodal energy exchange can be examined. The range- and frequency-dependent interaction of a single trapped mode exchanging energy with multiple continuum modes is identified within this framework. The goal of the analysis is two-fold. First, it provides insight into additional loss mechanisms that can be incorporated into current local mode models (c.f. Koch and Stotts, A Derivation of Energy Loss via Coupled Modes, 165th Meeting of the Acoustical Society of America, June 2–7, 2013). Second, a comparison can be made to previous descriptions of additional attenuation, such as Kirchhoff scattering loss. Varying the bottom roughness permits tests of the applicability of the Born approximation.
Evidence for a common scale O(0.1) m that controls seabed scattering and reverberation in shallow water regions
Authors: Holland Charles W.
Published: May 2013

Analysis of the spectral content of long-range reverberation yields two observations. First, there is a remarkably similar scale, O(0.1)m, between three diverse continental shelf regions. This is surprising given general understanding of the complexity and diversity of geologic processes. Second, there is strong evidence that the scale is associated with heterogeneities within the sediment. Thus, sediment volume scattering, not interface scattering, controls long-range reverberation from a few hundred Hertz to several kilohertz. This is also unexpected given that at long-ranges the vertical grazing angles are less than the critical angle, and hence, the penetration of the acoustic field into the sub-bottom is expected to be modest. The consistency of the scale, O(0.1)m, suggests an underlying feature or mechanism that is consistent across many ostensibly diverse geological settings. Neither the feature nor mechanism is known at this time. Several hypotheses will be presented. [Work supported by ONR Ocean Acoustics.]
Enhanced spatial analysis of room acoustics using acoustic multiple-input multiple-output systems
Authors: Morgenstern Hai , Rafaely Boaz
Published: May 2013

Standard acoustic measurements in enclosures typically employ single-input single-output (SISO) acoustic systems. The parameters obtained from these measurements describe features of energy decay and do not characterize spatial attributes of the enclosure. Directional analysis of enclosures became popular with the introduction of microphone and loudspeaker arrays. In particular, spherical arrays have been shown to be highly beneficial for spatial analysis. Spherical microphone arrays facilitate the estimation of the arrival direction of the direct and reflected sound, while the use of both loudspeaker and microphones arrays can support the estimation of both radiation and arrival directions, with the application of conventional beamforming methods. However, when several reflections are attributed to the same time bin in a discrete impulse response, reflection paths may not be uniquely determined by existing beamforming techniques. We present a new method to uniquely determine source and receiver directions for multiple reflections when time separation is unfeasible. The paper presents the formulation of the proposed method, also showing a simulation study to demonstrate the performance of the proposed method.
Spatio-temporal energy measurements in renowned concert halls with a loudspeaker orchestra
Authors: Tervo Sakari , Pätynen Jukka , Lokki Tapio
Published: May 2013

Room acoustical parameters are commonly measured as spatial averages from a few source positions to several listening positions. In addition, ISO3382-1:2009 suggests that subjective listener aspect could be predicted with a few parameters at mid frequencies. It is obvious that such recommendation is inadequate to describe all perceptual differences between concert halls or seating positions. Explaining multidimensional subjective impressions is not possibly with a few averaged numbers. In our opinion, to compare accurately between concert halls, the measurement positions should be exactly at the same distance from the sources in each hall. And sources should be numerous to represent the typical real source on a wide area—a symphony orchestra. In this paper, we compare renowned European concert halls with spatio-temporal visualization of sound energy. Measurements at identical distances in halls enable an accurate comparison of sound energy distributions. The analysis of measured spatial impulse responses is performed with spatial decomposition method (SDM). The spatial sound energy distribution is presented at different frequency bands as a function of time to visualize the cumulative energy before and after 100 ms.