bluuebunny/crossref_metadata_2025 huggingface dataset:
End-point detection of speech using spectral transition for captioning system
Authors: Koga Ayako , Fujikashi Yuki , Arai Takayuki , Kanedera Noboru , Yoshii Junko
Published: October 2013

In recent years, captioning video contents with text translations is increasingly necessary because of the burgeoning use of media internationally, resulting from the rapid development of communication technology. In addition, within one language, video captioning is very important for hearing-impaired people. However, the process of captioning videos, including speech and nonspeech decisions, is often done manually by translators at present. Therefore, an efficient automatic end-point detection of speech for captioning video contents has been proposed. We attempted to detect speech end-points based on acoustic landmarks that identify times when acoustic changes are prominent in the speech signals [K. N. Stevens, Acoust. Phonetics (1998)]. In this study, landmarks were obtained by combining the mean square for the regression coefficients of logarithmic envelopes of 1/3-oct bands in time, which resembles the parameter proposed by Furui to measure spectral transition [S. Furui, J. Acoust. Soc. Am. 80(4), 1016–1025 (1986)], with other ones such as the logarithmic power of speech signals. An experiment was carried out using the proposed technique for speech detection. Results showed a high correct rate and introduced the possibility of its application to an efficient video captioning system. [Work partially supported by Open Research Center Project from MEXT.]
The calculation of sleep apnea syndorome’s condition index using the snore noise
Authors: Ono Takahiko , Yano HIroo , Oota Tatsuya , Takasaki Yuuji , Kurosaki Sachiko
Published: October 2013

The polysomnograph analysis is used to measure an apnea hypopnea index (AHI) of sleep apnea syndorome. However, this method needs high grade devices and technique. To compensate the conventional method, the calculation of AHI using only the snore noise of the patient is invented.
Voice activity detection with voice/nonvoice classification using reliable fundamental frequency
Authors: Suk Soo-young , Kojima Hiroaki , Chung Hyun-Yeol
Published: October 2013

In this paper, we introduce a voice activity detection (VAD) algorithm to perform voice/nonvoice (V/NV) classification using a fundamental frequency (F0) estimator called YIN. Although current speech recognition technology has achieved high performance, it is insufficient for some applications where high reliability is required, such as voice control of powered-wheelchairs for handicapped persons. The proposed VAD, which rejects nonvoice input in preprocessing, is helpful for realizing a highly reliable system. Previous V/NV classification algorithms have generally adopted statistical analyses of F0, the zero-crossing rate, and the energy of short-time segments. A combination of these methods, a cepstrum-based F0 extractor, has been proposed [S. Ahmadi and S. S. Andreas, IEEE Trans. SAP. 7, 333–339 (1999)]. The proposed V/NV classification adopts the ratio of a reliable fundamental frequency contour to the whole input interval. To evaluate the performance of our proposed method, we used 1360 voice commands and 736 noises in powered-wheelchair control in a real environment. These results indicate that the recall rate is 97.4% when the lowest threshold is selected for noise classification with precision 97.3% in VAD. The proposed VAD, which rejects nonvoice input in preprocessing, can be helpful to realize a highly reliable system.
An implementation of realtime speech speed control system using hierarchical cascaded integrator and comb filters
Authors: Takei Hironobu , Nakamura Shogo , Saitoh Hiroto , Sukadana I. Nyoman
Published: October 2013

In this paper, our algorithm for speech speed control system and its DSP implementation are described. The system slows or speeds up speech without changing its pitch, which aids learning a foreign language, listening for older people, and so on. The proposed algorithm is based on the phase vocoder, which doesn’t need extraction of voiced fragments from a speech signal. And the reconstruction of time-scaled signal involves no smoothing operation between voiced and unvoiced sections. The speech signal is decomposed to bandpassed frequency channels so that the envelope of each of the channels is stretched. Then the stretched envelope is multiplied by the corresponding sinusoidal, and by summing up the channel components the speed controlled signal is reconstructed. Each band has very narrow bandwidth, which requires a large amount of computation with a FIR-based design. To achieve realtime operation of our system, a hierarchical structured cascaded integrator and comb (HSCIC) filter is also introduced, which can have highly sharpened characteristics with only a few multipliers.
Transient detection algorithms for speech coding applications
Authors: Szwoch Grzegorz , Kulesza Maciej , Czyzewski Andrzej
Published: October 2013

One of the methods for improving transmitted speech quality is efficient encoding of transients. The majority of currently used speech codecs employ the parametric approach, thus they do not differentiate between transients and other signal components. A sine+noise+transients signal model, proposed earlier as a tool for wideband audio signal encoding, seems to be a promising solution for enhancing signal quality in speech coding. Methods of transient identifying for an efficient transform-domain encoding of transients has to be chosen carefully. Different transient detection algorithms are described in literature, most of them optimized for musical signals. The methods do not ensure efficient transient detection in speech signal, due to the different character of musical signals and speech. This paper provides a comparison of several transient detection methods, mostly based on evaluation of signal energy, that are now primarily used for musical signals. Their efficiency and application to transient detection in speech are evaluated and discussed. Differences between transient detection in wideband musical signals and speech are pointed out. Modifications to existing algorithms are proposed, allowing efficient transient detection and encoding in the proposed, wideband speech codec. [Research funded by the Polish Ministry of Education and Science within Grant No. 3 T11D 004 28.]
Sound source classification using temporal pattern for mixed sounds
Authors: Taniguchi Toru , Shirai Katsuhiko , Tohyama Mikio
Published: October 2013

In speech information processing of real-world environments, how to deal with nonstationary acoustic noise such as background music or speech is a remaining problem while much progress has been made in the speech processing in stationary noise. Music or speech is a sound that has a complex spectral and temporal structure composed of many musical notes or phonemes, and this fact makes it difficult to estimate signals of music or speech separated from a foreground speech. For noise estimation of one-channel sound, only the prior knowledge of temporal or spectral structures of the noises is available besides the signal itself. Therefore, dealing with nonstationary noise of music or speech essentially requires the models representing the temporal and spectral structure of them. In our study, a classification method of nonstationary sounds using sinusoidal decomposition is proposed. In the method a classification of mixed sounds into sound categories, like speech, singing voices, or instruments, is performed based on temporal characteristics of spectral sequences analyzed by a spectral peak tracking algorithm. Temporal features are inspected by statistical analyses and discrimination experiments of sinusoidal sequences. The classification algorithm is evaluated by experiments of classification of mixed sound.
Multiple fundamental frequency estimation assisted by audio watermarking
Authors: Liu Yi-Wen
Published: October 2013

The success of computer segregation of sound sources from a single-channel mixture often relies on the estimation of multiple fundamental frequencies. Instead of solving the problem directly, this presented work utilizes audio watermarks to assist sound source segregation. Here, individual sources are assumed to be available for watermark embedding before mixing. Each source’s short-time spectral peaks are aligned to frequency gridpoints labeled with a digital ID [Liu and Smith, ‘‘Audio Watermarking Based on Sinusoidal Analysis and Synthesis,’’ International Symposium on Musical Acoustics, Nara, Japan (2004)]. A modified sound source is sinusoidally synthesized afterwards. In this sense, each source is embedded with a watermark. In an experiment, two sound sources are watermarked with two sets of gridpoints P and Q, respectively. Points in P and Q interleave and are spaced 10 cents of a semitone apart. The watermarked sources are then linearly added. To demix them, sinusoidal trajectories found in the mixture are segregated based on whether its frequency more often aligns to P or Q. Then, segregated signals are reconstructed by sinusoidal synthesis. Although the reconstructed signals do not sound like the original sources, the melodies of the original are successfully extracted from the reconstructed in monophonic pitch estimation.
Azimuth and elevation estimation for multiple sound sources using two-dimensional frequency domain binaural model
Authors: Chisaki Yoshifumi , Kawano Sho , Usagawa Tsuyoshi
Published: October 2013

Direction of arrival (DOA) estimation for multiple sound sources with only two sensors as the human being does is proposed. The frequency domain binaural model (FDBM) [Nakashima et al., Acoust. Sci. Technol. 24, 172–178 (2003)] was proposed as one binaural model. The method uses sound diffraction to estimate DOAs and segregate sound sources. The binaural model can estimate one-dimensional DOA, namely, azimuth. However, human beings can estimate not only azimuth of DOA but also elevation of that. In order to achieve an estimate of both azimuth and elevation simultaneously, a new DOA estimation algorithm based on interaural phase and level differences is proposed as a part of frequency domain binaural model. Results of DOA characteristics of the proposed binaural model estimation on multiple sound sources’ condition is compared with that of human auditory characteristics. As a result, it was confirmed that performance on accuracy of DOA estimation depends on power level ratio among sound sources. [Research was partially supported by the Ministry of Education, Science, Sports and Culture, Grant-in-Aid for Scientific Research (C), 2006, 18500135 and carried out by the Cooperative Research Project Program of the RIEC, Tohoku University (Grant No. H16.A02).]
Free listening-point synthesizing method in a large microphone array using acoustic transfer function
Authors: Panahpour Tehrani Mehrdad , Hirano Yasushi , Kajita Shoji , Mase Kenji , Nishino Takanori , Takeda Kazuya , Fujii Toshiaki
Published: October 2013

In this research a free listening-point synthesizing method is proposed by using the measured acoustic transfer function (ATF) in a large microphone array for future 3-D audiovisual systems. ATFs at the location of far-field microphones are estimated in frequency domain by using a recorded time-stretched pulse (TSP) signal broadcasted by speakers at the location of sources. Then an ATF pool is generated with a chosen density by synthesizing the estimated ATFs for each sound source. The ATF pool contains all the microphones’ locations, plus synthesized ATFs with a chosen density from sound sources to any point on the arc which passes all of the microphones. The free listening-point is generated by selecting the ATF for each source from the ATF pool. Then the source microphone sounds are convolved with the chosen ATFs and added. The proposed method is evaluated by an array with 38 microphones in the far field on an arc with r=4 m with 20 cm between each pair of microphones, and two microphones close to two sound sources with 1.5-m distance. Experimental results have been shown for the estimated ATFs and comparison of the synthesized arbitrary listening point with the actually recorded sound for a given point that shows satisfactory results.
Sensor signal processing circuit for echolocation using a delta-sigma modulated single-bit digital signal
Authors: Hirata Shinnosuke , Kurosawa Minoru K. , Katagiri Takashi
Published: October 2013

Ultrasonic sensors are suitable for perception of the environment of mobile autonomous robots. Most bats use echolocation that is a high-quality sensing system by their ultrasonic sensors [S. J. Gaioni, H. Riquimaroux, and N. Suga, ‘‘Biosonar behavior of mustached bats swung on a pendulum prior to cortical ablation,’’ J. Neurophysiol. 64(6), 1801–1817 (1990)]. The echolocation can estimate the distance to the object, its relative velocity, and its scale. But, technological application of echolocation necessitates a large computational requirement because delay of the echo signal from the generated signal is estimated from cross correlation between the generated signal and the echo signal. A new sensor signal processing for echolocation is proposed by using a delta-sigma modulated single-bit digital signal. A delta-sigma modulated single-bit digital signal is a single-bit high-speed data stream. [S.R. Norsworthy, R. Schreier, and G. C. Temes, Delta-Sigma Data Converters (IEEE Press, New York, 1997] Using delta-sigma modulated single-bit digital signal processing, multiplication and accumulation of cross-correlation can be changed into logical operations: EXOR, NOT, SUM. Simulations show the possibility of estimation of the distance to the object from correlation of the echo signal with the generated signal converted single-bit data stream, along with reduced computational and circuit-scale requirements.
Modeling of phased antenna array of about 500 microphones, detecting landing airplanes
Authors: Caronna Gaetano , Roselli Ivan , Testa Pierluigi
Published: October 2013

The University La Sapienza participated recently in the project SAFE-AIRPORT funded within the sixth framework program of the European Commission. This project aimed at developing a phased antenna array of about 500 microphones in order to acoustically detect and track flying airplanes. A preliminary field testing was performed at Rome-Fiumicino airport, however experimental data are not yet fully available. For this reason, a computer-aided model of the antenna phased array was developed and beam-forming was simulated with real input signals (acoustic emissions from a real landing aircraft with superimposed disturbing, spatially uncorrelated, pink noise of realistic level). Detection range was measured by applying classic shift and sum beamforming. An algorithm was tested in order to increase maximum detection range by exploiting information from more than one temporal frequency, assuming that disturbing environmental noise is uncorrelated on different temporal frequencies. Specific results are reported.
Compressive coding for multichannel audio signals using independent component analysis
Authors: Mihashi Tadashi , Takatani Tomoya , Miyabe Shigeki , Mori Yoshimitsu , Saruwatari Hiroshi , Shikano Kiyohiro
Published: October 2013

This study addresses a novel compressive coding for multichannel audio signals with independent component analysis (ICA). In recent years, multichannel audio contents are widely spreading, and thus the transmission quantity is increasing. Consequently, audio-signal compression is a problem demanding prompt attention. Conventional compression methods utilizing sparseness assumption among audio-source signals have a problem that the method is likely to distort original sources owing to the lack of sparseness in the mixed signals consisting of two or more sound sources. To solve this problem, we propose an audio-signal compression strategy using ICA; ICA assumes that the source signals are mutually independent, and can estimate the source signals from the mixed signals without a priori information by applying an unmixing matrix. The decomposition of mixed signals leads to an enhancement of sparseness among signals. Furthermore, we propose a mechanism to detect time-frequency grids where sparseness is not satisfied, and for the detected grids we directly transmit multichannel signals to mitigate the distortion. The reconstruction process utilizes an inverse projection of the unmixing matrix to make correct lateralization of each sound source. Experimental result shows that the proposed method can compress mixed audio signals more efficiently and achieve a higher quality than the conventional method.
Spatial coherence of the acoustic field for propagation ranges 500–1000 m in an East China Sea channel
Authors: Dahl Peter H. , Choi Jee Woong
Published: October 2013

Measurements of the vertical spatial coherence at ranges 500–1000 m and for transmission frequencies 2–20 kHz were made as part of the Asian Sea International Acoustics Experiment (ASIAEX) conducted in the East China Sea (depth 105 m). In an earlier work [J. Acoust. Soc. Am. 115, 589–599 (2004)], ASIAEX spatial coherence measurements corresponding to arrivals interacting once with the sea surface were modeled using the van Cittert–Zernike theorem in conjunction with model estimates of bistatic scattering scattering from the sea surface. The theorem relates the spatial coherence of an observed field to the distant source intensity distribution associated with this field. Here, the approach from the 2004 work is extended to study the full multiple path arrival involving bistatic scattering from the sea surface, and coherent reflection plus bistatic scattering from the seabed. Model and data comparisons are made for the vertical direction and predications are made for coherence in horizontal, and horizontal-transverse direction.[Work supported by the ONR.]
Statistics of the Parseval sum, match filter, and instantaneous single-frequency transmissions in an ocean waveguide
Authors: Andrews Mark , Chen Tianrun , Lee Sunwoong , Makris Nicholas , Ratilal Purnima
Published: October 2013

During the ONR Main Acoustic Clutter Experiment of 2003 on the New Jersey Continental Shelf, short duration broadband pulses with 50- to 150-Hz bandwidths were transmitted at 50-s intervals from a moored source array in the 300- to 1500-Hz range. The signals were measured by a towed horizontal receiving array at varying ranges from the source. The broadband data are processed in a variety of ways including the use of Parseval sum and match-filter, and as an instantaneous single frequency transmission. The ping-to-ping fluctuations in the measured acoustic data are a result of changes in the waveguide modal intereference structure due to both motion of the array and presence of time-dependent random internal waves. Our analysis shows that the Parseval sum and match-filter outputs for the direct arrival have significantly smaller standard deviations compared to the instantaneous single frequency transmission. We account for these observations by modeling the broadband acoustic field intensity in both a static waveguide and with Monte Carlo simulations in a fluctuating waveguide, as well as with the use of statistical distributions for the broadband acoustic field. We present an approach for empirically quantifying the range-dependent degradation in the match-filter output in a multimodal waveguide.
Ray-based acoustic localization of cavitation in a highly reverberant environment
Authors: Chang Natasha A. , Dowling David R.
Published: October 2013

Acoustic localization techniques are applied in a water tunnel to study low event rate cavitation in a vortical flow. The measured center frequency and bandwidth of the acoustic pulse from the collapse of a small isolated cavitation bubble are 10 kHz and 200 kHz, respectively, and the measured pulse duration is ∼15–20 μs. Its timing and exact waveform is not known. An array of 16 receiving hydrophones is used to estimate the source location in the test section of the water tunnel. The direct path signal used for this effort can be buried in background hydrodynamic noise from the water tunnel. Fortunately some of the direct-path signal arrival is distinct enough for acoustic localization. The signal coda is not used in the present effort. The received direct-path signals are cross-correlated between hydrophones to obtain arrival time differences. These arrival time differences are used in conjunction with a simple ray-based acoustic model to estimate the source location in three dimensions. Once the cavitation bubble is located, the signal can be back propagated and part of the free space signal estimated and correlated with high-speed video of cavitation bubble dynamics. [Work sponsored by ONR codes 333 and 321.]
Bistatic invariance for active sonar systems
Authors: Zurk Lisa , Quijano Jorge , Velankar Manish , Rouseff Dan
Published: October 2013

The concept of an invariant time-frequency relationship for passive acoustic intensity has been established through the consideration of interfering normal modes traveling in a shallow water channel. This formulation has received considerable attention in the past and has been applied to the interpretation of passive lofargrams. For bistatic active geometries, the acoustic energy propagates along two distinct pathways: source-to-target and target-to-receiver. In recent work we proposed that this more complicated propagation may also exhibit an invariant structure, thus suggesting potential for improved target tracking through the application of invariant-based track constraints. In this work, further exploration of this phenomenon is presented, with specific attention to the impact of the target scattering matrix that couples energy at the target between incoming and outgoing modes. Simulation results are presented for a variety of underwater environments to examine the effect on the invariant structure. Results are also presented and discussed from spectrograms produced from active sonar data obtained in shallow water environments.
Propagation of impulsive broadband signals in a coastal ocean setting during the 1996 Shelfbreak Primer experiment
Authors: Crocker Steven E. , Miller James H. , Potty Gopu R. , Dossot Georges A. , Lynch James F.
Published: October 2013

The 1996 Shelfbreak PRIMER experiment provided an opportunity to study the impulse response of a coastal ocean waveguide using broadband acoustic signals from explosive sources. Acoustic transmission paths in the vicinity of the shelfbreak included various orientations ranging from up-slope to along-slope, including interactions with a shelfbreak front. Acoustic receptions were acquired with a 16-element vertical line array (VLA). In addition, thermistor string data acquired at the receiving array suggest that a group of high-amplitude, internal waves traversed the test site during this portion of the experiment. Modal decomposition of broadband signal arrivals was performed to investigate variation of the impulse response during source deployments. Variations in the received spectra of individual normal modes were compared with numerical predictions for impulsive broadband signals propagating in range-dependent environments constructed using oceanographic data available for the test site. The potential for mode coupling and three-dimensional effects to shape the modal and spectral distribution of received energy is being investigated. Numerical and experimental results will be discussed. [Work supported by Office of Naval Research.]
Underwater propagation at 16 Hz from French Polynesia to New Zealand: Echoes from islands, banks, and seamounts, and attenuation by seamounts
Authors: Hall Marshall V.
Published: October 2013

Underwater signals from two nuclear shots fired in Fangataufa and three in Mururoa during 1995/96 were recorded off northern New Zealand, a geoid distance of 4660 km. The signals have been bandpass filtered at 16 Hz and found to contain up to six echoes. The causative scattering seafloor features were estimated from the travel times. For some echoes there was ambiguity in the feature and hence some uncertainty in the deflection angle (DA). Amongst the six echoes, the DA ranged between 20 and 56 deg. In the two Fangataufa signals, echo intensities relative to the intensity of the direct pulse ranged between −44 and −25 dB, and this relative intensity (RI) decreased with increasing DA. The highest feature on the direct path from Fangataufa, a seamount whose peak is 1.6 km deep, appears to have had little effect on the signal. In the three Mururoa signals, there were fewer echoes, and RI did not vary significantly with DA. The highest feature on the direct path from Mururoa was a seamount whose peak is 0.7 km deep. This seamount attenuated the direct signal at New Zealand by approximately 23 dB, as well as paths to features at small angles from the direct.
Features of the broadband acoustic propagation in very shallow water
Authors: Lee Seongwook , Park Kyu-Chil , Yoon Jong Rak , Lee Phil-Ho
Published: October 2013

In shallow water, acoustic propagation is severely influenced by the property and structure of the bottom. We present the results of acoustic measurement conducted in a very shallow water of about 11-m depth to characterize the features of broadband acoustic propagation. In the measurements using both of the ship and projectors as sources, the signal was received at four hydrophones, two of them were located in the water and another two were located in the sediment. Broadband propagation losses derived from ship noise revealed the frequency band that has high losses around 120–140 Hz. Analysis results on the possible causes of high losses at that frequency band are presented.
A stochastic model for underwater acoustic channel impulse function in the Kauai experiment
Authors: Song Aijun , Luo Jing , Badiey Mohsen
Published: October 2013

Modeling of underwater acoustic channel impulse function has been of interest to the research community. As is well known, the underwater acoustic wave propagation depends on a number of environmental parameters, such as the sound speed profile along the propagation track, sea surface condition, and source/receiver geometry. The fluctuations of these environmental parameters cause the variability of the acoustic wave propagation in the shallow water region. In this paper, a stochastic model for the channel impulse function is proposed. The characteristics of the each eigenray, such as arrival time, average intensity, and coherence time, are extracted from the Kauai experiment (2003). The correlation between the model parameters and the environmental variability will also be shown.
Arrival time change of long distance sound travel in the central Pacific
Authors: Ogasawara Hanako , Nakamura Toshiaki , Hachiya Hiroyuki , Fujimori Hidetoshi , Mizutani Koichi
Published: October 2013

Ocean acoustic tomography (OAT) is a valuable method for observations of oceanographic phenomena over wide areas. A tomography experiment using seven 200-Hz transceivers at depths around 1000 m was executed in the central equatorial Pacific Ocean during January through December 2000 to monitor ocean phenomena related to El Niño and southern oscillation (ENSO). Because sound that propagates in the ocean takes various paths depending on its incident angle, the sound arrival times showed time lags. Arrival times of the first arrival signals were used to evaluate fluctuations throughout the year. Positional data of transceivers calculated from transponders indicated that major fluctuations of arrival times resulted from location movements of transceivers. The arrival times were corrected by calculating the distance using the positional data to remove distance effects in each transmission. Tidal effects were observed from spectral analyses of the corrected arrival times throughout the experimental period. Reciprocal arrival time differences between one pair of transceivers showed different changes during mid-September through November in comparison to other reciprocal arrival times of several paths. This change might have resulted from some kind of oceanic variation. Acoustic observations are advantageous for detecting various changes over long periods.
Simultaneous estimation of reverberation times and their uncertainties from room impulse responses using a single-measurement procedure
Authors: Xiang Ning , Jasa Tomislav
Published: October 2013

Accurate measurements of reverberation times are of fundamental importance in room acoustics. A number of test procedures for characterizing acoustics in performing arts venues, quantifying acoustic properties of materials in chamber measurements, rely on experimental determination of reverberation times. In addition, decay-time estimation in acoustically coupled spaces has been found to be very demanding. Our recent work has demonstrated that model-based Bayesian approaches [Xiang et al., J. Acoust. Soc. Am. 110, 1415–1424 (2001); 113, 2685–2697 (2003); 117, 3705–3715 (2005)] can be very useful for such analysis in architectural acoustics measurements. This paper discusses the recent development of probabilistic tools for estimating both reverberation (decay) times and their uncertainties within Bayesian framework. This work shows that Bayesian probabilistic inference can be used as a useful tool for sound energy decay analysis in both single-space halls and coupled spaces. Bayesian decay analysis simultaneously provides architectural acousticians with reverberation times, diverse decay times, related derivations, and interdependencies to quantify uncertainties of the estimation from a single measurement of room impulse responses followed by Schroeder backward integrations.
Permissible number of synchronous averaging times to obtain reverberation time from impulse response under time-variance conditions
Authors: Satoh Fumiaki , Hayashi Yukiteru , Sakamoto Shinichi , Tachibana Hideki
Published: October 2013

In the measurement of room impulse response, the synchronous averaging technique and such new methods as the MLS and the swept-sine methods are being widely used to improve the signal-to-noise ratio. In actual measurement conditions, however, the air in a room is continuously moving and the temperature is changing to some degree. The measured value of the reverberation time in such a room tends to be shorter at higher frequencies when applying the synchronous averaging. Therefore, the assumption of a time invariant has to be carefully considered, and, on this point, some research has been conducted to date. We also have reported various research results concerning the impulse response measurement under the time-variance conditions. In this paper, the permissible number of synchronous averaging times for reverberation measurement is studied through some field experiments. In each field, many time impulse response measurements were taken between a fixed pair of sound source and receiving positions by the swept-sine method, without averaging. After the measurements, the characteristics and the extent of the time-variance under measuring were estimated by a short-term running cross-correlation function between each impulse response. The influence of the time variance on the synchronous averaging result was studied based on the estimated time variance.
Selection of receiving positions suitable for evaluating acoustical parameters
Authors: Akama Taeko , Suzuki Hisaharu , Omoto Akira
Published: October 2013

Many physical parameters show characteristics of large sound fields such as concert halls. Some of them are adopted in the Annex of ISO 3382. That definition is clearly provided in ISO. However, practical measurement methods for them remain obscure. Our research is intended to examine an effective selection method of receiving positions based on the distribution of acoustical parameters in a real field. For that purpose, impulse responses are measured at more than 1400 seat positions to elucidate the distribution of acoustical parameters in an existing concert hall. The acoustical parameters, which are reverberation time, early decay time, clarity, and center time at each seat, are then calculated for 500-Hz, 1-kHz, and 2-kHz octave bands. The distributions of reverberation time are quite even at all seats. However, the distributions of other parameters show symmetrical patterns at 500 Hz. At 1 and 2 kHz frequencies, the distributions show asymmetrical patterns in this hall. Based on the results obtained in this study, an effective method to select the receiving position can be proposed.
Hybrid measurement method in room acoustics using dodecahedron speakers and a subwoofer
Authors: Miyazaki Hideo
Published: October 2013

A dodecahedron speaker is usually utilized for measurement in room acoustics under the hypothesis of omni directional point source. But generally speakers used for a dodecahedron speaker cannot playback low-frequency sound such as under 100 Hz, which is important especially for auralization, while the one constructed of units with large diameter to support low-frequency sounds cannot be considered as an omni-directional speaker in high frequencies. To meet these requirements, a hybrid system combining a dodecahedron speaker and a subwoofer has been developed and actually used for measurements of impulse responses in acoustical design of concert halls. The summary of this method will be presented. The feasibility of this method will be also discussed while evaluating the measurement results in concert halls by changing measurement conditions such as speaker locations and comparing these results with those of conventional methods.
Sound source with adjustable directivity
Authors: Behler Gottfried K.
Published: October 2013

Omni-directional sound sources are used to measure room-acoustical parameters in accordance with ISO 3382. To record a detailed room impulse response (RIR) with the aim of auralization, an extended frequency range is required that is not covered by the often-used building acoustics sound sources. To obtain this target, a loudspeaker with dedicated sources for low, mid, and high frequencies was designed, providing a smooth omni-directionality up to 6 kHz and a usable frequency range from 40 Hz up to 20 kHz. However, a realistic auralization of sources like musical instruments is not possible with an omni-directional measured RIR. To include the directional characteristics of instruments in the measuring setup, the directivity of the sound source has to be frequency dependent and must be matched to the (measured) directivity of the real instrument. This can be obtained by using a dodecahedron loudspeaker with independently operating systems and an appropriate complex FIR filtering of the frequency response of each driver. The directivity is a result of parameters like magnitude and phase and the interference sum of all systems. To create the appropriate directivity, optimization algorithms are used to achieve minimum error between measured and adapted directivity.
Objective measures for evaluating tonal balance of sound fields
Authors: Takahashi Daiji , Togawa Kanta , Hotta Tetsuo
Published: October 2013

The purpose of this study is to derive objective measures, which can well represent the characteristics of the sound field regarding the tonal balance corresponding to our hearing sense. Two kinds of listening test were conducted in the form of paired comparison, in which subjects were tested using sound fields produced by convoluting some anechoic music sources with some impulse responses. In the first listening test, impulse responses were calculated theoretically for a simple structure of sound field having a direct sound and reflections, and, in the second test, impulse responses were measured at the various seats of existing concert halls. In the latter case, impulse responses which give almost the same reverberance were used for the listening tests. From this investigation, it is found that one objective measure called the DL (deviation of level) has a possibility of an effective measure, which can be used as an appropriate measure for evaluating the tonal balance of sound fields. The index DL is calculated from the data based on the logarithmic scale in both the frequency and magnitude. This fact is not inconsistent with the past findings that human response corresponds to a logarithmic scale of stimulus.
Measuring impulse responses containing complete spatial information
Authors: Farina Angelo , Martignon Paolo , Capra Andrea , Fontana Simone
Published: October 2013

Traditional impulse response measurements did capture limited spatial information. Often just omnidirectional sources and microphones are employed. In some cases it was attempted to get more spatial information employing directive transdudcers: known examples are binaural microphones, figure-of-8 microphones, and directive loudspeakers. However, these approaches are not scientifically based and do not provide an easy way to process and visualize the spatial information. On the other side, psychoacoustics studies demonstrated that ‘‘spatial hearing’’ is one of the dominant factors for the acoustic quality of rooms, particularly for theatres and concert halls. Of consequence, it is necessarily to reformulate the problem entirely, describing the transfer function between a source and a receiver as a time/space filter. This requires us to ‘‘sample’’ the impulse response not only in time, but also in space. This is possible employing spherical harmonics for describing, with a predefined accuracy, the directivity pattern of both source and receiver. It is possible to build arrays of microphones and of loudspeakers, which, by means of digital filters, can provide the required directive patterns. It can be shown how this makes it possible to extract useful information about the acoustical behavior of the room and to make high-quality auralization.
Towards a predictive model of noise-induced temporary threshold shift for an amphibious marine mammal, the California sea lion (Zalophus californianus)
Authors: Kastak David , Holt Marla M. , Mulsow Jason , Reichmuth Kastak Colleen J. , Schusterman Ronald J. , Southall Brandon L.
Published: October 2013

A California sea lion that had previously been tested under water was assessed for noise-induced temporary threshold shift (TTS) in air. One hundred ninety-two controlled exposures of octave-band noise centered at 2.5 kHz were conducted over a 3-year period. The noise was varied in level (to 133 dB SPL re: 20 μPa) and duration (to 50 min) to generate a variety of equal sound exposure levels (SELs). Behavioral psychophysics was used to measure hearing sensitivity at 2.5 kHz before, immediately following, and 24 h following noise exposure. The levels of threshold shifts obtained ranged up to 30 dB. In cases where TTS exceeded 20 dB, thresholds were obtained at regular intervals until recovery occurred. The average slope of the long-term recovery function was 10 dB per log(minute). Results show that the threshold shifts correlated with SEL; however, the equal-energy trading rule did not apply in all circumstances, with exposure duration contributing more than exposure level. Repeated testing showed no evidence of a permanent threshold shift at 2.5 kHz or octave higher. The amphibious sea lions appear to be equally susceptible to noise in air and under water, provided that the exposure levels are referenced to the subjects thresholds in both media.
Click and tone-pip auditory evoked potentials in a large marine mammal, the northern elephant seal
Authors: Houser Dorian S. , Finneran James J.
Published: October 2013

The use of auditory-evoked potentials (AEPs) to study the hearing of mysticete whales is challenged by access to animals, their large size, and proportionately smaller brain relative to odontocetes. One means by which AEP techniques can be adapted to these larger animals is by application to more readily available proxy species. The northern elephant seal (Mirounga angustirostris) is a large pinniped, potentially in excess of 2000 kg, with a thick dermis, large skull, relatively small auditory nerve, and a low-frequency vocal communication system. AEP collection in elephant seals provides similar challenges to those of the mysticetes but at a scale that provides a greater opportunity for success. AEP tests were conducted on northern elephant seals at Año Nuevo State Reserve, the natural haul-out site of the elephant seal. Subjects were chemically immobilized with tiletamine/zolazepam and chemical restraint was maintained with bolus injections of ketamine. Click-evoked potentials were collected from four weanling and two adult male elephant seals and tone-pip-evoked potentials were collected from a 2-year-old female. Results demonstrate that AEPs can be recorded from large pinniped species, providing a step towards the application of similar techniques to larger cetacean species.
Hearing frequency selectivity in four species of toothed whales as revealed by the evoked-potential method
Authors: Popov Vladimir
Published: October 2013

Frequency tuning curves were obtained using a tone-tone simultaneous masking paradigm in conjunction with the evoked potential recording. The masker was a continuous tone and the test was a sinusoidal amplitude-modulated (SAM) tonal signal, which evoked the envelope following response (EFR). The EFR was recorded in unanaesthetized animals from a head surface with the use of suction-cup electrodes. The obtained tuning curves featured very sharp tuning with Q(ERB) (quality estimated by the equivalent rectangular bandwidth) from 35 in Tursiops truncatus to nearly 50 in Delphinapterus leucas. This acuteness is several times better than in humans and many animals. The Q(ERB) dependence on probe frequency could be approximated by regression lines with a slope from 0.18 in Tursiops trucatus to 0.83–0.86 in Phocoena phocoena and Neophocoena phocoenoides. Thus, the frequency representation in the odontocete auditory system may be either near constant quality (in Tursiops) or near constant bandwidth (in porpoises). [Work supported by The Russian Foundation for Basic Research and Russian President Grant.]
Growth and recovery of temporary threshold shifts in a dolphin exposed to midfrequency tones with durations up to 128 s
Authors: Schlundt Carolyn E. , Dear Randall L. , Carder Donald A. , Finneran James J.
Published: October 2013

Auditory thresholds at 4.5 kHz were measured in a bottlenose dolphin (Tursiops truncatus) before and after exposure to midfrequency tones at 3 kHz. Experiments were conducted in relatively quiet pools with low ambient noise levels at frequencies above 1 kHz. Behavioral hearing tests allowed for thresholds to be routinely measured within 4 min postexposure, and tracked recovery for at least 30 min postexposure. Exposure durations ranged from 4 to 128 s at sound pressure levels ranging from 149 to 200 dB re: 1 μPa. Sound exposure levels ranged from 155 to 217 dB re: 1 μPa2/s. Temporary threshold shifts at 4 min postexposure (TTS4) of up to 23 dB were observed. All thresholds recovered to baseline and pre-exposure levels, most within 30 min of exposure. [Work supported by the U.S. ONR.]
Analysis of Australian humpback whale song using information theory
Authors: Miksis-Olds Jennifer L. , Buck John R. , Noad Michael J. , Cato Douglas H. , Stokes Dale
Published: October 2013

Songs produced by migrating whales were recorded off the coast of Queensland, Australia over 6 consecutive weeks in 2003. Approximately 50 songs were analyzed using information theory techniques. The average length of the songs estimated by correlation analysis was approximately 100 units, with song sessions lasting from 300 to over 3100 units. Song entropy, a measure of structural constraints and complexity, was estimated using three different methodologies: (1) the independently identically distributed model; (2) first-order Markov model; and (3) the nonparametric sliding window match length (SWML) method, as described in Suzuki et al. [J. Acoust. Soc. Am. 119, 1849 (2006)]. The analysis finds the songs of migrating Australian whales are consistent with the hierarchical structure proposed by Payne and McVay (Science 173, 585−597 (1971)], and recently confirmed by Suzuki et al. for singers on the breeding grounds. Both the SWML entropy estimates and the song lengths for the Australian singers were lower than that reported by Suzuki et al. for Hawaiian whales in 1976−1978. These lower SWML entropy values indicate a higher level of predictability within songs. The average total information in the Australian sequence of song units was approximately 35 bits/song. Aberrant songs (10%) yielded entropies similar to the typical songs. [Sponsored by ONR and DSTO.]
Ultra-high-speed imaging of bubbles interacting with cells and tissue
Authors: Versluis Michel , Marmottant Philippe , Hilgenfeldt Sascha , Ohl Claus-Dieter , Chin Chien T. , van Wamel Annemieke , de Jong Nico , Lohse Detlef
Published: October 2013

Ultrasound contrast microbubbles are exploited in molecular imaging, where bubbles are directed to target cells and where their high-scattering cross section to ultrasound allows for the detection of pathologies at a molecular level. In therapeutic applications vibrating bubbles close to cells may alter the permeability of cell membranes, and these systems are therefore highly interesting for drug and gene delivery applications using ultrasound. In a more extreme regime bubbles are driven through shock waves to sonoporate or kill cells through intense stresses or jets following inertial bubble collapse. Here, we elucidate some of the underlying mechanisms using the 25-Mfps camera Brandaris128, resolving the bubble dynamics and its interactions with cells. We quantify acoustic microstreaming around oscillating bubbles close to rigid walls and evaluate the shear stresses on nonadherent cells. In a study on the fluid dynamical interaction of cavitation bubbles with adherent cells, we find that the nonspherical collapse of bubbles is responsible for cell detachment. We also visualized the dynamics of vibrating microbubbles in contact with endothelial cells followed by fluorescent imaging of the transport of propidium iodide, used as a membrane integrity probe, into these cells showing a direct correlation between cell deformation and cell membrane permeability.
Formation of water pore in a bilayer induced by shock wave: Molecular dynamics simulation
Authors: Koshiyama Kenichiro , Yano Takeru , Fujikawa Shigeo , Kodama Tetsuya
Published: October 2013

The irradiation of a shock wave or ultrasound with micro-bubbles has the potential to make transient pores on cell membranes. Although such pores are believed to contribute to the molecular delivery thorough the membrane, the detailed mechanisms of the pore formation with shock waves and the subsequent molecular delivery through the pores into cells are still unclear. To investigate the mechanism at a molecular level, the molecular dynamics simulations of the interaction of the shock wave with a lipid bilayer are conducted. The water penetration into the hydrophobic region by the shock wave is observed in picoseconds. As a next step, structural changes of the bilayer containing water molecules in the hydrophobic region are investigated. The water pore is formed in 3 ns when the large number of water molecules is inserted. The lifetime of the water pore is more than 70 ns. The radius of the water pore is ca. 1.0 nm, which is three times larger than the Stoke’s radius of a typical anticancer drug (5FU). Finally, the diffusion of the anticancer drug in the water pore is investigated.