bluuebunny/crossref_metadata_2025 huggingface dataset:
High-altitude infrasound propagation experiment
Authors: Andre Bill , Bass Henry E.
Published: October 2013

Experimental verification of propagation predictions and sensor performance for infrasound frequencies that propagate long distance (greater than 100 km) are difficult to perform due to the rarity of controlled sources. If the source is elevated, the lower atmospheric pressure allows an explosion of a given charge weight to expand further during the supersonic phase, resulting in a lower fundamental frequency. A 50-lb charge exploded at 40 km yields a fundamental frequency similar to a 500-ton explosion at ground level. During the fall of 2005 and the winter and summer of 2006, experiments were conducted at White Sands Missile Range, NM, where an Orion rocket lifted a 50-lb explosive charge to altitudes between 30 and 40 km. Launches occurred near 0200 and 0600 MST. Portable infrasound arrays to record events were deployed in New Mexico, Texas, Arizona, and California to complement permanent arrays. Infrasound recordings were supplemented with extensive meteorological measurements. Data analysis is in a preliminary stage. The High Altitude Infrasound Propagation Experiment resulted from collaboration between several U.S. Army and Navy commands, University of Mississippi, Southern Methodist University, University of Hawaii, University of California at San Diego, University of Alaska at Fairbanks, BBN Technologies, SAIC, and Los Alamos National Laboratory.
Infrasonic scattering studies using time-domain parabolic equation (TDPE) and gravity wave models
Authors: Norris David E.
Published: October 2013

The time-domain parabolic equation (TDPE) model is useful in predicting infrasonic waveforms, as it can account for diffraction and scattering mechanisms. In this study, a split-step Fourier (SSF) implementation of the TDPE is used to study infrasonic propagation. Scattering is modeled using a spectral gravity wave model, and multiple gravity wave realizations are used to quantify the uncertainty in waveform properties introduced by scattering from the gravity wave inhomogeneities. TDPE predictions are compared to specific infrasonic events and conclusions regarding the dominant propagation mechanisms are presented.
Infrasound signals detected by the observation system in Isumi, Japan
Authors: Murayama Takahiko , Honma Motohiro , Nogami Mami , Otsu Takayuki , Amano Takafumi , Saitoh Yasuharu , Arai Nobuo , Sakamoto Toyomi
Published: October 2013

An infrasound observation system is installed in Isumi, Japan (35.3N, 140.3E) as the component of the International Monitoring System for CTBT. The observation system was deployed on November 2004. It comprises a six-element array. Five elements of the array are deployed as a pentagon with sides approximately 1.2 km long. The sixth element is located within the pentagon-shaped array. Until now, some interesting infrasound signals were observed. One of them was the infrasound wave produced by the large earthquake that occurred off the Pacific coast of North-East Japan on 16 August 2005. The infrasound wave-train, which had a spindle-shaped envelope, was detected following the arrival of the seismic wave. The arrival direction of the infrasound wave, which had maximum amplitude, indicated the direction of the land area, which had maximum intensity. The time difference between the on-set time of that infrasound wave and the origin-time of the earthquake coincided with the travel time for propagation in the atmosphere between the area that had maximum intensity and the observation site. Therefore, it was estimated that the observed infrasound waves were produced by shaking ground. In this paper, some remarkable infrasound signals including the above are discussed.
Comparitive study of wind noise reduction devices
Authors: Webster Jeremy , Raspet Richard , Yu Jiao
Published: October 2013

Spherical porous foam windscreens are commonly used to reduce wind noise in acoustic measurements. We have argued [J. Acoust. Soc. Am. 119(2), 834–843 (2006)] that the wind noise measured outdoors will lie between the predicted fluctuating stagnation pressure generated by the microphone interaction with the incoming air flow for bare microphones and the turbulence-turbulence interaction pressure for a very large wind screen. The pressure level measured in a moderate size screen will lie between these two extremes. Different transducer geometries and arrangements have been used in aerodynamic and meteorological studies to measure the ‘‘fluctuating static pressure’’ without interference from transducer generated wind noise. In this paper we compare measurements of wind noise using bare microphones, microphones in foam windscreens, streamlined probes, the Quad Disk, and microphones mounted in flat plates. The purpose of this study is to clarify the principles involved in wind noise generation by transducer-turbulence interaction. [Research supported by the U.S. Army TACOM-ARDEC at Picatinny Arsenal, NJ.]
Determination of elastic properties and interatomic force constants using inelastic neutron scattering
Authors: McQueeney Robert
Published: October 2013

Neutron scattering is the most powerful technique for measuring vibrational excitation spectra. With single-crystal samples, the complete phonon dispersion relation can be measured. At small phonon wavelengths, the slope of the dispersion curve is related directly to the elastic constants of the material. Information from shorter wavelength phonons can be used to determine parameters for interatomic force models or compared to first-principles calculations. Model parameters can then be used to calculate phonon densities-of-states and consequently connect to the thermodynamics. Polycrystalline samples yield less information, but the data is directly related to the phonon density-of-states for Bravais lattices. Measurements on polycrystalline samples have the advantage of being fast and are more useful for parametric studies of the lattice dynamics (with temperature, for example). These ideas will be used to discuss several interesting systems, most notably Pu metal and Fe-Ga magnetostrictive alloys. [This work is supported by the U.S. Department of Energy.]
Elastic constants, Blackman diagrams, and new lanthanide-actinide insights
Authors: Ledbetter Hassel , Migliori Albert
Published: October 2013

Using monocrystal elastic constants and Blackman diagrams, one can infer material interconnections, physical-property trends, and knowledge about interatomic bonding. After reviewing briefly the fcc-metal case, focus shifts to Ce, Pu, Th, with known monocrystal fcc elastic constants, the Cij. For fcc U and a few other cases, one can deduce the Cij. From these results about the actinides, there follow several conclusions: (1) elastic anisotropy ranges widely; (2) Poisson ratios are low; (3) interatomic bonding varies widely; (4) all show unusual negative Cauchy pressures; (5) a strong covalent-bonding component occurs, perhaps related to localized f-electrons; (6) δ-Pu behaves most oddly; (7) δ-Pu shares some bonding features with fcc alkali metals. [Work supported by DOE, NSF, State of Florida.]
Ultrasonic studies of hydrogen absorbing materials: Quasicrystals, nanocrystals, and intermetallic compounds
Authors: Leisure Robert G.
Published: October 2013

Resonant ultrasound spectroscopy was used to study several materials—nanocrystalline palladium, quasicrystalline Ti-Zr-Ni, and a number of intermetallic compounds—that have the ability to absorb considerable amount of hydrogen, and thus are of potential use for hydrogen storage. Elastic energies are relevant to the thermodynamics of such systems, which are often described by a lattice gas model. Elastic interactions between dissolved H atoms lead to a critical temperature and precipitation of a hydride phase below the critical temperature. The energetics of hydride precipitation depend on the elastic constants. Thus, it is important that the elastic properties of such systems be measured. In addition, these materials are of interest for several other reasons. The elastic constants and internal friction for the hydrogen-free materials were measured over a temperature range of 3 to 300 K, and in a few cases to higher temperatures. These results will be discussed and interpreted. [This work was supported by the U.S. National Science Foundation under Grant DMR-0070808.]
Direct observation of phonon relaxation by optical beating light scattering spectroscopy
Authors: Sakai Keiji
Published: October 2013

A novel dynamic light scattering system to observe elastic relaxation phenomena with hyper frequency resolution was developed. The principle of the measurement is based on the theory, which describes the dynamic structure factor of fluid under the condition of the frequency-dependent compressibility. The dynamic structure factor, which is usually composed of the Brillouin and Rayleigh triplet, is modulated and shows an additional central component that directly reflects the whole aspect of the relaxation. In the experiment, the output from a frequency-doubled cw-YAG laser was incident into the liquid sample and the power spectrum of the light scattered into the backward direction was analyzed by the optical beating spectroscopy technique. The sample is liquid acetic acid that is known to show a strong ultrasonic relaxation around 1 MHz due to the molecular association process. We could find, in the observed spectrum, the central component introduced by the phenomenon, whose relaxation frequency and strength can be obtained from the width and the intensity of the observed central peak, respectively. The results show very good agreement with those previously obtained by the conventional ultrasonic spectroscopy technique. The recent experiment of the thermal phonon resonance will also be shown.
High resolution Brillouin scattering in solid and gases
Authors: Yogi Takeshi , Sakai Keiji
Published: October 2013

The experimental system of our optical beating Brillouin spectroscopy was improved, which can observe the phonon propagation with the resolution of 1 kHz. The increased sensitivity and frequency resolution were demonstrated in the observation of phonon spectrum in solids of fused silica and also in gases, nitrogen, and freon CHClF2: for their weak light-scattering ability, these substances are very tough specimens for Brillouin scattering studies, and the classical spectrometer of Fabry-Pérot etalon has been so far used for the frequency analysis. The phonon peaks were observed with the present light beating system in nitrogen and freon near 300 K over the wave-number range from 2π=6.20×103 to 3.09×105 m−1, the pressure range from 14.8 to 177 kPa. The observed spectra were well fitted with the Lorentzian curve representing the phonon lifetime, and the phase velocity. The phase velocity of phonon in fused silica agrees well with that obtained by ultrasonic measurement, while the phonon dispersions obtained for gases show the relaxational behavior due to the coupling to the rotational and vibrational motion of molecules.
Building an English speech synthetic voice using a voice transformation model from a Japanese male voice
Authors: Iida Akemi , Kajima Shimpei , Yasu Kiichi , Arai Takayuki , Sugawara Tsutomu
Published: October 2013

This work reports development of an English speech synthetic voice using a voice transformation model for a Japanese amyotrophic lateral sclerosis patient as part of a project of developing a bilingual communication aid for this patient. The patient, who had a tracheotomy 3 years ago and had difficulty in speaking, wishes to speak in his own voice in his native language and in English. A Japanese speech synthesis system was developed using atr chatr 6 years ago and the authors have worked in developing a diphone-based synthesis using festival speech synthesis system and festvox by having the patient read the diphone list. However, it was not an easy task for the patient to phonate and, moreover, to pronounce words in a foreign language. We therefore used a voice transformation model in festival to develop the patient’s English speech synthetic voice which enables text-to-speech synthesis. We trained using 30 sentences read by the patient and those synthesized with an existing festival diphone voice created from a recording of a native English speaker. An evaluation including a listening experiment was conducted and the result of this voice conversion showed that the synthesized voice successfully reflected the voice quality of the patient.
Database size and naturalness in concatenative speech synthesis
Authors: Bunnell H. Timothy , Mantell James T. , Polikoff James B.
Published: October 2013

Unit concatenation TTS systems seek to maximize perceived naturalness by minimizing the amount of signal processing applied to the recorded speech on which they are based. To generate distinct suprasegmentals for a given segmental sequence (e.g., to convey variation in focus or emotion), it is necessary to record and store multiple instances of the same segments that vary in fundamental frequency and voice quality. At the expense of naturalness, concatenative systems can store a minimal segmental inventory and synthesize suprasegmental factors by manipulating f0 and voice quality via signal processing. Classic diphone synthesis (where only a single instance of each diphone sequence is stored) represents the limiting case of this strategy. The present study explores aspects of the trade-off between perceived naturalness and segmental inventory size using the ModelTalker TTS system. Twenty-five speakers each recorded about 1650 utterances. From these, databases were constructed that limited the maximum number of alternate diphones usable for synthesis in five conditions to 1, 5, 10, 20, and 40. Sentences were synthesized from these databases using either full or no f0 control. Results of listening tests wherein subjects rate the naturalness of each sentence will be presented. [Work supported by NIDCD.]
Optimization of target cost weights in concatenative speech synthesis with very short segments of 5-ms duration
Authors: Hirai Toshio
Published: October 2013

If a concatenative speech synthesis system uses more short speech segments, it increases the potential to generate natural speech because the concatenation variation becomes greater. Recently, a synthesis approach was proposed in which very short (5 ms) segments are used [T. Hirai and S. Tenpaku, 5th ISCA Speech Synthesis Workshop, pp. 37–42 (2004), http://www.ssw5.org/]. In that approach, the target cost (how close a database segment is to desired segment) was defined as the simple average of the root mean squares (rms’s) of the difference between the features of the database segment and the desired segment for simplicity. Therefore, it has been expected to optimize the weight of each rms. A Japanese speech database was used to optimize the weights and to evaluate its effects. For corpus construction, 150 utterances were selected from the database. Ten other utterances were selected randomly for generation of feature time series as natural targets for synthesis. Half of them were used for optimization using many weight combinations in synthesis to determine the optimal weight set that shows the minimum concatenation distortion. Distortion for the other half of the utterance synthesis with optimized weights was reduced 34.4% compared to the former approach.
Improving Japanese syllable unit selection using search space with small acoustical features variance
Authors: Moriyama Takaaki , Tenpaku Seiichi
Published: October 2013

Usually in concatenative text-to-speech systems, suitable units are selected according to acoustical features. However, a method to reduce calculation costs has been proposed, which is using syllable notations and positions along with the mora length [Murakami et al., Trans. IEICE J85-D-II(7), 1157–1165 (2002)]. An enlargement of the corpus to synthesize longer phrases lowers the quality of synthesized speech because of increased feature variance within syllable units with identical notation. Methods to produce detailed groups in which the feature’s variance is smaller than syllable units, and to synthesize speech using these groups, are proposed. To produce the groups, a mel-cepstrum distance is used as a measure for evaluation. In this method, the quality of synthesized speech that is given an ideal group sequence might improve, as the distance between units is close within the same group. As a first step to evaluate the proposed method, five groups were generated from a Japanese corpus; then, speech that was synthesized to replace one unit in natural speech to others in the same groups was examined. Results of this listening experiment indicated that an average of the naturalness for these speech types was greater than 80%.
Development project for screen reader interface dynamically hastens speech while giving emphasized information to the tactile sense
Authors: Ifukube Tohru , Kimura Shinta
Published: October 2013

In present screen readers for blind people, users are still required to continue listening to synthesized voices and also they might easily miss important points of a document. A development project of a new screen reader interface has been promoted in cooperation with Japan’s Ministry (METI). The interface named TAJODA can dynamically control the speech rate while giving rich texts to the tactile sense. The speech rate can be changed according to word units by clicking a button of the interface using a thumb. From evaluation tests using sightless people, the maximum speech rate was determined at around 2000 morae/minute. Seven vibro-tactile patterns were selected for presenting rich texts onto an index finger using a tactile matrix display (2×8). Both speech and tactile information are synchronized automatically through a USB interface. It was also ascertained that users can read some documents two to three times faster using the TAJODA interface [Asakwa et al., IEICE Trans. E87-D(6) (2004)].
Evaluating the perceptual quality of speech signals enhanced using the Modified Phase Opponency model
Authors: Deshmukh Om D. , Espy-Wilson Carol Y.
Published: October 2013

Previously, we presented a speech enhancement algorithm based on the auditorily motivated MPO model. It was shown that the MPO-based speech enhancement scheme is robust to fluctuating noise. Compared to the output of several other speech enhancement techniques, the MPO-enhanced speech signals exhibit the lowest increase in Itakura-Saito, log-area-ratio, and log-likelihood-ratio-based distortion measures as the SNR reduces. In the present work, the performance of the MPO speech enhancement is compared with that of the other techniques using subjective evaluations. Three normal-hearing subjects were recruited to perform two-alternative, forced-choice preference tasks. Each sentence was presented twice: one of them was enhanced using the MPO scheme and the other was either left unprocessed or enhanced using one of three other speech enhancement techniques. Sentences are 5-digits long and corrupted by additive subway noise at −5, 0, 5, or 15-dB SNR. Most of the time, the MPO-enhanced speech signal was preferred weakly. Moreover, the MPO-processed clean speech signal was preferred over clean unprocessed speech signal just as many times as the clean unprocessed speech signal was preferred, indicating that the MPO processing does not introduce any perceptual artifacts in clean speech. [Work supported by NSF BCS0236707 DOD H9823005C0425.]
Noise reduction of speech using multiband noise subtraction and adaptive filtering methods in two stages
Authors: Hamid Md. Ekramul , Ogawa Keita , Fukabayashi Takeshi
Published: October 2013

Real-world noise is mostly colored having nonuniform power distribution over the whole spectrum. A single-channel two-stage method with multiband processing technique is presented in this paper. Initially a noise dominant signal is estimated from the observed speech and obtained the first noise-reduced speech by subtracting the signal. The noise-dominant signal is estimated from the valleys of the spectrum whose amplitudes are tuned by the estimated noise degree. The bandwise estimated noise degree is obtained from a priori function as a function of a parameter obtained from the autocorrelation of the noisy speech. The outcome of the first stage is allowed to process by the second stage in bandwise to obtain more enhanced speech. Blind source separation technique is employed that consists of two adaptive filters. The least-mean-square algorithm based on steepest-descent method is implemented to update the filter coefficients in an iterative manner. The method addresses the situations in which the input signal-to-noise ratio varies substantially and a fixed number of iteration of the least-mean-square algorithm for each signal-to-noise ratio is time consuming. Therefore, a function has been proposed that estimates the minimum iteration number required for a given value of the noise degree of noisy speech.
Speech interface for operating information technological home appliances
Authors: Enoki Masanori , Noguchi Tadashige , Arai Tsubasa , Miyake Ayako , Yasui Kohei , Ota Kenko , Yanagida Masuzo , Ida Masaki
Published: October 2013

Information technological (IT) home appliances are becoming highly advanced by being connected to the Internet. Conventional push-button systems on remote controllers cannot suffice for IT home appliances because of their complicated requirements for versatile and complex functions. Speech input is expected to be a good substitute or supplement for push-button systems. Effects of introducing speech input to a TV system are evaluated, implementing facilities of on-line program retrieval besides ordinary functions of controlling the TV sets. One of the problems with introducing speech input is degradation of speech recognition rate due to being affected by surrounding noises and waves reflected on the surface of walls, ceiling, and so on. So, the test system introduced reduction of known noise, dereverberation scheme, delay-and-sum beamformer and blind source separation by independent component analysis as preprocessing for speech recognition. Effects of introducing these types of preprocessing are investigated comparing speech recognition rates in case of employing these types of preprocessing with those in the case of not employing these types of preprocessing. [Work supported by Knowledge Cluster Project, MEXT, and by Academic Frontier Project, Doshisha Univ.]
Hyperparameter estimation for speech recognition based on variational Bayesian approach
Authors: Hashimoto Kei , Zen Heiga , Nankaku Yoshihiko , Akinobu Lee , Tokuda Keiichi
Published: October 2013

A hyperparameter estimation technique for HMM-based speech recognition using variational Bayesian (VB) approach is proposed. In recent speech recognition systems, the ML and MDL criteria have been adopted for training HMMs and the context clustering, respectively. However, since the ML estimator produces a point estimate of parameters, the estimation accuracy is reduced with insufficient data. The MDL criterion is also affected by the amount of training data due to an asymptotic approximation. To avoid this problem, the VB method has been applied to HMM-based speech recognition. By using the VB approach, it is expected that the higher generalization ability is achieved and an appropriate model structure can be selected in the context clustering. In the Bayesian framework, the predictive distribution is obtained by integrating the likelihood function with respect to the prior density. Therefore, the performance of the Bayesian method depends on the setting of hyperparameters representing the prior distribution. However, the detailed investigation for hyperparameter estimation has not been conducted in the speech recognition field. This study proposes an estimation technique of hyperparameters based on the maximization of the marginal likelihood for HMM-based speech recognition. Experimental results show that the proposed technique can estimate appropriate hyperparameters from training data.
Comparison of visual features for audio-visual speech recognition using the AURORA-2J-AV database
Authors: Togo Takahiro , Nimura Yukitaka , Kitasaka Takayuki , Mori Kensaku , Suenaga Yasuhito , Miyajima Chiyomi , Takeda Kazuya
Published: October 2013

Various techniques for noise-robust speech recognition have been developed for many years. However, speech recognition in nonstationary noisy environments, such as in car conditions, is still a very challenging research problem. Audio-visual speech recognition is a promising technique for improving noise robustness in such adverse acoustic conditions. Our research group has recently developed an audio-visual speech database ‘‘AURORA-2J-AV’’ following the AURORA2 database protocol, which has been widely used for evaluating the performance of audio-only speech recognition techniques. The AURORA-2J-AV database consists of audio speech signals and two types of image sequences: color and infrared human face images recorded when multi-digit numbers are spoken by about 100 native Japanese speakers. Visual noise as well as acoustic noise is added to the database after recording for simulating various conditions. Audio-visual speech recognition experiments are conducted using the audio speech and infrared images of the AURORA-2J-AV database. Two visual features, correlation coefficients (CCs) between two successive image frames and principal component scores (PCSs) of lip images, are compared in various signal-to-noise ratio (SNR) conditions. Experimental results show that CCs outperform PCSs at high SNRs, while PCSs function better at lower SNRs.
Frequency domain blind source separation in a noisy environment
Authors: Mukai Ryo , Sawada Hiroshi , Araki Shoko , Makino Shoji
Published: October 2013

A prototype system for blind source separation (BSS) in a reverberant and noisy environment is presented. Our system uses a small three-dimensional array with eight microphones and realizes BSS in various situations such as the separation of many speech signals located in a three-dimensional space, and the extraction of primary sound sources surrounded by many background interferences. The mixed signals observed by the microphone array are processed by independent component analysis (ICA) in the frequency domain. The permutation problem is solved by normalized basis vector clustering, which is a generalized version of directions of arrival (DOA) clustering and has several advantages to the DOA based method. The system estimates the DOA of the source signals as a by-product of the separation process. Moreover, the system has the ability to distinguish primary target signals and ambient noise. Our system performs an on-the-spot BSS of live recorded signals. Live demonstration will be performed at the meeting.
Beyond the independent component analysis: New blind acoustic sound separation in real world via single-input multiple-output-based independent component analysis
Authors: Saruwatari Hiroshi
Published: October 2013

This review discusses a new generation of blind acoustic-signal separation in the real world, focusing on a recently proposed ICA method, Single-Input Multiple-Output (SIMO)-signal-based ICA (SIMO-ICA). The term ‘‘SIMO’’ is referred to as a specific transmission system in which the input is a single source signal and the outputs are its transmitted signals observed at multiple microphones. The SIMO-ICA consists of multiple ICA parts and a fidelity controller, and each ICA runs in parallel under the fidelity control of the entire separation system. In the SIMO-ICA scenario, mixed multiple sources detected at the microphones can be separated, not into monaural source signals but into SIMO-model-based contributions from independent sources as they are at the microphones. Thus, the separated signals of the SIMO-ICA can maintain the spatial qualities of each sound source. This attractive feature of the SIMO-ICA shows the promise of applicability and connectability with the other multichannel signal processing for achieving the higher performance beyond the conventional ICA methods. As the successful examples of SIMO-ICA’s application, three kinds of systems are described: (i) combination with inverse filtering for MIMO decomvolution, ii) combination with adaptive beamformer for higher-performance BSS, and iii combination with binary masking for real-time processing.
Informed acoustic source separation and localization
Authors: Knuth Kevin H.
Published: October 2013

Advances in Bayesian computational technology in the last decade have enabled the development of new source separation and source localization algorithms. These algorithms are greatly improved by the encoding of prior information about a specific problem in the form of the chosen relevant model parameters, the assignment of the likelihood functions, and the assignment of the prior probabilities of the model parameter values. I refer to such source separation algorithms as informed source separation for the reason that they are endowed with specific and often vital information about the problem. Furthermore, the Bayesian methodology allows source separation to be united with source localization simply by including the model parameters that are of interest to the researcher. Here, I will discuss the union of source separation and source localization under the Bayesian methodology, the incorporation of prior information, and the construction of an informed algorithm using the new computational technologies that allow us to estimate the values of the parameters that define these high-dimensional problems.
Sparseness-based 2ch blind source separation in reverberant environments based on direction-of-arrival estimation with a reliability measure
Authors: Izumi Yosuke , Ono Nobutaka , Sagayama Shigeki
Published: October 2013

We present a novel sparseness-based 2ch blind source separation (BSS) method based on robust direction-of-arrival (DOA) estimation that enables reasonable separation of source signals even in a highly reverberant environment. DOA estimation for BSS is not easy in reverberant environments since reflected sound waves behave as coherent interferences in the observed signals and make ambiguous the sound source direction. Our key idea to solve this problem is to find and integrate clean, reliable time-frequency fragments through the observation with a reliability measure. For that, (1) based on a diffused sound field model of reverberation, a theoretical correlation matrix is introduced for whitening the coherent interferences, (2) the DOA of each time-frequency bin is estimated with reliability by utilizing information of neighboring bins, and (3) the reliability-weighted DOAs are integrated over the whole time-frequency bins. After that, separated signals are obtained by time-frequency masking or inverse filtering. Experimental results including comparison with other methods are also reported.
Experimental results of a fast Fourier transform-based motion compensation of synthetic aperture sonar
Authors: Sawa Takao , Aoki Taro , Tahara Jyunichirou , Kamakura Tomoo , Suzuki Ryouji , Yamaguchi Hitoshi
Published: October 2013

Synthetic aperture and its motion compensation technique became available in sonar systems owing to dramatic developments of computer performance. A bottom observation test with synthetic aperture sonar was carried out at the gulf of Katagami in November 2005. The sonar on a tow-fish had detected a bottom and floats at the test. Accelerations and angular velocities of the sonar system had been recorded with external sensors on the tow-fish, too. A relative position and attitude of sonar were calculated from recorded motion data, and a new motion compensation method based on FFT was applied to the synthetic aperture sonar algorithm. In this report, details and capacity of this new method are shown through these experimental results. Sonar images with this new compensation were better than a conventional compensation that uses full bandwidth of motion frequency. Moreover, images were improved by adjusting bandwidth. The authors would like to thank Takeshi Hamazone, a captain of the mother ship, for his maneuvering the ship even in bad weather.
The study on data transmission with short positioning pulse in deep sea
Authors: Watanabe Yoshitaka , Ochi Hiroshi , Shimura Takuya
Published: October 2013

The ultra short base line (USBL) underwater acoustic positioning in deep sea with digital data transmission is being studied. The goal of this study is improvement of the accuracy and expansion of the functions of the USBL for autonomous underwater vehicles (AUVs) and submersibles due to utilization of the transmitted data. To achieve the goal, the reliable data transmission with short positioning pulse must be ensured. The direct sequence spread spectrum (DSSS) is applied. The frequency band width of signal is 3 kHz with a center focus on 12 kHz for the deep sea use. From the ocean experimental data, it seems that the multipaths from the neighborhood have a dominant effect on the transmission quality. The results of investigation about this matter by experiments and simulations are shown. [This study is supported by a Grants-in-Aid for Scientific Research from the Ministry of Education, Culture, Sports, Science and Technology in Japan.]
Development of a digital 3-D SBL underwater positioning system combined with GNSS compass
Authors: Miyamoto Yoshinori , Furusawa Masahiko , Uchida Keiichi , Motoyoshi Toshikazu , Yamaguchi Toshihiro , Miwa Katuji , Kakihara Toshiharu
Published: October 2013

A highly digitized underwater positioning system using the SBL method was developed. This system can measure highly accurate three-dimensional positions of marine organisms equipped with an ultrasonic transmitter (pinger). This system is combined with a GNSS (GPS) compass that measures the bearing, the location (latitude and longitude), and the commotion (rolling angle and pitching angle) of a boat or a buoy. Then the pinger positions are compensated for the commotion and are translated into the underwater absolute positions. The four arbitrary frequencies of the pingers are received by four wideband hydrophones. The received signals are immediately AD converted in high speed and processed by the quadrature detection and the digital filtering with a bandwidth of 3 kHz using FPGA. Finally, the receiving time differences are measured by the cross correlation, realizing a resolution of 0.01 ms. A position measurement experiment was performed in a water tank (length 15 m, width 10 m, depth 10 m), changing the position of a pinger, and it is confirmed that almost the expected accuracy could be realized. Moreover, this system was installed in a boat and the absolute positions of a fish were pursued to confirm the practicality.
Near-orthogonal signals
Authors: Piper John
Published: October 2013

A least-squares approach was used to construct near-orthogonal signals that can be useful in sonar applications. For the matched-filter case the solution can be written with Lagrange multiplier terms, which reduces to a simple eigenvector problem. Decoding filters are also examined, and these results are compared to the matched-filter case.
Predicted versus measured sensitivities of fiber-optic sensors
Authors: DeMetz Fred C.
Published: October 2013

Fiber-optic sensors are finding wide use in underwater and seismic applications due to their ease of construction, wide useful acoustic bandwidth, sensor deployment ranges from the light source of 10 km or more, robust temperature and static pressure tolerance, and immunity from requiring supporting electronics near the measurement points. The theoretical sensitivity for a number of optical sensor designs are compared with experimental data. Their signal-to-noise performance is assessed from the standpoint of minimum detectable signal, environmental background, and optical noise source.
Development of an acoustic cloud condensation nuclei counter
Authors: Zahui Marcellin , Deshpande Abhijit , Delene David
Published: October 2013

Theoretical development of an acoustic cloud condensation nuclei counter is presented. The proposed instrument will be able to count the number of aerosols present in a sample of air flowing through a growth chamber. The air condenses around aerosols and forms water droplets. The number of aerosols is determined by measuring the sound produced underwater by these droplets when the droplets strike a water surface at the bottom of the growth chamber with an impact velocity equal to either their terminal or maximum velocities. First, the terminal velocities of the droplets are calculated. Then, the maximum velocities that these droplets can sustain without breaking are calculated as a function of droplet diameter. Second, the sound due to droplet impact is estimated and experimentally verified. In this study the bubble sound is not considered because micro droplets falling with terminal velocities do not create bubbles. Also, when accelerated, the velocities are limited such that there is no bubble formation at impact. Finally, two approaches are considered for the instrument. One case uses one hydrophone to measure the total sound radiated by all the droplets. The other case is based on a grid of tiny hydrophones, each measuring a droplet impact sound.
Differences in dealing with the acoustics of large outdoor and indoor listening spaces
Authors: Raichel Daniel R.
Published: October 2013

It was perhaps fortuitous that Wallace Clement Sabine’s first consulting assignment from the Harvard Board of Overseers, the execution of which proved to be seminal in the history of architectural acoustics, entailed an indoor facility. The problems of dealing with outdoor performance areas are considerably greater than those encountered in indoor spaces. Indoor spaces provide surface reflections and absorption of sound, both of which can be manipulated through proper design to yield the desired results with respect to reverberation time and sound dispersion. In the case of outdoor performing areas, there are no reflective surfaces (except in the cases of certain amphitheaters), and the need for sound amplification is often necessitated. A reflective bandshell is generally required to direct the program material toward the audience. The lack of wall reflections tends to result in a lack of musical sonority that is usually present in a well-designed indoor auditorium. A number of specific problems with interior and outdoor spaces and modern approaches toward resolving acoustical problems of large listening areas will be discussed.
Prediction of reverberation by stochastic processes in a forest: Creation of reverberation in an outdoor performance space
Authors: Hanyu Toshiki , Sekiguchi Katsuaki
Published: October 2013

A forest is an outdoor space. For that reason, reverberation formulas of indoor spaces cannot be used when predicting forest reverberation. As a matter of fact, because the level decay curve of the reverberant sound produced in a forest is nonlinear, the concept of reverberation time is inapplicable. We clarified that reverberation of a forest is expressible as a Wiener process (Brownian motion), which is a stochastic process, and we derived the formula of reverberation of a forest based on that fact. In this study, Monte Carlo simulation was also carried out. Results clarified the following. (1) The energy decay curve of the reverberation sound of a forest becomes a straight line on the logarithmic scale of time and energy; its line gradient is −2. (2) At a point far from a sound source, the reverberant sound energy increases first and begins to decrease after some period. (3) When the mean-free path is short, the sound level of reverberation becomes high. Furthermore, we devised some methods of generating reverberation in outdoor space using the forest reverberation mechanism. For example, a proposed reverberation wall can produce reverberations using only one wall incorporating sound scatterers.
Large outdoor performance venues: Acoustical lessons learned
Authors: Cavanaugh William J. , Hoover K. Anthony
Published: October 2013

The Tweeter Center for the Performing Arts (formerly Great Woods) in Mansfield, MA, off Interstate 495, midway between Boston and Providence, accommodates 19900 patrons and offers 75 to 85 performances each year between 15 May and 15 October. The Pittsburgh Symphony Orchestra inaugurated Tweeter Center’s first concert season in June 1986, and The Center hosted the PSO as its summer home for three additional seasons. The Center has hosted performances from solo comedians to symphonic orchestras and rock bands, featuring such diverse acts as the Boston Pops, James Taylor, Phish, and The Kinks. The acoustical lessons learned over two decades of operation are described. The Center has served as a model for the design and operation of facilities in the 20000-capacity range, including the often difficult issues of controlling excessive amplified sound transmission to residential neighbors, as well as assuring good listening conditions for patrons and performers alike.
The Jay Pritzker Music Pavilion at Millennium Park, part 1: Design and construction
Authors: Talaske Richard
Published: October 2013

The acoustic and audio system design for the Pritzker Pavilion represents a leap forward in the outdoor listening experience. The venue offers superior sound experiences to audiences of 11000 using audio reinforcement and acoustic enhancement systems to support all types of musical performances, particularly for the resident Grant Park Music Festival orchestra. With reliance on the audio systems to project and envelop the audience in sound, the design of the stage enclosure was able to focus on the acoustic quality for the ensemble, a rare opportunity in outdoor facilities. The stage enclosure design includes variable acoustic devices and has an emphasis on intraensemble communication. This presentation will discuss the design of the stage enclosure, the trellis-supported audio systems, and the ways in which the facility is used by resident and visiting musical performers.
The Jay Pritzker Music Pavilion at Millennium Park, part 2: Research and testing
Authors: Miller Gregory , Talaske Richard , Harrison Byron , Way Evelyn , Wang Lily , Bradley David , Johnson Craig
Published: October 2013

The design of the Pritzker Pavilion includes a number of innovative acoustic and audio elements. Two of these elements, the resilient orchestra riser system and the acoustic enhancement system, will be discussed in this presentation. Research previously presented to the ASA [Hoffman et.al., 141st meeting of the ASA) was used in the design of the orchestra riser system at the Pritzker Pavilion. This presentation discusses the field tests that were conducted to assess vibration transfer across the surface of the riser system. Comparison is made to previous research and anticipated results. Subjective evaluations by orchestra members will also be discussed. The acoustic enhancement system is a specialized audio system, separate from the main reinforcement audio system, based on LARES digital signal processing and a distributed loudspeaker approach. Different settings have been developed over time to provide varying decay signatures that are employed on a music-program-dependent basis. Impulse response measurements were taken of unamplified sound, the reinforcement system, and the combination of the reinforcement system with the enhancement system. Testing was conducted with faculty and students from the University of Nebraska.
Acoustical design of the ‘‘Tenseishinbikai Hakko Shinden’’ Shrine in Shiga
Authors: Ikegami Masayuki
Published: October 2013

A shrine called the ‘‘Tenseishinbikai Hakko Shinden’’ was designed and constructed. It covers a large area and has seating for 3690 people. It is used mainly for religious ceremonies and occasionally for concerts. This report outlines its acoustical design and noise control measures. The acoustics design specifically addressed reverberation control appropriate for its use, in addition to echo-nuisance prevention. The flat fan shape of the building often engenders insufficient initial sound reflection and generates echoes. To avoid these problems, in consultation with the designer, an irregular reflective surface was used for the side walls, a rib-shaped acoustic absorption surface, and a folding-screen shaped diffusion surface was used for the rear wall. In addition to conventional machine room noise control, particular attention had to be devoted to glass walls, rain noise, etc. For the glass walls, the effects on the outside (sound generated in the event room) and the effects on the inside (outdoor traffic noise) had to be controlled. To control rain noise, the sound generated by rain on the metal roof had to be predicted; a suitable roof material was selected.
Sound around the New England Patriots’ stadia
Authors: Hoover Anthony K. , Carballeira Andrew , Moore Matthew
Published: October 2013

The New England Patriots wanted to build a new football stadium in Foxboro, MA, adjacent to the old one which would be demolished after the new one was opened. The Environmental Impact Statement (EIS) indicated that one benefit would be a 3.34-dB reduction of stadium noise at the nearest town line because the distance would increase by 980 ft. A procedure to test the resultant noise reduction was developed with keen interest from the Massachusetts Department of Environmental Protection. Touring rock concert sound systems would output high levels of pink noise, twice during sound-checks in the old stadium and then twice in the new stadium; resultant levels would be measured at several locations around the old stadium and then again around the new stadium. This paper will summarize the results of the measurements and will discuss interesting aspects of a test procedure designed by committee, including the importance of declining to be responsible for certain expectations of an EIS.
Improving sound qualities at the seats under deep balcony overhangs
Authors: Chiang Wei-hwa , Lin Wei , Wu Kun-tai
Published: October 2013

Source-receiver distances in a large hall could be reduced by increasing the balcony overhangs, which, however, may cause insufficient acoustical energy for the under-balcony seats. Besides, the reduction in the sense of reverberation cannot be easily mitigated. In this research various design strategies, including utilizing the technique of a flying balcony, were examined in a 1550-seat hall with deep balcony overhangs. Both BEM simulation and scale model measurements were performed. Optimizing the profile of the under-balcony space and utilizing ceiling and side reflectors near the platform were effective in enhancing early reflection. This is especially meaningful for high-frequency string sounds that radiate against the ceiling. Opening the rear part of the balcony that formed a partially flying balcony could compensate late reflections. Preliminary scale modeling results showed that approximately 15% increase in early decay time can be achieved with carefully designed ceiling geometry when the opening area was only 7% of the overhang area.
Art Myrberg’s contribution to the study of hearing in fishes
Authors: Fay Richard R.
Published: October 2013

Art Myrberg was trained to study behavior from an ethological perspective. When it came to studying the hearing of the vocal and communicating damselfish species of the genus Eupomacentrus, he surprisingly decided to use conditioning methods in combination with the best technology then known to produce an excellently designed animal psychoacoustic (laboratory) study that remains today, 26 years later, an exemplary model for students in this field [A. A. Myrberg and J. Y. Spires, J. Comp. Physiol. 140, 135–144 (1980)]. First, Myrberg well understood that sensitivity to acoustic particle motion was possible for these species, so he produced or simulated sound in the far, free field using a long tube with sound absorbers at the opposite end, and measured both sound pressure and acoustic particle motion. He documented that Eupomacentrus was sensitive to particle motion at 100 Hz, but was sound-pressure-sensitive at higher frequencies. He also understood that ambient noise was an important issue in hearing studies, and he did all that he could to measure and reduce it. The result was a ‘‘V’’-shaped audiogram with greatest sensitivity at 500 Hz, and almost no interspecific variation. We can have great confidence in these audiograms today as valid measurements of hearing ability.