bluuebunny/crossref_metadata_2025 huggingface dataset:
Speech enhancement during analysis-synthesis by harmonic plus noise model
Authors: Lehana Parveen K. , Pandey Prem C.
Published: October 2013

Speech analysis-synthesis in codecs and voice transformation often introduces distortion in the resynthesized output, particularly if the input is corrupted by noise. The harmonic plus noise model (HNM), particularly suited for voice transformation, divides speech signal into harmonic and noise parts. The harmonic part is synthesized by taking the summation of pitch harmonics up to a frequency called maximum voiced frequency, and the spectrum above this frequency is attributed to the noise part. Removal of nonharmonic components in harmonic part may result in noise rejection, while the errors in the estimation of harmonics may lead to distortion. Investigations were conducted to assess the speech enhancement obtainable through HNM on broadband noise corrupted speech signal. Listening tests showed that both the input and the resynthesized signals were perceived as distortion-free for SNR greater than 10 dB. In the 0–10-dB range, both the signals had no loss of intelligibility, but listeners rated the resynthesized signal as having higher quality. No enhancement was obtained for SNR values lower than 0 dB. Hence, for higher noise levels, spectral subtraction or other enhancement techniques should be employed along with analysis-synthesis.
An evaluation of hands-free voice activity detection algorithm based on weighted cross-power spectrum phase analysis and zero crossing detector
Authors: Tanaka Takamasa , Denda Yuki , Nakayama Masato , Nishiura Takanobu
Published: October 2013

VAD (voice activity detection) is necessary for recognizing hands-free speech. However, conventional VADs could not achieve enough detection performance in real noisy environments because they utilize only time-sequence information or only spatial-sequence information of captured signals. To cope with this problem, we have already proposed noise-robust hands-free VAD by composing the time-sequence and spatial-sequence information. The conventional ZCD (zero crossing detector) was employed to acquire the time-sequence information, and weighted CSP (cross-power spectrum phase) analysis that was a specialized DOA (direction of arrival) estimation algorithm for speech signal was also employed to acquire the spatial-sequence information. However, this algorithm can insufficiently improve the VAD performance in higher noisy environments. To overcome this problem, in this paper, we try to utilize talker direction information and attempt the environmental adaptation of our former algorithm. As a result of evaluation experiments in real acoustic environment, the proposed hands-free VAD could achieve a more effective performance than the conventional VAD and our former VAD. [Work supported by MEXT of Japan.]
A study of robust omnidirectional audio-visual talker localization algorithm with microphone array and omnidirectional image
Authors: Denda Yuki , Nishiura Tanaknobu , Yamashita Yoichi
Published: October 2013

In video conferencing environments, it is very important to localize the talker. However, conventional audio signal-based algorithms often suffer from audio interference, and conventional visual signal-based algorithms fail in the presence of visual interference. To deal with these problems, this paper proposes a robust omnidirectional audio-visual talker localization algorithm that not only exploits audio feature parameters, but also subordinately uses visual feature parameters. To achieve omnidirectional audio-visual talker localization, paired-omnidirectional microphones are employed as an audio sensor, and an omnidirectional camera is employed as a visual sensor. For robust talker localization, audio feature parameters are extracted using weighted cross-power spectrum phase (CSP) analysis and CSP coefficient subtraction, and visual feature parameters are extracted using background subtraction and skin-color detection. The talker is finally located by the fusing of weighted audio/visual feature parameters, and the weight of this feature parameter fusion is automatically controlled based on the reliable criterion of audio feature parameters. The results of localization experiments in an actual room revealed that the proposed audio-visual talker localization algorithm is superior to that of conventional localizers using only audio or visual feature parameters, but not both. [Work supported by MEXT of Japan.]
Noisy speech recognition using modification of spectral envelopes and smoothing and emphasis method by image filters
Authors: Midorikawa Yoichi , Akita Masanori
Published: October 2013

The purpose of this study is to propose an improvement of the speech recognition under the noisy environment. Automatic speech recognition (ASR) systems are most effective in noiseless environments. We propose a method for ASR with signals contaminated with colored noise using modification of the spectral envelope shape. Our previous study has proposed a method of speech recognition for the signals polluted with color noise by modifying spectral envelope shape by rules. The proposed method is based on cepstral analysis. We have proposed the modified rules for adding valleys are defined. These methods are applied to the enhancement system of noisy signals and the performance of the system is improved. However, these rules only make use of frequency domain. Therefore, the cepstral coefficients of neighboring frames become independent of time domain. In this study, we apply the smoothing and emphasis method by image filters to the modification of spectral envelope shape. As a result, performance of speech recognition is improved.
Large-scale continuous speech recognition system design using discriminative training
Authors: McDermott Erik , Nakamura Atsushi
Published: October 2013

Discriminative training is difficult to implement but essential to attaining state-of-the-art performance in automatic speech recognition systems today. Most of the discriminative training results for large scale recognition tasks (with vocabularies well over 10000 words) so far use the maximum mutual information (MMI) framework, but recent results for the minimum classification error (MCE) framework suggest that MCE too yields significant improvements in recognition accuracy and system compactness on large-scale tasks. MCE embodies rather well the general intuition that recognition system design should attempt to improve performance (i.e., recognition accuracy) directly, by optimizing a criterion function that is closely related to performance, rather than indirectly, by optimizing a criterion such as overall log likelihood that does not reflect performance. This presentation provides an overview of the MCE framework, and describes recent MCE speech recognition results obtained with both the MIT Galaxy system and the NTT Communication Science Labs SOLON system. The tasks examined include a 33000-word vocabulary telephone-based spontaneous speech weather information task, a 22000-word telephone-based name recognition task, and a 100000-word Japanese lecture speech transcription task.
Acoustic modeling with contextual additive structure for hidden Markov model-based speech recognition
Authors: Nakamura Kazuhiro , Zen Heiga , Nankaku Yoshihiro , Tokuda Keiichi
Published: October 2013

This study proposes an acoustic modeling technique based on an additive structure of context dependencies for HMM-based speech recognition. Context dependent models, e.g., triphone HMMs, are widely employed in speech recognition systems. Generally the dependency between phonetic contexts and acoustic feature distributions is direct, i.e., if a context is given, the Gaussian distribution is specified immediately. However, more complex dependencies can be considered. This study assumes an additive structure of acoustic feature components that have different context dependencies. For example, if the left and right phones are assumed as contexts of additive components, the generative process of the triphone feature is as follows: the additive components of left and right contexts are generated independently from each distribution and then added to the monophone feature. Since the output probability distribution is composed of the sum of the mean vectors and covariance matrices of additive components, a number of different distributions can be efficiently represented by a combination of fewer distributions. However, it is unknown which kinds of contexts have additive dependencies to acoustic features. Therefore, this study proposes a context clustering algorithm for the additive structure model. The proposed algorithm automatically extracts additive components by constructing multiple decision trees simultaneously.
Robust speech recognition for the control of wheelchairs by inarticulate speech of the severely disabled
Authors: Kojima Hiroaki , Sasou Akira , Suk Soo-Young , Lee Shi-Wook , Sadohara Ken
Published: October 2013

The goal of this research is to develop voice-controlled wheelchairs that can be operated by inarticulate speech affected by severe cerebral palsy or quadriplegia, for instance. In this case, principal factors obstructing recognition performance are significant pronunciation variation caused by difficulty in stable articulation and bad influences of variety of noise in the real environment. To cope with the pronunciation variations, pronunciation lexicons that consist of multitemplates of reference patterns are utilized. The reference patterns are represented with subphonetic codes, which can describe variations of inarticulate speech more precisely than ordinary phonetic transcriptions. Pronunciation lexicons are generated by generalizing coded samples into compact representation of templates based on DP and data mining. For noise robustness, a voice activity detection method is investigated in order to circumvent friction of microphone, cough, etc. A sound source localization using a microphone array is also integrated in order to reject sounds from outside of the wheelchair. These methods are integrated into a system that can be mounted on a wheelchair. A usability test operated by a severe cerebral palsy patient in the real environment results in 95.8% accuracy within 1404 samples for the recognition of the five-command set. [Work supported by MEXT.]
Selection of recognition candidates based on parallel recognition of sentences and partial words in spoken dialogue
Authors: Kawakami Makiho , Terashi Hiromasa , Nishida Masafumi , Horiuchi Yasuo , Ichikawa Akira
Published: October 2013

We have proposed a method that predicts a user’s utterances in spoken dialogue systems by recognizing prediction sentences of each dialogue state, thereby decreasing recognition errors. However, the conventional method might repeat recognition errors because it confirms the whole sentence even if it recognizes only a part of a long word, such as a compound word, correctly. For this study, we introduce a method using decoders based on prediction sentences and partial words obtained by dividing long words. The proposed method can confirm a user’s utterances by selecting a candidate to a partial matched word using recognition results of the prediction sentence and a divided partial word. We conducted experiments for 560 utterances of seven persons using a navigation system. Results showed that the word recognition accuracy was 81.1%. The rate at which recognition results by the prediction sentence and a divided partial word were an exact or partial match was 70.5%. The rate of recognition errors when the recognition results were an exact or partial match was 0.9%. Therefore, we demonstrated that it is possible to confirm a user’s utterances using the proposed method by selecting recognition candidates.
Multiple utterance prediction based on a tree structure of dialogue states
Authors: Terashi Hiromasa , Nishida Masafumi , Horiuchi Yasuo , Ichikawa Akira
Published: October 2013

A problem of spoken dialogue systems is that recognition errors increase when the recognition-word vocabulary is large. To solve the problem, we propose a method to predict a user’s utterances by recognizing prediction sentences of each dialogue state. The method can judge whether the user’s utterance is within the range of prediction using decoders based on a prediction sentence and a large word vocabulary. However, the conventional method performs processing to lead in a prediction frequently when a topic changes because even an utterance within a domain is treated as an utterance of the prediction outside. In this study, we propose a new method using multiple predictions by performing recognition based on all prediction sentences in a domain. The proposed method produces a tree structure for dialogue states in a domain and regards child nodes of an arbitrary node as within the range of prediction. We conducted experiments for 451 utterances of 15 dialogues by ten persons using the university guide system. As a result, recognition accuracy of prediction sentences was 97.7% and the judgment accuracy of utterance predictions was 88.2%; high performance was obtained using the proposed method. Results demonstrate that it is possible to perform flexible dialogue control.
Document-level optimization in speech recognition
Authors: Nakazato Rie , Sadamitsu Kugatsu , Yamamoto Mikio
Published: October 2013

Most ASR systems optimize scores at the sentence level because language models are designed to assign a meaningful probability to a sentence, but not to a document. However, in the last half-decade, NIPS people have specifically examined aspect-based global language models and developed generative text models such as the latent Dirichlet allocation (LDA), which can assign a meaningful probability to an entire document. We investigated a system to recognize all sentences in a read document, maximizing a global score including the document probability, in addition to assigning acoustic and local language probabilities. Document level optimization using global scores engenders a combinatorial problem in its implementation. Therefore, we used the rescoring framework with N-best recognition results for each sentence in a read document and the hill-climbing method to search for approximately optimal recognition results. In this study, we compare mixture of unigrams (MU), Dirichlet mixtures (DM), and LDAs as generative text models in experiments using read document speech data in Japanese Newspaper Article Sentences (JNAS) and academic presentation speech data in The Corpus of Spontaneous Japanese (CSJ).
Generating search query in unsupervised language model adaptaion using WWW
Authors: Kajiura Yasutomo , Suzuki Motoyuki , Ito Akinori , Makino Shozo
Published: October 2013

To improve the accuracy of an LVCSR (large vocabulary continuous speech recognition) system, it is effective to gather text data related to the topic of the input speech and adapt the language model using the text data. Several systems have been developed that gather text data from World Wide Web using keywords specified by a user. Those systems require the user to be involved in the transcription process. However, it is desirable to automate the entire process. To automate the text collection, we propose a method to create an adapted language model by collecting topic-related text from World Wide Web. The proposed method composes the search query from the first recognition result, and it gathers text data from the WWW and adapts the language model. Then the input speech is decoded again using the adapted language model. As the first recognition result contains recognition errors, we developed a method to exclude the misrecognized words using word-based confidence score and similarities between keywords. To evaluate the proposed method, we carried out adaptation experiments.
A method for estimating the degree of a speaker’s anger using acoustic features and linguistic representation
Authors: Arimoto Yoshiko , Ohno Sumio , Iida Hitoshi
Published: October 2013

Automatic speech recognition (ASR) systems are greatly demanded for customer service systems. With advanced interactive voice response systems, humans have more opportunities to have dialogues with computers. Existing dialogue systems process linguistic information, but do not process paralinguistic information. Therefore, computers are able to obtain less information during a human-computer dialogue than a human can during a human-human dialogue. This report describes a study of the estimation method of degree of speakers’ anger emotion using acoustic features and linguistic representation expressed in utterances during a natural dialogue. To record utterances expressing the users’ internal anger emotion, we set pseudo-dialogues to induce irritation arising from discontentment with the ASR system performance and to induce exasperation against the operator while the user makes a complaint. A five-scale subjective evaluation was conducted to mark each utterance with a score as the actual measurement of anger emotion. As a result of this, an emotional speech corpus was produced. We examine the acoustic features and features of linguistic representation of each utterance with reference to these anger score. Then we conduct experiments to estimate automatically the degree of anger emotion using parameters selected from those features.
Towards the detection of potentially hazardous situations in vehicle traffic using driver speech and brake pedal operation
Authors: Malta Lucas , Miyajima Chiyomi , Itou Katsunobu , Takeda Kazuya
Published: October 2013

A method for automatic detection of potentially dangerous situations in motor vehicle traffic is introduced. Unlike precedent works, which typically relied on camera arrays or road-traffic monitoring sensors to detect collision incidents, the proposed approach specifically incorporates changes in a drivers’ behavior, detected through driver speech and brake pedal operation. Experiments were performed using a large real-world multimedia driving database of 493 drivers, obtained from the Centre for Integrated Acoustic Information Research (CIAIR, Nagoya University). The drivers, who interacted verbally with a human operator, uttered expletive words to express negative feelings in 11 of the 25 situations that we selected as potentially hazardous. In 17 of them, sudden and intense compression of the brake pedal was observed. The proposed lexicographical speech-feature-based method also detected 33 false alarms to detect 80% of these 11 scenes. As for the other 17 scenes, our method based on two-dimensional histograms of brake pressure and its dynamics achieved an 80% detection rate for 473 false alarms. Analyses of data recorded while drivers interacted with a machine and a Wizard of Oz system as well as a rank of the most commonly uttered words in dangerous situations are also presented.
Computational complexity of a distance-based active search algorithm
Authors: Sugiyama Masahide
Published: October 2013

An efficient pattern (segment) search in a large-scale database is a very important technology in current widespreaded Internet society. To achieve efficient searches, many efficient pattern search algorithms have been proposed. Active search (AS) is one of them; it is based on skipping of search calculation. Initially, AS was formulated between two histograms using a similarity measure. In fact, AS can be formulated using L—p distance, where p is greater than 1. The search efficiency can be evaluated using a number of distance calculations. The first result in this paper shows that the L—1 distance in AS gives the minimum number of distance calculations among L—p distances. This property can be derived from a newly derived inequality between the L—p and L—1 norm on output probability space. A new distance can be derived from positive weighting of any distance. The second result of this paper is that the average skip width for any weighed L—p distance cannot be greater than the most efficient distance. Therefore, no positive combination of L—p can be more efficient than L—1. The two results indicate that L—1 distance is the most efficient for active searching.
Recovering the quality of speech degraded by reverberations in a room
Authors: Miyoshi Masato
Published: October 2013

Speech signals captured by distant microphones in a room are usually reverberated due to wall reflections. Reverberation may seriously deteriorate the signal characteristics, thus damaging the quality of such applications as hands-free telephony or automatic speech recognition. To eliminate such room reverberation effects, the inverse-filtering of the room transfer functions (RTFs) between a speaker and the microphones appears to be a very promising approach. When applied to a microphone system, this processing should work blindly because no explicit information on RTFs or source signals is available. The blind design of an inverse-filter set may be roughly classified into three approaches: (1) calculating an inverse-filter set of RTF estimates obtained from observed reverberant signals based on the ‘‘subspace method,’’ (2) calculating the filter set from observed signals and replicas of direct-sound signals using statistical properties inherent in speech signals, and (3) calculating the filter set directly from received signals, by whitening these signals without excessive degradation of the original signal characteristics. This presentation will provide an overview of these three approaches and introduce our speech dereverberation trials.
Multilayered spatio-temporal gradient analysis for acoustic blind source separation
Authors: Teramoto Kenbu , Khan Md. Tawhidul Islam , Torisu Seiichirou , Uekihara Akito
Published: October 2013

A novel blind source separation of a mixture of two or more voice signals has been proposed in the present paper. The separation system has been focused based on the spatio-temporal gradient analysis. The proposed algorithm utilizes the linearity among the signals: sound pressure of source signals, the three-dimensional (x, y, and z directional) particle velocity vector, and its gradient of the observed signals, all of which are governed by the equation of motion. Principally, as the mechanism of blind source separation uses no- priori information about the parameters of convolution, filtering as well as mixing of source signals, some simple assumptions such as the statistical independency of the linearly combined (mixed) observed signals containing zero mean as well as unit variance have been implied in the present separation algorithm. Therefore, the proposed method has successfully simplified the convoluted blind source separation problem into an instantaneous blind source separation problem over the spatio-temporal gradient spaces. An acoustic experiment with two female voices has been carried out to compare the simulated data as well. A eight-microphone array system has been adopted to evaluate the voice signals efficiently.
Study for blind source separation on time-frequency domain considering phase information
Authors: Sasaki Fumio , Yasuoka Masahito , Tanaka Osamu
Published: October 2013

The methods that can be done for the estimation of the number of source signals and separation of source signals using time-frequency information are proposed as a method of blind source separation. In these methods, some independent conditions are assumed in a time-frequency domain, and observed signals are expanded to time-frequency domain using wavelet analysis. However, these methods are not considered phase differences (difference of distance) between source signals and observed signals. A new method is proposed. The method can be done not only for the source separation but also the specification of the locations of source signals considering phase differences. In this method, it is necessary to assume a rather stronger independent condition than former methods. But, on account of this assumption, a function that becomes a real value only when phase differences are coincidence in time-frequency domain is determined. Using this function, first of all, the number of source signals is calculated, then the locations of source signals and source signals are calculated. The effectiveness of the method is shown using actual synthetic signals.
Single element synthetic aperture using an ocean glider
Authors: Dossot Georges A. , Miller James H. , Moore Kristy A. , Crocker Steven E. , Holmes Jason D. , Glenn Scott
Published: October 2013

The feasibility of using a single transducer element on an ocean glider to create a synthetic aperture is discussed. Acoustic data were collected on two Webb Slocum gliders deployed by Rutgers University during the Shallow Water Experiment (SW06) on the continental shelf off New Jersey. These gliders periodically surfaced for GPS fixes and data transfer via satellite phone. A synthetic aperture is created through coherent processing of the acoustic data as the glider travels through the water. A number of issues including changes in depth, nonsteady motion of the glider, and clock drift can limit the performance of the processing. However, the glider provides a low-noise and low-speed platform, potentially improving the signal-to-noise ratio. The geometry of the experiment provided for near-broadside recording of the 224- and 400-Hz WHOI tomography sources. An acoustic normal mode representation of the field provides the basis for processing of the data similar to the Hankel transform approach of Frisk and Lynch [J. Acoust. Soc. Am. 76, 205–216 (1984)] and Holmes et al. [J. Acoust. Soc. Am. 119, 3346 (2006)]. Effects of spatial variations in sound speed are examined. [Work sponsored by the Office of Naval Research.]
Processing data from a low-drag array towed by a glider
Authors: Hursky Paul , Porter Michael B. , Siderius Martin , McDonald Vincent K. , Gillcrist Mark , Granger Brian , Jones Ryan , Bratten Aaron , Huizinga Andy , Sullivan Peter T. , Briest Susan G.
Published: October 2013

Small underwater robotic vehicles such as gliders and AUVs have limited capabilities in terms of propulsion, speed, and power consumption, so it takes great care to design suitable sensor systems such as towed arrays for these platforms. Two such arrays were deployed during the Makai experiment, off the coast of Kauai in Hawaii. A 15-element line array with an acoustic aperture of 21 m was towed by a Webb Research Slocum glider. A 40-element line array with an acoustic aperture of 60 m was towed by a small work boat. We will describe the design and construction of these arrays and present results of processing data recorded on these arrays. We will also discuss potential applications for these arrays deployed from gliders and AUVs.
Higher frequency matched-field processing
Authors: Debever Claire , Kuperman William A.
Published: October 2013

We study matched-field processing (MFP) in the 3.5-kHz shallow water regime at a range of 4 km for which MFP is typically problematic. Data are used to construct both the cross-spectral-density-matrix (CSDM) and the weight vectors for the Bartlett, minimum variance distortionless response (MV), and white noise constraint (WNC) processors. There is sufficient data to construct the weight vectors as a function of time as measured around the CSDM construction time. While the MV processor degrades significantly over a few minutes, the WNC processor remains stable for 15 min or longer. We study these experimentally derived weight vectors to further understand the origin of the increased stability of the WNC over the MV and to gain further insight into how to construct a robust MFP processor. [Work supported by ONR.]
Steering vector sensor array elements with cardioids, hippopedes, and other beam shapes
Authors: Smith Kevin B. , Van Leijen A. Vincent
Published: October 2013

The purpose of this report is to study the impact on linear beamforming due to individual vector sensor element steering patterns. Standard, linear beamformers employ cardioid beampatterns for each vector sensor. In this work, we examine a class of vector sensor element steering patterns beyond the standard cardioid. The element weighting is nonlinear but nonadaptive, making it simple to implement in hardware processing. The new sensor steering patterns, referred to as hippioids, are products of cardioids and various orders of hippopedes. The angular resolution of individual sensors, and the impact on angular resolution from arrays of varying aperture, will serve as the performance measure along with peak-to-sidelobe levels. An example of the differences in vector sensor steering patterns is provided using measured, DIFAR buoy data. [Work supported by the Office of Naval Research, Code 321OA, the Naval Postgraduate School, and the Netherlands Defense Academy.]
Wideband acoustic communication using quadrature-phase shift keying and eight-phase shift keying
Authors: Ochi Hiroshi , Watanabe Yoshitaka , Shimura Takuya
Published: October 2013

In JAMSTEC, a short-range (up to 500 m) high-speed acoustic communication system for deep sea environment has been investigating. To widen a bandwidth, the new wideband projector was developed. It consists of two transducer elements and has a toroidal beam pattern. To reduce the acoustic radiation pattern of the rear side, the unit, which is like a lampshade, was attached to that projector. Moreover, the beam axis is tilted to the upper side by delayed input signal to the upper transducer element. When this projector is used in the vicinity of the sea bottom, a bottom-reflected wave can be reduced. Carrier frequency of this projector is 80 kHz, and its bandwidth is 40 kHz. The experiment was carried out at the depth of 1000-m area using this projector. Two omni-directional hydrophones were used for receiving. Quadrature phase shift keying (Q-PSK) and 8-PSK were used as its modulation method. The data of several transmission speeds [10 k symbol per second (sps), 12.5, 16, 20, 25, and 40 ksps] were obtained at the transmission range of 200–230 m. Under those conditions, error free communication was carried out for 60000 bits of data.
Continued Haro Strait path data processing
Authors: Tolstoy A. , Michalopoulou Z.-H.
Published: October 2013

Previous efforts have led to the estimation of candidate source-array (S-R) geometries for some Haro Strait data. In particular, nw014 (one broadband shot to the NW array) has been extensively analyzed with the result that source range, source depth, array phone localizations, water depth at the source (D1), and water depth at the array (D2) can now be restricted to sets based on the time-domain data alone, i.e., on the signal boundary reflections. Most recent efforts include FFTs of the data followed by MFP geoacoustic inversion at multiple high frequencies (480–520 Hz)—using the restricted sets—in order to estimate array depths and shapes as well as some of the bottom properties for the nw014 data. Array shape and phone localizations are extremely important but difficult to attain. This presentation will describe these new efforts.
Wave front design by acoustic contrast control
Authors: Choi Joung-Woo , Kim Yang-Hann
Published: October 2013

A way to design and generate desired wave fronts using multiple sound sources is proposed. In principle, any kind of wave front can be described as a sum of orthogonal basis functions, and the wave front of a specific shape can be generated by focusing multiple sources’ energy on a single basis function. Once the sound field is decomposed into a set of orthogonal functions, we can apply a conventional focusing algorithm to generate a desired wave front. Extending the concept of energy focusing, we can also generate a wave front packet, whose energy is concentrated on a group of orthogonal functions rather than single component. This enables us to generate a wave front within a desired bound. For this purpose, we employ acoustic contrast control, which focuses sound energy on a selected group of orthogonal functions by enhancing the energy difference between the focal region and others. Various example cases demonstrate how the proposed method can generate a group of planar and spherical wave fronts propagating to different directions. [Work supported by the NRL project of KISTEP and BK21 project initiated by Ministry of Education and Human Resources Development of Korea.]
A new optical-fiber-based underwater sensor
Authors: Guillot Francois M. , Trivett D. H. , Rogers Peter H.
Published: October 2013

A new type of underwater sensor based on optical fiber Bragg gratings is presented. This transducer can be used in a dipole or a quadrupole (shear sensor) configuration. The sensing principle of the device relies on the interference signal from two Bragg gratings written on the same fiber and illuminated by a tunable, narrow-band light source. The gratings are 5 cm apart, and they each reflect a portion of the incident light. The fiber is epoxied to two spacers separated by a 1-mm gap situated between the gratings. This assembly is then adhered to two plates connected by a hinge, which is located below the gap. One plate is held rigidly and the tip of the other (free) plate experiences transverse vibrations, when ensonified. These vibrations produce periodic gap length changes, which modulate the interference signal from the two gratings. The modulation is related to the amplitude of the sound wave and is monitored with a photodetector. Design optimization to increase the sensitivity and to reduce the noise floor is discussed, and the performance of a practical sensor is presented. [Work supported by ONR.]
Sound system design for acoustically challenging spaces
Authors: Mochimaru Akira , Hayashi Soichiro
Published: October 2013

Sound reinforcement (SR) systems take a critical part in controlling acoustic performance in large venues. Loudspeaker system design including consideration of room acoustics becomes crucial in providing audio information with good intelligibility and natural tonal balance throughout the audience area. There are many elements that can spoil acoustic performance of the sound system in large spaces. Relatively long reverberation time, number of speaker clusters with distances in between, size of each speaker cluster that creates unexpected phase cancellations and so on. Neither the single speaker product nor sound system design based on experiences can provide a perfect solution for these kinds of spaces. On the other hand, an advanced acoustic simulation system including consideration of relative phase between sound sources with capability of listening to the simulated results is now available. By using this technology with academically proven accuracy, speaker array design methodology called bass-array was produced. Bass-array allows the sound system designer to control radiation directivity characteristics of low frequency. Design of custom line-array to suit room configuration of the space is another deliverable of the technology. This technology in association with bass-array or line-array has been successfully applied for providing appropriate acoustic solutions in a number of acoustically difficult spaces.
Recording in an outdoor performance space
Authors: Harght Pamela
Published: October 2013

For both audio engineers and musicians, dealing with indoor and outdoor spaces poses several types of problems for recording and performance purposes. While recording indoors poses many problems, most of them (pertaining to room design, ventilation, placement of musicians) are controllable. However, dealing with outdoor spaces creates many issues entailing uncontrollable situations, specifically those entailing the environment. During the summer of 2006, the author spent 8 weeks at the Tanglewood Music Center in Lenox, MA on an audio engineering fellowship. Tanglewood is the summer home for the Boston Symphony Orchestra; it has both an outdoor performance space and a concert hall that opens for additional lawn seating. This paper discusses and compares the many issues of recording classical music outdoors as well as indoors, based upon experience, research, and interviews with musicians and members of the audio department.
Acoustics for the one percenter
Authors: Pfeiffer Scott
Published: October 2013

GMC adopted the one percenter from outlaw bikers who took pride in identifying themselves as the 1% of bikers that were making a bad name for the rest of the 99% that were law abiding. GMC and the outlaw bikers had the right idea. We all like to identify ourselves with the few, the proud, those that appreciate the special, the esoteric, the pursuit of greatness. Large venue projects present a special challenge to reaching these goals as the issues of getting 5000 or more people in and out begins to drive the design, natural acoustics become challenging or impossible, and the distribution of audio and video begins to depend on delay clusters and secondary distribution, which all begin as compromises. Doubt enters as the question is raised of how many patrons will notice. It is in these moments when finding the design that serves the one percenter makes all of the difference. Special examples of one percenters will be presented as a retrospective of large venues honoring the most successful projects in this type.
A tribute to Art Myrberg
Authors: Stadler John H. , Gruber Samuel L.
Published: October 2013

Arthur A. Myrberg, Jr. passed away in April, 2005, and will be remembered by many as a pioneer in the fields of fish bioacoustics and behavior. His distinguished career spanned more than 40 years, and saw him become a world-recognized expert in these fields, publishing more than 90 peer-reviewed articles, some of which are considered seminal. During more than 35 years as a professor of Marine Science at the Rosenstiel School of Marine and Atmospheric Sciences, University of Miami, he was a well-respected and admired scientist, teacher, lecturer, mentor, and advisor to 24 graduate students. This presentation will trace Art’s life from his days as a youth in Lulenecks, Illinois, through his college career and his work with Nobel laureate Konrad Lorenz at the Max Planck Institute, Seiweisen, Germany, culminating in his long career at the University of Miami.
A history of fish bioacoustic studies
Authors: Popper Arthur N. , Fay Richard R. , Tavolga William N.
Published: October 2013

Interest in fish sound production and hearing dates back to the writings of Aristotle (384–322 BCE) and Pliny the Elder (23–79 CE), both of whom noted not only that fish make sounds, but also speculated that fish must be able to hear despite their not having apparent ears. The fact that fish could hear was first demonstrated with the work of G. H. Parker in 1903 and later by Karl von Frisch. Studies on sound production were sporadic in the early part of the 20th century and often based on anecdotes until technology derived from antisubmarine warfare became available after World War II. This equipment enabled investigators to listen to fish and correlate sounds with behavior. It was not until the early 1960s that intensive investigations started on fish hearing. This paper will review the history of the study of fish bioacoustics and discuss the contributions of a large number of investigators including those of Arthur Myrberg. Indeed, Art’s research stands as a landmark in the field, as well as a fine example of true scholarship. Furthermore, his charismatic presentations of his work were instrumental in stimulating the investigations of other scientists.
Art Myrberg’s contributions to fish acoustic communication
Authors: Fine Michael L.
Published: October 2013

Arthur Myrberg was an ichthyologist and ethologist who trained with the Nobel laureate Konrad Lorenz. Art’s work exceeded the boundaries of communication (a signal produced by an animal for its own benefit) to include related phenomena such as acoustic attraction and repulsion of sharks, effects of noise, and the creation of the term interception to describe a communication signal picked up (e.g., intercepted) by another receiver for its own benefit. His work included a wide range of species (sharks, surgeonfish, damselfish, cichlids, gobies, and even invertebrates and marine mammals). He worked at sea and utilized diving for observation and experimental manipulation and then set up lab colonies for further observation and experimentation. What is perhaps his signal contribution to acoustic communication followed this paradigm in an investigation in four related Eupomacentrus species that produce similar chirps during a courtship dip. By playbacks of natural and altered sounds he and his students separated the subtle cues used by these fishes to permit intraspecific communication. This rather dry abstract fails to capture Art’s love of science, his ability to make outstanding presentations, and my value of his friendship.
Recent advances in fish acoustic behavior
Authors: Tricas Timothy
Published: October 2013

Sounds produced by fish provide important signals or cues in many behavioral contexts. Mechanisms of sound production are diverse and best described in fishes with specialized anatomical structures, but remain tentative for many species. The coproduction of low-frequency hydrodynamic and acoustic stimuli during behavioral interactions can excite the inner ear, lateral line, or both to provide complex acoustico-lateralis information. Recent behavior studies show that acoustic signals function in behavioral displays of aggression, species recognition, courtship, mate attraction, and spawning; in some cases it can function as a single modality signal. Sound stimuli often overlap with the hearing band of highest sensitivity (100–1000 Hz for most species), but in some species these do not match. Much data now exist on the diversity of sounds produced by fishes and environmental acoustic cues, but more evaluation is needed within the context of the animals natural environment. Examples include acoustic cues for recruitment of pelagic fish larvae to reefs, use of ultrasound to avoid predators, and infrasound stimuli. Comparative work on both sound production and sensory reception in fishes can be assessed in relation to their ecology to better understand the evolution of fish acoustic behavior.
Lateral line research: Recent advances and new opportunities
Authors: Coombs Sheryl , Bleckmann Horst
Published: October 2013

Computational fluid dynamics, digital particle imaging velocimetry, and engineered micro flow sensors, along with emerging collaborations between neurobiologists and engineers, have opened up new vistas for lateral line research in recent years. Engineering advances have increased our ability to measure the spatiotemporal characteristics of biologically significant, more complex flow signals and how these signals are sensed and processed by the lateral line. Following in the rich tradition of Art Myrberg’s work on sound communication in fishes, studies on hydrodynamic signal production and use by fish during interactive social behaviors are now feasible. In addition to these new opportunities, several recent studies have enriched the lateral line landscape. The lateral line sensing and encoding of vortex trails shed by swimming fish or obstacles in a stream have been investigated, along with associated behaviors (e.g., vortex trail following and obstacle entrainment). Blind Mexican cavefish have been shown to form cognitive maps of their surroundings using lateral-line based, hydrodynamic cues. Finally, the responses of goldfish auditory and lateral line systems to the same stimulus source have been documented to reveal how these two closely related systems might work in concert to extract complementary but different types of information from nearby hydroacoustic sources.
Investigating hearing in sturgeon: Getting closer to the origin of hearing in vertebrates
Authors: Meyer Michaela , Popper Arthur N. , Fay Richard R.
Published: October 2013

It has been hypothesized that hearing arose early in vertebrate evolution to use sound to glean information about the environment, including the presence of predators and prey. The importance of sound to sense the acoustic scene is especially true underwater where the visual horizon is limited. To create an acoustic scene from sound, an animal needs to localize and identify sound sources. We hypothesize that the sound localization capabilities of primitive bony fishes such as sturgeon are as capable as those of modern bony fishes such as goldfish and, perhaps, many terrestrial vertebrates. We present results from experiments investigating the neural mechanisms of directional hearing and frequency analysis in the auditory periphery of the sturgeon. Extracellular single units were recorded from the eighth nerve during stimulation with a three-axis shaker system that subjects the fish ear to linear accelerations in known directions at various frequencies. This system enables the characterization of both spatial and frequency tuning of the afferents innervating the ear. The results show that similar neural mechanisms for encoding sound direction and frequency exist in the periphery of modern and primitive bony fish, thereby supporting the hypothesis that basic auditory functions arose very early in vertebrate history.
Particle motion perception in cephalopods
Authors: Kaifu Kenzo , Segawa Susumu , Akamatsu Tomonari
Published: October 2013

In the 1980s, there still had been the argument whether cephalopods could detect sound stimulus or not. The ability of sound detecting (hearing ability or an ability to detect vibration) in cephalopods was proved around 1990, however, there have been few studies about sound detection in cephalopods afterwards. Based on morphological and behavioral studies, it seems that cephalopods do not perceive sound pressure. Provided that compressional waves do not stimulate the nervous systems of the animals, they do response to sound waves due to water particle movement. In this study, the ability to detect sound waves in cephalopods was investigated from the standpoint of water particle movement perception. An object would be well vibrated by sound waves when the length of the object is much smaller relative to the wavelength; consequently, when an animal detects water particle movement, the smaller the animal is, the higher frequency and weaker stimulus it would detect.
Acoustical imaging of nonbladdered animals using underwater acoustic camera
Authors: Iida Kohji , Takahashi Rika , Mukai Tohru , Tang Yong , Sato Masanori
Published: October 2013

An underwater acoustic camera using 3.5-MHz ultrasonic waves enclosed in a pressure-resistant case was constructed to observe underwater marine animals. This enabled the measurement of the size, shape, and behavior of living marine animals including internal organs by B-mode images in the detection range up to 240 cm. Observations were conducted for captive animals in a water tank and for natural animals in a field. The captive animals, including fish, squid, and jellyfish, were observed, and a three-dimensional internal structure of animals was reconstructed using multiple acoustical images. The most important contributors of acoustic scattering were the vertebra for nonbladdered fish, and the liver and reproductive organs for invertebrate animals. In a field experiment, the shape, size, and swimming behavior of wild animals were observed. The possibilities and limitations of the underwater acoustic camera for fishery applications were discussed. [Work supported by JSPS Grant-In-Aid for Scientific Research.]