bluuebunny/crossref_metadata_2025 huggingface dataset:
Robust speech parameter based on a psychoacoustical rule-base model
Authors: Nishimura Syunji , Chisaki Yoshifumi , Usagawa Tsuyoshi , Ebata Masanao
Published: October 2005

To realize a speech recognition system for our daily life, surrounding noise needs to be considered. Many studies have been done for parameters which are robust in a noisy environment. This paper presents a new parameter which is based on a psychoacoustical rule-base model. First the spectral local peaks of formants are calculated as a basic parameter. By performing this operation frame by frame, the parameter can be represented in a three-dimensional form like a spectrogram. The parameter is then modified by applying psychoacoustical rules to the sequence of peaks, corresponding to a so-called ‘‘auditory stream’’ in a speech. The parameter modification utilizes the characteristics of the parameter such as the synchronous onset or offset, the continuity of sequential peaks, and so on. The efficiency of the proposed parameter is evaluated on an isolated word speech recognition system with various SNR. As a result, the improvement due to the proposed parameter is at least 4 dB in SNR for white noise, in comparison with the performance of a LPC-based system.
Vowel identification from harmonic contours
Authors: Antrobus John S. , Betancourt Octavio
Published: October 2005

A set of automatic algorithms expresses the acoustic vowel signal as the ratio series, log(fj/F02/3), where fj are the first 32 integer multiples of F0, plus an additional 32 log(fj/F02/3) delta terms that represent vowel trajectories. F0 is measured on a window-by-window basis by an algorithm that eliminates all smearing due to conventional windowing algorithms. In order to reduce the dimensionality of this expression, the 64 terms are summarized by 11 + 11 terms from the cosine series. Using ten monosyllabic words spoken by 137 men, women, and children, vowel classification is within one percent of human accuracy. Because the model uses none of the circular definitions of formant measurement and makes only one assumption that is unique to speech, namely, the −log F01/3 offset, the log(fj/F02/3) contour is superior to a formant representation of voiced speech. Because a Euclidean classifier is as accurate as a quadratic discriminant function which uses more than ten times as many degrees of freedom, it is argued that the log(fj/F02/3) transform may be accomplished by a genetically acquired neural mapping of the acoustic signal that facilitates the learning of vowel categories by infants.
Robust speech recognition using singular value decomposition based speech enhancement
Authors: Lilly B. T. , Paliwal K. K.
Published: October 2005

Speech recognition systems work reasonably well in laboratory conditions, but their performance deteriorates drastically when they are deployed in practical situations where the speech is corrupted by additive noise distortion. One way to improve the performance of a speech recognition system in the presence of noise is to enhance the speech signal (and remove noise) prior to its recognition. Recently, a singular value decomposition based technique has been proposed for speech enhancement [Dendrinos et al., Speech Commun. 10, 45–57 (1991)]. In this technique, singular value decomposition was applied to an overdetermined, overextended data matrix formed from the noisy speech signal and a noise-free, low-rank approximation was obtained by retaining a specific number of singular values. This technique was applied here as a preprocessor for recognizing speech in the presence of noise and was found to improve the recognition performance significantly, especially at lower signal-to-noise ratios. [Work supported by the Australian Research Council.]
Word boundary detection under noise by using an adaptive noise canceler
Authors: Sakamoto M. , Ohshima Y.
Published: October 2005

An adaptive noise canceling technique was applied to word boundary detection problems observed in a word recognition task under noise in a car. Many word boundary detectors previously proposed only made use of sample statistics of the short-term energy of incoming signal and noise, the operation of which is often found unreliable and prone to errors with the existence of radio talk and background music. In this study, an adaptive noise canceler was employed that was based on a projection algorithm, and it was integrated into the signal processing front end of a word recognizer as the pre-processing to its word boundary detection, in order to reduce deletion errors and to improve accuracy of detected boundary of the utterances by suppressing the effect of co-channel components. Also observations were made of the behavior of the integrated boundary detector in terms of the estimated profile of the short-term energy, and its effect on the recognition accuracy were analyzed.
Investigation of subband-cross-correlation analysis for speech recognition under noisy conditions
Authors: Kajita Shoji , Takeda Kazuya , Itakura Fumitada
Published: October 2005

This paper describes subband-cross-correlation (SBXCOR) analysis for robust speech recognition under noisy conditions. The SBXCOR analysis is an extended signal processing technique of subband-autocorrelation (SBCOR) analysis that extracts periodicities present in speech signals. In this paper, the performance of SBXCOR is investigated using a DTW word recognizer, under simulated acoustic conditions on a computer and a real environmental condition. Under the simulated condition, it is assumed that speech signals in each channel are perfectly synchronized while noises are not correlated. Consequently, the effective signal-to-noise ratio of the signal generated by simply summing the two signals is raised about 3 dB. In such an ideal case, it is shown that SBXCOR is less robust than SBCOR extracted from the two-channel-summed signal, but more robust than the conventional one-channel SBCOR. The resultant performance was much better than that of smoothed group delay spectrum and mel-frequency cepstral coefficient. In a real computer room, it is shown that SBXCOR is more robust than the two-channel-summed SBCOR. The microphone setup for SBXCOR will be also discussed.
Speech recognition based on subword units
Authors: Noizumi Takuya , Mori Mikio , Taniguchi Shuji
Published: October 2005

Large vocabulary, isolated word recognition requires a large amount of training data proportional to the vocabulary size to characterize each individual word model. A subword-unit-based approach is a more viable alternative than the word-based approach to overcome the problem of the training data size, since different words can share common segments in their representations in the former. This paper deals with a couple of isolated word recognition systems where the subword-unit-based approach is commonly employed, though their methods of segmentation are completely different. In one system a hidden Markov model is used to decompose a word into subword units (segments), and frequency spectra of those subword units are fed to a recurrent neural network to yield a subword code sequence for the word. This sequence is then recognized hopefully as the original word by a set of hidden Markov models for isolated words. In the other system subword boundaries within a word are detected by finding peaks of the delta cepstrum of the word, and the resulting sequence of subwords is deciphered into the original word by means of concatenated hidden Markov models of isolated words. Those systems attain average recognition accuracies over 92%–96%.
Comparison of the effect of homophones on speechreading between Japanese and English
Authors: Hiki Shizuo , Kamikubo Emiko
Published: October 2005

The speech information conveyed by the shape of the mouth differs according to the characteristic phonetic system of each language. Besides the phonetic characteristic, a peculiar feature found in Japanese speech is that there are many homophenous words (words having a different sound but the same mouth shape), because the Japanese vocabulary comprises a large number of homophones (words having a different sound but the same mouth shape), compared with English. This results in a decrease of information obtainable through speechreading. In order to ascertain this feature quantitatively, first, the mouth shapes of vowels and consonants of Japanese speech are described symbolically. Detailed symbols are assigned to as many kinds of mouth shapes as are readable in a clear utterance. Using these symbols, the changes in mouth shapes in Japanese speech are described based on coarticulation rules and a combination of preceding and following vowels. Then, the ratios of homophenous words and homophones found in a basic vocabulary are estimated by referring to statistical data of the syllabic structure of Japanese words. By assembling these ratios, the nature of the information conveyed through speechreading is analyzed, and the negative effect of homophones on the speechreading of Japanese is discussed.
In search of an invariant representation for speech: The modulation spectrogram
Authors: Greenberg Steven , Kingsbury Brian E.
Published: October 2005

The ability to understand speech spoken by diverse speakers under a wide range of acoustic environmental conditions presents a central challenge to current theories of speech perception. Reverberation, background noise, and speaker variation all contribute to the heterogeneous acoustic realization of phonetic information from which listeners routinely derive linguistic information. Representations based on the detailed spectro-temporal properties of speech (e.g., the sound spectrogram) are not sufficiently stable under such conditions as to provide a reasonable account of the processes by which the brain decodes spoken language. A new representational format for visualizing speech, based on the distribution of the modulation spectrum below 16 Hz (and which emphasizes modulation frequencies centered around 4 Hz) across critical-bandlike channels, provides a high degree of stability under reverberant and low signal-to-noise-ratio conditions. This modulation-based representation appears to capture many essential properties of neurons in the primary auditory cortex, and is also well matched to the temporal dynamics of speech production. As such, it may provide a more principled basis for understanding the mechanisms by which human listeners decode the speech signal under natural acoustic conditions than finer-grained spectro-temporal representations.
Phoneme recognition using reference patterns constructed with discriminative training and DP matching
Authors: Okimoto Yoshiyuki , Makino Shozo
Published: October 2005

A modified LVQ2 algorithm (MLVQ2) for phoneme recognition is proposed. The length of a phoneme reference pattern made by the algorithm was fixed. It showed high phoneme recognition performances for isolated spoken words. However, it did not show high performances for continuous speech because the phoneme reference patterns with fixed length could not deal with a big change in phoneme duration. In this paper, a new construction algorithm of the reference patterns using discriminative training based on the MLVQ2 and DP matching is proposed. The initial multiple reference patterns are optimized as follows: At first, the correspondence between a frame of the input sample and a frame of the nearest reference pattern in the incorrect phoneme class is calculated by using the DP matching. Then, the corresponding frame vector of the reference pattern in the incorrect phoneme class is moved away from the frame vector of the input sample based on the MLVQ2. On the other hand, the corresponding frame vector of the reference pattern of the correct phoneme class is moved nearer to the frame vector of the input sample. The experimental result using the ASJ 503 sentences speech database showed a 81.3% phoneme recognition rate, increasing 1.7% compared to the previous method.
Feature and model compensation for robust speech recognition
Authors: Lee Chin-Hui
Published: October 2005

A mathematical framework based on maximum likelihood stochastic matching is proposed to perform feature and model compensation for robust speech recognition. Speech recognition is often formulated as a matching problem between the feature vectors extracted from a test utterance and a set of speech models or patterns obtained from some training corpra. It is well known that a speech recognizer often degrades in performance when the testing data are not acoustically similar to the training data. One way to improve is to find features that are invariant under all acoustic conditions and distortions. Some form of compensation is often required. The proposed stochastic matching approach assumes a structure or a form of the feature and/or model transformations. Together with a set of nuisance parameters, the transformations approximate the distortion in the test utterance. To decrease the acoustic mismatch between a test utterance and a given set of speech models, e.g., hidden Markov models, the stochastic matching algorithm estimates the nuisance parameters and then applies the feature/model transformations during speech recognition. Simple channel distortion can be approximated with linear transformations. For more complicated distortions, such as environmental, speaker, and combined mismatches, nonlinear compensating transformations are needed. These compensations give a significant performance improvement in speech recognition over the systems without them when utterances are affected by additive ambient noises and convolutional channel distortions.
Compensation for speech recognition in degraded acoustical environments
Authors: Stern Richard M. , Moreno Pedro J. , Raj Bhiksha
Published: October 2005

The accuracy of speech recognition systems degrades when operated in adverse acoustical environments. This paper discusses two ways in which more detailed mathematical descriptions of the effects of environmental degradation can improve speech recognition accuracy using both ‘‘data-driven’’ and ‘‘model-based’’ compensation strategies. Data-driven methods learn environmental characteristics through direct comparisons of speech recorded in the noisy environment with the same speech recorded under optimal conditions. Model-based methods use a mathematical model of the environment and attempt to use samples of the degraded speech to estimate model parameters. Two approaches to data-driven compensation, RATZ and STAR, are described, as well as a new approach to model-based compensation, referred to as the vector Taylor series (VTS) algorithm. Compensation algorithms are evaluated in a series of experiments measuring recognition accuracy for speech from the ARPA Wall Street Journal database that is corrupted by artificially added noise at various signal-to-noise ratios (SNRs). For any particular SNR, the greatest recognition accuracy obtained using a practical compensation algorithm is observed when that system is trained using noisy data at that SNR. The RATZ, VTS, and STAR algorithms achieve this bound at global SNRs as low as 15, 10, and 5 dB, respectively. [Work supported by ARPA.]
Multisensor-based speech processing for robust speech recognition
Authors: Zhao Y. , Yen K. , Zhuang X.
Published: October 2005

A multichannel signal-processing technique is integrated with automatic speech recognition for dealing with time-varying interference signals. Two microphones are employed for signal acquisition, each picking up a convolutive mixture of two source signals. A decorrelation adaptive filtering [Weinstein et al., IEEE Trans. SAP 405–413 (1993)] is first performed to restore the source signals; a cross-spectral analysis is next performed on the restored signals to determine the presence regions of each source signal [K. Yen and Y. Zhao, Proc. ICSLP (1996)]. The restored speech signals within the presence regions are inputted to an HMM-based speaker-independent continuous speech recognition system [Y. Zhao, IEEE Trans. SAP, 345–361 (1993)]. Five test sets were constructed from the TIMIT database under the SNR conditions of 20, 10, 0, −10, and −20 dB, each consisting of 78 sentence pairs. The acoustic coupling channels were simulated by FIR filters. The recognition vocabulary size was 853 and the task perplexity was 105. It was found that the multichannel integrated recognition system significantly improved recognition performance; above −10 dB, the word accuracies were close to that of the interference-free condition (91% word accuracy). [Work supported by NSF IRI-95-02074.]
Acoustic imaging by backward propagation: An overview of algorithm structure and configurations
Authors: Lee Hua
Published: October 2005

The backward propagation technique is known as one of the most widely used algorithms in acoustic imaging for its simplicity, stability, and resolving capability. This presentation provides an overview of the development of backward propagation based algorithms for various system configurations in a systematic manner, from the simple case of coherent systems with stationary sources to the more complex wideband pulse-echo systems for time-varying objects, in both monostatic and bistatic modes. With this treatment, it is now possible to establish an accurate assessment of systems’ performance in terms of computation complexity, algorithm structure, sensitivity, and limitations.
An ultrasonic ring transducer system for studies of scattering and imaging
Authors: Waag Robert C. , Liu Dong-Lai , Mast T. Douglas , Nachman Adrian I. , Jaeger Paul , Kojima Tadashi
Published: October 2005

A novel ultrasonic ring transducer and special control electronics have been developed for scattering and imaging studies. The transducer contains 2048 rectangular elements with a center frequency of 2.4 MHz and a −6-dB bandwidth of 70%. At the center frequency, the element size is 0.29 wavelength ×40 wavelength and the spacing is 0.37 wavelength. A multiplexer provides access to any contiguous 128 elements for transmission and any contiguous 16 elements for simultaneous reception. The transmit electronics have independently programmable waveforms. The receive electronics have time-varied gain functions independently programmable over the range 15–55 dB. Each receive channel includes a 20-MHz, 12-bit A/D converter. The electronics permit synthesis of arbitrary transmit and receive apertures. A novel ultrasonic wavefront design method has been implemented to determine element excitations using backpropagation of a user-specified field pattern. Pulse-echo compound images using constant f/1.0 transmit and receive apertures have been obtained for model scattering objects and an anthropomorphic breast phantom. Scattering measurements have been analyzed to obtain frequency- and angle-dependent average differential scattering cross sections of random media. The system is a useful facility for measurements of ultrasonic scattering for characterization of tissue, development of adaptive beam-formation techniques, and implementation of quantitative image reconstruction methods.
Broadband acoustic-field image enhancement in a noisy shallow ocean
Authors: Candy James V. , Sullivan E. J.
Published: October 2005

When a narrow-band source propagates in a shallow ocean environment, the ocean can be characterized by a waveguide resulting in a normal-mode solution. When the source is broadband, its spectrum can be decomposed into a set of distinct narrow-band lines, each supporting a distinct set of normal modes differing from frequency to frequency by both the number of modes and their respective wave numbers. Couple this propagation problem to a set of noisy hydrophone measurements obtained from a vertical array of sensors spanning the water column and a formidable problem in estimating the broadband acoustic field results. The field itself is essentially a space–frequency and once this space–frequency field can be estimated from the noisy measurements, then various postprocessors can be applied to estimate, detect, and localize the underlying source. Thus it is important, in the broadband case, to extract and enhance the broadband acoustic-field image. In this paper the development of a model-based image enhancement technique is discussed which employs the broadband normal-mode solutions using an optimal estimation scheme to provide the required enhancement. A by-product of the outputs of the processor, which is a space-varying Kalman filter, is the ability to not only enhance the image, but also to extract and enhance the corresponding broadband modal functions (one for each frequency) as well.
Development of transducers for a holographic 3-D imaging sonar
Authors: Ishihara Chiaki , Takano Osamu
Published: October 2005

This report describes the processes of development and the performances of the transducers applied to a real-time 3-D acoustical holographic imaging system based on the encoded wavefront technique. In this system, the acoustical wave modulated its phases by a system of Walsh functions, emitted from the transducers. Thus the performance of the broad bandwidth was needed for transducers to make clear the points of the phase modulation on the waveforms. Piezorubber film was selected as a piezoelectric material and it was laminated to improve the sensitivity of the transmitters. The receivers have a copolymer film (PVDF) as the piezoelectric material. The thin film and a preamplifier mounted close to the film give superior performances to the receiver. It is made clear that the directivity of the transducers agrees with the results of theoretical analysis of the acoustic field generated by circular plane vibration; hemispherical shapes of the transducers distorted the waveform in some conditions; the broad bandwidth will be realized to set the resonance frequency lower than the operational frequency.
A comparison of quantitative imaging techniques for ceramic materials
Authors: Chinn D. J. , Schneberk D. J. , Del Grande N. A. , Thomas G. H.
Published: October 2005

This work compares the effectiveness of five different imaging methods for porous preforms made from ceramic fibers. The methods are acoustic attenuation, x-ray tomography, radiography (x-ray attenuation), optical attenuation, and dual-band infrared imaging. The preform properties evaluated are fiber concentration, cracks, voids, surface defects, and contamination. Preforms with varying fiber concentration, preform thickness, and fiber type have been examined. All the methods allow density variations to be mapped. X-ray tomography has the highest contrast and sensitivity to defects but is slow if the entire preform is scanned. The attenuation methods are orders of magnitude faster and less expensive, but average over the thickness in the viewing direction and have less sensitivity to voids and cracks. Of the line-of-sight attenuation methods, radiography has the highest resolution and contrast. Acoustic imaging is most sensitive to surface aberrations. [Work performed under auspices of the U.S. Department of Energy by the Lawrence Livermore National Laboratory under contract No. W-7405-ENG-48.]
Quasi-three-dimensional inverse scattering computerized tomography for high-precision sound velocity measurement in tissues
Authors: Yamada Akira
Published: October 2005

A quasi-3-D inverse scattering technique is proposed to reconstruct sound velocity slices of the weak scattering tissues by transmitting and receiving acoustic waves around a single rotational axis. The reconstructing procedure proposed is as follows. First, a backpropagation operation is introduced using the holographic formula for a homogeneous medium. This backpropagation preprocessing operation contributes dramatically to the reduction of the final quantitative reconstruction errors. Next, after the Rytov transform over the marginal backpropagated plane, data are then filtered in the perpendicular direction to discriminate the horizontally scattered wave. Consequently, quasi-3-D sound velocity images on the horizontal sliced plane of an object can be obtained, using the conventional 2-D inverse scattering reconstruction procedure. Precision of the proposed 3-D reconstructed sound velocity image is investigated using the calculated scattered data from the spherical object. As a result, it is demonstrated that quantitative precision of the reconstructed 3-D velocity image is verified to be within 1% over the range of the velocity inhomogeneity from −10% to 10%. The results obtained are promising for actual clinical applications such as the human breast cancer screening test.
Two-step holographic reconstruction of an acoustic field
Authors: Tabei Makoto , Ueda Mitsuhiro
Published: October 2005

A method for the holographic reconstruction of an acoustic field from a finite sound source is discussed in detail. It is often experienced that the holographic reconstruction based on the angular spectral method gives unexpectedly erroneous reconstruction. This is due to the fact that the singularity of the transfer function in the spatial frequency domain cannot be treated accurately by the discrete system. [Waag etal ., ‘‘Cross-sectional measurements and extrapolations of ultrasonic fields,’’ IEEE Trans. Sonics Ultrason.SU - 32, 26–35 (1985)]. In this paper, a reconstruction is proposed that is based on the convolution of observed data and the transfer function rather than the direct manipulation of data in the frequency domain. It is shown that when the observed data are backpropagated to the plane that includes the finite sound source, the sampling interval of the data can be taken to be significantly larger than the half-wavelength. On the other hand, by taking the pixel interval of the reconstructed source image as less than or equal to the half-wavelength, the acoustic field at an arbitrary point in the space can be evaluated by the successive forward propagation of the source image. The effect of observing an aperture on the resolving power is also discussed.
Beam simulation error estimation using the scattering operator
Authors: Dubberley John R.
Published: October 2005

Beam simulation has been used to estimate scattering from large surfaces by calculating the scattering from the individual subsurfaces and then summing over these subsurfaces. Previously physical intuition and comparison to method of moments (MoM) calculations were used to determine beam simulation’s region of validity. Here the MoM operator is used for a direct comparison of beam simulation to MoM. The improved speed of comparison allows a more in-depth analysis of beam simulation at a lower computational resource cost. Scattering from one-dimensional random surfaces with Gaussian height distributions will be specifically examined to verify and clarify earlier beam simulation validations. Although one-dimensional surfaces are an oversimplification for most applications, they are well studied and are likely to give insight into higher dimensional applications.
Stimulus-response measurement of floor impact sound
Authors: Nash Anthony
Published: October 2005

In the area of floor impact testing, a number of excitation techniques have been used for evaluating the degree of sound transmission. The ultimate goal of the excitation is to replicate the force and time duration of a human footfall so one can reliably rank the acoustical quality of the floor/ceiling system. The most common excitation methods involve dropping some object from a known height so the kinetic energy delivered to the floor is consistent. The limitation of the energy method is that the force spectrum imparted to the floor can vary if the floor finish has unstable or nonlinear characteristics (e.g., a thick carpet). In order to control this parameter, research laboratories have occasionally used a force sensor to help calibrate the impact forces from dropped objects. This paper will discuss the concept of using a force sensor as one component of a stimulus–response impact sound measurement in the field. A unique 13-kg force transducer ‘‘platform’’ was constructed in order to perform the field experiments. The platform is capable of measuring large dynamic forces at frequencies up to 100 Hz. Some examples of two-channel field data (force versus sound pressure) will be presented to demonstrate the technique.
Scattering from rough pressure-release surfaces using the finite-difference time-domain method
Authors: Hastings Frank D. , Schneider John B. , Broschat Shira L.
Published: October 2005

The finite-difference time-domain (FDTD) method provides a robust and accurate means of studying a wide range of propagation and scattering problems. However, the FDTD method, though simple to implement and readily parallelizable, is computationally expensive. In this paper, a benchmark sea surface scattering problem [Thorsos, Reverberation and Scattering Workshop, Gulfport, MS (1994)] is used to show how the computational requirements of an FDTD solution can be mitigated. A contour-path approach is used so that the grid conforms to the continuously varying surface. This permits accurate modeling of the rough surface without requiring an excessive number of points per wavelength. Errors due to numerical dispersion are reduced by adjustment of the phase speed. Although the test-case problem has both transmitting and receiving arrays in the near field of the scattering surface, simple near-field to near-field transformations are used so that the transmitting and receiving arrays need not be included in the computational grid. These transformations significantly reduce the size of the grid and, correspondingly, the computational resources needed to obtain a solution. The problem considered here has been studied previously using time-harmonic techniques; however, an FDTD solution allows results to be obtained over a broad spectrum. The modifications needed to obtain broadband results are discussed. [Work supported by ONR.]
A Monte Carlo finite-difference time-domain technique for scattering from rough elastic bottoms
Authors: Hastings Frank D. , Schneider John B. , Broschat Shira L.
Published: October 2005

Scattering from elastic ocean bottoms has been modeled using approximate analytical methods such as perturbation theory and the Kirchhoff and small slope approximations. An alternative approach is to use a numerical technique. This paper describes one such technique, the finite-difference time-domain (FDTD) method, which has been used successively to model scattering from pressure release surfaces [Hastings et al., IEEE Trans. Antennas Propagat.43 , 1183–1191 (1995)]. Monte Carlo scattering strengths are obtained for a tapered incident beam insonifying rough granite and basalt surfaces with either a Gaussian spectrum or a modified power law spectrum. These results are compared with those of formally averaged scattering strengths obtained for several different approximate models. One benefit of using the FDTD approach is that it allows the addition of heterogeneities at little extra computational cost. However, the initial computational cost is great, and the presence of surface and subsurface waves may result in strong scattering at grazing angles despite the use of the tapered incident field. Additionally, grazing incident angles may require surface lengths that are too computationally expensive. These issues and how they affect the applicability of the FDTD method to elastic problems are discussed. [Work supported by ONR.]
Bottom scatter measurements and modeling from the California Continental Borderlands
Authors: Holland Charles W. , Neumann Peter , Rogers Andy
Published: October 2005

Bottom scatter measurements were conducted in several littoral and deep-water regions using explosive charges (processes from 100 to 2000 Hz) and low-frequency (processed from 175 to 325 Hz) coherent sources. The measurements provide an opportunity to compare bottom scatter data using coherent and incoherent sources. The bottom scatter collected by coherent and incoherent sources appears to be identical within measurement uncertainties. This important result implies that survey measurement techniques (which use incoherent sources) may be employed for developing databases that support coherent systems. A variety of seafloor mechanisms controlled the scattering depending upon frequency, grazing angle, azimuth, and environment. However, modeling results indicate that the dominant scattering mechanisms arise from sub-bottom features and not the water-sediment interface. One important scattering mechanism appears to be a sub-bottom horizon. This kind of mechanism is generally not treated in current scattering models. A modeling approach is presented that captures the angular and frequency dependence of the sub-bottom horizon scattering. [Work supported by SPAWAR and ONR.]
A wide frequency range model for seabed scattering
Authors: Ivakin Anatoliy N.
Published: October 2005

A unified model of seabed scattering applicable for a wide low- to high-frequency range is presented. The first-order perturbation solution has been obtained for the scattering amplitude of a randomly inhomogeneous seabed consisting of an arbitrary number of fluid sediment layers having slightly rough boundaries and covering an elastic half-space. The model for the second statistical moments of a scattered field involves both auto- and crosscorrelations between roughnesses of different interfaces as well as between fluctuations of different bulk parameters. Frequency-angular dependencies of the scattering strength for both monostatic and bistatic cases are calculated and analyzed for various seabed types. These dependencies are shown to be largely governed by transitions from one effective seabed type to another. For example, at sufficiently high frequencies, strong absorption causes the influence of stratification to be negligible, while, at moderate frequencies, interference effects due to sediment stratification become important. At lower frequencies, sound penetration of the sediment increases, and the effects of elastic scattering and reflection from the basement supporting both compressional and shear waves and corresponding interference become important. All these effects are considered in the frame of a unified model and their sensitivity to different medium and signal parameters is analyzed. Possible applications are discussed. [Work supported by ONR.]
Audibility of individual reflections in a complete sound field, III
Authors: Bech So/ren
Published: October 2005

This paper reports on the influence of individual reflections on the auditory localization of a loudspeaker in a small room. Two previous papers [J. Acoust. Soc. Am. 97, 1717–1726 (1995); ibid. (in press)] have reported on the influence of individual reflections on timbre of reproduced sound. The sound field produced by a single loudspeaker positioned in a normal listening room has been simulated using an electroacoustical setup. The setup models the direct sound, 17 individual reflections, and the reverberant field. The threshold of detection was measured using the method of adjustment for five reflections using three subjects for noise and speech. The thresholds have been measured for two simulated situations: (1) A loudspeaker with frequency-independent directivity characteristics and frequency-independent absorption coefficients of the room surfaces and (2) a loudspeaker with directivity according to a standard two-way system and absorption coefficients according to real materials. The results have shown that subjects can reliably distinguish between timbre and localization, that the spectrum level above 2 kHz for individual reflections determines their influence on the localization of the main source, and that only the first-order floor reflection will contribute to localization of the main source. a)Present address: Bang and Olufsen A/S, Peter Bangs Vej 15, DK-7600 Struer, Denmark.
Assessment of acoustical environments in malls in Saudi Arabia: A case study—Al-Rashid Mall
Authors: Khaiyat Sami A.
Published: October 2005

Most of the recently developed malls in Saudi Arabia have provided poor acoustical environments. Assessment of such environments is dis- cussed in detail in this paper through a case study; Al-Rashid Mall in Al-Khobar. Physical characteristics of the mall are described along with the special acoustical systems that are incorporated within the mall. One of these systems is the public address system which has many loudspeakers and feedback microphones that allow the system to automatically adjust the loudness of the sound signal produced according to the noise level. Unfortunately, this system has failed to provide the functions for which it was built. Acoustical characteristics such as background noise levels, reverberation time, and the sound transmission index (STI) were measured and analyzed in many locations. It was found that the mall has unexpectedly provided high background noise levels and reverberation time, as well as very poor speech intelligibility. Causes of such problems along with possible solutions are conceptually discussed. Finally, general social cultural factors that affect acoustical environments in malls in Saudi Arabia are briefly discussed along with a set of recommendations to overcome such factors.
Investigation on relationships between speech intelligibility on Japanese language and STI for the design of the sound system of an auditorium
Authors: Komoda Motoo , Takeuchi Matsumi , Umekai Katsuhiro
Published: October 2005

In order to clarify the applicability of STI as a parameter in sound system design, a working group was organized in JATET (Theatre and Entertainment Technology Association, Japan) in 1993. It has conducted a series of investigations on speech intelligibility in various aspects. This report summarizes the results of investigations on the following items: (1) STI round-robin tests with six measuring systems; (2) relationships between the intelligibility in listening tests and measured STI under the following different conditions—(a) acoustical conditions of the room, (b) types of loudspeakers, (c) listening levels, (d) frequency characteristics of signal sources, with and without microphone—(3) prediction of STI by computer simulation; and (4) a listening test by an auralization system.
Near-surface distributions of bubbles in the upper ocean, with applications to acoustical oceanography
Authors: Longuet-Higgins Michael S.
Published: October 2005

This lecture will review laboratory and field data on the distribution of bubble sizes due to rain and breaking waves and compare them with theoretical distributions where appropriate. Present gaps in our knowledge will be indicated. The applications to underwater acoustics are of two kinds: active generation of sound by bubble formation, and passive backscatter or absorption of incident sound waves. Each of these depends to some extent on the horizontal and vertical variations in bubble density. Inversely, one can obtain information on rainfall, breaking waves, and near-surface currents.
Viscous and elastic air bubble models of resonant scattering from fish swimbladders
Authors: Feuillade C. , Nero R. W.
Published: October 2005

Acoustic scattering from fish at low frequencies is enhanced by the resonance properties of the fish swimbladder. The resonances are due to excitation of the monopole motion of the fish bladders in much the same way that bubbles resonate, and ‘‘bubble-like’’ models have been a popular method for describing resonance scattering from swimbladder bearing fish. The response of the swimbladders is strongly damped by the viscous properties of the medium surrounding the bladder, while the resonance frequency is sometimes increased above that of an air bubble of equivalent volume and appears to be influenced by the elastic properties of the swimbladder wall. In this work the method by which both elastic and viscous effects modify the physical behavior of the swimbladder is described by comparing models of bubbles with elastic shells to bubbles with viscous shells. The formalism is applied to the interpretation of experimental data.
Sound magnification due to the collective oscillations of bubble clouds: A resonance phenomenon
Authors: Kolaini Ali R. , Sutin Alexander M. , Hobbs Christopher M.
Published: October 2005

The characteristics of the sound interaction with a bubble cloud produced by a series of hypodermic needles are studied both experimentally and theoretically. It has been observed that the sound-pressure level is magnified when the incoming waves are at resonance with the first collective mode (breathing mode) of the cloud. The magnification, as high as 20 dB, depends on the void fraction. A simple theory based upon a forced mechanical oscillator is developed. This theory suggests that when the incident wave frequency is close to the fundamental frequency of the cloud, the system is at resonance. As a result, a significant sound magnification occurs. The theory and the experimental observations are in a good agreement. In this paper, the resonance magnification is discussed in great detail and some practical applications of this phenomenon are offered. [Work supported by the Office of Naval Research.]
Suspended bubble clouds: Generation, theory, and experiments
Authors: Mueller Alan C.
Published: October 2005

A technique is presented in which small concentrations of a polymer additive are used to suspend air bubbles in a ‘‘gel’’ to form stable bubble clouds. Bubble clouds with volume fractions as low as 0.04 size distributions determined by using a CCD camera and image processing software, show clouds with most of the bubbles less than 100 mm in radius and means near 60 mm. A theoretical framework for the sound speed and attenuation as a function of the bubble number density distribution, which includes the shear-thinning properties of the gel, is presented. This theory indicates that the acoustics of bubble clouds in this gel are, over a large range in frequencies, almost identical to that of bubble clouds in pure water. Image processing along with low-frequency acoustic measurements performed in a long steel tube clearly show that only small changes in volume fraction and bubble size distribution occur over a matter of days. Results are presented comparing the theory and measurements for sound speed and attenuation over a three-decade range of volume fractions. [Work supported by DARPA SBIR Contract No. DAAH01-95-C-R164.]
Surface wave parameters for the prediction of breaking waves and acoustic backscatter
Authors: Hanson Jeffrey L. , Mandelberg Michael D.
Published: October 2005

Results from the acoustic surface reverberation experiment (ASREX) and critical sea test 7 (CST-7 phase 2) are examined to determine if surface wave parameters can be used to augment wind speed in air–sea process models. The wave parameters are extracted from directional wave spectra using a fully automated spectral partitioning approach. Isolated directional wind–sea spectra provide the basis for estimating the total rate of energy dissipation by breaking waves. The method employs Phillips’ equilibrium range theory and extends Felizardo and Melville’s recent work obtaining wave dissipation estimates from one-dimensional wave spectra. Preliminary results are encouraging; wave dissipation reduces the scatter in whitecap fraction observations by several orders of magnitude when wind speed is replaced with wave dissipation as the independent variable in the standard power-law model for whitecap fraction. It will be demonstrated that the success of this approach is partially attributed to a wind history bias that is automatically compensated for in the wave parameters. Results of similar on-going investigations using an extensive ambient noise and surface scatter observation set, collected during ASREX by the University of Miami, will also be discussed. [Work supported by ONR Code 32OA.]
The effect of water column stability on acoustic backscatter from the ocean bubble layer
Authors: Bekker Kathy K. , Gilbert Kenneth E. , Goodman Ralph R.
Published: October 2005

Low-frequency (200–1000 Hz) surface backscatter measurements reported by Percy et al. and Brownet al. show levels that are 10 to 20 dB less than the levels measured by Chapman–Harris, Chapman–Scott, and Ogden–Erskin under similar wind-speed/sea-state conditions. The Percy/Brown measurements were generally made under conditions in which the surface water was warmer (less dense) than the underlying water and consequently was gravitationally stable (buoyant). The other experiments cited were generally done in winter conditions where the surface water was either isothermal or colder than the underlying water, and, as a result, the water column was either marginally stable or unstable. Preliminary low-angle backscatter predictions using inputs estimated from Thorpe’s upward-looking sonar measurements give a level difference of over 10 dB between stable and unstable conditions. Since scattering from the sea surface and high void fraction bubble plumes should not depend strongly on the water column stability, an experimental demonstration of a marked dependence of low-angle backscatter on the gravitational stability of the water column would point to advected microbubble clouds as the primary scattering mechanism. The details of the calculations and planned experiments will be discussed. [Work supported by the Office of Naval Research.]
Difference frequency generation due to collective resonant oscillations of a bubble cloud
Authors: Sutin Alexander M. , Kolaini Ali R. , Prosperetti Andrea
Published: October 2005

Bubbly liquids exhibit a strong acoustic nonlinearity that can be used to generate low-frequency sound in a parametric-array arrangement. To study this possibility, two different bubble clouds have been generated in our laboratory. One is produced by a series of hypodermic needles giving a high void fraction bubbly medium with uniform bubbles sizes of about 1.1 mm radius. The other cloud is produced by means of a jet device that generates much smaller bubbles and a smaller void fraction. With the first cloud, the generation of waves with frequency around 200 Hz (corresponding to the natural frequency of the cloud’s breathing mode) was observed. The pump frequencies were around 2.2 kHz, close to the resonance frequency of the individual bubbles produced by the needles. For the jet-generated cloud the observed difference frequency was around 2 kHz for pump frequencies ranging from 5 to 28 kHz. The generation of the difference frequency over this broadband is related to the broad bubble size distribution generated by the jet. A theoretical model based upon nonlinear sound generation by bubbles resonating with the pump frequency will be presented to account for the observations. [The work was supported, in part, by Office of Naval Research and by RFBR (Grant 94-05-16755), Russia.]
Sound-speed measurements in the surface wave zone during ASREX
Authors: Melville W. K. , Terrill Eric J.
Published: October 2005

Wave breaking at the ocean–atmosphere interface injects air into the surface waters in the form of bubbles, altering the acoustical properties of the water. In situ sound-speed measurements at 3, 5, and 10 kHz were obtained in this dynamic near-surface layer using a moored sound-speed profiling buoy. The buoy was deployed in the North Atlantic during the winter of 1993-94 and obtained more than 40 days of data as part of the ASREX experiment. The record spans several storms, many with winds and significant wave heights in excess of 15 m/s and 7 m, respectively. Surface sound-speed measurements made in the first 7 m of the water column reveal large sound-speed reductions that are a result of air injection due to breaking waves. The data reveal the variable nature of the surface layer, both in short time scales O(seconds-minutes) that result from individual air-injection events, as well as longer time scales O(days) from one storm to another. Results of correlating the sound-speed field with environmental conditions will be discussed. The relationship between low-frequency, nondispersive sound speed and void fraction of air in water [A. B. Wood, A Textbook of Sound (1941)] make the resulting data useful for both oceanographers and acousticians. [Work supported by ONR.]