bluuebunny/crossref_metadata_2025 huggingface dataset:
Mutual information as an acoustic matching measure for speech recognition
Authors: Ozeki Kazuhiko
Published: October 2005

In recent speech recognition technology, the matching measure between a hypothesis and the corresponding speech segment is usually defined on the basis of HMM likelihood. As is well known, however, the likelihood is a relative measure, and some kind of normalization is necessary when hypotheses corresponding to different speech segments are to be compared. The aim of this paper is to show that the mutual information, or, equivalently, the likelihood normalized by the probability of the speech segment, is a better acoustic matching measure than the likelihood. An ergodic HMM was exploited to estimate the speech probability. An all-phone model was also tried as a speech probability estimator for comparison. An HMM-based connected word recognition algorithm was employed to generate recognition hypotheses. Those hypotheses were scored according to the above matching measures. An ART 200-sentence English speech database was used for the experiment. Evaluation was conducted from various points of view: recognition rate, word-end detection power, etc. The results show that the mutual information calculated with an ergodic HMM significantly outperforms the likelihood as an acoustic matching measure.
Acquisition of language models based on HMnet
Authors: Suzuki Motoyuki , Makino Shozo , Aso Hirotomo
Published: October 2005

Word n-grams and ergodic HMMs were generally used as statistic language models obtained from a large amount of training samples. These models can express short distance correlations between words, however, it is difficult for them to express the long distance correlations between words. In order to solve this problem, a construction algorithm is proposed for a statistic language model based on the hidden Markov network (HMnet). HMnet can express long distance correlations between words. To show the effectiveness of HMnet, it is compared with n-grams and ergodic HMMs by simple experiments. Training and test samples were randomly generated by a stochastic finite state automaton. HMnet showed lower perplexities than word n-grams and ergodic HMMs at an optimum number of states. However, an algorithm is needed for calculating the optimum number of states. If the test set perplexity is estimated from training samples, the optimum number of states is determined as the number with the minimum test set perplexity. So an estimation algorithm of the test set perplexity from training samples has been proposed. From the experimental results, the estimated values were nearly the same values as those for the test set perplexities.
Comparison between beam search and A* search methods for isolated/continuous speech recognition
Authors: Nakagawa Seiichi , Ida Masaki
Published: October 2005

In speech recognition, accomplishing high accuracy recognition and real time recognition is an important objective. Recently, large vocabulary continuous recognition systems have been developed, yet an accurate and efficient search algorithm is needed that can get an optimum candidate without matching all possible paths. In this paper, the evaluation of recognition performance utilizing the beam search and the A* search method will be described. Although the beam search does not guarantee an optimum result, it is rare that an optimum result cannot be gotten if use is made of suitable beamwidth and pruning threshold, and then the search can operate quickly because of reduced search space. In experiments on both isolated spoken word recognition and continuous speech recognition, the beam search method was superior to the A* search for recognition accuracy and speed. In addition, from the fact that there exists an optimum path in the region of the beamwidth in almost all cases, we propose an A* search method with beamwidth constraints.
New processing by neural networks for speech visualization and its performance in reading many words
Authors: Watanabe Akira , Tomishige Shingo , Nakatake Masahiro
Published: October 2005

To realize speech visualization by which one can read any word, the processing used to extract features on manner and place of articulation by neural networks has been improved. First, a necessary and minimum number of frames and those positions to maximize the automatic recognition rate have been searched for four three-layered perceptrons, which means the networks for manner of articulation, nasals, plosives, and fricatives. The results show that two or three frames within the interval of 100 ms are necessary. Next, since the coefficients of the networks in a part around the top of words are somewhat different from those in the middle or the end, two kinds of network coefficients have been estimated for the top of words and for the others, respectively. By using the outputs of the networks for speech visualization, consonantal patterns appear naturally without using any recognition technique. Finally, reading tests have been carried out using 520 words spoken by each of ten speakers, for a total of 5200 words. Considering that the subjects were not told the answers at all during the tests and that the training time was only 30 h, the correct answers reached 90%–92%.
Design and data collection for a spoken dialog database in the Real World Computing (RWC) program
Authors: Tanaka Kazuyo , Hayamizu Satoru , Yamashita Yoichi , Shikano Kiyohiro , Itahashi Shuichi , Oka Ryuichi
Published: October 2005

The RWC program is constructing substantial databases for advancing and evaluating research and development conducted under its program and related domains. In this presentation the motivation of this effort, a basic design of spoken dialog databases, and the current status of data collection work are described. At the first stage, some fundamental data collection has been carried out to determine several environmental conditions and data-filing specifications. Here, two topics are selected for the dialog: one was dialogs between car dealers and customers, and the other was dialogs between travel agents and customers. Professional dealers and agents were employed to produce reality in the conversations. To date, 60 samples of dialogs were recorded and 48 of them were filed into CD-ROMs which included about 10 h of speech waveforms with transcriptions and labeling-related information. The speech data are almost completely spontaneous but are of good quality in the acoustic-phonetic sense. The CD-ROMs are ready for distribution as a database to be used in research. [Work supported by the RWC program, MITI, Japan.]
Pronunciation variability in the Switchboard corpus
Authors: Fulop Sean A. , Keating Patricia A.
Published: October 2005

This paper first describes a project on manual phonetic labeling of some 2075 words from the Switchboard corpus (a 3 million word corpus of unscripted telephone conversations, recorded and orthographically transcribed by Texas Instruments, and available from the Linguistic Data Consortium). Multiple (from 10 to 40) tokens of 72 lexical items were transcribed; the transcription system used was an extension of the TIMITBET, designed to allow a narrower transcription, particularly of consonants. Intertranscriber agreement was assessed using the Oregon Graduate Institute’s metric for transcription accuracy, and comparison with their results for their English telephone corpus will be provided. A number of phonological facts will then be elucidated from the transcriptions. To facilitate this, a database of contextual information has been created for each phoneme in the dictionary forms of the words; the database includes both lexical contextual factors and the actual context in which a given token appears. The scheme is similar to that of Withgott and Chen [Computational Models of American Speech (1993)], who used such a contextual database to develop phonological rules for TIMIT. Here, the phonological rules which emerge from Switchboard will be discussed and compared with those already found in TIMIT.
KIDS: A database of children’s speech
Authors: Eskenazi Maxine S.
Published: October 2005

A database of children reading age- and reading-level-appropriate text aloud has been collected. This labeled data, to be distributed in the near future, was primarily intended to be used in CMU’s LISTEN tutor, which employs speech recognition to monitor children’s reading and then helps correct errors. The speaker population was therefore chosen to represent good and poor readers and to incorporate dialects of the speakers for whom the reading coach is intended. Phonemic balance could not be achieved (although it has been calculated) since the primary concern in recording children reading is to present sentences that can effectively be read by first through third graders. The text is a series of sentences that was adapted from text in the Weekly Reader series—most of the adaptation concerned the lack of the accompanying images. The text was chosen for its intrinsic interest and widespread use. Several trial recording sessions were used to develop a protocol that kept extraneous noises produced by the children at a minimum. This and other problems inherent in recording children reading will be discussed. Novel techniques developed for labeling this kind of speech will also be presented. [This work was funded by NSF Grant No. IRI-9528984.]
Speaker recognition model using two-dimensional mel-cepstrum and self-growing LVQ
Authors: Kitamura Tadashi , Hashimoto Kazue , Miyajima Chiyomi
Published: October 2005

This paper describes a speaker recognition model using TDMC and self-growing LVQ. Two-dimensional mel-cepstrum (TDMC) consists of averaged and dynamic spectral features of the two-dimensional mel-log spectra in the analyzed interval. A self-growing algorithm is used for VQ. In the beginning of this algorithm, there is no centroid and the first input feature vector is set to the first centroid. Then an input vector is classified into a specified centroid. When the amount of data in the centroid becomes greater than a predetermined threshold, the centroid is divided into two so that each centroid has the same number of data. By these procedures the number of centroids increases gradually while the amount of the data in each centroid become almost equal. In this study, text-dependent speaker identification experiments for 30 speakers were carried out. Each speaker recorded 10 digits three times during five sessions. Each speaker model is created using the speech data of two sessions. The area, the number of centroids, and a combination of averaged and dynamic features of TDMC were studied. The experimental results have shown that the combination of averaged and dynamic features is very effective and the proposed model gives an averaged identification score of 95%.
HMM-based speech synthesis with various voice characteristics
Authors: Masuko Takashi , Tokuda Keiichi , Kobayashi Takao , Imai Satoshi
Published: October 2005

This paper presents a text-to-speech synthesis system based on continuous density HMM which can synthesize speech with various voice personality characteristics. An algorithm for speech parameter generation from HMM has been proposed, and it is shown that by using differential parameters as dynamic features, a smoothly varying speech parameter sequence according to the statistical information of static and dynamic features modeled by HMMs can be generated. A framework of HMM-based text-to-speech synthesis system using this algorithm is also shown. In this paper, an approach to voice conversion for the HMM-based text-to-speech synthesis system is described. In the approach, voice conversion is achieved by changing the parameter of HMMs which are used as speech units in the system. To transform the voice characteristics of synthesized speech to the target speaker’s, a speaker adaptation technique to the HMMs was applied. From the subjective experimental results, it is shown that speech can be synthesized easily with various voice characteristics by transforming HMM parameters, and investigating the relation between the voice characteristics and the amount of the adaptation data.
Speech synthesis using a nonlinear modeling of vocal folds vibration effect
Authors: Ohmura Hiroshi , Tanaka Kazuyo
Published: October 2005

From a theoretical viewpoint, the vocal folds vibration affects the vocal tract transfer characteristics through nonlinear time-varying interaction between the glottis and vocal tract. However, ordinary speech processing systems have neglected those effects under the assumption of linearity of production systems. But if one tries to improve the voice quality to be more natural or human-like, it becomes crucial to investigate and model such effects. For this purpose, analytic exeriments were conducted on the vocal folds vibration effects [Ohmura et al., ICSLP96]. Based on these experimental results, two types of speech synthesis models are proposed: one is a wave function model in which formant energy dumping is given by a time window function. The other is a second-order nonlinear differential equation in which the formant energy dumping pattern is controlled by its friction term. Its performance was confirmed by a preference test on speech wave reconstruction. A speech sample set used in this test contains 60 sentences uttered by four speakers. Evaluators listened to two kinds of reconstructed speech samples: one is synthesized by the proposed method and the other is synthesized by an ordinary LPC-based method. Results show that the proposed method is clearly superior in its quality.
A method for controlling the speech rate in speech synthesis and its perceptual evaluation
Authors: Ohno Sumio , Fukumiya Masamichi , Fujisaki Hiroya
Published: October 2005

It is well known that speech rate varies both globally and locally in natural discourse due to various factors such as contrastive stress, syntactic boundaries, emotion, etc. While the global speech rate can be clearly defined by the durations of utterances and pauses, the local speech rate has not been well defined. The present authors have proposed a rigorous and quantiative definition for the relative local speech rate and showed an objective method for its measurement [S. Ohno and H. Fujisaki, Proc. EUROSPEECH’95, Vol. 1, pp. 421–424 (1995)]. Based on the analysis of changes in both global and local speech rates found in a speech material consisting of readings of a story at various speech rates, the present paper proposes rules for controlling the global and local speech rates in order to produce a synthetic discourse to fit exactly in a specified time interval. The validity of the method has been tested and confirmed by perceptual experiments using synthetic discourse of various durations generated from a natural discourse by analysis–resynthesis.
High-quality speech signal manipulation by Fourier synthesis
Authors: Reetz Henning
Published: October 2005

Many speech research applications require manipulation of certain acoustic parameters (e.g., F0, formants, duration) and high-quality resynthesis. Unfortunately, most existing resynthesis methods are either of rather poor quality (e.g., LPC), or require careful parameter adjustment (e.g., formant), or allow manipulations only within a limited range (e.g., PSOLA). Although a resynthesized Fourier analyzed signal is theoretically identical to the original signal, manipulating values in the frequency domain lead to discontinuities in the phase domain and distort the resynthesis. The human ear is, however, insensitive to phase relations in a signal. Thus high-quality resynthesized signals are possible by estimating appropriate phase values preventing phase jumps. The presented method is based on this principle. The method is as follows: The Fourier is computed and either: (1) the harmonics are shifted by keeping the envelope constant (to perform F0 manipulations); (2) the envelope is manipulated by keeping the distances of the harmonics constant (to perform spectral shape manipulations); or (3) frames are repeated (to perform duration manipulations). For the resynthesis, the phase values for the manipulated signals are computed for each Fourier frequency such that no sudden changes occur.
Generation of wideband speech from bandlimited speech by use of an autocorrelation function
Authors: Suzuki Jouji , Nagami Kayo , Yashima Hiroyuki
Published: October 2005

A high quality of speech sound is requested in speech communication circuits. This requirement, however, is not satisfied in telephone and mobile communication circuits. This paper proposes a new method to generate a lower speech signal from bandlimited speech (300–3400 Hz). A short-time autocorrelation function (ACF) is calculated on a speech waveform where the amplitude is compressed with a square root. A period is determined from the ACF. The period of the ACF is employed as an output signal. This signal is low-pass filtered and added to the original bandlimited signal. On the other hand, the higher band signal is synthesized by a pitch excitation formant synthesizer with fixed formant frequencies. The synthesized signal is filtered with BPF (3.4–7 kHz) and added to the original. Those processes are repeated with every pitch period. Short sentences uttered by two males and two females are assessed with a preference test. Synthesized speech is compared with bandlimited natural speech where the lowest frequency is varied between 0 and 300 Hz. The hearing test reveals that the proposed method provides a speech signal that is equivalent to the bandlimited signal where the limit of low frequency is 100–200 Hz.
A method for registration of a new voice in a text-to-speech synthesizer
Authors: Saito Takashi , Sakamoto Masaharu
Published: October 2005

In conventional text-to-speech systems, synthesis unit inventories are prepared in advance through a laborious process of speech data gathering, analysis, and manual segmentation, and users of such systems cannot add unit inventories of new voices. This paper describes a method for registration of a new speaker’s voice to synthesis unit inventories in a text-to-speech system, by providing users with a function of registering their voices like a training function commonly used in speech recognition systems. An approach taken here is as follows: (1) create an initial unit inventory as a reference unit inventory by manual segmentation, (2) for new speakers, let them give utterances following a guidance synthetic speech of the reference speaker, whose pitch range is adjusted to those of new speakers, (3) segment the new speech database automatically by a phonemic alignment technique combined with constraints given by the segment information of the reference unit inventory. Experimental results are shown for a waveform-concatenation-based TTS system, which was recently developed by the authors [Saito et al., Proc. ICASSP’96, 381–384 (1996)].
Automatic acquisition of speech units for recognition and very low bit coding
Authors: Saito Minoru , Masukata Mikio , Nakagawa Seiichi
Published: October 2005

In this paper, a method is described to acquire speech units automatically which have almost the same duration length as a syllable (or a phoneme). Optimal speech units are obtained following the procedure that the number of units is given beforehand and the pattern of units is updated successively based on a minimum distortion criterion. To investigate whether the speech units used in this procedure correspond to syllables, speech units were acquired automatically on condition that both the number and length of units are equivalent to syllables. As a result, acquired speech units corresponded to about half of syllables. When these units were used for 216 word recognition experiments, the obtained recognition rate was higher than that for the syllable-based method. And a word recognition rate of about 98% or so was achieved by increasing the number of units. At this time, the bit rate was still less than 60 bps, so this method may be applied to the model for ultra very low bit coding.
Quantization of vector sequences using statistics of neighboring input vectors
Authors: Tokuda Keiichi , Kobayashi Takao , Masuko Takashi , Imai Satoshi
Published: October 2005

A vector quantization method using statistics, i.e., mean and covariance, of neighboring input vectors (or linear transform of those) is proposed. In the proposed method, the distance between the input vector and the codeword is measured by the output probability defined by the statistics, and the output vector sequence is determined so that the output probability of the output vector sequence is maximized. The codebook can be trained by a conventional training procedure based on the statistically defined distance or a simplified distance. The value of the output vector sequence is determined by a high-order set of equations. Fortunately, it can be shown that the set of equations can be solved by a fast algorithm. In the conventional vector quantization methods, adjoining output vectors can change discontinuously; quantizing speech spectral parameter vectors causes perceivable glitches in the synthesized speech, whereas, in the proposed method, the change of adjoining output vectors is controlled appropriately according to the statistics of neighboring input vectors. Through an example of vector quantization of the LSP coefficients obtained from natural speech, it is shown that the proposed method can improve objective and subjective performance of vector quantization.
Effect of the mobile terminal acoustic characteristics on speech quality
Authors: Kubo Masato , Ono Shigeru
Published: October 2005

The mobile terminal handsets for the digital cellular phone systems use low bit rate speech CODEC. Although the speech quality of the handsets is strongly related to the speech CODEC performance, the effect of the handset acoustic characteristics on reconstructive speech quality cannot be negligible. This is because the equantizers of the speech CODECs are generally designed based on predetermined speech databases. Moreover, the speech databases are designed by using the ideal acoustic characteristic instruments. However, it is difficult to realize the ideal acoustic characteristics in real handsets. This paper discusses the acoustic characteristics realizable and suitable for low bit rate speech coding. The acoustic characteristics were prescribed in terms of input–output linearity and frequency characteristics. Six types of the acoustic characteristics for the receiver and three types for the transmitter were prepared, and a subjective evaluation test based on MOS was carried out over the speech stimuli created through the above nine types of the acoustic characteristics. The evaluation results showing which acoustic characteristics are suitable for the mobile terminal handset will be presented at the conference. Moreover, noise canceler property is critical to improve the speech quality. This paper also discusses the preferable time-varying property.
Automatic scoring of pronunciation quality for language instruction
Authors: Franco Horacio , Neumeyer Leonardo
Published: October 2005

The purpose of this work is to develop methods for automatic assessment of pronunciation quality, to be used as part of a computer-aided language instruction system. The basic pronunciation scoring paradigm [Bernsteinet al., ICSLP 90, Kobe, Japan] uses hidden Markov models (HMMs) to generate phonetic segmentations of the student’s speech. From these segmentations, different machine scores are obtained based on HMM log-likelihoods and durations. The machine scores are evaluated based on their correlation with human scores on a large database. Previous approaches were based on statistical models built for specific sentences. The current algorithms were designed to produce pronunciation scores for arbitrary sentences, i.e., sentences for which there is not training acoustic data [Neumeyer et al., ICSLP 96, Philadelphia]. This approach allows great flexibility in the design of language instruction systems because new pronunciation exercises can be added without retraining the scoring system. Initial results showed that duration-based scores outperformed HMM log-likelihood scores. Recently, it was found that HMM-based scores can be significantly improved by using average log-posterior probabilities of phone segments. Correlation went up from r=0.48 to r=0.84, which is similar to that of duration-based approaches. This level of performance approaches that of human raters.
Helium speech database and some new aspect of diver’s speech
Authors: Nakai Takayoshi , Suzuki Hisayoshi , Ishida Keizo
Published: October 2005

Speech is distorted in pressurized helium-air. Many aspects, except for upward shifts of formant and pitch in the helium speech, however, have not been known yet. A helium speech database system including more than 5 000 words and messages of divers was developed. The vibration of the cheek, the neck, and the nose were also recorded for some utterances. They were taken at the simulation dives up to the depth of 300 m at JAMSTEC since 1982. The helium speech and the normal speech were compared in terms of (1) duration of phonemes, and (2) pitch frequencies and formant trajectories in two different hearing/speaking conditions: the first condition is a case when a diver spoke as he was listening to his own unscrambled speech, and in the other condition the diver was not hearing his unscrambled speech. As a result, the shorter duration of word-initial consonants, higher pitch by psychological stress, and rough movement of formant trajectories were observed in addition to the well-known formant and pitch frequency shifts. As a special property of helium speech, it is newly discovered that there exists a quasiformant below the normal first formant, and that it may be caused by the sound from the nostrils.
Imaging the hydraulic structure of the seabed through crosswell acoustic tomography
Authors: Yamamoto Tokuo , Kuru Murat , Nye Tom
Published: October 2005

The hydraulic structure of the seabed (porosity, permeability and shear strength) is important with regard to many physical processes including the storage and transport of pore fluids, the consolidation and subsidence of man-made islands, the liquefaction of foundations, and the occurrence of catastrophic earthquakes and tsunamis. Propagation of acoustic waves through sediments is affected by the hydraulic properties of the sediments which may be modeled by poroelastic theories by Biot (1956), Dvolkin etal . (1993), and Chesnokov et al. (1991). Crosswell tomography measurements enable one to image the structure of sound speed and attenuation within the earth, from which the hydraulic structure is found based on the poroelastic theories. The hydraulic structures of the seabed revealed from four case studies will be presented and their implications to the earth processes of storage and transport of pore fluids, consolidation, liquefaction, and earthquake will be discussed. Acoustic waves are scatted from the seabed. It will be shown that the seabed structure can be extracted from measured acoustic bottom scatter. It will be also shown that the disturbances made in the seabed for instrumentation can act as vertical waveguides which may be mistaken as a false Biot slow wave. [Work sponsored by ONR, NSF and KSC.]
Acoustic scattering by targets in porous ocean sediments
Authors: Lim Raymond
Published: October 2005

Biot theory appears to be a viable model for wave propagation in porous ocean sediments, but difficulties in accurately determining the required material parameters leave some interesting predictions based on this formalism in doubt. For example, parameters recently reported for sand sediments off Kings Bay, GA and Panama City, FL [N. P. Chotiros, J. Acoust. Soc. Am. 97, 199–214 (1995)] suggest that Biot’s slow wave can be excited in these sediments with a propagation speed high enough to significantly enhance sonar transmission of coherent acoustic wave energy into the bottom at shallow grazing angles. While buried hydrophone measurements confirm higher sound-pressure levels than expected from fluid sediment models, the interpretation of these measurements can be ambiguous. To remove some of these ambiguities, numerical modeling of the scattering by simple targets both in free-field and in buried configurations is suggested. By investigating the scattering by porous targets or elastic targets in a porous background, the dynamics unique to a coupling to Biot’s slow wave can act as a fingerprint of this wave’s generation. This will be demonstrated with examples calculated using an exact scattering solution of Biot’s equations. [Work supported by ONR.]
An idealized model for the frame elasticity of sand
Authors: Chotiros Nicholas P. , Mautner Adrienne M.
Published: October 2005

An idealized model of the poroelasticity of dry- and water-saturated sand is hypothesized. The model addresses the nature of the grain–grain contact, and it is based on frame shear and bulk moduli inverted from measured acoustic and shear wave speeds as a function of pressure. The experimental data are from E. Hamilton [J. Geophys. Res. 76, 579–605 (1971)] in which both the compressional and shear wave speeds were measured as a function of the differential pressure in dry- and brine-saturated sand sediments of grains sizes approximately 0.137 and 0.7 mm. The inversion is based on Biot’s theory. The results suggest quantifiable effects due to adhesion between grains and pore fluid, partial lubrication of the grain–grain contact by the pore fluid, and a solid grain–grain contact area that changes with applied pressure. [Work supported by ONR, Code 321 OA.]
A generalized acoustic model for marine sediments
Authors: Boyle Frank A. , Chotiros Nicholas P.
Published: October 2005

Current acoustic backscatter models are limited in applicability to specific types of marine sediments. A fine-grain sediment in which particles are suspended can generally be treated as an acoustic fluid, whereas an elastic model is more appropriate over a rigid bottom. Sandy sediments have been modeled successfully as Biot poroelastic media [F. A. Boyle, and N. P. Chotiros, J. Acoust. Soc. Am. 98, 531–541 (1995)]. Since it is unclear how to treat a sediment that straddles the boundary between sediment types, a unified theory for bottom backscatter is needed, from which a seamless model can be developed. Due to the porous nature of marine sediments, a poroelastic model is considered to be most appropriate. Critical parameters that need to be determined, particularly frame moduli and permeability, are inverted from acoustic measurements, to give empirical values as a function of porosity. [Work funded by NRL Code 321 OA.]
Mandrel effects on tube waves
Authors: Hsu Chaur-Jian , Bailey Robert , Kostek Sergio , Johnson David Linton
Published: October 2005

The characteristics of tube waves in water-filled boreholes with and without an elastic cylindrical mandrel in the borehole are compared. With a mandrel, the tube waves are slower, more dispersive, and more sensitive to the formation shear modulus than without. Additionally, the characteristics of tube waves in a borehole in the presence of a liquid-saturated porous mandrel were also studied. The tube waves propagating in the annulus couple with the slow waves propagating in the porous mandrel. This coupling and the viscous dissipation of the slow waves introduce attenuation and additional dispersion to the tube waves. Whether the mandrel is porous or not, there are two modes which are weakly dispersive in the quasistatic limit. Very roughly, the two modes consist of an extensional mode in the rod and a fluid-based mode in the annulus which are coupled together.
Acoustic nonlinearities in porous media
Authors: Ostrovsky Lev A.
Published: October 2005

This is a brief review of theoretical and experimental studies of acoustic nonlinearities in porous media. Like gas bubbles in water, pores may increase the effective nonlinearity parameter of a medium by several orders at a rather small porosity. For porous solids, such phenomena can be observed in the case of ‘‘waterlike’’ media when the shear modulus is small compared to the longitudinal elastic moduli. Experiments with porous polymers have confirmed the occurrence of very strong nonlinearity and have shown a good agreement with the theory. Another problem is grainy media with fluid filling of the space between grains. Such a system is known to possess anomalously big nonlinearity due to the contact deformations (Hertz law). Here, the presence of air bubbles in the intergrain fluid is taken into account. It is shown that the presence of free gas may further increase the nonlinearity of the structure. Finally, some theoretical and experimental studies of nonlinearities in a water-saturated seabed with bubbles are outlined. Estimates made for some artificial and natural materials show a possibility to use the acoustic nonlinearity as a sensitive parameter to evaluate the medium structure.
Temporal prediction of acoustic backscatter from sediments at 10–250 kHz
Authors: Sternlicht Daniel D. , deMoustier Christian P.
Published: October 2005

A temporal model of high-frequency (10–250 kHz) seafloor acoustic backscatter has been developed which simulates the average squared echo envelope from a monostatic echo sounder. Tests comparing the model to measured backscatter echoes from pelagic and terrigineous sediments are presented, and the diminishing contribution of the sediment volume with increasing acoustic frequency is demonstrated. The temporal model was developed to assist in the identification of sediments by statistically comparing its output to the echo amplitude time series measured with bottom mapping sonars. The model incorporates the sonar geometry, electroacoustic transfer function and transmitted signal characteristics, the ocean volume parameters, the statistical geoacoustic parameters which describe the water/bottom interface and the sediment volume, and the multiscale roughness statistics of the interface for the bottom types being investigated. [Work supported by ONR.]
Measurements of acoustic wave speeds in a fluid-saturated sandy bottom
Authors: Simpson Harry J. , Houston Brian H.
Published: October 2005

A new laboratory has been created at NRL to study the physics associated with the scattering from buried and partially buried objects in a saturated porous media environment. The laboratory consists of a 600 000-liter deionized water tank that is instrumented with precision robotic scanners and sources and receivers for broadband (0.5 kHz–5 MHz) acoustics research. Presently, the tank is filled with a 3.0-m-deep bottom of 212-mm-mean-diam sand that has been filtered and cleaned. Preliminary measurements of the acoustical wave sound speeds propagating in this saturated porous media are presented. In addition to time-delay measurements, similar to those done by Chotiros [J. Acoust. Soc. Am. 97 (1995)], a novel synthetic array was used to map the wavefronts propagating through the bottom. From these two-dimensional synthetic array measurements, wave speeds are evaluated using analytical techniques that include wave-number frequency and ray path analysis. Attenuation in the porous media and reflection and transmission from the interface are also studied. [Work supported by NRL.]
Effects of arbitrary pore-size distribution on the reflection and transmission coefficients for a porous seabed
Authors: Turgut Altan
Published: October 2005

Plane-wave reflection and transmission coefficients were calculated for a porous viscoelastic seabed with arbitrary pore-size distributions based on a previously developed theory [Yamamoto and Turgut, J. Acoust. Soc. Am. 83, 1744–1751 (1988)]. A sensitivity analysis indicated that, for a fixed permeability value, arbitrary pore-size distribution seems to broaden the frequency dependency of plane-wave reflection and transmission coefficients. Assuming a single pore-size Biot model, performance of global optimization inversion algorithms was investigated using the synthetic chirp signals reflected form a porous seabed with an arbitrary pore-size distribution. Inversion of porosity, shear modulus, and permeability was primarily attempted by assigning representative values to other Biot parameters which have secondary effects on the plane-wave reflection coefficient within their physical bounds. Perfect agreement was obtained between inverted and true values of porosity and shear modulus. However, estimated permeability values were always smaller than true permeabilty values due to the assumption of single pore size.
Scattering from a bubble in a saturated sediment
Authors: Kargl Steven G. , L. Williams Kevin , Lim Raymond
Published: October 2005

The breathing mode for a gas-filled, spherical cavity in a saturated, poroelastic sediment is investigated. The analysis entails the scattering of an incident displacement field that preferentially excites the monopole resonance of the cavity. The main result demonstrates that a gas-filled, spherical cavity in a saturated poroelastic medium exhibits two distinct breathing mode resonances. These resonances arise from the two distinct longitudinal modes of propagation in saturated poroelastic medium as described by Biot’s theory. Preliminary results from laboratory measurements involving a synthetic sediment will be presented. The experiments utilize spherical glass beads (250–300 μm) saturated with an immersion fluid whose optical index of refraction matches the corresponding index for the glass. [Work supported by ONR.]
Horizontal inhomogeneities and acoustic scattering by granular poroelastic ocean sediments
Authors: Yelton Dennis J. , Chotiros Nicholas P.
Published: October 2005

The effect of horizontal inhomogeneities in sediment properties on acoustic scattering is modeled as a problem of diffraction by random apertures over a randomly stratified medium, by numerical simulation. Apertures of varying geometries were investigated. The ocean sediment is modeled in two steps. (1) The granularity is modeled as an ensemble average of a randomly layered poroelastic material, in which each layer represented a granular material of a particular grain size, and the thickness of each layer was related to the associated grain size and porosity by a conservation-of-mass relationship. (2) Lateral variations are approximated by performing a coherent ensemble average of results from several realizations of the randomly stratified medium in which the incident and scattered sound wave enter and exit the sediment interface through an aperture. Poroelastic medium parameters were chosen to represent water-saturated sand. The mean and standard deviations of the grain-size distribution were chosen to match existing experimental data. Scattered signals were computed for a selection of aperture functions and at various grazing angles. The angular variations in the scattering strength were compared. [Work supported by ONR Code 321 OA.]
Determination of acoustic properties of marine sediments
Authors: Dunlop John I. , Thomson P. Philip
Published: October 2005

The reflectivity properties of the seabed depend primarily on the acoustic properties of the upper layer sediments. Laboratory measurements may be made to an acceptable precision on samples abstracted from the seabed and mathematical modeling carried out to predict the acoustic properties of the sediment. However, due to disturbance of the cores in handling and storage, it is also desirable to make ‘‘on-board’’ measurements on fresh samples, often to a lesser precision. Laboratory measurements of sound speed, attenuation constant, and shear wave velocity were made on cores taken from the North West Shelf of Australia. These measurements were made at several frequencies in the range 90 kHz to 5 MHz using a three-transducer speed frame. Supplementary measurements were also made using a more portable two-transducer 50-kHz pulse transit timer suited to on board operation. The measured acoustic properties were fitted to mathematical models based on Biot–Stoll theory of acoustic propagation in water-saturated unconsolidated sediments allowing estimations of the unmeasured Biot parameters. Predictions were then made of the acoustic reflectivity properties of the seafloor.
Development of a heavy and soft impact source for the assessment of floor impact sound insulation
Authors: Tachibana Hideki , Tanaka Hiroshi
Published: October 2005

In order to examine the impact sound insulation performance of building floors against heavy and soft impacts such as those of children jumping and running, a method whereby a heavy and impact source (an automobile tire) is dropped is specified in JIS A 1418, in addition to the standard method of using the ISO tapping machine. In order to improve the latter method, a research project has been carried out in the Architectural Institute of Japan. After various trial experiments, a new heavy and soft impact source (a rubber ball, weight: 2.5 kg; size: 183 mm in diameter; impact time: 20 ms) has been developed. In this paper, the basic impact characteristics of the source and the experimental results obtained in field measurements are introduced in comparison to the existing impact sources (automobile tire and the tapping machine).
Proposal for a standard heavy impact source for the measurement of floor impact sound insulation performance
Authors: Inoue Katsuo , Kimura Sho
Published: October 2005

The floor impact sound generated in multifamily dwellings frequently has a peak in the low-frequency band (20–30 Hz). Because of that, in order to assess the floor impact sound insulation performance in real-life situations, it is necessary to measure floor impact sound using a standard impact source, with an impact force characteristic corresponding to actual impact sources. However, in the current ISO standards, a ‘‘tapping machine’’ that is light and stiff with a weak impact force is the only one that has been approved as a standard floor impact source, which cannot be considered adequate as a method of measurement in solving problems of floor impact sound. Actual case examples of the measurement of impact force performance of floor impact sources generated in actual housing units are presented in this paper, together with a verification of the usefulness of the heavy floor impact source stipulated by current JIS standards in Japan and a proposal for the introduction of the same impact force characteristics as ISO standards.
Some consequences of including low frequencies in the evaluation of floor impact sound
Authors: Rindel Jens H. , Rasmussen Birgit
Published: October 2005

A method for including frequencies down to 50 Hz in the evaluation of floor impact sound has become available with the new version of ISO 717-2. In addition to the single number quantity for rating the impact sound insulation, a new spectrum adaptation term has been defined. The method has been studied by the Acoustics Group of NKB (Nordic Committee on Building Regulations). The new method has been applied to a large number of recent measuring results from the Nordic countries. It was found that the spectrum adaptation term for the extended frequency range depends on the type of floor construction, and light floor constructions are evaluated less favorably than heavy constructions. Comparison with data from a 14-year-old Swedish survey suggests that the extended frequency range leads to a higher correlation with subjective evaluation of impact noise. The consequences of applying the extended frequency range in future building regulations or in a system for sound classification of dwellings have been considered. However, there are several problems to be solved, among which are a lack of available data for floor constructions at low frequencies, an increased measurement uncertainty, and the fact that some floor constructions are evaluated too favorably by the new method.
Floor research project at NRC Canada: An update on progress—Measurements with experimental impact devices
Authors: Warnock A. C. C.
Published: October 2005

A major thrhust in this floor project is the collection of low-frequency impact information, below 100 Hz, using special impactors: a tire-dropping machine, two experimental rubber balls supplied by H. Tachibana, and a walker. The frequency range for the standard ISO tapping machine is also extended at 50 Hz. A preliminary analysis of the data collected so far in the project will be presented. Factors that were expected to change the shape of the impact spectrum drastically at low frequencies, so far, have had little or no effect. Changing joist depths from 185 to 285 mm had negligible effect on low-frequency transmission. Changing I-beam depth from 235 to 460 mm did have a significant effect. The best wood joist floor tested to date is still worse than a 150-mm concrete slab. So far in the project, the most effective remedy for excessive low-frequency impact sound is mass.
Review of research reports on floor impact sound in Japan
Authors: Fukushima Hirokazu , Yasuoka Masahito
Published: October 2005

The authors have reviewed technical papers on floor impact sound reported in journals and proceedings published by the Architectual Institute of Japan, the Acoutical Society of Japan, and the Institute of Noise Control Engineering Japan. Research on this subject started in the 1960s in Japan, with considerable delay from the development in this area in North America and Europe. However, in the last decade, approximately 25 papers have appeared on this subject every year, and the total number of the relevant papers in Japan has been found to exceed 400. The number of research works on floor impact sound in Japan is characterized by the use of a soft and heavy impact source (a tire for compact cars). As is widely known, this source is the standard impact source unique to Japan. However, approximately half the recently reported papers deal with noise produced by the tapping machine. This change seems to reflect the Japanese consumers’ general choice of floor lining changing from carpets to wooden floor coverings. This report will discuss the Japanese trend in research on floor impact sound, not only through the number of papers but also through the introduction of relevant social background in Japan.
Impact force characteristics by footsteps and the vibration characteristics in housing floors
Authors: Watanabe Hideo , Kimura Sho , Inoue Katsuo
Published: October 2005

Various experimental researches were conducted dealing with physical characteristics and sensation in order to be able to express quantitatively appropriate hardness in terms of comfortability in residential floors. From among these, the results of analyses of impact force time characteristics of a sole by footsteps and the production of a prototype impact source involving the impact force of the heel, which is a representative impact force generated by footsteps are indicated in this paper. Then, using the impact source to measure the amount of dynamic displacement in wooden floors of old construction and in the direct-pasted wood flooring commonly used in recent multifamily housing, the dynamic spring coefficient of traditional wood-construction floor structure, which had never before been clarified, was quantitatively determined.
30 years of resilient furring channels
Authors: Harris David A.
Published: October 2005

This paper reviews the development, testing, and present day use of the venerable resilient furring channel, one of the most successful acoustical products invented. Billions of feet have been produced and ‘‘it’s still going.’’ Wall and floor/ceiling systems with resilient furring channels were, and still are, among the most practical techniques for improving the airborne and impact sound transmission characteristics of wood frame structures. Performance data on typical assemblies will be presented along with tips to maximize field performance.