bluuebunny/crossref_metadata_2025 huggingface dataset:
Vibration modes of the violin forced via the bridge and action of the soundpost
Authors: Saldner Henrik O. , Molin Nils-Erik , Jansson Erik V.
Published: October 2005

The action of the soundpost in a violin is not well understood and fundamental knowledge is needed. Therefore low-frequency forced modes, operating deflection shapes, 250–600 Hz of a violin were investigated. Harmonic, sine wave, excitation was applied via the bridge (resembling playing) or via the sound from a loudspeaker. The latter excitation gives a reciprocal measure of radiativity. The modes were investigated with TV-holography and electro-acoustical admittance measurements. A carefully selected violin was investigated without soundpost, with soundpost, and with the soundpost position shifted toward the centerline. Answers were sought to three questions: Which are the violin modes forced via the bridge? To what extent do they radiate sound? What is the action of the soundpost? The observed modes were mostly not pure normal modes but mode combinations. Generally there is a nodal line close to the soundpost position of the plates. Schelleng has hypothesized that the main action of the soundpost is to make the violin nonsymmetric. The first top plate mode with soundpost can be looked upon as a combination of the first symmetric top plate mode and the first asymmetric mode without soundpost. The first mode is an efficient radiator and the second mode transfers the string vibrations via the bridge to the top plate efficiently. The asymmetric first mode with the soundpost is thus both excited and radiates sound. Our experimental results make the hypothesis plausible, but show that body (global) modes play an important role too. In the range 500 to 600 Hz, a body mode, C3, constitutes a major part. The C3 mode has a nodal line close to the normal soundpost position and is especially interesting as it seems to form a quality mark of a good violin.
Impulse responses for feedback to the driver of a musical wind instrument
Authors: Ayers R. Dean
Published: October 2005

This paper generalizes the linear part of Robert T. Schumacher’s approach for modeling wind instruments [R. T. Schumacher, Acustica 48, 71–85 (1981)]. A reinterpretation of that work shows that the general problem is one of deconvolving an infinite series of multiple reflections, rather than attempting to define an anechoic termination for the input end. In order for the deconvolution to yield the desired results, the reflection impulse response from which the series is generated must not include long-period multiple reflections itself. That condition is satisfied automatically in an air column that begins with a cylindrical segment of appreciable length. For air columns that are tapered or more complicated at their input ends, the impulse responses needed are already available from previous work [R. D. Ayers, J. Acoust. Soc. Am. 98, 81–87 (1995)]. Calculated impulse responses for a primitive trumpet are presented and discussed.
Inertial cavitation produced by pulsed ultrasound in controlled host media
Authors: Deng Cheri X. , Xu Qihong , Apfel Robert E. , Holland Christy K.
Published: October 2005

The purpose of this work is to provide measurements in well-characterized media in order to better understand inertial (transient) cavitation phenomena. Focused transducers with megahertz center frequencies (2.5 MHz, 4.3 MHz) and a clinical diagnostic ultrasound system (4.0 MHz) were used to generate pulsed ultrasound to induce cavitation. An improved active cavitation detector which utilizes a highly focused transducer with much higher center frequency (30 MHz) was used to measure the threshold of inertial cavitation. In order to study the effect of the concentration of nucleation agents on cavitation thresholds, experiments were conducted in distilled water seeded with polystyrene particles. Inertial cavitation thresholds were measured for various concentrations of polystyrene particles. It was observed that the threshold decreased from 2.5 MPa at concentration of about 106 particles/ml to 1.6 MPa at a concentration of about 109 particles/ml. The effect of the concentration is not significant for smaller changes of the concentration. Measurements of the cavitation thresholds were then made in specially developed phantom materials to study the effect of viscosity on the cavitation threshold when surface tension and other mechanical properties of the materials are kept relatively constant. Experimental results show that the threshold increases with increasing viscosity, consistent with theoretical predictions. Cavitation was also detected in water seeded with polystyrene particles using a clinical ultrasound system at an acoustic pressure of 3.84 MPa. Results are comparable with those obtained in the laboratory using a 4.3-MHz focused transducer.
Development of an analysis method for acoustic–structural coupling systems using the BEM for the acoustic field: The validity of the method developed and comparison with the FEM acoustic field
Authors: Hagiwara Ichiro , Kozukue Wakae , Okada Hiroshi
Published: October 2005

A formulation is developed for the analysis of coupled acoustic–structural systems using the boundary element method (BEM) in order to apply the coupled method to the open field in addition to the closed field. In this formulation the most general mode superposition method by Ma-Hagiwara, namely, the high- and low-order truncatable mode superposition method, is used where the higher and the lower modes are truncated. The propriety of this formulation is verified by comparing it with a theoretical solution in a one-dimensional problem and with the solution by a method in an interior noise problem, which was developed using the finite-element method (FEM).
Vibration and sound radiation of viscoelastically damped plates supported in elastic frames
Authors: Kanazawa Jyun-ichi , Yamaguchi Yoshio , Akamatsu Katsuji
Published: October 2005

Modal properties of viscoelastically damped plates that were clamped at the edges between sandwich rectangular frames were obtained by FEM analysis and modal strain energy analysis. The modal damping loss factors were calculated at various temperatures using the reduced-temperature nomogram of the applied damping material obtained from the measurements based on the Oberst beam test method. Modal frequencies and loss factors of test plates were measured by the half-power bandwidth method in a thermostatic chamber and were compared with the analytical results. There are small differences of the modal frequencies between the analytical and experimental values. The differences are assumed to be due to the boundary condition at the interface of plate and frames. The loss factors show good agreement for plate modes. The vibration of the point-exited plates was calculated using the modal properties. The transfer functions between the applied forces and the vibration responses were compared with measurements. Finally, the radiated sound field was evaluated from the vibration responses by BEM analysis, and acoustical intensity measurements were carried out at the surface of the plates. Mean-square velocity and radiated sound power vary inversely as loss factor.
Analytical prediction of sound transmission through automotive door seal systems
Authors: Gur Yuksel , Morman Kenneth N.
Published: October 2005

A finite element-based acoustic-structure interaction modeling procedure is developed to determine the sound transmission loss characteristics of automotive door seals. The model accounts for the effects of seal system compression, the density and constitutive behavior of the materials used in the seal construction, individual seal cross-sectional geometry, and spacing between layers of seals. The results of a sensitivity analysis conducted using the modeling procedure reveals that the overall transmission loss of door seal systems is most sensitive to changes in: seal material density, geometry of the seal in the compressed deformed configuration, and geometry of the cavities formed by the space between seals of adjoining layers. An important implication arising from the sensitivity analysis for the design of door systems for closing efforts is that, without changing seal system compression load deflection behavior, the sound transmission characteristics may be altered substantially by changing the spacing between layers of seals in the system. The results of the sensitivity study and an application to a prototype door seal system demonstrate that the new modeling procedure can be used to effectively eliminate much of the ‘‘cut and try’’ from the present process of designing door seal systems.
Active control of turbulent boundary layer noise transmission through an aeroelastic plate
Authors: Frampton Kenneth D. , Clark Robert L.
Published: October 2005

The reduction of sound transmission to the interior fuselage of modern jet aircraft is gaining increasing concern in an ever-competitive international market. In an attempt to identify effective, weight-saving noise control strategies, active control of sound transmission through elastic plates has received much attention in the literature. While many of these investigations have demonstrated promising results, none of them have considered the effects on plate dynamics by convected fluid loading. Past investigations into the effects of convected fluid loading on plates (i.e., aeroelasticity) have demonstrated significant effects on plate response in the high subsonic and supersonic Mach number range. This is the typical operating range for current and future commercial aircraft. The objective of this investigation is to demonstrate the effectiveness of active control on the transmission of turbulent boundary layer noise through an aeroelastic plate and to highlight the importance of including convected fluid loading in the model. Development of a fully coupled model will be presented for a plate subjected to convected fluid loading on one side and acoustic loading on the other will be presented. Finally, the effectiveness of active control employing co-located piezoelectric transducers on the plate will be presented.
Spectral representation of speech using mel-generalized cepstral coefficients
Authors: Koishida Kazuhito , Tokuda Keiichi , Kobayashi Takao , Imai Satoshi
Published: October 2005

In mel-generalized cepstral analysis, the model spectrum can be varied continuously from AR to cepstral modeling by changing the value of a parameter γ and choosing an appropriate model spectrum. Furthermore, the spectrum represented by mel-generalized cepstral coefficients has a frequency resolution similar to that of the human ear. Although it is expected that the mel-generalized cepstral coefficients are a useful parameter for speech coding and speech synthesis applications, the filter stability may become unstable after quantization of the mel-generalized cepstral coefficients. This paper presents a spectral representation of speech using mel-generalized cepstral coefficients. The filter stability can be easily ensured after quantization of the proposed spectral parameters. First, the statistical distribution of the proposed parameters is shown. Second, the computational complexities for calculating the proposed parameters from mel-generalized cepstral coefficients are investigated using a speech database. Finally, the quantization and interpolation properties of the proposed parameters are compared with those of LSP. Experimental results show that proposped parameters have better performance than LSP.
Optimizing acoustic parameters for phonetic features
Authors: Bitar Nabil N. , Espy-Wilson Carol Y.
Published: October 2005

In developing acoustic parameters based on phonetic features for speaker-independent speech recognition, it is important that the parameters (1) be based on relative measures as opposed to absolute ones to minimize speaker-dependent effects and (2) be selected from a pool of candidates according to some ‘‘goodness’’ criteria. In this study, place-of-articulation phonetic features are targeted to help classify obstruent sounds. It will be shown how acoustic phonetic knowledge and statistical analysis were combined to go from qualitative definitions of the acoustic correlates for phonetic features to computational algorithms for extraction of the relevant acoustic properties using discriminant analysis and classification trees. Further, it will be shown how using relative measures reduces speaker-dependent effects, specifically gender, on the acoustic parameters while honing on the phonetic information contained in the speech signal. [Work supported by NSF Research Grant No. IRI-9310518.]
Feature space reduction through nonlinear identity mapping
Authors: Ito Hironori , Takeda Kazuya , Itakura Fumitada
Published: October 2005

Reducing the dimension of acoustic feature space is realized using a wine-glass-type neural network, which has fewer number units in the middle layer than in the input and output layers, trained for the identity mapping. Three-layer wine-glass-type neural networks, which have 32 units for both input and output layers and two to five units for the middle layer are trained so as to map the input 32 LPC derived log spectrum vectors to the identical output vectors. After 500 iterations of backpropagation learning, the network can map the input to itself by 2.7 dB (five units) to 4.14 dB (two units) signal-to-deviation ratio of log spectrum for the training data. For the evaluation, DTW isolated word recognition experiments are performed using 132 similar city name utterances of a male speaker. Using the output of the middle layer units as the reduced feature vector, the recognition accuracy are 75.6% (two units) to 88.6% (five units). Since the accuracy using 16 cepstral coefficients derived from the original speech is 89.4%, the effectiveness of nonlinear identity mapping for reducing the feature dimension is confirmed.
An application of Dempster and Shafer’s probability theory to speech recognition
Authors: Kobayashi Tetsunori
Published: October 2005

The Dempster and Shafer probability theory is applied to calculation of the likelihood function for speech recognition. HMM is now the major technique for speech recognition. It, however, has some limitations because it adopts the Bayse-based likelihood function. In the Bayse theory, the values of likelihood functions are valid only in comparative situations. The values themselves are meaningless. Therefore, they are not applicable to spotting or branch pruning. Besides, there is no guiding principle for merging the likelihoods from different information sources. To solve these problems, the Dempster and Shafer probability is adopted. The DS theory can combine likelihood functions from multiple sources. It also treats the information of ‘‘do not know’’ or ‘‘cannot decide.’’ The values themselves can represent the certainty of the evidence. Here, it is adopted in three parts. First, the frame-level phonetic likelihood functions are calculated by merging the likelihood functions from multiframe features. Second, the segment-level phonetic likelihood functions are calculated by merging segmental and durational likelihood functions. Finally, the phonetic sequence is decided by using multi-segmental-level likelihood functions. All merging processes are performed by the DS theory. Thus the new speech recognition method, whose likelihood function is applicable to spotting and branch pruning, is realized.
Training section tolerant HMM in concatenated training
Authors: Matsuo Hiroshi , Ishigame Masaaki
Published: October 2005

Concatenated training of phoneme HMMs can use speech data without hand labels for training HMMs, but it has a tendency to decrease the recognition rate because of an improper training section. The results for speaker-independent isolated word recognition experiments showed that non left–right HMM is tolerant of the quality of the training section in concatenated training than the conventional left-to-right HMM. Non left–right HMM has a structure where state transitions within a phoneme are ergodic and state transitions between successive two phonemes are left-to-right. Non left-right HMM and the conventional left-to-right HMM show much the same performance as long as the training section is given properly by hand labels. Since the training section contains undesirable data, the recognition rate for both HMMs decreases, but the decrement of the recognition rate for non left-right HMM is smaller than the decrement for the conventional left-to-right HMM. Similar tendency was shown when the training section was determined by a phoneme duration model independently from the hand labels.
Speaker-independent phone modeling using speaker specific phoneme alignment
Authors: Tonomura Masahiro
Published: October 2005

Efficient parameter sharing is essential in the design of robust phoneme context-dependent HMMs for unseen phonetic context in the training data. Maximum likelihood success state splitting (ML-SSS) [H. Singer and M. Ostendorf, Proc. ICASSP, 601–604 (1996)], an iterative algorithm to design the topology of HMMs, is convenient for processing a speaker-independent database as it reduces the influence of speaker variation on the topology. However, to reduce the cost of iterative likelihood calculation, accurate phoneme boundaries are essential for this algorithm. This paper proposes a phoneme boundary estimation method for a speaker-independent database. First, for the individual speakers in the target database, speaker-adaptation HMMs are trained with embedded re-estimation using data that is to be used for boundary estimation. Then, phoneme boundaries in each speaker’s speech data are estimated with a Viterbi algorithm using the speaker-adaptation HMMs of each speaker. By applying this phoneme boundary decision method to build speaker-independent models, it was confirmed that phoneme recognition error was reduced by 8.3% compared to conventional speaker-independent models. This result shows the effectiveness of the automatic topology derivation and shows that phoneme boundaries estimated by the proposed method are accurate enough for model training.
Recent progress in the INRS speech recognition system
Authors: O’Shaughnessy Douglas D. , Li Zhishun , Farhat Azarshid , El Meliani Rachida , Vergin Rivarol , Héon Michel
Published: October 2005

The INRS large-vocabulary continuous-speech recognition system employs a two-pass search. First, inexpensive models prune the search space; then a powerful language model and detailed acoustic–phonetic models scrutinize the data. A new fast match with two-phone lookahead and pruning speeds up the search. In language modeling, excluding low-count statistics reduces memory (50% fewer bigrams and 92% fewer trigrams); with Wall Street Journal texts, excluding single-occurrence bigrams and trigrams with counts less than five yields little performance decrease. In acoustic modeling, separate male and female right-context VQ models and a bigram language model are used in the first pass, and right-context continuous models and a trigram language model are used in the second pass. A shared-distribution clustering uses a distortion measure based only on the weights of Gaussian mixtures in the HMM model. Testing the system with a 5000-word vocabulary, the word inclusion rate (i.e., correct word retained in the first pass) is about 99%; word recognition accuracy is about 91%. Keyword spotting with new types of fillers retains accuracy with 1.2 false alarms/hour/keyword. [Work supported by NSERC-Canada and FCAR-Quebec.]
A dialog control strategy based on the reliability of speech recognition
Authors: Niimi Yasuhisa
Published: October 2005

A dialog control strategy was proposed based on the reliability of speech recognition. Receiving an utterance including several items of information, the dialog system using this strategy accepts those items that have been recognized reliably, while it rejects the rest of the items and asks the speaker to utter an utterance including the unaccepted items. The operation of the dialog system was mathematically analyzed under the assumption that the system can compute the reliability R(I) of each item of information, independently of the position of the item in a sentence and accept the item if R(I) is greater than a threshold t, and that the probability p that an item be accepted and the probability q that the accepted item be correct are given by a function of the threshold t. The analysis showed that the operation of the system could be described by a Markov process, and derived two important quantities N and Pac to evaluate the performance of the dialog system: N is the average number of turns taken between the speaker and the system until the system accepts all the items and Pac is the probability that all the items will be recognized correctly.
A timing management method in multimodal dialog systems
Authors: Itou Katunobu , Hayamizu Satoru , Tanaka Kazuyo
Published: October 2005

A new method is proposed which manages timing of individual module actions and the whole system for multimodal dialog systems [Hasegawa et al., Proc. IJCAI, 82–87 (1995)]. Many integration methods proposed so far for dialog systems were based on the flow of information or the dependency of processes. Even when they treated the timing, they described only the time relation among multiple inputs. However, implementation of smooth dialog requires managing time relations between inputs and outputs and time behavior of modules. The proposed method introduces a time management server which manages the time aspect of information flow between modules. Each module is an independent client to the server. It gives an implementation framework to treat interrupt and integration of multiple inputs while maintaining smooth dialog. [Work supported by Real World Computing Program (MITI, Japan).]
Discourse segmentation for radio news
Authors: Suzuki Yoshimi , Fukumoto Fumiyo , Sekiguchi Yoshihiro
Published: October 2005

One of the typical problems in speech recognition of discourse which consists of different subject domains is to identify what kinds of subjects are included in the discourse. In this paper, radio news is focused on, and a method is proposed for dividing it into several subject domains. First, a method for the classification of newspaper articles in order to make referential subjects is described. Each subject is characterized by a χ2 vector which is calculated based on word frequencies. Then, by using χ2 vectors, a method for dividing radio news and classifying it into suitable subject domains is shown. Two experiments have been conducted. The first experiment is concerned with the clustering technique for newspaper articles. The data used contained 60 articles. They were classified into six referential subject domains. The results showed that the correct ratio was 96.6%. The second experiment is concerned with the segmentation for radio news and was conducted by using χ2 vectors which were obtained in the first experiment. Radio news utterances were manually translated into the romaji strings which nearly correspond to phoneme strings and were then used. The results showed that the precision, recall, and correctness attained about 90%, 30%, and 70%, respectively, and demonstrated the effect of the method.
Classification of news speech articles by keyword spotting
Authors: Sakurai Mitsuyasu , Ariki Yasuo
Published: October 2005

This paper proposes a method to automatically classify news speech articles using the word spotting technique. In word spotting, keyword probability Ps(w) ending at time t is computed by subtracting an accumulated probability of the best syllable sequence from the best syllable sequence + keyword probability. Keywords are extracted by finding the local peaks of the keyword probabilities. In article classification, the topic contribution of word (TCW) P(n|w) is computed in advance using ASAHI newspaper classification indices. Then a topic probability P(n)=ΣwP(w)×P(n|w) is computed using the TCW P(n|w) and keyword probability Ps(w) is substituted into P(w). This topic probability is the integration of acoustic word probability Ps(w) and a priori knowledge probability TCW. The news article can be classified by finding the highest topic probability P(n). The classification of news speech articles was carried out using 5-min NHK news programs over 30 days. The phoneme HMMs were trained using one-day news (5 min). In the experiment, 77 articles were classified using 183 keywords. The numbers of phoneme HMMs and syllables are 40 and 108, respectively. The classification rate is at present 49.4% (38/77). The correct rate of word spotting was 54.9% (567/1033) and false alarm was 1.2 fa/kw/h.
Multilingual speech recognition based on phonological elements
Authors: Williams Geoff , Martindale Geoff , Kaye Jonathan , Terry Mark
Published: October 2005

Phoneme-based approaches to ASR can achieve good performances in medium-sized vocabularies for individual languages. Transfer to a different target language, however, generally produces poor results. An alternative, phonologically based model which uses a set of seven universal subsegmental elements as recognition targets is presented. The elements are phonologically monovalent and are postulated to have a direct encoding in the acoustic signal. Speech segments are expressed in terms of elements and recognition proceeds by identifying the elements present in the speech signal. This model, evolved from government phonology theory, forms the basis of a single recognition module which can be applied to most languages by selection of the appropriate lexicon and linguistic constraints. Initial experiments to test this approach have used perceptually based front ends and a MLP classifier. For each element, individual neural nets were trained on the TIMIT (American-English) database. These classifiers were then tested on Spanish, German, and British-English (isolated words recorded by native speakers in nonstudio conditions). For elements recognized on TIMIT, average transfer rates to these languages were of the order −6%. The results are encouraging in terms of transfer to different languages. Results for Mandarin Chinese and Japanese will also be presented.
Optimization of speaker recognition model based on dynamic features of two-dimensional mel-cepstrum
Authors: Miyajima Chiyomi , Kitamura Tadashi
Published: October 2005

This paper describes an optimization of TDMC-VQ based speaker recognition model. Two-dimensional mel-cepstrum (TDMC) is defined as the two-dimensional Fourier transform of a mel-frequency scaled logarithm spectra in the frequency and time domains. It consists of averaged and dynamic spectral features of the two-dimensional mel log-spectra in the analyzed interval. It has been demonstrated that TDMC is very effective for word recognition and robust for a noisy environment. The speaker model is based on vector quantization of the TDMC parameter and is created by a self-organizing algorithm. In this study, TDMC and a conventional mel-cepstrum were used as the speech parameter. An optimization by a weighting function applied to the components of the speech parameter and a combination of the two parameters were studied. In order to evaluate the proposed model, text-independent speaker identification experiments for eight female speakers were carried out. Each model learns speech data of about 18-s length and identifies 200 input sentences. The experimental results have shown that the proposed model with an optimal weighting function gives better performance than the usual method and gives 100% identification performance for 1.4-s input speech. The model also shows robustness for time interval.
Cepstral analysis of ‘‘cold-speech’’ for speaker recognition: A second look
Authors: Tull Renetta G. , Rutledge Janet C. , Larson Charles R.
Published: October 2005

Speech affected by colds continues to be an issue in speaker recognition technology. This study is a continuation of the ‘‘cold-affected’’ speech project [R. G. Tull and J. C. Rutledge, J. Acoust. Soc. Am. 99, 2549(A) (1996)] which compared ‘‘cold-speech’’ and normal/healthy speech of a male subject to analyze differences in vowel durations and mel-cepstral coefficients. This new study analyzes the speech of two additional male subjects during a cold and after a cold to test speaker intrasession (‘‘cold-speech’’ within the same recording session, normal speech within the same session), and speaker intersession (‘‘cold-speech’’ versus normal speech on different days). The sentence being used for recording is the recitation of numbers: ‘‘1 2 3 4 5 6 7 8 9 10.’’ The lower-order mel-cepstral coefficients are chosen as parameters (independent variable) in this study. The research for this study shows that there are patterns in the coefficients (c2 and c3) of ‘‘cold-speech’’ that are different from the patterns of the same coefficients in normal speech. These differences suggest that the cold, as a temporary communications disorder, introduces another level of intra- speaker variability that needs to be addressed in the design of speaker recognition systems. Confronting the ‘‘cold-speech’’ through manipulation of the mel-cepstral coefficients may improve a system’s ability to recognize a speaker with a cold as the customer, rather than as an imposter.
Language and dialect identification by syllabic spectral features
Authors: Li K.-P.
Published: October 2005

A speaker-based automatic language identification system has been studied to discriminate both languages and dialects. Syllabic spectral features are extracted automatically by an artificial neural network. The system makes no use of phonetic segmentation for recognition or training. High performance has been obtained for monologue telephone recordings. The system has been applied to several different databases to study the discrimination possible among different languages, dialects, and accents with such limited phonetic–acoustic information. The system performance is also related to individual speaker characteristics and the size of reference population for each language, and the length of testing samples. This study will report experimental results related to performance and data environments. It will also report an attempt by others to compare recognition performance of trained or expert human listeners with automatic recognition results.
Speech synthesis by rule based on VCV waveform synthesis units
Authors: Koyama Takao , Koizumi Nubuo
Published: October 2005

In recent years, several systematic speech synthesis methods in Japanese, that are based on the pitch synchronous waveform superposition method (PSOLA), have been proposed. Generally, the phonemic or the syllabic unit is used as the synthesis unit in these methods. Since these methods require the phonemic concatenation in the phoneme boundary position, the superposition of noise becomes a common problem. To solve the noise problem, a method is proposed based on VCV synthesis units. VCV synthesis units are concatenated in the vowel steady section. To improve the VCV method which does not consider using the unvoiced vowel originally, unvoiced vowels as the exceptional segments are regarded, and included the segments in the waveform database. When the VCV unit is divided into more primitive parts, it is useful to take the spectrum distortion into consideration when choosing segments. The method has been evaluated with 50 synthesized voices, and it was found that there is a correlation between evaluated score and spectrum distortions.
Multilevel approach to rule-based speech synthesis using quasiarticulatory parameters
Authors: Williams David R. , Stevens Kenneth N. , Carlson Eric , Bickley Corine A.
Published: October 2005

Rules are being developed for controlling a formant synthesizer (HLsyn) in which ten higher-level parameters control changes in vocal-tract natural frequencies, areas of four orifices, and other aspects of articulation. The input to the rules is a phonetic sequence labeled in terms of distinctive features. The rules operate in four stages. First, the time locations of a sequence of landmarks are established. The landmarks are of three types: locations of vowel nuclei and offglides, locations of amplitude minima for glides, and times of formation and release of consonantal constrictions. Second, formant trajectories between vowel landmarks are derived. Third, these vowel-to-vowel formant movements are modified by the effects of consonantal constrictions, working from left to right. Fourth, the effects of secondary consonant articulations are added, including introduction of a velopharyngeal opening, spreading or constricting glottal movements, and influences of segmentally related vocal fold stiffening and vocal tract expansion or contraction for obstruents. Trajectories of the parameters are, for the most part, constructed from quarter-cycle segments of sinusoids. Examples of synthesis of simple VCVCV utterances illustrate the vowel-to-vowel coarticulation and context-dependent vowel formants that result from application of the rules. [Supported in part by NIH Grant No. MH52358.]
Comparing techniques for synthesizing emotional speech
Authors: Henton Caroline
Published: October 2005

Comparative attempts to incorporate vocal emotions into synthetic speech using two different speech synthesizers are presented. On a Macintosh computer running the concatenative speech synthesizer MacinTalkPro2 it is possible to generate emotions in synthetic speech, using a very limited number of parameters [C. Henton, J. Acoust. Soc. Am. 95, 3010(A) (1994); C. Henton and B. Edelman, Multimedia Tools and Applications 3, 1–25 (1996)]. These parameters are: average speaking pitch; pitch range; pitch movement; speech rate; volume; duration; and silence. Traditional formant synthesizers, such as Digital Equipment Corporation’s DECtalk, allow for emotional affect to be added through the manipulation and interaction of a great many more acoustic parameters. The outcome of using the same limited number of parameters to create the same emotions in both systems will be analyzed. Output from the two speech synthesizers reading the same emotionally colored passage will be demonstrated. Improvements and suggestions for how to incorporate emotional synthetic speech into applications according to the sophistication of the user, the speed/memory limitations of the synthesizer and/or CPU, and the intended use for the application will be offered. The need for different gender-specific emotional profiles for adults and children will also be explored.
Synthesis of wideband speech from narrow-band speech based on piecewise linear mapping
Authors: Nakatoh Yoshihisa , Tsushima Mineo , Norimatsu Takeshi
Published: October 2005

On the analog telephone and the mobile telephone system, the bandwidth of the speech is limited from 300 Hz to 3.4 kHz. This paper proposes a new recovery method of wideband speech (50 Hz–7.4 kHz) from narrow-band speech based on piecewise linear mapping. In this method, first, the narrow-band spectrum envelope of input speech is transformed to a wideband spectrum envelope using linearly transformed matrices which are associated with several spectral spaces. These matrices were estimated by speech training data, so as to minimize the mean-square error between the transformed and the original spectra. Second, the residual wave with a wideband spectrum is generated from the residual wave of input speech by the nonlinear computation. Finally, the reconstructed wideband speech is synthesized from the transformed spectrum envelope and the generated residual wave. This algorithm is compared to the following other methods: (1) the codebook mapping [Y. Yoshida and M. Abe, Proc. of ICSLP94, 1591–1594 (1994)]; and (2) the neural network. Through the evaluation by the spectral distance measure, it was found that the proposed method achieved a 0.67-dB maximum lower spectral distortion than the other methods. Perceptual experiments indicate a good performance for the reconstructed wideband speech.
Shadowing—A possible method for assessment of synthetic speech
Authors: Lyberg Bertil , Hällgren Åsa
Published: October 2005

Today synthetic speech is often based on concatenation of natural speech such as polyphones. There is a need for methods for assessing the overall speech quality of this type of synthesis. In this investigation a method based on shadowing is presented. The subjects listen to spoken utterances produced by the system to be evaluated. At the same time the subjects repeat the utterances as fast as possible. The time difference between the incoming and the repeated speech is measured at the vowel onsets and the averaged time difference over the utterances is used as a measure of the speech quality. The shadowing technique mirrors the ability of the subjects to predict the future of the incoming utterances at every point in time. This technique can therefore be used as a means to judge the accuracy of incoming cues, such as segment duration and the fundamental frequency contour. Results of this method will be presented for different types of speech synthesis and the sensitivity of this method for prosodic cues will be shown.
LPC-based voice transformation from adult to child
Authors: Wong Darrell , Lange Robert , Long Russel , Story Brad , Titze Ingo
Published: October 2005

An effective transformation of sentence level voiced speech from young to aged, or male to female, requires the parametrization and subsequent modification of various components in speech. A glottal excited linear predictive model (GELP), similar to that developed by Childers and Hu [D. G. Childers and H. T. Hu, J. Acoust. Soc. Am. 96, 2026–2036 (1994)] was used as a basic platform for experiments in gender, age, and vocal quality transformation. This model permits a parametrization initially into excitation and articulation components, which were then separately modified. This study describes the changes in pitch, prosody, glottal pulse shape, and formant structure necessary to evoke a perceptual transformation from an adult male to a prepubescent child. The transformation was performed at the sentence level. A time-domain glottal pulse model which is sensitive to gender/age was used, along with a male-to-child vocal-tract pseudoarea warping. [Work supported by NIH Grant DCO2532-01.]
Characterizing human ability to discriminate talkers over low data rate voice coders
Authors: Schmidt-Nielsen Astrid
Published: October 2005

Talker discrimination over low data rate voice coders (2400 bits/s) was evaluated in three separate experiments using a paired comparisons task. Test materials were Harvard sentences spoken by ten male and ten female talkers from the Boston area. Listeners were asked to decide whether two different sentences were spoken by the same person or by two different people. They then judged how dissimilar the two voices were using a five-point scale. The ability to discriminate among the voices (measured by d′) was compared to the subjective judgments of the perceptual distance between the voices. The effect of different types of voice coders on talker discriminability and on the perceived talker space was compared with uncoded speech. Comparisons with traditional measures of intelligibility and voice quality suggest that higher intelligibility or quality scores are not necessarily related to better talker discrimination, but dissimilarity ratings of different talkers were more closely related to voice quality scores. Listeners from two different parts of the country also perceived the talker space differently. [Work supported by NAVSPAWARSYSCOM.]
Speech enhancement by LPC cepstrum vector quantization
Authors: Kobatake Hidefumi , Yoshida Katsu , Suzuki Hideta
Published: October 2005

This paper presents a new speech enhancement method. It consists of two processing steps. The first processing step is the estimation of speech spectrum from noisy speech, for which a VQ method is adopted that uses two LPC cepstrum codebooks. One of the codebooks is generated from a clean speech database and the second one corresponds to that of noisy speech. Each entry of the noisy speech codebook corresponds to one of the entries of the clean speech codebook. Extracted VQ code of noisy speech is transformed into a clean one using the correspondence between the noisy and the clean codebooks. This method, based on the pairing of two codebooks, makes it possible to recover attenuated or lost formants. The second processing step is to restore the speech waveform. A modified Wiener filtering has been adopted. Also developed is a VQ method of noisy speech using only the clean speech codebook. The estimated LPC cepstrum coefficients of noisy speech are transformed into clean ones using an experimentally determined mathematical model. The noisy speech codebook depends on the signal-to-noise ratio and noise characteristics. The second method can be applied to any noise conditions. Experimental results showed the effectiveness of the proposed method.
Multimedia telecommunication quality with temporal discrete degradation
Authors: Aoki Hitoshi , Yamazak Tetsurou
Published: October 2005

The importance of considering the time-variant factors in the evaluation of multimedia telecommunication quality is discussed from the viewpoints of future multimedia networds and services. The quality degradation caused by bit erorrs in LD-CELP and ADPCM codecs was measured under conditions of variable speech-sample duration, a varying number of errored bits, and variable error-burst duation. The degradation in the mean-opinion score (i.e., the DMOS) for communication with variable speech-sample duration was quantified for both the number of errored bits and the error-burst duration. The results due to the number of errored bits were significant at a significance level of 1% under all conditions, as well as due to the interaction between the number of errored bits and the error-burst duration under some conditions. The effects of the quality factors were different among codecs. For example, the longer the error-burst duration, the better the quality in LD-CELP, while in ADPCM the duration did not have any effect. These experiments identified several items that should be considered in developing a quality-evaluation method, including determining such testing parameters as speech-sample duration, a measure for expressing degradation, and a procedure for subjective evaluation.
Telephone operator assisting system
Authors: Inoue Naomi , Nakamura Makoto , Sakayori Shin’ichi , Yamamoto Seiichi , Yato Fumihiro
Published: October 2005

KDD has developed a telephone operator assisting system, which provides operators with automatic database inquiry using speech recognition. An outline of the system is described and the results of a field trial are evaluated. There have been many speech recognition systems for practical use. A fast recognition method is one of the important subjects for practical systems. For this purpose, various methods are proposed. [Sagayama et al. ‘‘New Approaches Toward Fast Speech Recognition,’’ Proc. Acoust. Soc. Jpn. 25–28 (1996) (in Japanese)]. However, it takes some frames to output recognition results after finishing speech input, because any method deals with a few 100-ms pause following speech, which is a margin to detect the speech end. Therefore, a new fast recognition method is proposed. The basic idea of this approach is to output results when inquiry keywords are decided, even before detecting the speech end. This approach makes it possible not only to reduce speech recognition time, but also to recognize speech including out-of-vocabulary words and nonspeech sounds at the end of a sentence. Evaluation results show that (1) speech recognition time was reduced 30% and (2) high recognition accuracy was achieved: 95% for only expected inputs, 92% for all inputs. Fifty percent of unexpected inputs included out-of-vocabulary words at the end of a sentence, and most of them can be recognized correctly using this method.
Geoacoustic modeling using the Biot theory
Authors: Stoll Robert D.
Published: October 2005

Since its first application to geoacoustic modeling of the seafloor in 1970, the Biot theory has proved to be a powerful tool for studying the acoustic response of a wide variety of sediments. By modeling the sediment as a porous, viscoelastic medium filled with a viscous fluid, the effects of changes in frequency-dependent compressibility of various components and the influence of different local mechanisms causing energy loss may be studied in an integrated manner. Using the principle of viscoelastic correspondence, it is shown how a variety of local losses resulting from ‘‘hidden degrees of freedom’’ can be incorporated into the overall model using viscoelastic operators. In particular, modeling the response of fine-grained, high-porosity sediment with a ‘‘cardhouse’’-type structure is compared with the approach required to model coarser sediment with lower porosity such as sand. As another example, the influence of gas bubbles including bubble resonance effects is included in the overall response by defining the fluid compressibility to be a complex, frequency-dependent parameter. [Work supported by ONR, Code 321OA.]
Poro-elastic and poro-acoustic propagation modeling
Authors: Collins Michael D. , Lingevitch Joseph F. , Kuperman W. A. , Schmidt H. , Siegmann William L.
Published: October 2005

Until recently, sound propagation modelers have treated the ocean bottom as an acoustic or elastic medium. This paper will discuss some recent progress in poro-elastic propagation modeling. Biot’s theory of poro-elasticity has been incorporated into normal mode, wave-number integration, and parabolic equation models. Models have also been developed for poro-acoustic media, the limit of Biot’s theory in which shear waves are neglected. A series of benchmark problems that emphasize poro-elastic and poro-acoustic effects has been designed to test the accuracy of the models and their implementations. In addition to the propagation models, techniques have been developed for solving scattering problems and inversion problems involving poro-elastic layers. [Work supported by ONR.]
Time domain finite difference methods for range-dependent Biot media
Authors: Stephen Ralph A.
Published: October 2005

The physics of sound propagation in saturated, porous (Biot) media differs from propagation in fluids and elastic/anelastic solids because of the existence of a second compressional wave, the ‘‘slow’’ wave. Many environments in bottom-interacting ocean acoustics consist of mud and sands which can be modeled as saturated, porous solids with low shear moduli. In order to understand the physical mechanisms responsible for acoustic propagation, attenuation and scattering in porous media, and to determine if the ‘‘slow’’ wave plays a role in these processes, we have extended our numerical scattering chamber (NSC) code to solve Biot’s equations in two-dimensional Cartesian coordinates. The NSC formulation is based on the method of time domain finite differences and is applicable to the rough and laterally heterogeneous structures that are found on the seafloor. Within the same NSC code we can compare acoustic, elastic, anelastic, and Biot models for realistic environments with sub-bottom heterogeneities and seafloor roughness with scale lengths comparable to acoustic wavelengths. All multiple interactions and mode conversions are included in the solutions. [Work supported by ONR.]
Normal-incidence, high-frequency bottom penetration in the soft gassy sediments of Eckernforde Bucht, Germany
Authors: Wilson Marcia A. , Stanic Steve
Published: October 2005

In June and July of 1994, a series of high-frequency bottom penetration measurements were made in the gassy sediment of Eckernforde Bucht, Germany. A multielement hydrophone array was buried in the sediments and insonified by a normal-incidence signal. Measurements were made of the water–sediment insertion loss and signal level as a function of frequency (15–40 kHz) and hydrophone depths (0, 14, 28, 71, 122, 152, and 158 cm). Results showed that hydrophones located in a gassy sublayer had signal levels that were higher than the signal level at a hydrophone located just below the sediment–water interface.