bluuebunny/crossref_metadata_2025 huggingface dataset:
Influence of nonuniform transducer velocity distribution in the near-field echo formation from a cylindrical rod
Authors: Lawu Tjundewo , Tabei Makoto , Ueda Mitsuhiro
Published: October 2005

The echo formation in the near field of an ultrasonic transducer is strongly influenced by the amplitude distribution of the transducer. In this paper, the velocity amplitude on the face of a plane circular transducer was estimated using on-axis transient radiation of the actual transducer. The velocity amplitude can be obtained by deconvolving the pressure waveform with the electrical characteristics of the transducer and then integrating the result with respect to time. Using the estimated velocity amplitude, the scattering of wideband ultrasonic pulses from a cylindrical rod was then numerically calculated by applying a mathematical description of the echo [Lawu et al., J. Acoust. Soc. Am. 98, 2809–2818 (1995)]. Results of these estimations agree well with those obtained experimentally. The proposed method improves the accuracy of the echo waveforms scattered by an object in the near field of the transducer.
A new configuration of dummy head microphones using neural networks
Authors: Eto Masahiro , Hamada Hareo
Published: October 2005

A binaural recording and reproduction system using a dummy head microphone (DHM) has been used in various applications such as virtual reality systems, transaural reproduction systems for entertainment, and evaluations of many sound fields. However, the size and weight of the DHM have been inconvenient for many practical uses. An alternative to an ordinary DHM was proposed which consisted of a microphone array and a neural network controller. Thanks to the electronic configuration of the new DHM, many advantages are expected. For example, not only will it include a downsized microphone, but also several advantages, including multiple HRTF configurations, could be realized. In this paper, an efficient algorithm is used in the learning process of the neural network. The results of subjective evaluation for perceived directions compared to those of the ordinary DHM will be shown.
An STFT-CS method for estimation of temporarily changing room transfer functions under noisy conditions
Authors: Okuno Takatoshi , Hirobayashi Shigeki , Nakajima Hirofumi , Tohyama Mikio
Published: October 2005

A robust echo canceller using an STFT-CS (short-time Fourier transform and cross spectrum) method under noisy conditions was proposed [J. Acoust. Soc. Am. 99, 2505(A) (1996)]. This method is developed by using an algorithm which estimates the room transfer function (RTF) using a frame average of the cross spectrum between the input and error signals. Taking the cross-spectrum frame average reduces the RTF estimation errors produced by surrounding noise. The robustness and convergence properties of STFT-CS and other conventional methods under noisy conditions are investigated. STFT-CS is robust to surrounding noise, but it does not rapidly follow temporarily changing RTFs due to taking the frame average. Scattering waves from listeners (and/or talkers) cause significant changes in the RTF. Thus the scattering sound fields from a sphere are investigated as an initial step to develop an RTF prediction method for STFT-CS under noisy conditions. The time waveforms and frequency characteristics of the scattering waves are analyzed numerically, and the RTF prediction errors are evaluated experimentally for STFT-CS and conventional methods under noisy conditions.
Sounds of insect of flight
Authors: Sakayanagi Masako , Ohnuki Takeshi , Kawachi Keiji
Published: October 2005

Aeroacoustics has been introduced since the momentum equations of fluid were transformed into the equations of sound transmission by Lighthill. Although the transmission from a sound source can be analyzed, little attention has been paid to a source with unsteady motion. An insect (Bumblebee: Bombus terrestris) was used as a sound source, and the mechanism caused by vibrating wings was analyzed. The sound produced by flying insects consists of a fundamental frequency, which corresponds to the wing beat itself, and many harmonics, which are caused by the physical characteristics of the wings and their motion through the air. Measuring the distribution of the acoustic radiation, an attempt was made to decompose the sound into noises by wing thickness, pressure fluctua- tion, and vortex motion. The result showed that the insect fluttering noise was mainly based on the pressure fluctuation on the wing surface.
Low-frequency impedance measurement using the two-microphone tube technique
Authors: Peng Chaoying , Morrey Denise , Sanders Phillip
Published: October 2005

As progress is made toward environmental protection and developing lightweight vehicles, noise control in automobiles is becoming increasingly significant. For designing a quieter vehicle, the knowledge of acoustic impedance of the trim materials is crucial to the prediction of the interior noise level in the passenger compartment. Since automobile noise is usually dominated by low frequencies, 50 to 300 Hz, the knowledge of low-frequency impedance of trim materials is particularly important. This paper presents the use of the two-microphone tube technique to measure low-frequency impedance. A purpose-built impedance tube was developed based on theoretical analysis of the evaluation accuracy of impedance. The measurement results are reported for the tests on some typical trim materials, carpet, underlay, seat foam, and acoustic foam, over the frequency range of 100 to 600 Hz. It has been found that the location of microphones is crucial to the accuracy of measurement. Analytical and experimental results showed that the ideal microphone location is that both microphone spacing and the distance from the tested material to the nearest microphone takes the value of a quarter wavelength. Care should be taken to avoid a microphone spacing of a half-wavelength of the frequency of interest.
A real-time, polyphonic physical modeling synthesizer
Authors: Leary Andrew B.
Published: October 2005

Physical modeling is a new synthesis technique which creates sound using mathematical models of actual musical instruments. These models are computationally efficient and allow real-time control and musical expression through a small number of natural parameters. Algorithms that simulate plucked strings, woodwind, brass, and other instruments have been developed and implemented in a real-time, DSP-based synthesizer. These algorithms are described and parameter estimation results based on analysis of real instruments are presented. Results show that physical model parameters and corresponding excitation signals can be found from analysis. Models so derived are not only accurate representations of the sound, but are also expressive and controllable like the instruments they model. A commercial, polyphonic, multitimbral music synthesizer is described which uses physical modeling as one of its synthesis methods. In this keyboard, custom DSPs provide an unprecedented amount of synthesis power and flexibility. Practical issues of implementation and real-time control are discussed and demonstrated. For example, real-time control of pitch, damping, dispersion, and pick location are demonstrated for a guitar model. A short demo that shows the expressiveness and realism of physical modeling is presented.
Efficiency issues in model-based synthesis of plucked string instruments
Authors: Karjalainen Matti
Published: October 2005

Efficient model-based synthesis of string instruments is typically based on digital waveguides or extensions of the Karplus–Strong (KS) model. The detailed relation of these two cases is analyzed in this presentation by deriving the KS model from a more physical waveguide formulation. The equivalence of these models is discussed and their computational efficiencies are compared. For most instruments and playing conditions KS-type models are preferable from the point of view of fast real-time implementations. In some cases the waveguide model is necessary for detailed synthesis, such as for instruments exhibiting nonlinearities, since it is capable of properly simulating true wave propagation. Methods to extend the KS model in order to include beats and some other nontrivial behaviors are discussed. Finally, the capacity of present signal and RISC processors to implement real-time sound synthesis is estimated. [Work supported by the Academy of Finland.]
In search of efficient physics-based piano tone synthesis methods
Authors: Van Duyne Scott A.
Published: October 2005

The piano is a complicated, subtle tone-producing device. Much work has been done toward understanding its physical and acoustical mechanisms. On the other hand, excellent results have been achieved in commercial sound synthesis, particularly through the sampling and processing of real piano sounds. The present work builds on digital waveguide string synthesis methods to develop physics-based synthesis algorithms of piano and percussion sounds through appropriate physical models chosen with an eye toward psychoacoustics-based simplifications and toward mathematical reformulation into efficient, physically calibratible digital filter structures. Recently developed physics-based sound synthesis algorithms include: a shared-loss string coupling structure modeling two-stage decay, and a simplified string stiffness filter design, both calibratible from recorded sound data; a time-varying wave digital piano hammer or mallet filter directly modeling nonlinear felt stiffness and hysteresis; membrane, plate, room, and piano soundboard models using a 2-D, or 3-D digital waveguide mesh designed for high-speed multiply-free parallel hardware implementation; and a highly simplified commuted piano synthesis method, which commutes the high-order soundboard filter backward through the strings and linearized hammer filter, replacing it with its own impulse response, which in turn is synthesized directly with enveloped noise.
New developments in time-domain modeling of plates for the simulation of string instruments
Authors: Chaigne Antoine J. , Lambourg Christophe
Published: October 2005

Pertinent simulations of string instruments, for the purpose of sound synthesis or instrument making, require at least three major conditions: a deep understanding of the physics, accurate measurements of elastic and geometric properties, and good mathematical properties of the discrete time modeling. Today, the modeling of complete string instruments appears to be a rather unrealistic objective. However, one intermediate step consists of a thorough investigation of the vibrations and radiation of plates, which are essential in the production of sound by such sources. This presentation will concentrate first on the development of appropriate damping models for both isotropic and anisotropic plates, using a generalized viscoelastic formulation, in order to account for the complex, but clearly audible, influence of frequency-dependent decay times on timbre of most materials. Other refinements, such as the time-domain modeling of inhomogeneities, stiffening of plates, and plate–cavity coupling, which are known to be significant in the physics of string instruments, will be addressed. Sound examples will be played, highlighting the results and limits of the state of the art for these current investigations.
Digital waveguide modeling of woodwind toneholes
Authors: Scavone Gary P. , Smith Julius O.
Published: October 2005

Woodwind tonehole parameters measured by Keefe [D. H. Keefe, J. Acoust. Soc. Am. 72, 676–699 (1982); ibid. 88, 35–51 (1990)] are converted to traveling-wave scattering parameters suitable for digital waveguide implementation. Digital filters are designed to approximate the reflection and transmission transfer functions implied by the Keefe data. The final result is that each tonehole is modeled using a single, first-order digital filter. The digital waveguide discrete-time simulation technique requires significantly less computation time than traditional multiconvolution time-domain methods, without loss of accuracy. The results of a digital waveguide six-hole flute bore implementation are compared to Keefe (1990), with excellent agreement.
Reducing computational complexity in struck string physical models
Authors: Bilbao Stefan
Published: October 2005

Chaigne and Askenfelt recently published a paper in which they described a physical model of a struck string [J. Acoust. Soc. Am. 95, 1112– 1118 (1994)]. The model is based on a finite difference approximation to a particular partial differential equation, and involves updating transversal displacement at discrete locations along a string. A nonlinear hammer is also incorporated into the model. It is shown in this paper that, if it is desired to know the output only at one particular point (or perhaps a weighted linear combination of points) along the string, then the number of operations required in order to compute the synthesized output can be reduced. This is done by expressing Chaigne and Askenfelt’s model in state space form, and by then performing appropriate coordinate transformations to ‘‘modal‘‘ coordinates. What is more, computational complexity in the modal coordinates will be independent of the number of spatial derivatives modeled, so that accurate modeling of dispersion can be achieved at no extra cost. The coordinate-changing method applies equally well to the case of spatially varying media, and is simply extended to models which are higher order in time.
Automatic transcription of simple polyphonic music
Authors: Martin Keith D.
Published: October 2005

A novel computational system has been constructed which is capable of transcribing piano performances of four-voice Bach chorales written in the style of 18th century counterpoint. The system is based on the ‘‘blackboard’’ architecture, which combines top-down and bottom-up processing with a representation that is natural for the stated musical domain. Knowledge from auditory physiology, physical sound production, and musical practice has been successfully integrated in the current implementation, enabling the system to infer the presence of octaves, thereby overcoming a common pitfall of many traditional bottom-up transcription systems. The architecture and incorporated knowledge will be presented, along with an appraisal of the system’s performance and comparisons with other systems in the literature. [Work supported by NSF.]
Progress in the pulse-tube refrigeration and the thermoacoustic theory in Asia
Authors: Tominaga Akira
Published: October 2005

There are a wide variety of thermoacoustic phenomena. They are classified as thermally induced spontaneous oscillations, prime movers, refrigerators, and others. Recent progress in pulse-tube refrigeration is remarkable in Asia. Common features of these phenomena are oscillation of fluid bounded by solid matter and axial gradient of temperature. The traditional viewpoint of the regenerator cannot describe functions of the stack and the pulse tube because it neglects the irreversible process. Linear network analysis neglects effects of the axial gradient of temperature. The thermoacoustic theory developed by N. Rott includes both the irreversible process and the temperature gradient, and it was supported by experiments on the stability limit of the spontaneous oscillation. To understand the thermodynamic aspects of the theory, concepts of heat flow and work flow were introduced. Interchange between these flows are related to the axial gradient of temperature. Extension of the thermoacoustic theory to include the progressive wave in addition to the standing wave made it possible to understand devices equipped with the regenerator by the help of a computer code THERMOACOUSTICA.
Thermoacoustics at high power density
Authors: Swift Gregory W. , Gardner David L.
Published: October 2005

Some applications of thermoacoustics will require high power density to reduce the cost and size of the hardware needed for a given total power. Such high power density can be reached by using high acoustic amplitudes, but this leads to significant differences between the actual operation and predictions based on the acoustic approximation. To date, all such differences have resulted in reduced efficiency. Experimental results for several thermoacoustic engines operated with acoustic pressure amplitudes near 10% of mean pressure will be presented. Causes of deviation from acoustic-approximation predictions, including entrance effects, turbulence, and harmonic content, will be discussed. Experimental and theoretical approaches to understanding these deviations will be proposed.
A model for transverse heat transfer in thermoacoustic devices
Authors: Mozurkewich George
Published: October 2005

Improved understanding of heat transfer in thermoacoustic heat exchangers is desirable for the design of practical thermoacoustic devices. Such heat transfer is transverse (as opposed to axial) with respect to the stack pores. Current thermoacoustic analyses [e.g., Swift, J. Acoust. Soc. Am. 84, 1145–1180 (1988)] cannot treat time-average transverse heat transfer because they assume that the time-average temperature of the gas in any section is the same as that of the immediately adjacent section of the stack. The present analysis lifts this restriction by augmenting Swift’s treatment of the oscillating temperature fields in gas and stack with an analogous treatment of the time-average temperature fields. Because the goal is to develop physical understanding, rather than to obtain quantitatively precise results, simplifying assumptions are introduced. The main assumption is that the gas and stack can be treated as essentially one-dimensional entities, within which any transverse temperature variations may be neglected. Within the approximations, the equations may be solved analytically. Results compare favorably with recent numerical work by Cao et al. [J. Acoust. Soc. Am. 99, 3456 (1996)]. Applications to thermoacoustic heat exchangers will be presented.
An experimental thermoacoustically driven thermoacoustic refrigerator
Authors: Hofler Thomas J. , Adeff Jay A. , Heatter Tom , Atchley Anthony A.
Published: October 2005

The construction of a thermoacoustically (heat) driven, thermoacoustic refrigerator (TADTAR) is nearing completion. The design utilizes a novel resonator topology and high-efficiency primary heat exchangers. Numerical models suggest a cooling power level of at least 500 W over a temperature span of 40 °C. The design should be easily scalable to much higher cooling power levels. Experimental data characterizing the prime mover portion of the engine will be presented, possibly along with some preliminary refrigeration data. [Work supported by ONR.]
Modulation rate discrimination by normal-hearing and hearing-impaired listeners
Authors: Grant Ken W. , Summers Van , Leek Marjorie R.
Published: October 2005

Sensorineural hearing loss has been consistently associated with deficits in frequency analysis in addition to the loss of sensitivity. However, temporal processing may be relatively normal in many hearing-impaired (HI) individuals. For these listeners it might be possible to improve speech recognition by coding certain dynamic spectral cues (e.g., variations in voice fundamental frequency, formant frequency transitions, etc.) as temporal modulations of simple carrier waves. For this signal-processing strategy to be successful, listeners must be able to discriminate changes in the rate of temporal modulations. In the present study, modulation detection and rate discrimination were examined at three different standard modulation rates (80, 160, and 320 Hz) and at various modulation depths for both normally hearing (NH) and HI subjects. Measures of pure-tone frequency discrimination were also obtained at these same test frequencies. For all subjects, modulation rate discrimination thresholds increased with increases in standard rate and decreases in depth of modulation. For HI subjects, modulation rate discrimination thresholds were approximately equal to pure-tone frequency discrimination thresholds at 80 Hz but were much worse at higher rates. These results suggest that for these subjects, transforming dynamic spectral speech cues into dynamic temporal cues would be of little benefit. [Work supported by NIH.]
Objective and subjective measures of hearing aid sound quality
Authors: Chiu Eva , Jamieson Donald G. , Parsa Vijay , Cornelisse Leonard
Published: October 2005

Subjective assessments of the severity of hearing aid distortion were obtained using both rating scale and paired comparison measures, with sentence materials. Data were collected under 30 hearing aid conditions, representing a broad range of real hearing aid distortions. These data were related to a range of objective (engineering) measures of hearing aid distortion, including techniques based on pure tones (THD, IMD), measures using broadband noise stimuli having certain speech-like characteristics, and measures derived from real speech. Data from both normal and hearing-impaired subjects were stable and orderly, with good agreement between datasets collected using paired comparison and rating scale methods. Correlations with behavioral data were significantly better for certain speech-based objective measures than electroacoustic measures based on noise or pure-tone signals. [Work supported by the Natural Sciences and Engineering Research Council of Canada.]
Aging of hearing ability
Authors: Yanagida Masuzo
Published: October 2005

The average life span of the Japanese has been drastically extended in recent decades thanks to developments in medical science and treatments. Considerations for the aged are demanded in every aspect of social life including voice announcements by synthetic speech. The real situation of hearing synthetic speech by aged persons is clarified through dictation tests on synthetic speech generated by a commercial synthesizer system using texts from Murphy’s law for 58 subjects aged 10 to 80 years old. In order to investigate the effects of hearing loss and language ability, dictation results were compared with results of hearing tests for sinusoidal waves and language ability tests by various types of written and oral tests. Dictation scores of primary school children and aged persons show larger variances compared with those for subjects 20–59 years old. Rates of mishearing by aged persons having hearing difficulty sometimes exceed 50% in mora units, although those of primary school children remain within 18% at the most. The correlation coefficient between dictation score and comprehension score was 0.76. Occurrence rates of phonemic confusion on plosive sounds by subjects with hearing difficulty show clear differences between voiced plosives and unvoiced plosives. [Work supported by STA, Japan.]
Phoneme recognition by a non-native English-speaking cochlear implantee using the WSP, MSP, and spectra speech processors: A case study
Authors: Copmann Kathryn S.
Published: October 2005

This study presents the performance of a non-native English-speaking adult using the Nucleus 22 Channel cochlear implant. Included are the subject’s (1) history, (2) promontory stimulation results, (3) audiological information, (4) pre- and post-implantation speech recognition test results, (5) post-implantation auditory phoneme recognition using the wearable speech processor (WSP), the minispeech processor (MSP), and the Spectra processor, and (6) the subject’s subjective reports of benefits in terms of communication and social interactions. The subject, a highly motivated and intelligent male who, presented with a profound bilateral, progressive sensorineural hearing loss of unknown etiology, was 59 years old when the right ear was implanted. Prior to implantation the subject was aided in the left ear, but had never been aided in the right ear. The subject’s performance using each of three processors is presented for medial vowel recognition and medial consonant recognition using auditory stimulation only. Results are displayed through confusion matrices generated based on the subject’s responses to these acoustic stimuli. In summary, the vowel and consonant phoneme recognition of a subject with the Nucleus cochlear implant using various processors is presented. Improved phoneme recognition is demonstrated with filter banking versus feature extraction coding strategies.
Learning and generalization in auditory backward masking
Authors: Wright Beverly A. , Johnston Paul A. , Reid Miriam D.
Published: October 2005

Excessive auditory backward masking has been observed in elderly persons and in children with language impairments. The purposes of this project were to determine (1) whether practice can improve the ability to detect a backward-masked signal and, if so, (2) whether this improvement generalizes to other masking tasks. These issues were examined in six normal-hearing adults using an adaptive, 2IFC procedure. Each subject was asked to detect a tonal signal (10 ms, 1000 Hz) presented immediately before a noise masker (300 ms, 200–1800 Hz, 40-dB SPL spectrum level), on 900 trials per day for ten days. Before and after the training period, subjects were tested on five other masking tasks in addition to the trained task. A one-way ANOVA revealed a significant learning curve over the ten days of training [F(9,45)=5.1, p<0.001], with the average signal threshold decreasing from 63 to 53 dB SPL. Comparison of the pre- and post-test results showed significant threshold reductions in other backward-masking tasks, but not in simultaneous- and forward-masking tasks. These results indicate that (1) the interference from an auditory backward masker can be reduced with practice, and that (2) this learning is specific to backward masking. [Work supported by the James S. McDonnell Foundation and NIDCD.]
Feature extraction of phonemes from autocorrelation functions of auditory filterbank outputs
Authors: Sato Koichi , Toyama Jun , Shimbo Masaru
Published: October 2005

Features extracted from auditory filterbank outputs are useful as a front end for speech recognition systems. If an auditory filter has a center frequency (CF)[Hz], the extracted feature has been, in most cases, used at the CF[Hz] itself. In the case of inputting an f [Hz] pure tone into the auditory filterbank, the feature spreads across a wide frequency, though the output frequencies of any filters are f [Hz]. Therefore, the feature must be used not at CF[Hz], but at a special frequency which depends on the input wave. An autocorrelation function is useful for extracting such a special frequency. If the function has peaks at every T [s], the output of the filter mostly includes the component of 1/T [Hz]. Thus the special frequency and the feature can be extracted from the peaks of the autocorrelation function. As a result, the features extracted from all filters are not spread across a wide range of frequency. At low frequencies (less than about 1 kHz), the features are distributed among the pitch of the voice and its harmonies. At high frequencies, where pitch harmonies are not separated, the features are distributed among voice formants.
Plane-wave acoustic scattering by two strongly interacting shells above coincidence
Authors: Gaunaurd G. C. , Huang H.
Published: October 2005

The scattering of plane sound waves incident in arbitrary directions on two elastic, fluid-filled, spherical shells in water is analyzed exactly. The shells can be close to each other, and thus can be strongly interacting. The analysis is performed in a broad frequency band that extends beyond the midfrequency (or ‘‘coincidence’’) enhancement region of the backscattered echoes. It is then possible to exactly account for the many, very complex features present at high frequencies within the scattering cross sections of multiply interacting shells. The incident and scattered wave fields are expanded in normal-mode series and the addition theorem for the spherical wave functions is used to determine the exact expression for the sound fields scattered by each shell in the presence—and repeatedly interacting with—the other, referred to as coordinate systems at the centers of either shell. Numerical evaluations involve the solution of a truncated, ill-conditioned, coupled, complex, transcendental, matrix system by means of the Gauss–Seidel iteration method. The large signature enhancement present for the end-fire configuration, above the coincidence region, is determined and explained as a novel elasto-structural acoustic effect caused by the focusing and reradiation of the a0 Lamb wave in the shells.
Full-field visualization of ultrasonic waves in elastic solids using phase-stepping digital shearography
Authors: Bard Benjamin A. , Gordon Grant A. , Wu Shudong
Published: October 2005

Measurement of surface displacements, due to elastic wave propagation, has traditionally been studied using single-point measurement techniques, such as ultrasonic transducers or interferometers. Full-field methods, excluding scanning techniques, are uncommon, especially outside of research environments—examples include optical holography and Schlieren photography. Laser-modulated phase-stepping digital shearography is a full-field, common-path, interferometric method which can be used to quickly visualize ultrasonic wave fields in industrial settings. A shearography system will be described that provides not only whole-field images of the ultrasonic waves, but quantitative displacement data within these fields. Images of the displacement fields produced by Lamb and quasi-Rayleigh waves were recorded as they traveled along surfaces of several plates and bars. Evidence of wave reflections, multimode propagation, and flaw interaction will be discussed. [Work supported by the Penn State University Applied Research Lab and the Navy Manufacturing Science and Technology LINC Program.]
Real-time estimation of sentence structure from prosody
Authors: Ichikawa Akira , Imiya Atsushi , Shimizu Nagayuki , Horiuchi Yasuo
Published: October 2005

New methods are proposed by which the sentence structure of utterances can be estimated from the prosody under real-time conditions. The conventional Hough transform algorithm is modified so that it will be able to extract approximated straight lines of pitch patterns in real time. The sensitivity of the approximation can be established at will. So, parallel transform processes with different sensitivities can give multilevel components of an utterance structure and can make a tree in real time. The tree is not exactly a grammatical one, but rather a psychological one or one of the speaker’s intentions. Using this structure, a sentence tree matching algorithm is developed to recognize the sentences. The trees are transformed to regular trees by adding dummy nodes to express trees as lists, and the editing distance of the tree is computed recursively by a list-matching technique. Each of the extracted lines of fine sensitivity are classified into codes by vector quantization. Speaker’s intention states or dialogue control information in the conversational speech will be related to certain sequences of the codes. One hundred and twenty-eight spontaneous conversations (total length is about 21 h) were collected from 64 subjects. The performance of the proposed methods was checked by the CORPUS.
Good prosody facilitates comprehension
Authors: Sanderman A. A. , Collier R.
Published: October 2005

Previous research showed that prosodic features are perceptually conspicuous in marking boundaries and that a suitable implementation of these features improves the quality of synthetic speech in terms of acceptability. Assumably, listeners use prosody to compute the informational structure of the input speech. The hypothesis is that differently phrased utterances may lead to differences in cognitive load during comprehension. A method was designed in which a kind of verification task is combined with a question-answering task. The stimulus set consisted of structurally ambiguous sentences. The expectations were as follows: (a) when listeners receive a question followed by an appropriately phrased utterance, they will react more rapidly than when it is followed by an utterance without phrasing; (b) in the latter situation reaction times (RTs) will still be shorter than if an inappropriately phrased utterance is presented. The results confirmed these expectations for the situation where the prosodically intended reading corresponded to the most likely interpretation of the ambiguous sentences. However, when the least likely interpretation was the prosodically suggested one, it appeared that a question followed by an inappropriately phrased utterance gave the same RTs as one without phrasing. But an appropriately phrased utterance still produced the fastest RTs.
Effects and limitations of prosody on the resolution of syntactic ambiguity in Japanese
Authors: Misono Yasuko , Mazuka Reiko , Kondo Tadahisa , Kiritani Shigeru
Published: October 2005

The role of prosodic cues in interpreting syntactically ambiguous Japanese sentences was examined with special reference to the interaction with semantic and pragmatic biases. Syntactically ambiguous sentences with varying degrees of semantic and pragmatic bias were constructed. The degree of bias in each sentence was evaluated through visual presentation experiments. At the same time, auditory presentation experiments were conducted using speech samples recorded with prosody maximally favoring each of the possible interpretations of the sentences. The results showed that the prosodic cues could influence the interpretation of a sentence even when the sentence was semantically strongly biased, refuting the view where the prosodic cues play a role only when another source of information such as semantics and pragmatics failed to specify the interpretaiton of the sentence. The results of the present study also showed the limitation of the prosodic cues in that the prosodic biases alone were not sufficient to fully determine the interpretation of the sentences even when the sentence biases were neutral.
Temporal organization of segmental and prosodic features of speech in the word accent of common Japanese
Authors: Fujisaki Hiroya , Ohno Sumio , Ueno Tohru , Tomita Osamu
Published: October 2005

The acoustic–phonetic characteristics of the word accent in common Japanese are based on the moraic structure by which segmental and prosodic features are temporally coordinated. The present study investigates the temporal organization of these features with an emphasis on the influence of ‘‘moraic phonemes.’’ The speech material consists of words of varying lengths, phonetic constituents, and accent types embedded in a carrier sentence. The timing of the prosodic features was represented by the onset/offset of the accent command, estimated from the F0 contour using a quantitative model for the process of F0 contour generation. The timing of the segmental features was represented by the acoustic–phonetic boundaries, detected by referring to the frequency–time–intensity patterns. While the timing of the accent command is strongly correlated with the onset of the vowel rather than with the onset of the mora-initial consonant, segmental constituency, especially the presence/absence of the moraic phonemes, is found to exert strong systematic influences on their relative timing. The results are formulated as a quantitative model of timing for use in speech synthesis.
Prosodic domain effects in real-word English sentences
Authors: Keating Patricia A.
Published: October 2005

Fougeron and Keating [J. Acoust. Soc. Am. 97, 3384(A) (1995)] showed that there is an effect of position in prosodic domain on the articulation of English /n/ in reiterant speech, using electropalatographic (EPG) contact as the articulatory measure. Fougeron and Keating [Proceedings of the Autrans Workshop and Seminar, pp. 93–96 (1996)] showed a similar effect on the articulation of French /t/ and /n/ in real-word sentences. The present study adds to this research effort by examining real-word English sentences. In each of these sentences a sequence of segments spans one of four prosodic boundaries: syllable, phonological word, small phrase, or intonational phrase. The corpus includes several sentences for each boundary level. Several American speakers will read this corpus while EPG and acoustic data are collected. The amount and duration of EPG contact for /t/ and /n/ will be compared across the four conditions. Based on previous work, these measures are expected to distinguish three or four levels. The larger size of this sample will allow better study of intra- and interspeaker variation. [Work supported by NSF.]
A new control model based on rising and falling fundamental frequency
Authors: Moriyama Takaaki , Ogawa Hitoshi , Tenpaku Seiichi
Published: October 2005

The fundamental frequency (F0) control in speech production has not yet been understood completely. There exists a rich variety of F0 patterns depending on a speech production style (i.e., emotional influence, dialects, speaker’s intentions and habits, and so forth). These F0 patterns are controlled by the laryngeal mechanism. Part of this mechanism, the contraction of the cricothyroid muscle (CT) makes F0 rise, and the sternohyoid muscle (SH) movement causes F0 to fall. Up to now, few studies were made on F0 falling, but, in order to generate variousF 0 patterns, it is believed that it is necessary to control both the rise and fall of F0. Therefore a newF 0 control model is proposed which has two parameters. The first parameter, Tau1, represents F0 rising, and the second parameter, Tau2, represents F0 falling. In addition, some primary results on controlling F0 by using the proposed model are included in this report. Currently, progressive studies on F0 patterns of the Osaka dialect are being made. TheF 0 falling patterns in the Osaka dialect are more frequently observed. By applying the proposed model to the utterance of the Osaka dialect, the flexibility of the proposed model could be examined.
Effects of task instructions on voice F0 responses to auditory feedback pitch shifts
Authors: Burnett Theresa A. , Singh Shajila A. , Kiran Swathi , Larson Charles R.
Published: October 2005

Particular manipulations in auditory feedback pitch differentially affect voice F0 [Burnett et al., J. Voice (in press)]. Subjects who sustained constant-pitch phonation of the vowel /a/ with auditory feedback set at a masking-loudness level appeared to produce a multicomponent voice F0 change following upward or downward directed shifts in feedback pitch. The first response appeared to be relatively automatic, with a latency of approximately 150 ms and a consistent response direction opposite the stimulus direction. The second, more variable response appeared to arise from a voluntary effort to adjust voice F0. Until now, the characteristics of this second response were hidden by its close temporal proximity to the first response. The purpose of this study was to separate and further study these two responses through manipulation of task instructions. Subjects were alternately instructed to ignore the auditory feedback pitch shifts, change their pitch in the same direction as the pitch shift, or change their pitch in the opposite direction. Through this method, the duration of the first response and the latency of the second response were determined. The results are consistent with and expand upon the previous findings of two voiceF 0 response components.
Production of accentual prominence patterns in English by Mandarin Chinese speakers
Authors: McGory Julia Tevis
Published: October 2005

This study investigated productions of accentual prominence patterns in English by native English speakers (ENG) and by two groups of Mandarin Chinese speakers differing in amounts of English language experience, M1 (less-experienced) and M2 (more-experienced). Because the falling intonation patterns of declarative English sentences are acoustically similar to high-falling tone 4 in Mandarin, and low-rising intonation patterns of interrogatives are similar to low tone 3, transfer of these tonal constructs by Mandarin speakers into productions of English intonation patterns is expected. The stress-pair, memorizes/memorial was produced with three levels of accentual prominence, from most prominent to completely unaccented, in declarative and interrogative contexts. Comparisons of f0, duration, and intensity levels between the unstressed and stressed first and second target syllables were made. Preliminary results suggest that M1 use acoustic characteristics of prominence from their native language while M2 perform more like ENG. M2 and ENG subjects produced progressively greater changes in f0, duration, and intensity within three levels of accent. In M1 productions, similar rises in f0, durations and intensities were evident in the stressed syllable of target words regardless of accent level and type indicating that transfer of Mandarin prosodic constructs occurs in the early stages of language acquisition.
Boundary tones and focus realization in African American English intonations
Authors: Jun Sun-Ah , Foreman Christina
Published: October 2005

Intonation of African American English (AAE) has been impressionistically described as having a final level rising and an early rise contour different from that of mainstream American English (MAE) [Tarone (1972)]. But to our knowledge, no systematic description of AAE intonation has been done based on phonetic experiments. In this study, tone patterns occurring at the final edge (nuclear pitch accent, phrase tone, and boundary tone) of different sentence types were examined. Tonal patterns were compared based on the framework of Pierrehumbert and her colleagues [Pierrehumbert, 1980]. In addition, the location of focus was varied to examine how focus interacts with the early rise or final level rise in AAE. Three dialogue sets acted out by female speakers of AAE and MAE were recorded. Preliminary results show that both groups use the same final edge tone patterns for declaratives and wh-questions, but different patterns for yes/no questions (L*, H-, upstepped H% for MAE vs H*, H-, level H% for AAE) and vocatives. For focused sentences, AAE very often added postnuclear pitch accents when nonfinal words were focused. Furthermore, AAE used a sharp final fall preceded by a high tone (H+L*L−L*) when they have a negative attitude toward the sentence or situation.
An analysis of the high and low plateau of Japanese English
Authors: Ueyama Motoku
Published: October 2005

The tonal contour of focus in English intonation is H*L−L% in declaratives and L*H−H% in interrogatives, within the framework developed by Pierrehumbert (1980). Consequently, L- or H- show low- or high-plateau patterns, respectively. Japanese intonation does not permit such total deaccentuation after focus because of the need of preserving lexical pitch accents characterized by a sharp rise and fall. In earlier work on intonation, not much attention was paid to the interaction of the intonation of a first and second language. This study will analyze high- and low-plateau patterns of Japanese English. The speech of beginning and advanced Japanese learners of English was compared to the speech of native speakers of American English. The production data show the following patterns. The high plateau was often produced as a low plateau by both levels of Japanese learners due to the sharp pitch fall after the high peak. In contrast, the low plateau was produced as a sequence of downstepped H*. These patterns can be attributed to either the effect of Japanese intonation or possible universal constraints on intonation. These hypotheses will be further evaluated using additional data on Italian learners of English.
Acquisition of contrastive emphasis
Authors: Jannedy Stefanie
Published: October 2005

In English, broad and late narrow focus have identical accent patterns: Pitch accents are aligned with the last accentable constituent within an intonation contour. Under early narrow focus, the pitch accent is claimed to shift to an earlier location and the last accentable syllable is deaccented. This study examined acquisition of the narrow focus pattern, using a picture-naming task, in which either the object or its color differed from the previous picture. Subjects were asked to use adjective–noun sequences to name each picture and then to name the difference from the previous picture. Spontaneous utterances from 42 children (age 3–10) and six adults were obtained with implicit and explicit contrast (narrow focus) on either adjective or noun, and labeled prosodically for pitch-accent type and location. A subset of the utterances was used in a perception experiment: 18 naive listeners judged either the adjective or the noun to be contrastively emphasized. An ANOVA on the contrast-type. Listeners consistently rate the adjective to be more emphasized when implicitly contrasted. Yet, a majority-rule count on the tonal transcriptions for accentuation on this subset shows almost exclusively ‘‘noun-accented’’ ratings, suggesting that nouns were not completely deaccented when narrow focus was on the adjective.
The perceptual and acoustical analysis of speech intelligibility in Mandarin-speaking adults with cerebral palsy
Authors: Liu Huei-Mei , Tseng Chin-Hsing
Published: October 2005

This study investigated the relationship between specific acoustic features and perceptual judgment of speech intelligibility of Mandarin-speaking young adults with cerebral palsy. Use of the minimum phonetic contrast task allowed for intelligibility analysis and correlated acoustic analysis according to specific articulators spatial and temporal movements. Selected task contrasts included vowel height, vowel frontness, aspiration–unaspiration, affricate–fricative, stop–nasal, nasal, and stop articulation place. Speech materials consisted of 90 bisyllabic words. Perceptual judgments from item identification and scaling procedures reflected the diversity of speech intelligibility in 20 cerebral-palsied adults. Vowel contrasts yielded higher perceptual accuracy than consonant. All phonetic contrasts correlated well with speech intelligibility. However, multiple regression analysis revealed that three contrasts (aspiration–unaspiration, affricate–fricative, and vowel frontness) accounted for 99% intelligibility variability. Seven acoustic parameters were examined; they were F1, F2−F1, VOT, frication duration, percentage of burst appearance, nasal characteristic, and burst spectrum. F1, F2−F1, nasal characteristic, and burst spectrum were found to differentiate the cerebral-palsied subjects from the normal speakers. Results from multiple regression predicted intelligibility scores with 74.8% accuracy by acoustic parameters related to F2−F1, VOT, and percentage of burst appearance. These findings are discussed in relation to specific areas of production deficiency in the cerebral-palsied population.
Variability of voice quality ratings
Authors: Kreiman Jody , Gerratt Bruce R. , Bell Theodore S.
Published: October 2005

Surprising variability emerged when the distribution of the quality ratings that individual voices received in data sets that met conventional standards for rating reliability was examined. Whatever quality was rated or scale used, individual raters were self-consistent in judgments of voices near scale end points, but consistency for nonextreme voices varied widely by voice. Raters differed in how they defined scale endpoints. Pairs of raters did not agree consistently in ratings for any voices, and groups of raters taken together never agreed well in their ratings for any but the most extremely pathologic stimuli. Traditional measures of rating reliability do not document this variability, and may be misleading when used for this purpose. These findings suggest that quality scales are probably better treated as categorical, rather than as interval, with category boundaries defined individually for each listener. Ratings averaged across listeners appear to have little meaning, and ‘‘moderately pathological’’ is probably undefinable perceptually. These findings are also consistent with evidence from nonparametric multidimensional scaling [Kreiman and Gerratt, J. Acoust. Soc. Am. (in press)], which suggests that the distinction between severe and mild-to-moderate pathology is the only perceptually valid dimension for pathological voice quality. [Work supported by NIDCD.]