bluuebunny/crossref_metadata_2025 huggingface dataset:
Circumferential wave propagation in a fluid filled borehole: Theory
Authors: Orton M. , Plona T. J.
Published: October 2005

For our mathematical model we consider a right circular cylinder or borehole filled with fluid and surrounded by a homogeneous isotropic, elastic solid—the formation. An infinitesimally thin line source is located in the fluid parallel to the borehole axis and very close to the borehole wall. The formation compressional and shear velocities are both faster than the fluid velocity. We calculate the early time response for different angles between source and receiver. This is done with great accuracy using the first terms of the ray expansion. The arrival structure apparent in the synthetic waveforms is explained. The propagation characteristics of the circumferential creeping waves are derived from the dispersion curves for the corresponding traveling wave representation. Our results show that for most fast formations, typical borehole sizes, and transmitter-receiver spacings between 60° and 120°, circumferential formation dominated arrivals can be separated from fluid dominated modes provided the center frequency of the signal is sufficiently high. The model for the no loss case predicts that only the circumferential compressional and pseudo-Rayleigh waves are detectable and that the circumferential shear wave is highly attenuated.
Effects of noise on the cardiovascular system: Appraisal of epidemiologic evidence
Authors: Thompson Shirley J.
Published: October 2005

The epidemiologic evidence relating noise and cardiovascular disease was examined by a team of audiologists, cardiologists, and epidemiologists through critical analysis of the extant world literature. While relationships were reported for various cardiovascular parameters, high levels of noise were found to be most strongly associated with elevated blood pressure. Foreign studies revealed prevalence ratios for hypertension ranging from 2.2 to 9.7 among industrial workers exposed to noise of 90–116 dB. Use of the cross-sectional design and failure to analyze blood pressure as a continuous variable limited risk quantification. Deficiencies existed in at least four aspects of study design: noise exposure characterization, health outcome specification, sample selection, and data analysis. Multiple regression and multiple adjustment procedures commonly applied in cardiovascular research were used in only one study. Confounding factors and strong risk modifiers were seldom considered. Retrospective-cohort studies of occupationally exposed workers, intervention studies with noise abatement and hearing protection programs, and other recommendations aimed at improving study designs and developing statistical models for future research are discussed. [Work supported by EPA.]
Vibration of crystal oscillator devices in an airborne radar environment
Authors: Rossman Herman
Published: October 2005

In airborne radar applications of oscillators, vibration can cause frequency modulation (FM) of the resonator. The incidental FM initiated by the motion can produce sideband levels sufficiently high so that radar targets are masked. Furthermore, narrowband vibration (spikes) will cause false targets (false alarms) to be displayed on the radar screen. The acceleration sensitivity of crystal controlled oscillators will be discussed. The emphasis will be on describing how resonators influence the level of vibration an oscillator is permitted to experience and still perform within specification. In addition to this, methods for reducing the vibratory motion, caused by structureborne and airborne (acoustic) excitations, will also be discussed.
Theory of vocal registers: Review of the literature
Authors: Large John W.
Published: October 2005

For artistic voice production in Western music culture the singer is expected to develop and maintain a pleasing quality over a wide range of fundamental frequencies and intensities. To cope with the demands the singer employs different mechanisms called registers, which, especially prior to training, may exhibit marked differences in timbre. Whether a trained vocal artist chooses to use the characteristic timbre of a particular register or to try to make it blend with the timbre of an adjoining register depends on several factors including (1) the mood or character of the literary text and the music, (2) the performance practice of a particular style, and (3) the tenets and techniques of a specific vocal method. This review examines the literature of vocal registers with regard to number, names, ranges, sources, pedagogy, equalization, safety, and aesthetics. Four primary lines of investigation into the nature of vocal registers were revealed: (1) laryngeal adjustment, (2) supra-glottal coupling to the larynx, (3) subglottal coupling to the larynx, and (4) formant enhancement of different partials. García's laryngeal adjustment definition of vocal registers is modified to include nonlaryngeal determinants.
Voice quality in singing: An acoustic, electromyographic, and glottographic study
Authors: Estill Jo , Baer Thomas , Honda Kiyoshi , Harris Katherine
Published: October 2005

The present study was designed to investigate contributions of the larynx and vocal tract to the production of different singing voice qualities. The subject, a trained singer, produced isolated phonations of the vowel /i/ on each of five frequencies (196, 294, 392, 587, and 784 Hz) in six qualities. In previous studies, the six qualities have been named Speech, Falsetto, Sob (lowered larynx), Twang (nasalized), Opera, and Belting. EMG electrodes were inserted into four supralaryngeal muscles: levator palatini, palatopharyngeus, middle constrictor, and geniohyoid. Electroglottographic and photoglottographic signals were recorded simultaneously with the EMG. For each quality, the activity in all muscles generally increased with frequency, although there were differences in pattern across qualities. For example, the geniohyoid was most active for Opera and Belting, presumably to enlarge the pharynx. There were consistent differences in the acoustic spectra among qualities at all but the highest frequency. Relationships among the spectral characteristics of the acoustic signal, the muscle activity patterns, and the glottographic signals will be discussed. [This work was supported by NIH Grants NS-13870 and NS-13617 to Haskins Laboratories.]
The relationship between bright timbre, acoustic features, and larynx position: A comparison of different singing voices
Authors: Wang Shi-gian
Published: October 2005

This study demonstrates that the bright voice heard especially in the high notes of tenors can be produced in different professional singing techniques (such as covered voice, Chinese singing voices, Western early music singing voice etc.) with different larynx position and corresponding differences in acoustic features. The data supporting these claims come from video demonstration and spectral analysis of the various professional singing voices. My results are different from that of Shipp and Izdebski [J. Acoust. Soc. Am. 58, 1104 (1975)], who claim that professional singers usually maintain a lowered larynx position throughout their vocal range and that nonsingers generally positioned their larynx upward with an increase in frequency and tend to maintain a level at or above the larynx rest position throughout their range. My results also differ from those of Sundberg [STL-QPSR 1/1972; Sci. Am. No. 3 (1977)] who said that the larynx lowering typical of male professional singers explains the extra singing formant (between F3 and F4). Sundberg also claims that the extra singing formant explains brilliant timbre and the difference between the untrained singing voice and the speech as compared to the professional singing voice.
The role of signal processing in echolocation
Authors: Titlebaum Edward L.
Published: October 2005

The type of signals transmitted by various echolocating mammals are reviewed. These signals are discussed in light of the operating environments for each mammal. Consideration is given to the relative motions of the transmitter, target, and receiver, clutter and reverberation distributions, and target complexity. Each of these considerations is shown to have an effect on the choice of transmitted signal and receiver structure. Signal processing may be viewed as an attempt to optimize the signal-to-interference ratio, a parameter which includes all of the considerations discussed. It is concluded that no single signal is optimum for all situations. However, many signal classes have robust properties in these various environments.
Echolocation by bats and other animals in air
Authors: Simmons James A.
Published: October 2005

Bats use a biological sonar for remote sensing and spatial perception, emitting frequencies from 5 to over 200 kHz in a variety of frequency-modulated, constant-frequency, and harmonically-structured signal patterns. Using echolocation, bats (Eptesicus fuscus) can detect 5-mm spheres at distances up to 3 m, and 20-mm spheres at up to 5 m. Range (echo-delay) related gain control of echoes regulates echo strength to amplitudes for target-feature processing and helps to reject interference from a variety of causes. The accuracy of bearing (azimuth) determination is ±1.5° and of elevation determination is ±3°. The distance to targets is perceived from echo arrival time with an accuracy of 0.2 mm (1–2μs). Signal processing of echoes yields the capacity to perceive multiple targets, and adaptive changes in signals further control interference and match the bat's acoustic-imaging system to the requirements of different tasks. These capabilities are achieved by Eptesicus with 20–100 kHz, multiple-harmonic FM signals. Many other species of bats use long constant-frequency signals in a Doppler-sensing sonar which detects both overall velocity and target fluttering movements. Frequency accuracy of 0.02% is achieved using high-Q filters. A few species of birds echolocate with clicks in the 1–10 kHz range, and the fruit-bat Rousettus, echolocates with 10–40 kHz clicks. Across all echolocating species, the capacity to detect and avoid small obstacles is primarily determined by emitted wavelengths.
Lateralization: A comparison of five psychophysical procedures
Authors: Baumann J. N. , Yost W. A. , Dye R. H.
Published: October 2005

The lateralization performance of two subjects was compared across five psychophysical procedures. The procedures used were single interval, same-different (SD), two-interval forced choice, four-interval SD, and four-interval 2AFC. Psychometric functions were determined for each procedure by measuring d′ for phase differences of 2, 4, and 6 degrees (interaural time differences of 11, 22, and 33μs). The stimuli were 500-Hz tones of 250-ms duration, presented at 70 dB SPL. In order to facilitate comparison, d′ was not corrected for number of observation intervals, so that ratios of d′ could be formed and compared to those predicted by the Theory of Signal Detection (TSD). In general, performance was found to be superior with the 4-interval 2AFC task and worst with the single interval task. Differences between the psychometric functions produced with four-interval tasks and the single-interval task were larger than predicted by TSD. These discrepancies are discussed in terms of additional cues for motion and location provided by the four-interval tasks.
Conductive hearing loss and interaural discrimination under headphones
Authors: Zurek P. M.
Published: October 2005

Studies in the literature indicate that persons with extreme bilateral conductive hearing loss tend to have worse-than-normal interaural time and phase jnd's, normal interaural amplitude jnd's, and normal MLD's. Although some degradation of performance is expected for conductive cases because of poor interaural isolation and the consequent mixing of the two sounds at each cochlea, little attention has been given to quantitative modeling. We present an analysis of the effects of blocking the normal conductive pathway and thus increasing the relative level of sound conducted via the skull bones. Under conditions of interaural symmetry, little cross-over attenuation, and dominant bone-conducted components, interaural phase shifts in the stimulus result in inter-cochlear amplitude differences and interaural amplitude differences result in inter-cochlear phase shifts. The rates at which these two variables are converted to each other allows one to understand the data in terms of the vector resultants at the cochleas without assuming any processing abnormality. [Work supported by NIH.]
The rate of acquisition of pitch from sinusoidal and harmonic stimuli
Authors: Patterson Roy D. , Milroy Robert , Peters Robert
Published: October 2005

We compared the time taken to extract melodic pitch from a ten-component harmonic complex, with pitch in the range 80–130 Hz, and from its constituent sinusoids taken individually. A four-note, random melody was created from the first five notes of the diatonic scale; listeners had to indicate which note had changed on the second presentation of the melody. Note duration was varied from a nominal 10 to 80 ms. The low pitch of the harmonic complex was extracted faster than the low pitch of its constituent sinusoids (100, 200 Hz), but slower than the high pitch of its constituent sinusoids (400, 800 Hz).
Perception of harmonic intervals made by simultaneous complex tones
Authors: Houtsma A. J. M.
Published: October 2005

Most studies on pitch perception for complex tones deal with either isolated sounds (pitch matches) or with sequential pairs of such sounds (jnd or melodic interval experiments). Little attention has been given to the question how the auditory system perceives a sound of two or more simultaneous complex tones. Informal observation of musical behavior shows that if a chord, triad, or harmonic interval is played by a group of musical instruments, the various notes usually retain their identity in the perceived sound image. This poses the question how the auditory system parses and groups the spectral components of the total sound in order to compute the correct (simultaneous) pitches. An experiment was performed in which musically trained listeners were asked to identify harmonic intervals made up of two simultaneous complex tones, each comprising three successive random harmonics, by playing the intervals back on a keyboard. Tones were presented both diotically (all six spectral components to both ears) and dichotically (one three-tone complex to each ear). Data show that under most conditions e.g., most of the dichotic ones, the task can be done very well, but that diotic performance depends strongly on details of the total spectral pattern. This suggests that peripheral spectrum and central spectrum impose separate constraints on musical processing of complex sounds.
Combination-tone cancellation phase: Comparison of psychophysical and ear-canal measurements
Authors: Furst M. , Rabinowitz W. M. , Zurek P.
Published: October 2005

Subjects listened to a coherent three-tone stimulus, F1 < F2 < F3, with levels in the range of 45 to 75 dB SPL, where the frequencies were chosen so that the combination tone Fct = 2F1 − F2 = 2F2 − F3. With this stimulus, psychophysical cancellation of the pitch Fct can be obtained (without adding a cancellation tone), by adjusting the level and phase of F2—presumably resulting from 2F1 − F2 interacting with 2F2 − F3. After subjects performed such cancellations, the sound pressure of Fct in the ear canal was measured as a function of the phase of F2. Our hypothesis was that if ear-canal sound pressure at Fct derives from inner-ear mechanical activity primarily associated with the cochlear place characteristic for Fct, and if psychophysical cancellation eliminates activity at this place, then cancellation of the percept should result in cancellation of Fct in the ear canal. Results indicate that as the phase of F2 varied, the level of ear-canal Fct changed systematically by 10 to 15 dB. However, the phase for which the ear-canal Fct level was minimum was not equal to the phase of psychophysical cancellation. Thus it appars that the ear-canal acoustic distortion product is not a simple correlate of the perceptual combination tone. [Work supported by Myron Bantrell Fellowship.]
Analog versus digital measurement of aural acoustic immittance
Authors: Robinette Larry N. , Thompson David J.
Published: October 2005

Static and dynamic measurements of aural acoustic immittance (AAI) were made in couplers and in real ears using two analog and two digital instruments. Comparisons were made among instruments, and between the analog and digital approaches, with regard to accuracy, range, reliability, and validity (proper resolution of immittance components, measurement of real-ear immittance, stability of calibration in hard-walled couplers) of measures. The influence of environmental parameters (temperature, barometric pressure, shape of test cavity) and temporal constraints (total system latency, hysteresis) was also investigated. Relative advantages and disadvantages of the two measurement approaches for clinical determination of pressure-immittance functions and the acoustic reflex were cited. Results indicated promise for implementation of digital AAI testing, particularly with regard to dynamic measurements. [Work supported by RR and D of Veterans Administration.]
Epidemiologic study of foundry and shipyard workers exposed to vibration from hand-held pneumatic tools
Authors: Behrens Virginia , Taylor William , Wasserman Donald
Published: October 2005

A total of 451 workers in three foundries and one shipyard representing all workers using pneumatic chipping and grinding tools, together with a control population, were surveyed for the signs and symptoms of vibration syndrome. The results of a questionnaire and medical examination (to exclude all other causes of Raynaud's Phenomenon) showed that hourly workers at the shipyard had a prevalence of vibration syndrome of 19% and a latency of 17 years while foundry hourly workers had a prevalence of 28% and a latency of 1 year. Incentive foundry chipper and grinder groups had prevalences that ranged from 40% to 67% while latencies ranged from 1.5 to 3 years. Vibration measurements proceeded simultaneous to the medical study, and showed acceleration levels on the pneumatic chipping hammer of 2400 g(rms) on the chisels and 30 g(rms) on the handles in the frequency range of 6.5–1000 Hz. Pneumatic grinders varied depending on maintenance conditions from 0.5 to 79 g(rms) over the same frequency range. Recommendations are given for minimizing the vibration syndrome problem.
Analysis of a nonlinear cochlear feedback mechanism
Authors: Koshigoe Shozo , Tubis Arnold
Published: October 2005

The implications of a nonlinear electromechanical cochlear feedback model [S. Koshigoe, W. Kwok, and A. Tubis, J. Acoust. Soc. Am. Suppl. 1 72, S47 (1982)] are explored. The model is shown to be qualitatively compatible with data on stimulated and spontaneous emissions, combination-tone psychophysics, and changes in cochlear response induced by COCB stimulation and variation of the EP. [This work was supported by NSF.]
Measures of basilar membrane motion obtained from intact cochleas of guinea pigs via fiber optic lever
Authors: Cook R. O. , Hamm C. W. , Konishi T.
Published: October 2005

Quantitative measures of the displacement vibratory pattern of the basilar membrane (BM) were obtained from the third turn of anesthesized guinea pigs via a single 60-μ fiber probe whose output intensity was amplitude modulated by reflection from the cochlear partition. The cochlea was penetrated via an 80-μ hole drilled into the scala tympani as far as possible from the BM. Since the penetration was nearly perpendicular to the BM, the fiber tip was ground at a 45° angle and coated with gold to effect the 90° turn necessary to direct light onto the BM. After in situ calibration, the fiber was fixed in the hole with a dental fixative. Both input/output and frequency response functions were obtained. For frequencies near the best frequency of the site (ca. 500 Hz), the technique allowed displacement resolution corresponding to 20 dB SPL or less at the eardrum. Preliminary data indicate that significant amplitude nonlinearity and physiological vulnerability begin at moderate SPL at this site.
Sound pressure changes in the ear canal produced by local damage to the inner ear
Authors: Khanna S. M. , Galley M. J.
Published: October 2005

Cats were prepared for round window (RW) cochlear microphonic (CM) measurements. To provide access to the basilar membrane, the round window membrane was removed. The sound pressure level (SPL) at the tympanic membrane (TM) was measured using a modified Sokolich acoustic driver and probe microphone system. CM was measured between 0.1 and 30 kHz at 50, 60, and 70 dB SPL. A soft probe was advanced through the round window opening to touch the basilar membrane (BM) at a selected spot and then withdrawn. The SPL and the CM measurements were then repeated. Touching the BM with the probe produced local damage as indicated by loss in CM in a limited frequency region. The ear canal measurements showed an increase in SPL in the same frequency region. The damage to the cochlea produces an appreciable change (up to 10 dB) in the ear canal SPL indicating that the input impedance of the inner and the middle ear depends on the physiological condition of the cochlea. [Supported by NIH Grant 2R01 NS 03654.]
Impact noise: An evaluation of the equal energy hypothesis
Authors: Roberto M. , Hamernik R. P. , Salvi R. , Henderson D.
Published: October 2005

Four groups of chinchillas were exposed to impact noise. All groups received the same total energy of exposure; however, the intensity and total duration of the exposure was varied. The impact was 200 ms in duration and presented at the rate of 4 per sec at intensities of 107, 113, 119, and 125 dB peak sound pressure level. The exposure durations varied from 5 days to 1.9 h. Evoked response thresholds were obtained before and after exposure at test frequencies between 500 and 16 000 Hz. Thirty days after exposure final audiograms were obtained and the animals were sacrificed for surface preparation histology. There was a systematic increase in the maximum amount of TTS as a function of impact noise level particularly at the lowest test frequency. The three lowest levels produced relatively little PTS while the highest level produced as much a 40 dB PTS at 2 kHz. Average inner and outer hair losses were calculated over octave intervals along the cochlea for each group of animals. In general, there was an increase in the amount of hair cell loss as a function of level, particularly in the 4-kHz octave band region of the cochlea. On the basis of the limited range of parameters studied, both the threshold shift and hair cell loss fail to support the equal energy hypothesis. [Work supported by NIOSH 1-RO1-OH-00364.]
The nonscattering of sound by sound
Authors: Westervelt Peter J.
Published: October 2005

A recent claim [D. H. Trivett and P. H. Rogers, J. Acoust. Soc. Am. 71, 1114–1117 (1982)] that a cw plane wave is scattered by a pulse is subject to the same criticism [P. J. Westervelt, J. Acoust. Soc. Am. 29, 199–203, 934–935 (1957) and M. L. Lighthill, Math. Revs. 19, 915–916 (1958)] leveled at an earlier claim [U. Ingard and D. C. Pridmore-Brown, J. Acoust. Soc. Am. 28, 367–369 (1956)] that scattering occurs from the interaction region of two cw sound beams. The pulse wavefunction employed by Trivett and Rogers does not satisfy the homogeneous wave equation, a fact which gives rise to spurious sources responsible for their nonzero result. The correct result is approximated by Esq. (2t) or Eq. (22) [P. J. Westervelt, J. Acoust. Soc. Am. 59, 760–764 (1976)] multiplied by the longitudinal and transverse taper factors of the pulse. At θ = 0 when the pulse and the cw wave are travelling in the same direction, this solution has an infinite directivity index indicating that the scattered wave never levels the interaction region; this situation, unachievable in practice, should not be called a parametric array and contradicts recent results [D. H. Trivett and P. H. Rogers, J. Acoust. Soc. Am. Suppl. 1 72, S40 (1982)]. About thirty years ago I repeated Ingard and Pridmore-Brown's experiment using their apparatus and convinced myself that nonlinearities in the electrostatic transducer were responsible for their nonzero results.
Acoustic radiation stress in anisotropic solids
Authors: Cantrell John H.
Published: October 2005

The radiation stress associated with acoustic waves propagating in solids of arbitrary crystalline symmetry is a tensor of second rank for which each of nine components is generally nonzero. Expressions for the radiation stress tensor are derived in terms of generalized nonlinearity parameters of the solid and the energy density of the acoustic wave. For waves propagating in isotropic solids and in the pure mode propagation directions of single crystals the tensor reduces to diagonal form in the appropriately transformed frame. The generalized expressions hold for shear wave as well as for compressional wave propagation in anisotropic solids.
Self demodulation of a pulse of finite amplitude: Does a double hump really exist?
Authors: Odero D. P.
Published: October 2005

During the last meeting of the ASA, some results were presented using a line and cone transducer excited by a pulse modulator [G. L. Sackman, R. L. Eyman, P. J. Lastrange, and A. H. Seeman, J. Acoust. Soc. Am. Suppl. 1, 72, S98 (1982)]. This transducer employs the theory of a parametric transient effect presented by O. R. Berk tay [J. Sound Vib. 2(4), 435–461 (1965) and confirmed by previous experiments (M. B. Moffett, P. J. Westervelt, and R. T. Beyer, J. Acoust. Soc. Am. 69, 339–363 (1971)]. It was found that the “double hump” was in the secondary beam cross section. This effect is in fact due to the nonlinearity specific to the hydrophone (LC10) used during this first experiment. Several different hydrophones, TR225, BK8103, F50D, do not show this “double hump” phenomenon at the secondary frequency, but a nearly Gaussian like cross section that follows closely the shape of the primary beam. On leave from SINTRA-ALCATEL, Dept. DSM. Z. I., Les Paluds, Aubagne, 13400 France.
Tuning the longitudinal mode of piano strings
Authors: Conklin Harold A.
Published: October 2005

The primary motion of piano strings is transverse to their length, but longitudinal string vibration also is present when a piano is played. This has been found to influence significantly the perceived tone quality of the instrument. The frequency of the longitudinal mode depends upon string length, material, terminating conditions, and also, in the case of wrapped strings, upon the ratio of the mass of the wrapping to the mass of the core wire. The longitudinal mode is essentially independent of string tension. By designing the piano and its strings so as to place the frequency of the first longitudinal mode in certain preferred relations to the frequencies of the transverse modes, improved tone quality, better musical pitch definition, and increased evenness of tone from note to note have been obtained. Recorded examples will be presented.
Experiments on air modes within the violin and their effects on the sound of the instrument
Authors: Hutchins Carleen M.
Published: October 2005

A violin has been constructed with 16 holes which are 6 mm in diameter in its ribs. The free top and back plates are both tuned to have mode ♯5 at 360 Hz and mode ♯2 at 180 Hz. (See “Acoustics of Violin Plates,” Sci. Am. October, 1981.) Response curves are made both inside and outside of the instrument with the holes open as well as plugged with corks in various sections of the ribs. The implications of this work for further research will be discussed and the violin demonstrated.
Musical/acoustical design of the NX clarinet
Authors: Benade A. H.
Published: October 2005

A simple, failsafe mechanism separates the B4♭ tone hole function from that of a single register hole. This permits redesign of the clarinet's acoustics as follows. Accurate alignment (<5¢) of mode 1 and mode 2 resonances over the entire low register scale becomes possible, assuring good tone and response in the low register. The register hole is sized such that the “off-node” pulling up of pitches at both ends of the second register scale just offsets the effect of embouchure slackening needed to give best tone and response via reed resonance effects. Register hole dissipation is also reduced. A suitable taper angle plus a set of small vent holes lower down, let the conical bell function as an accurate impedance and radiation surrogate for an extended tone hole lattice. Dissipation via adjacent-hole, and hole entrance-exit interaction effects and turbulence are minimized by systematic relocation of holes at the upper end (zigzag around tube), by reproportioning for maximum tone hole length/diameter ratio on all holes (especially at the lower end of the bore), and by systematic rounding of all sharp edges on the tone hole and joints. The instrument is mechanically simple, full toned and docile, requiring little adaptation by the player. The good features of early 19th century clarinets (bell taper, bore and register hole relation, and low complex-flow interaction effects) had a major influence on the design of this instrument, whose adjustment was guided by formal playing-condition diagnostic techniques.
Theoretical analysis of timpani normal modes for arbitrary kettle shapes
Authors: Christian R. S. , Davis Robert E. , Tubis Arnold
Published: October 2005

The method of boundary integral equations is used to set up practical calculations of the normal modes of an air-locked timpani membrane for arbitrary kettle shape. Preliminary results are presented for the model frequencies and associated spatial distribution functions for kettle shapes in current use.
Digital simulation of audio devices: A research tool
Authors: Gibian G. L. , Leidecker H. W.
Published: October 2005

The growing interest in digital reproduction of audio information motivates further research regarding digital representation of various analog components for sound reproduction and signal processing. Hardware currently available “off the shelf” can be assembled to create a useful device for experiments involving digital signal processing: the device in its “null state” is an A-to-D followed by a D-to-A converter, accompanied by the necessary filters, amplifiers, and clocks. Using such a device whose input is perceptually indistinguishable from its output, digital simulations of physical systems can be inserted in the signal path to investigate our understanding of the systems and to predict the effects of modifications to the systems.
Loudness matching in the sound field of a room
Authors: Lindevald I. M. , Benade A. H.
Published: October 2005

Two loudspeakers are placed several meters apart in an acoustically good lecture room (V = 987 m3, Trev≃1 s). The signal is fed alternately to the speakers via a soft switch and individual 40 dB (0.1 dB least count) attenuators. The burst durations are each 1 s. The observer sets the level of speaker A, and the subject is instructed to adjust the level speaker B to a matching loudness. He is free to move around in the room while carrying out the match. Preliminary results indicate that about 65% of the time subjects match loudness to better than 2 dB. About 40% of the time subjects match loudness to better than 1 dB. Since there is a ± 5.6 dB rms Rayleigh-distributed spread in the point-to-point transmission of acoustical signals in a room, the listener's performance implies accumulation of the equivalent of some half-dozen statistically independent samples of the sound field for each matching sequence. [Work supported by NSF.]
Perception of musical and nonmusical triads
Authors: Roberts L. A. , Mathews M. V.
Published: October 2005

Fundamental frequencies of just major triads are in ratios of successive integers 4 to 5 to 6 which produces many coincidences of harmonics. This triad has an unambiguous “fundamental bass” (Rameau and Terhardt. We examined judgments of major (4:5:6) and minor (10:11:12) triads as well as nonmusical triads using ratios of nonsuccessive integers (3:5:7 and 5:7:9) which likewise have coincidences of overtones and unambiguous fundamental bass notes. By using nonmusical triads, we can observe intonation sensitivities apart from biases due to musical experience. We varied the middle note of each triad by −30, −15, +15, and +30 cents thus producing beating harmonics and ambiguous fundamental bass notes. Subjects were presented with pairs of triads and selected (1) the triad that is more “in tune,” (2) the triad they prefer, and (3) the triad that is more smooth. Listeners divided into two distinct groups: one showed an inverted V function, in which ratings are highest for the “just” triads; the other showed an inverted W function in which the triads deviating from just intonation by 15 cents are given the highest ratings.
The use of the phase vocoder in the analysis of ensemble sounds
Authors: Dolson Mark B.
Published: October 2005

An important step in the synthesis of rich and interesting artificial timbres has been the analysis of natural instrument timbres. In this research, we analyze ensemble timbres (e.g., a violin section playing in unison) with a similar motivation. As in many studies of solo timbre, we present tones via an additive model in which the time-varying amplitudes and frequencies of individual partials are estimated by means of the phase vocoder. However, we show that changes in the actual amplitude or frequency of a given partial can significantly distort the phase vocoder estimates. Furthermore, we show that this distortion can be eliminated by the use of a tracking phase vocoder which automatically adjusts the analysis frequency to match the instantaneous frequency of the partial being analyzed. With the tracking phase vocoder, we analyze actual violin ensembles and show how they can be simulated by applying appropriate amplitude modulation to the individual partials of a solo tone. Lastly, we show how errors can arise in using the phase vocoder to scale the pitch or timing of the ensemble sounds.
A speech signal preprocessor with wide dynamic range and weak signal enhancement properties
Authors: Lin Wen C. , Luo Tzieh-Tsai
Published: October 2005

This paper describes the design, analysis, and experimental evaluation of a speech signal preprocessor. The design utilizes analog and digital hardware as well as software techniques to achieve a wide dynamic range of operation and to enhance the signal resolution for the weak signal. For saving the memory storage area, the recognition logic for ONSET/OFFSET of speech signal is implemented with analog and digital hardware. Two features, namely energy and zero crossing count, are used in the decision making process. The hardware contains the following major elements: a 755N 6-decade log-amplifier by Analog-Device Corp., anti-aliasing filter, a precision full-wave rectifier, analog switches, and a 12-bit Analog-to-Digital converter. The anti-logarithmic operation is implemented with a software subroutine. Its result can then be used as natural data for further processing. The design problems and solutions, and the mathematical analysis for weak signal enhancement are discussed and experimental evaluation results are described. The system has been used to acquire phonetic data; it yields very good results.
A tool for studying the fine time frequency structure of speech
Authors: Lienard J. S.
Published: October 2005

Analyzing speech with a very short time window (a few milliseconds) could help detect and measure some salient features, apparent with the usual methods. The initial difficulty in this type of analysis is finding precisely those instants where the spectrum is to be measured, and the way to measure it. In order to study these problems, a program has been written, which allows the user to process a 76 ms segment of speech by a 13 linear phase bandpass filterbank, and to post process it in various ways: half- or full-wave rectification, linearization of the envelope between adjacent peaks, low-pass and bandpass filtering, nonlinear smoothing taking into account some time masking effects, and calculation of the coherence function that reflects a simultaneous raise of energy in several spectral bands [cf. Lienard and Manceron, J. Acoust. Soc. Am. Suppl. 1 69, S17 (1981)]. The goal is to define the speech signal as a string of short time spectra (“pulses” or “events”) corresponding to the glottal pulses, explosions, and flaps of some consonants, and instantaneous peaks of noise. Examples will be presented, and the implications of such an analysis in the interrelations of spectrum, pitch, and voicing will be discussed.
On the acoustical characterization of vowels
Authors: Eskenazi Maxine , Lienard Jean-Sylvain
Published: October 2005

Since the work OF Ladefoged and Broadbent [J. Acoust. Soc. Am. 29, 98–104 (1957)], it is generally accepted that the identification of a specific vowel is linked to the relation of all of the vowels in a language. The type of acoustical characterization chosen for the vowels of a given language, and the result obtained in its use, depend in part on the quantity of phonetically different vowels in that language. A substantial database has been constituted, including several tokens for each of 13 French vowels pronounced by 30 male and female speakers, in order to study the acoustical features of vowels. Results of antomatic recognition tests based on a statistical characterization of smoothed, amplitude-independent spectra [Eskenazi and Lienard, GALF-AFCET Seminar, Toulouse, Sept. 81, pp. 54–69] are compared to the results of perceptual recognition experiments on the same data, and discussed in light of some other vowel identification experiments [Komatsu et al., IEEE-ICASSP Paris, May 82; Cole et al., Artificial Intelligence Conf., Pittsburgh, Aug. 82; Caelen and Caelen, GALF-AFCET Seminar, Toulouse, Sept. 81, pp. 129–143].
Real-time extraction and display of voice fundamental frequency of electro-glottographic signals
Authors: Hasegawa Akira , Herbert Eugene , Mahshie James , Brandt Fred , Mars Marguerite
Published: October 2005

Dedicated front-end hardware was constructed to enable real-time extraction of voice fundamental frequency (f0) from electroglottographic signals. The cycle-by-cycle voice f0 period was measured by a digital circuit with clock frequencies of 1 MHz and 100 kHz. The temporal f0 information was transferred to PDP 11/34A digital computer via a 16-bit parallel interface, and converted into frequency by table lookup. The f0 patterns are displayed on a CRT screen via a graphics video interface with on-board memory for 512×512 pixels. Three types of displays, (1) f0 level, (2) f0 histogram, (3) f0 contour across time, are currently used for monitoring and modification of the voice f0 of the hearing-impaired individuals. Detailed performance characteristics of the f0 extractor and the results of its use will be described together with the f0 extraction and display schemes. [Work is supported by the National Institute of Handicapped Research and the Gallaudet Research Institute.]
Fundamental frequency declination over changes in length
Authors: Gelfer Carole E. , Harris Katherine S. , Collier Rene , Baer Thomas
Published: October 2005

The study of F0 declination affords the opportunity to study speech organization over large linguistic and temporal units. Of particular interest is how changes in utterance conditions are reflected as variations in the acoustic realization of F0. We have examined fundamental frequency, subglottal pressure and laryngeal muscle activity during utterances that varied systematically in overall length. Specifically we find that, while initial F0 values increase as a function of length, the rate of decline in F0 remains essentially invariant. That is, a speaker may prepare for and/or signal anticipated utterance length by increasing initial F0, but overall declination does not appear to be reorganized on the basis of this initial adjustment. We find too, that both tricothydroid and Ps activity correlate with the magnitude of the initial F0 changes and that the overall decline in Ps mirrors the F0 declination. Our impression is that a speaker has some expectations regarding changes in fundamental frequency over time, and that these changes are anticipated by making appropriate adjustments in initial conditions. These adjustments may, in fact, represent the extent to which declination is under speaker control. [Work supported by NINCDS and BRS grants.]
Fundamental frequency perturbation caused by voiced and voiceless stops in Japanese
Authors: Kawasaki Haruko
Published: October 2005

The study investigates the effect of voicing of stops on the fundamental frequency (F0) of the following vowels in Japanese, particularly, how this effect interacts with phonemic accentual patterns. A speaker of Standard Japanese recorded/Can/syllables where C was/p,t,k,b,d,g,m,n/. F0 was measured for each glottal period after the consonantal release, or after the vowel onset in the case of aspirated stops. The results showed a dip in F0 around 40 ms after the release of voiced stops, while F0 generally stayed higher after voiceless stops. To examine the interaction between consonantally-induced F0 variations and accentual patterns, the F0 contours in C1VC2V words with /t,d,n/ as C1 with either high-low or low-high accent were analyzed. In the high-low accent, the low initial F0 after a voiced stop caused F0 to peak later in time than after a voiceless stop. This pattern was not observed in the low-high accent, where the initial dip was followed by a fast increase in F0 after a voiced stop, while the pitch rise was more gradual after a voiceless stop.
Sound absorption coefficient measurement using time delay spectrometry
Authors: Bareham John , Gilbert John
Published: October 2005

The sound absorption coefficients (α) of acoustical materials can be obtained for normal and random incidence using classical methods. Using the relatively new technique of Time Delay Spectrometry (TDS)α can be measured for any arbitrary angle of incidence. Furthermore this method can be conducted in-situ. The TDS method is described and typical results are presented. The results of measurements obtained by this method are discussed in relation to those obtained by classical methods.
Closed-cell plastic foams for sound absorption
Authors: Kurtze Guenther
Published: October 2005

Sound absorbing materials according to present standards are porous, open-cell structures acting by their flow resistance. Their prototype is a thin foil with a flow resistance equal to ρc, the product of density and speed of sound in air, mounted at a distance of 1/4 wavelength in front of a rigid wall, having 100% absorption for normally incident sound waves. Sound can also enter into closed-cell foams like PE foam, provided their ρc is sufficiently low. Absorption inside these materials, however, cannot be caused by friction—since there is no air flow—but only by deformation of the plastic structure due to alternating pressure. Experimental results thus obtained are generally disappointing. High internal losses in plastics require shear deformation rather than compression. Special structures of absorbers are described in which the wanted shear deformation is obtained even for normally incident sound. Their absorption is compared to classical sound absorbers. Applications for engine noise reduction in the automotive industry are shown where closed-cell structures are of special importance.
Fractional derivatives and complex modulus with a new algorithm for order mechanical properties
Authors: Rogers Lynn
Published: October 2005

The Bode diagram technique, highly developed in feedback control system analysis, is used to synthesize operators for the stress-strain constitutive equation. The operators include fractional derivatives. A generalized “fractional” Maxwell model is developed where the usual dashpots are replaced with hybrid elements whose force is proportional to the fractional derivative of displacement. The associated complex modulus curves accurately and efficiently represent experimental data. The new algorithm is a scheme to determine parameter values for representation of the complex modulus by a ratio of factored polynomials in the square root of minus one times reduced frequency. A large number of factors may be used to permit a very close approximation. This representation is associated with a Maxwell and a Kelvin model which are used to obtain stress relaxation and creep compliance and also the discrete relaxation and retardation spectra.
Frequency-dependent dynamic properties of rubber materials with a random distribution of voids
Authors: Varadan V. K. , Varadan V. V. , Ma Y.
Published: October 2005

A self-consistent multiple scattering approach using the T-matrix method with suitable pair-correlation function has been employed to study the frequency-dependent properties of rubber materials containing a random distribution of voids. Numerical results indicate that for a certain range of concentration of voids, the resonance of the voids controls the frequency-dependent effective properties such that the imaginary part of the effective wavenumber becomes greater than the real part. The so-called “superviscous” propagation in which the propagating wave may be damped out over a distance smaller than a wavelength, seems to occur over a bandwidth controlled by the concentration of voids (c). The bandwidth increasing with concentrations in the range 0 < c < 0.10. For c > 0.10, this kind of behavior does not exist for the material system considered.
Active acoustic absorption—Where does the energy go?
Authors: Angevine O. L.
Published: October 2005

Active attenuation of noise in three dimensions around a complex source is an accomplished fact [M. J. M. Jessel and O. L. Angevine, in Inter-Noise 80 Proceedings, pp. 689–694]. But this author continues to receive questions regarding the theory such as “Where does the energy go?” When cancellation sources are added, the existence of a primary sound field changes the acoustic impedance seen by these cancellation sources, as pointed out by Levine [R. C. Levine, Am. J. Phys. 48, 22–31 (1980)]. The use of cancellation sources having a cardioid radiation pattern makes it possible to avoid influencing the acoustic impedance seen by the primary source, however, thus the secondary sources become acoustic absorbers. [M. J. M. Jessel, Acoust. Lett. 4(9) (1981)]. Further consequences of this approach are discussed.
A model of the combined effects of forward and simultaneous maskers
Authors: Jesteadt Walt
Published: October 2005

In previous papers, we have discussed a series of experiments concerning combinations of sinusoidal forward maskers and broadband-noise simultaneous maskers. These experiments have demonstrated that (1) the combined effects of forward and simultaneous maskers are greater than would be predicted on the basis of a linear sum of their individual effects, (2) the excess masking is greatest when the individual muskets are equated, (3) excess masking increases as the amounts of masking produced by the individual maskers increase, (4) for a given level of either masker, excess masking occurs over a broad range of levels of the other masker, and (5) there is less excess masking when the forward masker is below the signal in frequency. We have found that the compressive model described by Penner [J. Acoust. Soc. Am. 67, 608–617 {1980)] provides an excellent account of the first four effects. Our data yield estimates of the two key parameters in the model that are very close to those reported by Penner and Shiffrin [J. Acoust. Soc. Am. 67, 617–627 {1980)]. The model is less successful, however, in accounting for data obtained with forward maskers that differ from the signal in frequency. [Work supported by NIH and NSF.]