bluuebunny/crossref_metadata_2025 huggingface dataset:
Single auditory nerve fiber activity and behavioral thresholds in pathological cochleas
Authors: Salvi R. , Hamernik R. P. , Henderson D. , Giraudi D. , Parkins C.
Published: October 2005

Four chinchillas were exposed to 4 kHz CF octave band of noise at 86 dB for five days. After a recovery period of approximately six months, behavioral audiograms were obtained and auditory nerve fiber activity was recorded. The animals were sacrificed and the cochleas embedded in Araldite resin to obtain surface preparations and 1μ sections of the Organ of Corti. Permanent threshold shifts ranged from 5 to 20 dB between 4 and 8 kHz. VIII nerve recordings showed single fiber thresholds were elevated up to 70 dB between 4 and 14 kHz, while higher and lower CF fibers were essentially normal. Spontaneous activity in some of the animals was equal to or greater than normal. Cochleagrams all showed narrow lesions of inner and/or outer hair cells over approximately a 1-mm distance. A comparison of the three realms of data revealed the following: (1) The greatest loss in sensitivity from the noise exposure was seen in the single nerve fiber thresholds while the smallest change was seen in the behavioral threshold shifts; (2) Based on the frequency-place maps of the chinchilla cochlea, the range of fibers with elevated thresholds exceeded the extent of the outer hair cell lesions. A number of anatomical changes were seen that effectively increased the extent of the damage found in the cochleagram. These changes include distortions in the surface topography of the Organ of Corti affecting the orientation of IHC; missing pillar cells in the presence of normal OHC and the orientation of IHC; missing pillar cells in the presence of normal OHC and IHC populations and protrusion of the IHC cuticular plate into the subtectorial space. [Work supported by NSF and NIOSH.]
Auditory-nerve correlates of acoustic trauma
Authors: Ruggero M. A.
Published: October 2005

Single-unit recordings were made in the acoustic nerves of normal chinchillas and of chinchillas exposed one to two months previously to high intensity sounds. From 50 to 500 tuning curves were obtained in each animal. A systematic inverse relationship exists between threshold at best frequency (BF) and level of spontaneous activity. In normal chinchillas thresholds for high-spontaneous activity neurons, when plotted against BF, lie within a narrow (<10 dB) locus that approximately parallels the normal behavioral audiogram. In contrast, animals exposed for 2 h to 105-120 dB SPL warbled tones yield threshold-BF plots of irregular contours, with elevated thresholds in one or more frequency regions. Abnormal tuning curves have shortened tips and sometimes hypersensitive tails. Surface preparations of all relevant cochleas were examined to ascertain the presence or absence of every hair cell. Extensive outer-hair cell loss is always correlated with gross threshold elevation or total absence of responsive neurons within one or more ranges of BF. On the other hand, severe physiological abnormalities are often accompanied by only minimal hair cell loss. [Work supported by NIH (NS12125), Deafness Research Foundation and University of Minnesota Graduate School.]
Speech waveform coding: Techniques and performance
Authors: Crochiere Ronald E. , Kato Yasuo
Published: October 2005

With the evolution of large-scale integration of digital logic, a renewed interest has developed in methods of digital encoding and transmission of speech signals. Coupled with these advances, a number of new concepts and directions have emerged in the area of speech waveform coding, leading to more efficient ways of representing speech digitally. Also, a greater need has developed for defining and understanding issues of speech coder quality. This session, therefore, will focus on these topics and will present a broad coverage of new concepts in the design and implementation of speech waveform coders. In the second part of the session attention will be directed at issues of measurement and assessment of coder quality.
Frequency domain techniques for speech coding
Authors: Crochiere R. E. , Tribolet J. M.
Published: October 2005

Frequency domain techniques for speech coding have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a filter bank (sub-band coding) or by a suitable transform (transform coding) and then encoding them using adaptive PCM. Three basic factors are involved in the design of these coders: (1) the type of transform or filter bank, (2) the choice of bit allocation used by the quantizers, and (3) the control of the step-size of the quantizers. This paper reviews basic aspects of the design of these three factors, particularly as they apply to low bit-rate adaptive transform coding, and presents results of recent improvements on these techniques. Principles of short-time spectral analysis are used to explain basic properties and issues in the design of the transforms or filter banks. Principles of spectral estimation and models of speech production and perception are used to illustrate how the side information in the coder can be most efficiently represented and utilized in the design of the coder to control the dynamic bit allocation and quantizer step sizes. Examples of designs and performance of adaptive transform coders will be presented for bit rates in the range of 8–16 kb/s.
Propagation of sonic booms in the thermosphere
Authors: Gardner John , Rogers Peter H.
Published: October 2005

A nonlinear theory for the long-range propagation of sonic booms through the thermosphere has been developed. A realistic atmosphere is employed, and consideration is given to such factors as nonlinear stretching and decay of the wave, the effects of the caustic, the winds, the linear acoustic attenuation, and the increase in Mach number due to the decreasing density at high altitudes. Detailed results are presented for the case of the Concorde SST in straight, level, and steady flight at 17.5 km and a velocity of Mach 2. We predict maximum ground level pressures of 0.3 Pa with an N-wave “period” of about 10 s. The sound level is a minimum along the flight track with the maximum signal strength occurring about 300 km off the flight track. The strongest received signal travels initially downward and reflects off the surface of the ocean to thermosphere. The wave turns around at an altitude of 160 km and is returned back to the ground at a horizontal distance of 320 km from the launch point. The acoustic Mach number of the wave never exceeds 0.2. Ninety percent of the wave's energy is attenuated below 100 km, with 99% attenuated by the time the wave reaches the turning point.
Evaluation of two ADPCM codecs for toll-quality speech transmission
Authors: Mermelstein Paul
Published: October 2005

Speech transmission at 32 kbps by ADPCM coding offers a potential saving of a factor of 2 in bits transmitted per conversation. To evaluate the extent to which such codecs introduce perceptible degradation in the speech signal, two systems, one employing a third-order fixed predictor, the second a third order adaptive predictor were simulated. Evaluation through relative preference tests shows a slight degradation in quality compared to Mu-255 8 bit log PCM companding under high loading conditions (>20 dBm0) and a slight improvement in quality under low loading conditions. Comparison of our subjective preference results with a modified segmental measure of signal-to-noise ratio agrees with the findings of McDermott et al. [Bell Syst. Tech. J. 57, 1597–1618 (1978)] for simpler codecs, namely that the segmental SNR is a significantly better predictor of the subjective quality of coded speech than the SNR measured over an entire utterance.
Calculation of impedance for a refracting bottom
Authors: Stickler D. C.
Published: October 2005

It has been shown that the field reflected from both a nonrefracting half space and from a refracting layer depends not only on the incidence angle but also on the frequency and height of the source above the interface. This suggests that measurements be designed to exploit rather than ignore these effects. The measurement of the impedance—i.e., the ratio of the pressure to the normal component of the acoustic velocity—at the ocean-bottom interface can exploit these effects and does not require detailed information about the source spectrum or the transfer functions of the sensors. The characteristics of the impedance function for a point harmonic source, as a function of incidence angle and frequency are described for both a nonrefracting bottom and refracting layer. A discussion is given of the techniques needed to make these calculations. For the nonrefracting case there is a characteristic peak in the impedance associated with the critical angle. For the refracting layer, there is a characteristic peak associated with a caustic in the refracting layer. The behavior of these two characteristics is contrasted. [Work supported by ONR.]
Velocity gradients in the oceanic crust
Authors: Wipperman L. K.
Published: October 2005

Most marine crustal refraction lines have been interpreted in terms of a few relatively thick, constant velocity layers by using the well-known slope-intercept method. Recently, several researchers have used both arrival amplitudes and travel times to constrain candidate velocity-depth functions. They discovered that the combined amplitude and travel time distribution required crustal velocity-depth functions consisting of a series of velocity gradients and, occasionally, low-velocity zones. The presence of velocity gradients in the ocean crust has important consequences for the propagation of low-frequency sound in the ocean and for the geology of the oceanic crust. In this paper travel time inversion methods for determining crustal velocity gradients are discussed and illustrated using air gunsonobuoy refraction data. Careful picking and inversion of the travel time branches actually observed in the refraction data yield velocity-depth functions similar to those from a combined amplitude and travel time analysis. [Work supported by ONR.]
Amplitudes in seismic profiling data: Multilayer interference and attentuation effects at 4 and 6 kHz
Authors: Tyce Robert C. , Mayer Larry , Spiess F. N.
Published: October 2005

Utilizing the Deep Tow instrument of the Marine Physical Labortory at Scripps Institution of Oceanography, quantitative profiling data have been collected for a number of widely separated areas of the sea floor. Data have been collected at both 4 and 6 kHz, along with transponder navigated cores for comparison of core physical properties with acoustic data. Attentuation values have been calculated from the acoustic data which show good agreement with values measured directly. However amplitudes observed for buried reflectors are often too great to be accounted for by expected impedance changes, when attentuation is considered. Thus constructive interference is suggested. This conclusion is supported by 4 and 6 kHz data from a common area which show different reflectors. Detailed physical property analyses have been performed on core samples from a survey area in the equatorial Pacific and used to calculate impedance profiles. Convolution of the 1 ms profiler wave form with these impedance profiles gives reflection profiles which are in excellent agreement with observed acoustic profiles from the area, but which are poor representations of the actual structure.
Digital equalizer of a dummy head system using a direct approximation technique of log magnitude response
Authors: Hamada H. , Ogura Y. , Miura T.
Published: October 2005

When we intend the whole-space stereophnic system, we must design this system to be simulated for original listening conditions. For this purpose, we need several equalizing networks [M. R. Schroeder and D. Gottlob, J. Acoust. Soc. Am. 56, 1195–1201 (1974)]. Since these equalizers have rather complicated responses, we constructed them by digital systems. The digital equalizers described here have a form of digital filter which presents the best mean square approximation. To determine the coefficients of these filters, a direct approximation technique of log magnitude response for digital filter, proposed by I. Imai [IEEE Trans. Acoust., Speech Signal Process. 25, 127–133 (1977)], was utilized. The resulting digital equalizers are recursive type and realized in a cascade connection of very simple second-order-type recursive digital filters (number of multipliers is two in this second-order type filter). These equalizers have several important advantages, i.e., fewer multipliers are needed in these constructions, since we used the multipliers in time sharing fashion. And these equalizers result in easier control of the characteristics of the system.
A program to reduce permanent hearing loss caused by otitis media in Eskimo children
Authors: Gunn W. J.
Published: October 2005

From 25% to 40% of Eskimo children suffer from Otitis Media at any given time. The complications of this condition include permanent hearing loss as well as many other serious problems. The Alaskan State Health Department and the Center for Disease Control are engaged in a joint effort to determine the most effective ways to combat this problem through health education and screening. The program design, acoustical equipment problems encountered, and preliminary results will be discussed.
Selective perceptual dysacusis for pitch—the “Tin-Ear” phenomenon
Authors: Chaney R. B. , Hanson T.
Published: October 2005

Following reports by clinical patients that they could not distinguish between large changes in frequency (two to three octaves in some cases), and further, that amplification devices with high-frequency emphasis actually depressed intelligibility scores in these patients, an attempt was made to quantify this unusual form of dysacusis. Stimulus pairs representing frequency intervals from 12 to 4 octaves were presented for same-different judgments by a patient with a moderately severe sensorineural high-frequency hearing loss. Results indicated essentially no frequency discrimination above 1 kHz in the right ear and above 3 kHz in the left. Selective frequency amplification depressed intelligibility scores as the response curve “invaded” the involved frequencies in that ear. Implications for auditory theory and selective amplification are discussed.
Ear dominance and the pitch of the dichotic intertone
Authors: Gregory A. , Efron R. , Yund E. W.
Published: October 2005

The intertone—the pitch perceived when two tones of slightly different frequency are presented, one to each ear via earphones—was matched to a variable frequency binaural tone following Odenthal's method [Acta Physiol. Pharmacol. Neerl. 12, 453–496 (1963)]. Subjects whose ear dominance for pitch had been previously determined by other methods [E. W. Yund and R. Efron, J. Acoust. Soc. Am. 59, 889–898 (1976)] matched the intertone with a frequency which was appreciably closer to the component of the dichotic chord presented to the dominant ear. The deviation of the pitch of the intertone from the center frequency was greater, the stronger the subject's ear dominance. The effects of various parameters (SPL, Δf, center frequency, and ΔI) on the pitch of the intertone are interpreted in terms of the shape of the resulting binaural excitation pattern.
Dichotic pitch pattern perception by musicians and nonmusicians across three response modes
Authors: DeFosse C. O. , Pinheiro M. L.
Published: October 2005

Musicians and nonmusicians were tested with dichotic pitch pattern triads which consisted of three tone bursts of 200 ms each with interburst intervals of 150 ms. Frequencies of the tone bursts were 630 and 1065 Hz for one set of pitch patterns and 1500 and 1935 Hz for the other set. There was an intrastimulus interval of 10 s. Subjects responded using three different response modes: (1) manual response by pressing buttons on two response boxes, (2) hummed response, and (3) verbal-linguistic response by labeling each tone in the pattern as “high” or “low.” Musicians performed significantly better than nonmusicians on all three response mode conditions. Right and left ear performances did not shift significantly in direction of ear advantage across the three response modes for either experimental group. Percentage of correct identifications did not vary significantly across response modes. Neither musicians nor nonmusicians demonstrated statistically significant ear advantages. Stimulus set bias was present in both experimental groups in favor of the higher pitched pattern set (tone bursts of 1500 and 1935 Hz). [Work supported by NIH.]
Ear dominance for dichotically presented complex and pure tones
Authors: Robinette L. N. , Nábělek I. V.
Published: October 2005

Ear dominance for nine subjects was determined for dichotically presented amplitude modulated (AM) sinusoids as a function of modulated and modulation frequencies. Ear dominance for five of these subjects was also determined for pure tone pairs centered around one of the modulated frequencies and around one of the modulation frequencies, so that ear dominance for pure tone pairs could be compared to that for AM signals in the frequency region of AM signal energy and in the frequency region of periodicity pitch of the AM signals. All signals were presented in a 2AFC paradigm such that competing tonal sequences were heard at each ear. Subjects reported the pitch sequence heard for each trial, and estimates of ear dominance were then inferred from the data. Ear dominance was observed for one-third of the subjects tested with AM signals, and it decreased as the center frequency of the test signals was increased. No definite effect of modulation frequency (component frequency spacing) was observed. All subjects tested with pure tone signals exhibited ear dominance to some degree. In general, ear dominance for AM signals was more closely related to the one pure tones centered around the modulation frequency than to that for pure tones centered around the modulated frequency.
Listeners' successful search for melodies scrambled into several octaves
Authors: Dowling W. J.
Published: October 2005

Distorting familiar melodies by randomly assigning notes to different octaves makes recognition difficult. This difficulty can be lessened by preserving the contour (ups and downs) of the original melody. These experiments (using complex tones) show that cuing with either the unscrambled melody itself or its title also improves performance. In experiment 1 listeners heard a “straight” version of a familiar or an unfamiliar melody, followed by a scrambled version of either the same melody or a lure. Hit rates for scrambled melodies averaged 0.80, while false alarms to lures averaged 0.40. In experiment 2, scrambled familiar melodies or lures were preceded by accurate or spurious titles, respectively. Hit rates were 0.80 and false-alarms 0.20. These results show that listeners can use a memorized, pitch-and-time-flexible, “melody template” in recognizing scrambled melodies, just as they can in recognizing an unscrambled melody temporally interleaved with other tones in the same pitch range. The template is a list of relative tone “chromas,” which can be used in the recognition of scrambled melodies just as in the transposition of the familiar melodic intervals of the NBC chimes.
Hemispheric differences for verbal and nonverbal stimuli in Japanese and English speaking subjects assessed by Tsunoda's method
Authors: Uyehara J. M. , Cooper W. A.
Published: October 2005

This investigation compared the performances of Japanese and English speaking subjects on Tsunoda's DAF keytapping task, which he purports assesses cerebral dominance for language [T. Tsunoda, “Functional differences between right and left cerebral hemispheres detected by the keytapping method,” Brain Lang. 2, 152–170 (1975)]. Twenty-three native speakers of Japanese and twenty-three English speaking subjects were employed in this study. Each subject was administered the DAF keytapping task utilizing the vowel /a/ and a 1000-Hz pure tone. The results revealed an effect of intensity, whereby an increase in DAF/SAF ratios resulted in an increase in performance disruption. No differences were observed between the two language groups, ears, or between the 1000 Hz pure tone and the vowel /a/. Only 15.5% of the subjects showed the hypothesized right ear advantage for the vowel and a left ear advantage for the tone when subjected to an analysis similar to Tsunoda's. All other subjects revealed variable degrees of advantage to either ear, to the same ear, or to neither ear. It was concluded that no significant functional auditory asymmetry was observed for either group, suggesting that the DAF keytapping task may not be a sensitive procedure to determine cerebral dominance for language among normals. [Work supported by Purdue Research Foundation.]
A comparison of hearing thresholds for pure tones and narrow speech bands
Authors: Franklin B.
Published: October 2005

The phrase “ah ah clap clap clap clap,” spoken by a female, was filtered into five narrow speech bands with a high-frequency and low-frequency cutoff for each band—250 Hz (125–375 Hz), 500 Hz (250–750 Hz), 1000 Hz (750–1250 Hz), 2000 Hz (1500–2500 Hz), and 4000 Hz (3000–5000 Hz). The frequency range of the bands was varied in order to maintain equal energy levels for each of the five filtered speech bands. Thresholds for pure tones (250, 500, 1000, 2000, and 4000 Hz) and the speech bands were obtained on easy-to-test hearing-impaired teenagers. Their pure-tone and speechband audiograms were remarkably similar at each test frequency. Speech-band audiograms were subsequently obtained for many hard-to-test, low-functioning children and adults for whom audiograms could not be obtained using conventional pure-tone audiometric procedures. The effectiveness of using speech bands to determine hearing thresholds may be that speech is a more meaningful auditory stimulus than pure tones for this population.
Tempo discrimination of formant-transitions by hearing-impaired listeners
Authors: Revoile S. , Holden L. D. , Pickett J. M.
Published: October 2005

In an earlier study on hearing for formant transitions, we found differences in the discrimination of transition tempo (rate of frequency change) as a function of the frequency-size of transitions which were rising to a steady state. In the present study tempo discrimination was measured for transitions that differed in temporal position and in frequency direction relative to a steady-state phase. Nineteen hearing-impaired and five normal-hearing listeners discriminated transition tempo in a 3IFC response paradigm. The transitions were initial or final, and were rising or falling in frequency, with reference to the steady-state frequency. For transitions covering the same frequency range (e.g., transitions of 100 Hz to or from a 1000-Hz steady state), tempo discrimination was better for initial transitions than for final. Better transitions of different frequency-directions, but sharing the same position relative to the steady-state (e.g., final transitions downward in frequency from 1000 to 900 Hz versus upward in frequency from 1000 to 1100 Hz) tempo discrimination was better for the transitions downward in frequency. Hearing-impaired listeners showed poorer mean transition-tempo discrimination than normal-hearing listeners; marked individual differences occurred in both groups.
Sound insulation of domestic roofing systems—a laboratory study
Authors: Cook K. R.
Published: October 2005

The sound insulation of common Australian domestic roofing systems was investigated using a 103 m3 room below a 203 m3 transmission room. Both rooms of non-rectangular shape satisfied recommendations of ISO/R 140. Measurements were conducted in the frequency range 100 Hz to 10 kHz. Initially, measurements of transmission loss (TL) were made using a basic ceiling with 9 mm plasterboard cladding, followed by tests when various commercially available in this were added. The enhancement of TL by infill was negligible at low frequencies. Various types of roof component, such as concrete tiles, asbestos-cement, and galvanized steel were then studied. Due to the great number of air gaps, the TLs varied little with frequency and were low in magnitude despite large differences in cladding surface density. Finally, various combinations of roof and ceiling samples were investigated. Although inclusion of ceiling infills yielded significant increases in TL values, the enhancements at low frequencies leads to the conclusion that, for acoustic protection against traffic noise, the addition of infills is a rather fruitless exercise unless close attention is paid to proper sealing.
Acoustical design of contemporary television broadcast and recording studios
Authors: Campanella A. J.
Published: October 2005

A series of television broadcast and video recording studios were designed with acceptable acoustical performance and reasonable cost. Special design measures were applied to control studio construction costs within the constraints of video taping and broadcast requirements. Experiences in the design and construction of two such installations including four studios are related. Preliminary results indicate that room-to-room isolation is the most critical design criterion while immunity to external noises—though important—remains a secondary consideration. Reverberation time requirements are modest because of the increasing use of close-microphone techniques. Novel use of monitor loudspeakers for audience participation in the same studio volume is common. Air handling systems can be adequately quiet when carefully designed.
Nonlinear interactions between acoustic fields and rotational flows at boundaries
Authors: Clark J. A.
Published: October 2005

A new amplitude-dependent effect has been detected in the near field of sound waves radiating into water. The waves, which were produced by electromagnetic sound sources, typically had the form of 50-μs pressure pulses with peak pressure levels that could be continuously varied over a range from 220 to 240 dB re 1 μPa. The waveforms were measured with piezoelectric transducers and also by optical holographic interferometry. Large distortions in the typical pulse shape were observed as the source strength was varied through an intermediate range of amplitudes. This acoustic effect appears to be induced by rotational flow occurring near the sound source boundaries. In many cases the rotational flow could be directly observed. Changes in the sound source design which altered the rotational flow characteristics were also found to strongly influence the distortion of the acoustic waves. In most instances the effect produced an enhancement in peak pressure levels and an increase in the directivity of the sound field. This interaction effect could have significant implications for the design of large-amplitude, low-frequency sound sources.
Strong interaction of vorticity and acoustic waves
Authors: Vaidya P. G. , Simcox C. D.
Published: October 2005

Equations of fluid mechanics have been used in the past to derive expressions for the vorticity, thermal, and acoustic waves. It has been generally assumed that these waves interact and produce source terms which are of second order and above. It is shown in this paper that there are several cases of practical interest where this assumption is incorrect. In particular, the interaction of sound and vorticity waves is considered. The equation for the vorticity wave shows the presence of terms which under certain conditions become secular. This leads to strong interaction of the vorticity and acoustic waves at the first order itself.
Percussion instruments of China and Indonesia
Authors: Rossing Thomas D. , Ross Richard
Published: October 2005

Percussion instruments play a very important role in Oriental music. The gamelan or orchestra of Java or Bali, for example, includes several types of metallophones, gongs, cymbals and drums. We have studied the modes of vibration, timbre, and tunings of a number of these instruments, and we compare them with percussion instruments of Western origin. Although they show considerable variation in tuning, Indonesian instruments tend to follow one of two tonal systems, known as pélog and sléndro. Included in the orchestra which accompanies Chinese opera are gongs, drums, bells and cymbals. Gongs are frequently constructed so that they sound one pitch when struck, then rise or fall about a semitone, large gongs (hsiao-lo) usually falling in pitch and small gongs (da-lo) rising in pitch.
Synthesis by linear prediction of the bamboo tone of Shakuhachi
Authors: Toyama F. M. , Takasugi T.
Published: October 2005

The study was concerned with synthesis of bamboo tones of shakuhachi entirely by the technique of linear predictive coding which is commonly used in speech synthesis. The bamboo pipe was modeled as a linear digital filter. In applying the linear prediction method, the noise source from the lips and the transfer function proper to the shakuhachi were separated from the bamboo tones by means of predictive coefficients. From our preliminary experiments, this noise source function was found to be quasiperiodic. For synthesizing, a pulse train on which white noise was superposed was used as the source of the sound, which was amplitude modulated by the power of the bamboo tone. This pulse train was combined with the predictive coefficients for synthesizing. The results were satisfactory, recreating its natural tone quality.
Pre-Columbian ocarinas, pipes, and whistles
Authors: Rawcliffe Susan
Published: October 2005

The instruments of the Pre-Columbian American cultures indicate a high degree of sophistication with an evolving period of experimentation and study and an empirical understanding of acoustical principles. The greatest variety of ocarinas and whistles were made by the Mayans. As indicated by the proportions of the aperture, most air-ducted ocarinas and pipes I have observed in photos and through actual playing show a preference for an “open” tone, as contrasted to a more “reedy” tone. The addition of a chamber around the mouthpiece greatly affects the tone quality while allowing the sound to be “flipped” from one chamber to another. This type shows up in many forms. A ball was often added to a chamber to produce a vibrato, as in the modern police whistle. A ball was also used in a closed tube to raise and lower the pitch; by changing the angle of the tube while playing and thus rolling the ball back and forth, the tube was shortened or lengthened. Multiple pipes evidence very sophisticated tunings and tone control. The multiple pipes which I have made generate strong combination tones. Pre-Columbian multiple pipes must function similarly. My own double whistles and the double Pre-Columbian whistles which I have played generate one, two, and more combination tones which can be distinctly heard. The Pre-Columbians must surely have been aware of combination tones long before they were discovered by Tartini.
The natural evolution of musical scales
Authors: Yadzinsky Edward
Published: October 2005

The 12-tone chromatic scale shall be shown to be a natural and evolutionary phenomenon, due to the pitch/interval responses of human hearing and the integral harmonic series of vibrating strings, air columns, and vocal chords. The important implication is that if music were to begin all over again, from day one, the same scale system of 12 chromatic tones would result. Credibility is gained by considering the origins of the pentatonic scale, in use since primitive times and precursor to the major and chromatic scales. The pentatonic scale is polytenetic, having evolved independently in the earliest cultures of Europe, Japan, Africa, Americas, Oceania, etc. Historically the Pythagoreans suspected the universality of such musical scales which consequently influenced their search for a numerical universal “harmony” (i.e., harmony of the spheres). Specific examples of vibrating strings under uniform tension (rates of vibration being reciprocal to respective lengths) can be seen in relationship to Pythagorean efforts ot represent the universe in simple numeric/geometric expressions. Beyond such historical considerations, comprehensive evidence supports the premise that the 12-tone scale system of music, rather than being invented or contrived, is a natural, evolutionary and universal manifestation.
Diagnosis and control of noise in alder rapid transit cars
Authors: Remington P. J. , Myles M. M.
Published: October 2005

In the United States there are a number of rapid transit systems still utilizing 20–30 year old transit cars. Although still functional many of these old cars are quite noisy, especially when compared with cars of more recent vintage. In a study for the New York City Transit Authority extensive diagnostic tests were performed on two types of older rapid transit cars presently operating on that system. The tests have shown that the noise reaching the car interior is primarily airborne and due to the propulsion system and wheel/rail interaction. The propulsion system was also found to contribute strongly to wayside noise. When the car is operating in a tunnel most of the noise reaches the car interior by way of the side doors, whereas when operating above ground most of the noise is transmitted through the floor. Based on the test data, an analytical/empirical model of noise transmission to the car interior has been developed and used to estimate the effect of various noise control treatments on car interior noise.
On the ground-borne noise propagation from a subway
Authors: Tokita Y. , Oda A. , Shimizu K. , Kimura K.
Published: October 2005

A prediction method of noise in the house caused by subway was studied by means of hypothetical theory and measurement. This procedure was done by two steps. (1) Supposing that waves propagate through the various impedance soil layers from the imaginary vibration line source placed under the subway construction to the ground surface. Coefficients in the formula are decided experimentally with measured values. (2) The relation between the noise in the house and ground vibration led by field measurements. The acceleration level (Lv) on the ground is expressed as follows: Lv = Lo − 10 log ( ∑ i ri) −  ∑ i αiri +  ∑ i Ki, where Lo is the imaginary source level; ri, the propagation path length in the i'th soil layer; αi, the damping factor of i'th layer; and Ki, correction terms. The relation of ground vibration (Lv) and noise (LA) in the house is expressed as LA = 0.88 Lv − 17. These relations may be available in the case of comparatively shallow tunnel construction. [Works supported by Teito Rapid Transit Authority and Yokohama City.]
Ground vibration from rail rapid transit train operation
Authors: Nelson James T.
Published: October 2005

Vibration data obtained for various rapid transit subway configurations are discussed with regard to the Washington Metropolitan Area Transit Authority (WMATA) and Toronto Transit Commission (TTC) systems. Measurement locations included the subway structure, ground surface, and at nearby buildings. The data were analyzed in 1/3 octave bands with a range of 3.15–1250 Hz. The subway configurations tested included circular rock tunnels, circular earth tunnels with steel and concrete liners, horseshoe rock tunnels, and cut-and-cover double box subways. Evaluation of the WMATA continuous floating slab performance characteristics are included, and comparisons are made between the continuous (WMATA Metro) and discontinuous double tie (TTC) floating slab trackbeds which represent the state-of-the-art in rail transit train vibration isolation. Surface ground vibration characteristics of the WMATA Metro system are also compared with those of the TTC system.
Acoustic emission generated by dislocation mechanisms during the deformation of metals
Authors: Carpenter S. H. , Heiple C. R.
Published: October 2005

A review of acoustic emission data generated by dislocation processes during the deformation of metals will be presented. The information, developed from an extensive review of the acoustic emission literature and from current unpublished results, will be analyzed to identify source mechanisms and characteristics of the emissions. The analysis will concentrate on the influence of the metallurgical condition of the test material on the observed acoustic emission. The effects of testing parameters and techniques will also be examined. Examples of major concerns in these areas will include the effects of crystal structure, purity, grain size, heat treatment, prior mechanical work and deformation orientation with respect to crystal axes or crystallographic texture; difference in acoustic emission obtained during tensile, compressive and biaxial deformation; and a comparison of acoustic emission data presented as ringdown counts, burst rate, and RMS voltage. Some examples of acoustic emission results mistakenly attributed to dislocation mechanisms will also be discussed briefly. [Portions of this work were supported by NSF, AFOSR, and DOE (contract EY-76-C-04-3533).]
Structure response to free-field acoustic excitation using the statistical energy analysis method
Authors: Wong D. M.
Published: October 2005

The statistical energy analysis (SEA) method has been used very successfully in the prediction of structural response to reverberant acoustic fields. This paper will extend the method to the free field acoustic excitation. The coupling loss factor (an important parameter in the SEA method) between the plate and the exciting acoustic field was derived. The analytical results compare reasonably well with test data obtained during the Space Shuttle (OV-101) acoustic test conducted at Dryden Flight Research Center. In that test two F-104 jets were used as sound sources. Accelerometer readings were obtained both at the cargo bay door and the side wall. Also, the correlation constants of the pressure field were obtained from the test.
A method to estimate collision point of a rod
Authors: Ikeda Osamu , Sato Takuso , Yamamoto Jiro , Manabe Akira
Published: October 2005

A method to estimate a collision point on a metal rod is presented, which is based on spectral analyses of the signal detected by a single accelerometer attached to the rod. First, the transfer function from a forcing point to the detector is derived, where space-variant and dispersive characteristics of the rod are taken into consideration in connection with varying diameter, the change of surrounding condition along the axis, and various transmission and reflection loss characteristics. Then an algorithm to estimate collision point is derived by using the obtained transfer function. The fundamental experimental results obtained by using a 17S aluminum rod of 2003 mm length and 10 mm diameter show that the estimated positions agree with the actual collision positions with a precision of less than 1 cm [O. Ikeda, T. Sato, J. Yamamoto and A. Manabe, J. Acoust. Soc. Am (to be published)].
Measurement of the effects of fluid loading on the dispersion of flexural waves in a plate
Authors: Jarvis James L. , Wilson O. B.
Published: October 2005

Dispersion curves for axially symmetric flexural waves in a circular aluminum plate were determined using both direct phase measurement and measurement of normal mode frequencies. Measurements were conducted with the 8 mm thick plate in air and with one surface loaded by water in an anechoic tank over a frequency range of 20–80 kHz. At frequencies larger than the critical frequency (at which the free plate flexural wave speed is equal to the sound speed in water) the phase speeds with the plate in air and in water are nearly equal and agree well with the Mindlin-Timoshenko theory. Below critical frequency, water loading causes significantly lower values, about fifteen percent here. The speed in the water-loaded plate appears to change abruptly at or just above the critical frequency, in reasonable agreement with the theory of Kurtze and Bolt [Acustica 9, 238–242 (1959)]. Significance of these results will be discussed. [Supported by the Naval Undersea Warfare Engineering Station, Keyport, WA.]
Effects of fluid loading on the transmission of free waves across a rib
Authors: Crighton D. G. , Maidanik G. , Eisler T.
Published: October 2005

The line drive admittance and the transfer admittance have been evaluated in closed form for a plate or membrane. The evaluation takes into account the influence of fluid loading. Both admittances are necessary for the evaluation of the transmission of free waves across a rib. In this calculation the rib is characterized by an impedance, and the influence of fluid loading can thus be ascertained. Computations illustrating this effect in a number of cases of interest are presented and discussed. In the case of the membrane the phenomenon associated with the critical frequency is introduced by assuming the tension to be frequency dependent.
The sensitivity of LPC synthesized speech quality to the imposition of artificial pitch, duration, loudness, and spectral contours
Authors: Sanders William , Gramlich Carolyn , Levine Arvin
Published: October 2005

In attempting to achieve more natural sounding synthesized speech, it is important to determine how speech quality is sensitive to the manipulation of spectral information and prosodic components: pitch, duration (rhythm), and loudness, in synthesizing sentences. To assess the impact of each of these on the perceived naturalness of synthesized speech, we have modified a set of recorded and LPC-analyzed sentences, so that one or more of these components is rule-generated from information available from a lexicon of individually recorded and LPC-analyzed words. Substitution of lexical spectral information for the observed sentential spectra, or imposition of a syntactically generated pitch contour [A. Levine, J. Acoust. Soc. Am. 60, S76(A) (1976)], are two of the modifications we will be testing here. Results will be reported from an experiment in which listeners judged the naturalness of these modified sentences. The usefulness of evaluating modified natural speech as a method for studying the quality of synthesized speech will also be discussed.
Optimal quantizations of coefficient vectors in LPC speech
Authors: Buzo A. , Gray R. M. , Gray A. H. , Markel J. D.
Published: October 2005

Recently developed algorithms for determining locally optimum vector quantizers for a given bit rate, distortion measure, and data training sequence are applied to vectors of reflection coefficients from LPC coded speech files with Itakura-Saito and spectral deviation distortion measures. The resulting quantizers with 7 or 8 bits per frame for 10 dimensional reflection coefficient vectors, yield comparable average spectral distortion to nonvector quantizers using far more bits. Tapes of the low bit rate synthetic speech will be played to illustrate the usefulness of the algorithms, the distortion measures, and the vector quantizers. [Work of first two authors supported by AFOSR.]
The acoustical radiation force in intense sound field
Authors: Leung E. , Wang T. G.
Published: October 2005

The acoustical radiation force on a small sphere has been reported by Rudnick [J. Acoust. Soc. Am. 62, 20 (1977)]. His experimental results are in good agreement with King's calculation [Proc. R. Soc., London A149, 212 (1934)]. We report here an investigation of acoustical radiation force including the effects of sound scattered by sample size, shape, and surface condition. We have found that in general our results agree with King's calculation in the region close to a pressure node or antinode, but deviates in the region between the pressure node and antinode (x ≃ 18λ). Comparison with Westervelt's calculation [J. Acoust. Soc. Am. 22, 319 (1950)] will also be discussed. [Work supported by NASA.]
A low bit rate vocoder based on an improved cepstral method
Authors: Imai S. , Abe Y.
Published: October 2005

By using a log spectrum approximation filter modeling a true log spectral envelope, a low bit rate and high-quality vocoder was realized. The true log spectral envelope is represented by the low-quefrency portion of the cepstrum modified by the low-quefrency components involved in the rectified fine structure of the log spectrum. It can be obtained through an iteration of DFT-rectifying-IDFT for the higher quefrency portion of the speech cepstrum. In this vocoder, the speech synthesis is not based on the homomorphic method. The synthesis filter can be obtained without transforming the cepstrum to an impulse response, and the computation for the filter coefficients is very simple. The spectral envelope information is transmitted in the form of a differential cepstrum corresponding to the difference of two adjacent log spectral envelopes, and the data rate is very low.
Some comparisons between the articulation rates of LPC and diphone or monophone-based synthesis by rules
Authors: Braun H.-J.
Published: October 2005

In order to make a quick decision about the quality to be expected from a speech coding or synthesis system, a special articulation test procedure for the German language has been developed. The basic test elements consist of 100 VC or CV transitions which are spoken in a C-VC-V or V-CV-C environment. These elements were selected to contain all most probable transitions of the German language (about 57%) and to be representative of the probability of the 40 single sounds used. Bandlimited speech, LPC speech, as well as synthetic speech have been tested by this procedure. The synthesizers which have been used are a diphone-based formant synthesizer (SAMT) developed at our laboratory and a German version of the VOTRAX synthesizer (VS-6.G2). Articulation rates and confusion matrices give detailed information about the quality of the synthesis and coding systems for the German language. In comparison to similar measurements with the Japanese language, the articulation rate is strongly influenced by vowel confusions. This is due to the fact that there are more than 20 discernible vowels in the German language and only 5 in the Japanese language.
Subjective quality testing of a new source model of LPO vocoders
Authors: Huggins A. W. F. , Schwartz R. M. , Viswanathan R. , Makhoul J.
Published: October 2005

Our source model, reported at an earlier meeting, excites the LPO speech spectrum with a low-frequency band of pulses mixed with a high-frequency band of noise. Pulses are low-pass filtered and noise is high-pass filtered at the same frequency, to yield a flat source spectrum. The cutoff frequency of the filters, a continuous variable, replaces the usual binary voiced/voiceless decision. Thirty-six phoneme-specific test sentences were processed through a single high-quality vocoder (5 kHz bandwidth, 11 poles, no quantization, 100 frames/s), which was excited in turn by both the usual pulse/noise source and by the new source. Subjects rated the resulting speech separately on eight-point buzziness and breathiness scales. The results show that the new source model greatly reduces perceived buzziness, occasionally at a cost of slightly increased breathiness. Any remaining inadequacies can probably be ascribed to the algorithm that extracts the cutoff frequency during analysis, rather than to the model itself. [Work supported by ARPA-IPTO.]