bluuebunny/crossref_metadata_2025 huggingface dataset:
Quarry blast acoustic wave (concussion)—response of structures and human annoyance
Authors: Taylor A. G.
Published: October 2005

Quarry blasts were monitored during 1973–1974 at several locations in Southern Ontario to determine if the acoustic wave (concussion) was likely to be of significance from the points of view of structural damage and human annoyance. The monitoring instrumentation used included sonic boom microphone-carrier system and FM tape recorder. The characteristics of the blast wave, overpressure, spectrum, and duration were analyzed using a storage oscilloscope and real-time analyzer and were compared with the characteristics of sonic booms. The two phenomena were shown to be similar in spectral content to most energy in the infrasonic region; the overpressure at several thousand feet from a blast can be similar to that of a sonic boom, whereas the duration of the pressure perturbation is several times longer for a blast than for a sonic boom. It is concluded that damage and annoyance criteria developed from sonic boom research may reasonably be applied to quarry blasts. It was also found that in many instances the acoustic wave can be of greater significance than ground vibration induced by the same blast.
Optimization of duct acoustic liners of finite length
Authors: Beckemeyer R. J. , Sawdy D. T.
Published: October 2005

Lining design procedures generally ignore the effects of discontinuous impedance changes on reflection and modal redistribution of incident energy. The present study, spurred by preliminary work of Lansing and Zorumski [J. Sound Vib. 27, 85–100 (1973)], involves identification of the mechanisms of optimal suppression by segmented liners, including impedance discontinuity effects. The mathematical model involves rectangular ducts in the absence of mean flow. Segmented acoustic fields are represented by duct eigenfunction expansions, and mode-matching is used to ensure continuity of the total field. Extensive parameter studies were used to identify attenuation mechanisms and to define preliminary liner configurations. These are used as starting values in conjugate-gradient optimization studies. Optimal segmented liners are presented and it is shown that, if the sound source is sufficiently well-defined, conventional infinite duct optimum attenuation rates can be improved upon by taking advantage of wall impedance discontinuities. Finally, follow-on work which will involve experimental verification and extension to the case with mean flow in the duct is described. [Supported by the NASA Lewis Research Center under Contract NAS3-18550.]
Application of small-amplitude theory to the analysis of practical engine exhaust silencers
Authors: Eriksson L. J.
Published: October 2005

Traditional problems in the application of small-amplitude theory to the analysis of practical engine exhaust silencers are discussed. Insertion loss measurements obtained in an anechoic chamber using a 27-in. 3-displacement, single-cylinder, four-stroke engine as a noise source are compared to the analytical results from lumped parameter as well as distributed parameter models. Included in this study are comparisons for a variety of exhaust and tailpipe configurations, a series of expansion chambers with a variety of lengths and diameters, and several more typical silencing elements, including pass and plug silencing sections. It is shown that although, quantitatively, the agreement between the analytical models and experimental results is poor, qualitatively, many features predicted by the small-amplitude theory are observed. This is particularly true for a good distributed parameter model of the silencing system. Based on these results, a series of recommendations is made for improved silencer design.
A noise and yet
Authors: Ryan William W.
Published: October 2005

The Coast Guard under Title 33, Chapter I, of the Code of Federal Regulations is designated to provide sound signals as aids to navigation. Cannons and bells were the first types of sound to be used. Through the years the sounds have changed. A tape recording of most of the sounds through the evolutionary years will be played. Some of these sounds are more readily accepted by the public than others. The Coast Guard is trying to provide the most pleasing sounds with the least amount of “noise pollution” from each signal, while providing the maximum signal necessary to the mariner. A summary of the noise control efforts being used in aids to navigation will be presented.
Short-term complaint system concerning community noise near a fossil-fuel power station
Authors: Thompson A. R. , Hoover R. M.
Published: October 2005

A communication system by which local residents living near a fossil-fuel power station could complain about the types of sounds propagating across their properties is described, as are the results of the program. This complaint system was in operation 24 hours a day for one month, during which time a consultant lived in one of the residences surrounding the station. In this way, he was situated in the same environment as the complainants and was able to make observations and measurements on a continual basis. Thirty-one complaints were received during the monitoring period with calls coming from each of the three nearby residential areas. The 31 calls were identified to be principally due to four noise sources—high-pressure safety valves venting steam to the atmosphere, large induction motors driving the station circulating water pumps, induced draft fans, and main power transformers.
Measurements of short-term variations in sound-velocity profiles
Authors: Larsen H. L.
Published: October 2005

Variations in the sound-velocity profile were measured at a fixed site near San Clemente Island. The measurements were derived from a series of salinity-temperature-depth (STD) drops and velocimeter drops to depths of 450 m over a period of approximately Eight hours. The means and standard deviations of temperatures, salinities, and sound velocity during this period were computed. Acoustic propagation loss predictions based on a ray trace model were computed for the velocity profiles. Contour plots of equal values of predicted propagation loss at depths to 450 m and ranges of 1500 m were prepared to illustrate the variations in propagation conditions which result from the measured short-term variations in sound-velocity profiles. Significant changes in predicted propagation conditions were noted in relatively short periods of time.
Scattering from a random two-fluid interface
Authors: DeSanto John A.
Published: October 2005

The problem of calculating the Green's function for two semi-infinite fluid media separated by a random interface is considered. The fluids have different (constant) densities and sound speeds. Gaussian statistics are assumed for the interface height. Continuity conditions at the interface are used to relate the Green's functions in the two media. Their relationship to the flat surface results is shown. Using these continuity conditions and Green's theorem in both media (with a single free-space Green's function), mixed surface-volume integral equations are derived for the coordinate-space Green's function. These equations simplify in Fourier transform space and, using diagram methods, further integral equations are derived for the mean and second moment of the Green's function (Dyson and Bethe-Salpeter equations, respectively). Some scattering examples are considered, and the relation of our results to other formulations of the problem is discussed.
Atmospheric absorption of sound: validity of a two-relaxation-process model
Authors: Evans L. B.
Published: October 2005

A comparison of the predictions of the generalized kinetic description and the simpler two-relaxation-process model for absorption of sound in air has been made [H. E. Bass, J. E. Piercy, L. B. Evans, and L. C. Sutherland, J. Acoust. Soc. Am. 56, S1(A) (1974); L. C. Sutherland, J. E. Piercy, H. E. Bass, and L. B. Evans, J. Acoust. Soc. Am. 56, S1(A) (1974)]. The two models are compared over the temperature range of 0°–40°C and from 1% to 100% relative humidity. The agreement between the two models is found to be quite good in most cases, with some exceptions which are easily explained. A comparison is also made to available experimental data which shows the validity of the models.
Comparison of hearing aids with amplitude compression
Authors: Nábĕlek Igor V. , Robinette Larry N.
Published: October 2005

Intelligibility tests using a Modified Rhyme Test were performed with commercially available hearing aids with various values of time constants of compression and of harmonic distortion during overshoots. For normal-hearing subjects, signal-to-noise ratios of −2 and +2 dB were used. Subjects with sensorineural hearing impairment listened in quiet at their most comfortable listening level and at two additional sound-pressure levels (10 and 20 dB lower than the first one). Different hearing aids affected the performance of both groups of listeners similarly. At low signal-to-noise ratios or low sound-pressure levels, better performances were observed for shorter time constants of compression and larger compression ratios. At higher signal-to-noise ratios or higher sound-pressure levels, the occurrence of harmonic distortion during overshoots deteriorates the performance, with greater deterioration occurring for impaired hearers than for normal hearers. [This research was supported by the NIH Biomedical Sciences Support Grant (FH 07088) and by the Prosthetic and Sensory Aids Service of the Veterans Administration.]
Some statistical properties of linear prediction coefficients for speech signals
Authors: Meister S.
Published: October 2005

Using speech material consisting of three groups of 24 sentences each, including statements and queries, spoken by three adult male speakers, coefficients were produced by an eight-pole linear-predictive coding autocorrelation method. The nonparametric Smirnov test statistic was used to test the hypothesis that the probability distribution function of the coefficients were homogeneous across speakers and across sentence groups, with separate tests for the voiced, unvoiced, and combined sample. With few exceptions it was found that the distribution functions were significantly different across different speakers uttering the same sentence group, and across different sentence groups uttered by the same speaker. Most of the computed Smirnov statistics were significant at the 0.001 or 0.002 level. This result held for the voiced, unvoiced, and combined samples. Results indicated the reflection coefficients from a single speaker or a single sentence group cannot be used as a basis for statistically describing the entire population of reflection coefficients.
Nuclear stress rule in French
Authors: Vaissière J.
Published: October 2005

Fundamental frequency (F0) contours for isolated French sentences and two texts read by five native speakers have been analyzed, and these contours have been schematized subjectively by ideal patterns, using basic elements such as those described by Maeda [J. Acoust. Soc. Am. 56, S33(A) (1974)]. It has been found that long words (words containing more than two syllables) indicate one of four typical patterns, P1, P2, P3, or P4. These patterns may be described by using six attributes: initial rise (Ri) at the onset of the words, failing (F), sustaining (S) during the middle portion of the words, continuation rise (Rc), peak followed by a strong fall (P), and lowering (L); the last three attributes occur at the end of the word. Then, P1 = Ri + F + Rc, P2 = Ri + S + P, P3 = Ri + S + L, and P4 = Ri + F. A word in a sense group (a group composed of closely related words in a sentence) receives one of the four patterns, depending first on the tendency of the particular speaker (each speaker using one pattern more often than the remaining), second on the structure of the sense group, and third on the position of the sense group in the sentence. A rule similar to the Nuclear Stress Rule for English can be used to predict all possible combinations of the four patterns for the words inside a sense group.
Some musically oriented acoustics demonstrations
Authors: Benade Arthur H.
Published: October 2005

The redundant, though irregular, transmission of sound from source to listener in an ordinary room provides our ears with a large amount of useful information which often permits them to perform far better, in such a room, than one would expect on the basis of psychoacoustic experiments done in anechoic chambers, or by the use of headphones. Some musically oriented illustrations are provided to support this observation.
A mini-course for non-science majors on sound reproduction and psychoacoustics
Authors: Apfel Robert E.
Published: October 2005

This eight-lecture mini-course is one of six sections of a course at Yale University entitled “Perspectives on Technology.” The student chooses three sections for the equivalent of a one-semester course. In a large sense the course might be retitled “John Tyndall—Revisited and Updated,” because Tyndall's style of providing an empirical understanding of acoustical phenomena through extensive use of demonstrations is closely followed. In our first year 270 Yale undergraduates (roughly 5% of all undergraduates) took this section. Organization, many handouts, meaningful demonstrations, continuity, and some showmanship were responsible for the success of this mini-course.
Visual and audible demonstration of complex acoustical phenomena: computer movie of jet noise nd computer-generated thunder
Authors: Ribner H. S.
Published: October 2005

In the past, ripples on a water surface have been used to model acoustic phenomena—Doppler effect, diffraction, etc. Here we have used a computer to draw moving groups of sound waves at successive small time intervals. The result is a computer movie on jet noise generation. We can also listen to the rumble and roll of synthetic thunder generated by a computer program. Starting with a digitized photo of a lightning stroke, the computer models the physics of the sound-generation process in terms of a sequence of overlapping sonic booms. The sound is generated by the tape-recorded output of the computer. [Supported by National Research Council (Canada) and U. S. Air Force Office of Scientific Research.]
Lecture-experiment on sound-level judgments
Authors: Gales Robert S.
Published: October 2005

In a lecture to a lay group of 17 elementary school teachers, the teachers were asked at the outset to estimate the sound level in decibels of an electric shaver operating ca a small table. The estimates were distributed approximately uniformly from 2 to 100 dB. Following the one-hour lecture, in the course of which a sound-level meter was used to demonstrate the measurement of the sound level (A weighted in all cases) of the room ambient, speech, and high-level rock music, the members of the audience were again asked to estimate the sound level of the same electric shaver. The estimates were approximately normally distributed with a mean of 62 dB. The mean of the measured sound levels was 65 dB. Fifty-three percent of the estimates were in the 60- to 70-dB range.
How to measure noise pollution
Authors: Young Robert W.
Published: October 2005

In 1929 when the New York City noise survey was in progress, Dr. Harvey Fletcher from the Bell Telephone Laboratories explained in a radio talk “How Noise is Measured and Why.” It continues to be helpful to illustrate how a noise level of 90 decibels really sounds. (He used a recording of applause at one of the receptions for Charles Lindbergh.) He demonstrated a reduction of noise level of 10 decibels. (He did not say 10 dB.) He gave the elements of “a piece of apparatus” that today we call a sound-level meter. He led to the easy conclusion that noise level is the same as sound level, and that you measure it with a sound-level meter. There was no entanglement with logarithms; simply “the decibel is a unit used to describe an amount of noise.”
Violin with a graphite-epoxy top plate
Authors: Haines Daniel W. , Chang Nagyoung , Hutchins Carleen M.
Published: October 2005

A violin has been constructed with a graphite-epoxy sandwich plate replacing the usual spruce top plate. This substitute plate is similar in design to that previously described for a guitar application [D. W. Haines, N. Chang, and D. A. Thompson, J. Acoust. Soc. Am. 55, S49(A) (1974)]. Modern methods of violin design and construction produced an instrument with qualities which compare favorably with those of superior violins constructed entirely of wood. Frequency response curves and mode shapes are presented of the detached graphite-epoxy plate. Shown also are frequency response curves of the complete instrument. Comparisons with similar tests of high-quality wood plates and instruments are also given. The new violin will be available for examination. [Assistance from Mr. Stan Cross of Hercules Inc., Mr. John C. Schelleng, and Dr. Karl Stetson. Supported by National Science Foundation Grant GK-35913.]
Digital tone synthesizer and beat indicator for tuning musical instruments
Authors: Widener M. Ward
Published: October 2005

Using a stable quartz oscillator and low-powered digital logic, a musical tone synthesizer has been developed to produce scales with any tempering of intervals based on any selected tuning reference pitch. A clock frequency of 3.5 MHz permits A-440 to be set closer than 0.11 Hz, or 0.43 cents. The logic includes a 12-bit presettable binary counter which is programmed to divide by a selected number of clock pulses. The binary number is encoded in a standard computer card and interpreted by a special mechanical card reader. The first octave can be divided into 1988 intervals. In addition to the equally tempered scale, cards have been encoded for Pythagorean, mean tone, and just intonations in a wide variety of reference pitches. The direction and magnitude of tuning errors are shown on a solid-state display driven by a type of ring counter. The result is a compact portable instrument that is both accurate and programmable to the pitch requirements of the user.
Evaluation of an automatic speaker verification system over telephone lines
Authors: Rosenberg A. E.
Published: October 2005

An automatic speaker verification system accessed by test customers from their own telephones over dialed-up lines has been evaluated. The test population consisted of over 100 male and female speakers who called up nominally once a day over a period of five months. The operation of the system is based on an acoustic analysis of a fixed, sentence-long utterance resulting in a set of functions of time. These functions are compared with stored reference functions to determine whether to accept or reject an identity claim. Details have been described elsewhere [A. E. Rosenberg and M. R. Sambur, IEEE Trans. Acoust. Speech Signal Proc. (Spring 1975)]. The sytem has been implemented on a NOVA 800 laboratory computer. Telephone line access to the computer is via a DATA-PHONER hookup. Identity claims are made by keying an identification number on a TOUCH-TONER dial. Instructions and responses to the customer are made by means of a programmed voice-response system. Reference data was computed off-line and updated with the analysis data of accepted utterances. The evaluation indicated an error rate of approximately 10% for new customers mad approximately 5% for adapted speakers.
Electrical characteristics of hearing aids as predictors of speech understanding
Authors: Smaldino J. J.
Published: October 2005

Multiple regression analyses were performed to determine optimal prediction of speech understanding from a knowledge of 31 discrete measurements of hearing aid performance. A moderate prediction of speech understanding [R = 0.65] was obtained by pooling measures of (1) low frequency response, (2) gain, (3) smoothness of frequency response, and (4) transient distortion. The data indicate that identification of complex patterns of electrical interactions may be necessary to precisely predict speech understanding.
Effect of prosodic boundaries on the performance of a speech understanding system
Authors: O'Malley M. H. , Cole A.
Published: October 2005

Major syntactic phrases in careful speech are often delimited by prosodic features (length, silence, F0 contour). It is generally assumed that if such prosodic features could be detected, they could be used to guide the syntactic analysis phase of a speech understanding system. In the present experiment, a working system (HEARSAY I vocal chess at Carnegie-Mellon University) was modified so that phrase boundaries which were marked by more than 200 msec of silence were detected. The syntax was then modified so as to make use of these prosodic boundaries. The effect of these modifications on system speed and accuracy was measured with careful speech in which most phrase boundaries were present. Speed was not significantly affected. Accuracy as measured by the percentage of words correctly recognized (with the semantic component disabled) increased from 50% to 62%. The paper discusses some of the factors which affect system performance.
Intelligibility of voice transmission through a satellite relay system
Authors: Milner Paul , Golab Joseph S.
Published: October 2005

An extensive series of measurements of speech intelligibility of voice modems designed for a satellite relay link to air and sea-going craft has been underway since 1973. Measurement parameters for these tests are the system functions associated with ATS-6 satellite L-band and C-band transmission. These include comparative evaluations of intelligibility versus C/N0, angle of elevation, antenna systems, range, and sea-state. For these tests, a number of modifications and adaptations of standard intelligibility testing procedures were developed. These include using a large number of scramblings of the PB-50 word lists (ANSI S3.2-1960) recorded by four professional talkers, two male and two female, in an anechoic room. Because of time considerations in conducting the flight tests, the lists were recorded at a reading rate of 2.5 sec with no carrier phrase Experiments showed that highly trained listeners [P. Milner, “Advantages of Experienced Listeners in Intelligibility Testing,” IEEE Trans. Audio Electroacoust. AU-21, 161–165] have no difficulty performing the open response task at this rate and that correlation between this method and the usual method is high. [This work performed under U. S. Dept. of Transportation Contract No. DOT-TSC-604.]
Test-retest performance of males and females on a verbal dichotic listening task
Authors: Johnson L. K. , Ryan W. J.
Published: October 2005

Verbal dichotic listening performance was investigated in 25 adult males and 25 adult females with normal and symmetrical hearing. Subjects were retested approximately one month after their initial test and performance analyzed on the basis of ten measures. A mean right-ear superiority was demonstrated by beth males and females, as well as the combined group of males and females. Significant mean differences between male and female performance were found for eight of the ten measures on at least one testing occasion. While initial test performance was found to be significantly and positively correlated with retest performance on all measures for males, females, and the combined group, mean differences in test-retest scores were found in at least one group on seven of the ten measures. In addition, listener reliability varied as a function of the measure used and the sex of the listener. These results were interpreted as justification for the distinction between male and female performance and evidence of the instability of an individual's response to verbal dichotic stimulation.
Speech perception and production errors in dysphasic children
Authors: Stark Rachel , Tallal Paula , Curtiss Barbara
Published: October 2005

In studies previously reported, children with developmental dysphasia were found to have difficulty in perceiving brief duration acoustic cues in speech. Thus, they were unable to identify and discriminate initial stop consonants. Twelve developmental dysphasics who had participated in the previous studies and 12 matched control children were tested for their ability to imitate (1) steady-state vowels, (2) diphthongs, (3) consonant-vowel syllables, and (4) consonant-vowel-consonant nonsense syllables produced by the examiner. Secondly, subjects were required to produce the names of pictured objects, i.e., single-syllable words comprising stop-consonants in the initial and final position as well as words comprising primarily vowels and diphtongs. Results demonstrated that the dysphasics who failed to discriminate between speech sounds which incorporate brief duration acoustic cues also failed to produce such speech sounds correctly. Those dysphasics whose perception of brief acoustic cues was less impaired in the previous studies were less impaired in their production of speech sounds incorporating brief acoustic cues. All subjects proved able to produce correctly longer duration vowel sounds, which are easily perceived. Short vowels and diphthongs were difficult for those children whose rate of processing problems were most severe.
Critical bandwidth in lateralization: effect of rise-fall time
Authors: Scharf Bertram , Florentine Mary
Published: October 2005

The onset time difference, ΔT, needed to lateralize dichotic tones toward the leading ear was measured as a function of the frequency difference, ΔF, between the tone in one ear and the tone in the other ear. Six listeners judged the locus of a sound image produced by two simultaneously terminated 30-msec tone bursts (near 80 dB SPL). With the center frequency set at a geometric mean of 2000 Hz, the threshold, i.e., the ΔT at which 75% of the judgments were correct, was measured at ΔF's from 0 to 1600 Hz. For a rise-fall time of 10 msec, the median threshold hovered around 0.12 msec when ΔF was less than or equal to 300 Hz, which is the critical band at a center frequency of 2000 Hz; beyond 300 Hz, threshold increased with ΔF. Similarly, for rise-fall times of 1 and 5 msec, auditory lateralization was independent of ΔF for tones within the same critical band, but worsened as ΔF widened beyond the critical band. [Research supported by NIH Grant No. NS 07270.]
Measurements of the distortion tone 2f1−f2 in normal-hearing and high-frequency-hearing-loss subjects
Authors: Sachs R. M. , Wightman F. L.
Published: October 2005

We made pulsation threshold and cancellation measurements of 2f1−f2 distortion on three normal-hearing subjects and on three subjects with a steep high-frequency hearing loss. In one condition, we varied the level (L1) and frequency (f1) of the lower frequency primary tone, with the other tone fixed. In another condition L2 only was varied. With f1 fixed and L1 varied, the level of 2f1−f2 is maximum for some particular value of L1 (L1max). For L1 above and below this level, the level of 2f1−f2 is reduced. Similarly, as L2 is varied, 2f1−f2 level is maximum at L2max. With subjects having a high-frequency hearing loss, the two maxima normally seen are less pronounced or absent. Typically, an asymptote is observed rather than a maximum. These effects occur when f2 is at the lower edge of the hearing loss in a region of normal hearing. This suggests that information at cochlear places basal to the f2 place can play an important role in perception of combination tones. For normal-hearing subjects, as f1 is decreased, L1max increases. If L1 is kept at L1max as f1 is decreased, the phase of the 2f1−f2 cancellation tone is nearly constant. This is in contrast to the rapid phase shift observed as f1 is decreased with L1 fixed.
Improved binaural sound reproduction
Authors: Wallace R. L.
Published: October 2005

When sound is picked up by a mannequin head equipped with Zwislocki inner ears and molded replicas of real human pinnae, it can be reproduced with a high degree of realism by using special headphones designed to produce a flat-frequency response as measured at the eardrum. A live demonstration of such pickup and reproduction will be given. Tape recordings of music recorded in this way will also be available for listening.
Microstructure of the pure-tone threshold
Authors: Thomas Ian B.
Published: October 2005

A painstaking determination of pure-tone threshold versus frequency in normal ears using a headphone reveals a remarkably consistent microstructure of hills and valleys. A large number of determinations of the frequency at the peak of each of the threshold hills reveals a standard deviation of less than 10 Hz for stimulus frequencies from 200 to 5000 Hz. The widths of the intervening valleys correspond remarkably closely to commonly accepted critical bandwidths for the stimulus frequencies concerned. Must surprisingly of all, the perceived pitch of the stimulus jumps quantally from one valley to the next at intensities very close to threshold and remains constant as stimulus frequency is varied within the “passband” of the valley. If intensity is increased just slightly above the top of a hill at the hill frequency, a mixture of the pitches including interactions is heard. If intensity is increased a few decibels still further, an apparently smooth variation of a single pitch is observed as stimulus frequency is varied from side to side of the hill frequency. Hill frequencies are in general different for the two ears. Intensity difference from top of a hill to the bottoms of the adjacent valleys is typically 12 dB. These findings strongly support the place theory of hearing for pure-tone stimuli.
Frequency discrimination in four temporal paradigms
Authors: Yost William A. , Berg Kathy
Published: October 2005

The frequency discrimination of a 20-msec target tone was measured in four temporal paradigms: a single-interval procedure; a two-alternative forced-choice procedure; a same-different procedure; and a method-of-adjustment procedure. In each paradigm the 20-msec target tone was preceded or followed by a 500-msec 800-Hz interfering tone. The interfering tone occurred 50 msec or 5 msec before the onset of the target tone; or the interfering tone occurred 100 msec or 5 msec after the offset of the target tone. In each of the four paradigms the value of Δf for the target tone was varied around 800 Hz to obtain an estimate of the frequency discrimination threshold in the method-of-adjustment procedure and estimates of psychometric functions in the other three procedures. In all paradigms except for the single-interval procedure, no significant differences in observers' frequency-discrimination behavior occurred among the four temporal orderings of the target and interfering tones. The thresholds for Δf from these three paradigms were the same as those obtained without an interfering tone. In the single-interval task, that temporal order in which the target tone preceded the interfering tone by 5 msec produced approximately a fivefold increase in the value of Δf required for discrimination over that required for discrimination for the other three target-interference tone temporal orders.
Acoustics as applied to petroleum and mineral exploration
Authors: Luehrmann W. H.
Published: October 2005

A review of acoustic techniques for exploration of petroleum and minerals is presented. The suitability of various transducer systems is discussed with respect to data acquisition, processing, and analysis.
Effect on fundamental frequency contours of modality operators
Authors: Allen J. , O'Shaughnessy D.
Published: October 2005

In a previous paper [J. Allen and D. O'Shaughnessy, “Fundamental Frequency Contours of Auxiliary Phrases in English” J. Acoust. Soc. Am. 56, S32(A) (1974)] the authors described effects on fundamental frequency contours which arise due to “modality operators” such as negatives, past tense modals, sentential adverbs, and subject quantifiers. In the present paper, the corpus is extended to examine these effects more thoroughly, and to introduce additional discourse-level constraints such as sentence embedding, presupposition, and focus. Results are presented which show the utility of fundamental frequency contours for establishing the “discourse thread” of a text, and which contribute to the formulation of a comprehensive model for fundamental frequency description.
Applications of nonlinear smoothing to speech processing
Authors: Rabiner L. R. , Sambur M. R. , Schmidt C. E.
Published: October 2005

In this talk a simple nonlinear smoothing algorithm which is appropriate for many speech-processing applications is described and evaluated. In many applications linear smoothers, or low-pass filters are entirely adequate. However, simple linear smoothers generally fail to provide adequate smoothing for data which exhibit both local roughness and sharp discontinuities. The proposed nonlinear smoothing algorithm can effectively smooth such data by using a combination of median smoothing routines and linear filtering. The concept of double smoothing as a refinement on the smoothing algorithm is also discussed. Examples showing the application of the nonlinear smoothing methods to typical speech parameters are included in this talk.
Fine structure of the sound-speed profile in ocean bottom sediments from in situ measurements
Authors: Shirley D. J.
Published: October 2005

Compressional-wave sound-speed profiles are obtained in situ in marine sediments using a profilometer which mounts on standard sediment corers. The measurements are made at 200 kHz as the corer dynamically penetrates the bottom. The data are recorded on magnetic tape in a waterproof package, allowing measurements in water depths up to 5 km. Profiles measured in the equatorial Atlantic Ocean off Brazil show significant fine structure in the sediment section, including both high-speed sandy layers and low-speed, high-porosity layers. Signal levels measured simultaneously with the sound speed indicate that attenuating sediments are associated with many of the layers indicated by the sound-speed profile. An anomolous high-strength, high-porosity layer was observed in one area at 6-m depth in the sediment. [This work was sponsored by the Office of Naval Research, Code 483.]
Turbulent boundary-layer noise in the interior of aircraft operating at varying altitudes
Authors: Bray Don E.
Published: October 2005

The effect of altitude on the interior noise level of commercial aircraft has been studied. Changes in noise levels predicted by several authors are compared to data observed on regularly scheduled flights. A constant velocity was assumed. Data collected at front seating positions are generally seen to vary in proportion to the first power of the density change with altitude. At midsection and rear seating positions, the noise level appears to vary according to the density changes squared.
Influence of multitube mixer nozzle geometry on CTOL-OTW jet noise shielding
Authors: von Glahn U. , Groesbeck D.
Published: October 2005

Acoustic shielding benefits for CTOL over-the-wing (OTW) applications were obtained experimentally with various multitube nozzles using a simple board to represent a wing. Six nozzles consisting of two to thirteen 0.93-in.-diam tubes were tested. Shielding surface lengths of 5.9 to 21.4 in. were used with each nozzle. In addition, the effect of spacing between tubes on jet noise shielding was studied using a six-tube nozzle. Farfield noise data were obtained at 90° from the jet axis and with a jet exhaust velocity of 660 ft/sec. The jet noise shielding benefits obtained with the various nozzle-surface configurations are discussed in terms of sound-pressure-level spectra. These data are then correlated in terms of a parameter that includes consideration of the number of tubes and the local peak velocity in the flow field near the trailing edge of the shielding surface. In addition, the sound-pressure-level amplification trends caused by jet-surface interaction noise at low frequencies are discussed for the various configurations.
Traffic noise propagation in city streets
Authors: Donavan P. R. , Lyon R. H.
Published: October 2005

At the 88th Meeting of the Acoustical Society, procedures were presented by the authors with which sound levels in typical city street geometries can be predicted. Using the methods underlying these procedures, a number of results have been obtained which apply generally to urban propagation of noise from flowing traffic and individual point sources. One set of results were the relationships between sound level down a street and variation of street width ratios. Differences in propagation into a side street from flowing traffic for four-way and tee intersections have also been established. Additional results have been obtained which indicate the variation in sound level across a street near primary and secondary street intersections. Another problem which has been considered is that of the noise associated with the start-up of traffic at intersections containing control devices. This problem was examined in terms of both traffic density and traffic control device cycle time. [Work supported by the National Science Foundation.]
Optimal linear interpolation in linear predictive vocoders
Authors: Viswanathan R. , Makhoul John , Russell William
Published: October 2005

In low-bit-rate linear predictive (LP) vocoders, the process of parameter interpolation at the receiver helps in smoothing the roughness in the synthesized speech which is normally associated with infrequent parameter updating. Simple linear interpolation has been used almost exclusively in LP vocoder systems. We propose an optimal linear interpolation scheme that requires the transmission of an extra parameter per frame, α : 0 ⩽ α ⩽ 1. The value of α determines that point along the line used for linear interpolation which minimizes the mean-square error with respect to the actual parameter values at the interpolation point. This, of course, requires the computation of the actual parameter values at the transmitter, but only the value of α need be transmitted. The transmission of α requires on the order of 150 bits/second. Experimental results have shown the proposed interpolation scheme to be superior to simple linear interpolation, especially during rapid transitions in the speech signal.
Voice-excited predictive coding system for low-bit-rate transmission of speech
Authors: Atal B. S. , Stover V.
Published: October 2005

Voice-excited speech analysis synthesis systems eliminate two of the most difficult tasks in speech analysis, namely, voiced-unvoiced decision and pitch analysis. They can therefore perform better than pitch-excited systems under practical speaking conditions. However, the transmission rates for voice-excited systems are usually much higher than for pitch-excited systems, owing to the additional bits needed for encoding the baseband. This disadvantage can be overcome by efficient encoding of the baseband. This paper presents a voice-excited predictive coding method for speech transmission at 2400 to 3600 bps. The analytic signal corresponding to a baseband of 300 to 900 Hz is square-rooted to reduce its bandwidth to 300 Hz and is encoded at 1200 bps by predictive coding. At the receiver, the wide-band excitation is derived by nonlinear distortion and inverse-filtering of the baseband. The speech signal is sampled at 6.67 kHz, and its spectral-envelope information is transmitted every 20 msec by 10 area parameters obtained by the linear prediction method. These parameters and the rms value of the speech signal are encoded at 1800 bps, yielding a transmission rate of 3000 bps for the speech signal.
Triple-function voice coder (TRIVOC)
Authors: Roberts J. E. , Smith C. P. , Wiggins R. H.
Published: October 2005

The first-formant region of the speech spectrum must be characterized with a high degree of accuracy in order to attain good speech quality and naturalness in a narrow-band speech processing system. This consideration led to the concept of using linear predictive coding for the first-formant region, vocoder channels for the upper spectrum, and detection and coding of the excitation function, to establish a narrow-band voice digitizer. This concept has advantages in the tradeoff of speech quality versus data rate, in comparison with the use of either channel vocoding or linear prediction alone. Such a triple-function voice coder (TRIVOC) has been implemented and investigated using a CSP-30 Digital Signal Processor. Processing algorithms have been established to permit comparison of various combinations of baseband and vocoder channels, LPC coefficients, and data rates ranging from 2400 to 4800 bits per second. Subjective results have confirmed the superior speech quality attainable by this method. Several versions of TRIVOC processing are presented, together with some results of subjective assessment of performance.
Automatic recognition of liquids and glides
Authors: Kashyap R. L. , Mittal M. C.
Published: October 2005

This study describes a new algorithm to recognize liquids (/l/, /r/) and glides (/w/, /y/) spoken by a number of speakers. To make the recognition problem more general, /i/ and /u/, the closest vowels to the glides, are also included in the study. We consider the sequence of time samples of each phoneme as an auto-regressive (AR) process of order r. These r coefficients (estimated by a least-square technique) and signal-to-noise ratio of this process will be taken as raw features for the phoneme under consideration. These raw features contain most of the information (speaker-dependent, speaker-independent, and noise, etc. ) that was contained in the phoneme. Clearly, only the speaker-independent information is required to carry out the above task. To extract the speaker-independent information and reduce the dimension of the raw feature space, we perform a linear transformation. The resulting features (in the lower-dimensional space) are then used for classifying the phonemes by a decision theoretic method, namely, the nearest-mean classification rule. Each of the six phonemes was spoken 10 times by each of the four speakers (two male and two female). We chose r as 8. The correct classification rate of about 85% was observed.
Interaural amplitude jnd's and frequency selectivity
Authors: Zurek P. M. , Leshowitz B.
Published: October 2005

Interaural amplitude jnd's were measured for conditions in which a pure tone of a fixed frequency, f0, was presented to one ear and a tone of frequency f0+Δf to the other. Interaural amplitude sensitivity was found to decrease as Δf was increased. This result suggests that the binaural interaction takes place within a limited frequency region. It was also found that the frequency region for which a binaural interaction is possible increased with signal frequency in a manner reminiscent of monaural critical-band findings. Finally, the range of Δf over which the task could be performed was considerably larger than that usually reported for binaural fusion of pure tones. For example, for f0=3000 Hz, an interaural amplitude jnd of less than 5 dB was obtained for Δf=1000 Hz.
Ohm's law: another look
Authors: Mills W. M. , Gilliom J. D.
Published: October 2005

In view of predictions of Ohm's acoustical law, this study explores the ability of the human observer to analyze multifrequency tonal configurations. The observer performed a monaural YES-NO detection task in which the signal frequency on each trial was randomly selected from a fixed set of eight frequencies. This frequency uncertainty could be reduced by utilizing a contralateral cue presented 700 msec prior to each observation interval. Cue types were (1) a single tone cue with the same frequency as the potential detection tone; (2) a three-frequency cue with the signal frequency as the middle component; and (3) a two-frequency envelope cue always bounding the signal frequency. When the cue consists of a single tone, detection performance is comparable to that obtained with fixed signal frequency. Performance is lower when the cue is a three-tone complex, containing the relevant frequency. Removing the relevant frequency from the cue reduces performance still further. The human observer appears to have the ability to hear out components of a tonal complex, although that ability is something less than perfect.
Ultrasonic wave propagation in animal bones
Authors: Adler Laszlo , Cook K. V.
Published: October 2005

Several experiments using ultrasonic wave propagation in freshly frozen dog tibia were carried out. Longitudinal sound-velocity measurements indicate significant directional variations in its value which may be due to structural variation of the biphasic material. The presence of shear waves in bone has been demonstrated by using a contact shear transducer. The shear velocity in the dog tibia is 1.75 × 105 cm/sec, while the longitudinal velocity is between 3.21 × 105 cm/sec and 3.62 × 105 cm/sec, depending on the direction of propagation. Both fractured and unfractured bones were used in the experiment. It has been demonstrated that echoes reflecting from the interface of the fracture can be used to detect fractures in bones. [Research sponsored by the U. S. Atomic Energy Commission under contract with the Union Carbide Corporation and aided by Grant No, In-89F from the American Cancer Society.]
Stress-induced changes in the velocity of guided waves
Authors: Egle Davis M. , Bray Don E.
Published: October 2005

This paper describes a series of experiments to evaluate the stress-induced changes in the wave speed (acousto-elastic effect) of guided waves. Pulsed ultrasonic waves having frequencies ranging from 0.1 to 5 MHz are introduced through variable wedges into rectangular bars machined from the head of railroad rail. The bars are stressed axially, in tension or compression, causing changes in the velocity of propagation of the various wave modes. The speed changes are measured by a pulse-overlay technique. Application to the measurement of longitudinal stress in rail is discussed.
Flow noise and self-excitation of pipes
Authors: Singhal Vijay K.
Published: October 2005

Measurement of flow noise in smooth pipes indicates that it behaves as a monopole at very low flow rates. As the flow velocity is increased, the sound pressure level varies as the sixth power of velocity and then as the eighth power at higher velocities. Eventually, choking of acoustic energy occurs in the upstream direction and saturation due to nonlinearity occurs in the downstream direction. The excitation of acoustic modes in the pipe in the presence of obstacles that shed vorticity leads to intense sound radiation. Some experiments conducted at high subsonic flow in circular pipes with crosswise cylindrical obstructions are described. The coupling of acoustic modes of a side branch resonator with the axial modes of the pipe is explored. Observations on how to make a nonresonating side branch cavity are presented. [This work was supported by the U. S. Navy (Office of Naval Research) under Contract N00014-67-A-0204-0019.]
Determining tongue body motion from the acoustic speechwave
Authors: Hafer Edward H. , Coker Cecil H.
Published: October 2005

A technique is described for estimating vocal-tract areas and articulatory parameters from speech sounds. Voiced sounds are analyzed by the covariance method to obtain predictor coefficients. Three or more resonance frequencies are determined; the bandwidth estimates are discarded. Corrections are made to account for inductive yielding walls of the vocal tract. Using a gradient technique, parameters of an articulatory model are manipulated to obtain best fits to the analyzed data. This method avoids the usual tenuous assumptions about vocal-cord spectra and about the distribution of losses in the vocal tract. Results of applying the method to vowels and short words with velar consonants are presented.