bluuebunny/crossref_metadata_2025 huggingface dataset:
Simultaneous and time-staggered dichotic speech perception after total hemispherectomy: evidence of release from competition
Authors: Berlin C. I. , Berlin H. L. , Lowe-Bell S. S.
Published: October 2005

In normals, time-staggering a pair of dichotically presented CVs leads to a “lag effect.” After anterior temporal lobectomy, the lag effect disappears, but the lesion-ipsilateral ear still shows the effects of some residual competition by virtue of its continuing to perform at less than 100% intelligibility. By contrast, after total hemispherectomy, the lesion ipsilateral ear tends to perform at or near 100%, implying that no residual competition, either phonetic or acoustic, exists between channels at the central analyzer. The impact of this observation on the nature of the dichotic listening task and the lag effect will be discussed.
Word recall in aphasia
Authors: Kewley-Port Diane
Published: October 2005

Memory for sentences and strings of digits is notably impaired for asphasic patients, This experiment attempts to specify this deficit in terms of the concepts of short-term store (STS) and long-term store (LTS) as developed by Waugh and Norman [Psych. Rev. 72 (1965)]. Using their probe memory paradigm, word lists were tape recorded for presentation to patients with mild-to-moderate aphasia and the normal subjects. The lists consisted of 10 words followed by a tone and a probe word. The subject attempted to recall the word following the probe word on the list. Since differences in serial recall between ears have been reported for monaural presentation of word lists, listerners were presented word lists randomly to either the right or left ear. Results demonstrated that most asphasic subjects in comparison to normals have an impoverished STS function and no evidence of a LTS function. Equivocal results for recall from the right- and left-ear presentations are discussed with regard to dichotic listening experiments using brain-damaged subjects.
Progress report on measurement of acoustic energy density in enclosed spaces
Authors: Sepmeyer L. W. , Walker B. E.
Published: October 2005

Preliminary measurements of the acoustic energy density, given by E = 12ρv2 + 12(p2 / ρpc2) have been made for pure-tone excitation of a large reverberation room. Measurements at eight independent positions in the room show a variation of approximately half that encountered when measuring p2 at the same positions. Additionally, average values of E obtained are close to twice the average values obtained for 12(p2 / ρc2), as predicted, and values of the standard deviation of p2 are very close to the average value of p2, also as predicted. Although the method used for the preliminary measurements was slow and laborious, instrumentation is presently under development which will allow the measurements to be made rapidly and inexpensively. Details of this instrumentation and proposed applications are discussed, together with additional data.
Effect of moving diffuser on the variance of sound power radiated into the reverberation chamber
Authors: Tichy Jiri
Published: October 2005

The qualification procedure of reverberation chamber for the measurement of sound power at single frequencies as defined by ASHRAE Standard considers both the spatial variability of sound pressure squared and the variability of the radiated sound power as a function of frequency. Theoretical considerations under idealized conditions allow prediction of the corresponding variances. This paper presents the results of accurate measurements of both the variance of sound pressure squared and sound power as a function of frequency. Considerable deviation from the theoretically predicted values has been found. The extent of these deviations and the effect on the qualification procedure are discussed.
Practice rooms of the Harvard University Music Building: a case history
Authors: Crabtree John B.
Published: October 2005

In 1969, Harvard University began planning a large addition to its Music and Music Library Buildings. The addition contains some 30 teaching studios, 20 practice rooms, an electronic music studio, a large rehersal room, a small ensemble room, and a huge area of library storage space. Upon completion and occupancy of the building, all of its facilities were found to be in excellent acoustical condition, with one exception: the sound isolation characteristics of the practice rooms, which were, by any standards, a dismal failure. What went wrong? This question is answered by tracing the history of the project from its conceptual stages through its actual completion and occupancy, including objective comparisons between the design criteria and the as-built measurements. Some thoughts and comments concerning possible remedial measures are presented.
Method for determining sound power levels of large machinery
Authors: Angevine O. L. , O'Donnell R. F.
Published: October 2005

The procedure as outlined in ISO Recommendation R1680, Part II, was used to determine the octave-band sound power levels of a large machine using sound pressure level data recorded close to the machine. A 5500 BHP pipeline engine-compressor unit was selected for testing this method. Using a precision sound level meter and octave-band analyzer, sound pressure levels were recorded along paths around and over the machine. Mean sound pressure levels found, employing these data, were used to calculate sound power levels for the 31.5–16k Hz center-frequency octave bands. Power levels were also calculated using only the readings taken in the horizontal path around the unit. This was done in order to check the effect on the final results of omitting the data taken overhead, a difficult procedure for larger machinery. Both results were compared with values calculated using the Diehl Correction Factor method, which requires a different set of test data taken over two surfaces.
Ground reflection and standard procedures for measuring noise
Authors: Olson N. , Embleton T. F. W. , Piercy J. E.
Published: October 2005

The effect of interference between direct and ground reflected rays is examined for standard methods of measurement of the noise emitted by road vehicles, snowmobiles, and other devices powered by internal combustion engines, such as chainsaws and lawn mowers. It is shown that cancellation leads to particularly erratic measurements for accelerating large trucks where the propagation is over asphalt, and for all devices where the propagation is over a porous surface, such as grass or snow. For propagation over asphalt, placing the microphone on the asphalt surface eliminates these effects, best considered as transmission artifacts. No such simple cure is available over porous surfaces. Good agreement is achieved between experiment and theory.
Use of graphic level recorders as indicating instruments. I. Meeting the specifications of a sound level meter
Authors: Webster W. J. , Farinacci J. W.
Published: October 2005

The recent trend in environmental noise control has been toward regulations based on sound level measurements. Such regulations typically require the use of a sound level meter satisfying certain specifications of ANSI S1.4-1971. If another acoustical instrument, such as a graphic level recorder, is to be used in the enforcement of these regulations, it is essential that it meet the same performance specifications. The ability of two well-known graphic level recorders to the meet the “fast” and “slow” response characteristics for indicating instruments has been determined experimentally. For each combination of potentiometer dynamic range and chart paper width, it was necessary to find settings for the remaining controls which would result in the specified tone-burst response and overshoot. It was also necessary to test the resulting frequency response to insure it remained linear over the operating range of a sound level meter. In some cases it proved impossible to obtain the response of a Type 1, or precision sound level meter, but the less stringent Type 2 specifications generally could be met. However, a comparison of a graphic level recorder set to Type 1 specifications to a sound level meter of the same type disclosed a noticeable difference in response to tone bursts. This observation prompted a more comprehensive investigation of the performance of the graphic level recorder which is detailed in a companion paper.
Average sound level, sound exposure level, and noise dose
Authors: Young Robert W.
Published: October 2005

The practical enforcement of any specification concerning sound, whether for the sound output of a sonar or the insulation of a house against aircraft noise, is facilitated by the reporting of sound measurements in the fewest possible ways. The possible ways are those covered by definitions of terms and measurement procedures available in internationally accepted standards. Definitions are here offered for three recently developed quantities; means of calculating one from the others are described. Average sound level is a sound level that is typical of the various levels that exist at a certain place during a stated period of time; technically, average sound level is the level of a time-mean-square A-weighted sound pressure, unless some other rule of averaging is specified; it may be called equivalent sound level; the unit is the decibel. Sound exposure level is the level of sound accumulated during a stated time interval or event; unless another rule of integration is specified, sound exposure level is the level of the time integral of squared A-weighted sound pressure; the unit is the decibel. Noise dose is the duration of a specified constant sound level that would lead to the same sound exposure level as an actual time-varying sound; noise dose is expressed usually as a fraction or percentage of a specified duration, like eight hours.
Statistical correlates of the energy-equivalent sound level
Authors: Driscoll D. A. , Webster W. J. , Haag F. G. , Farinacci J. W.
Published: October 2005

The increased use of several energy-averaged single number measures of environmental noise (e.g., Leq, LDN, and LNP) suggests a careful comparison with the more easily obtained statistical quantities (e.g., L5, L10, etc.). In this report the relationship between statistical and energy averaged quantities is examined for several theoretical distributions and for several categories of environmental noise. For a normally distributed sample of sound levels, the L10 exceeds the Leq by as much as 3 dB for standard deviations up to 12 dB [H. Von Gierke, NTID 73.4 (1973)]. A similar pattern is seen for a large sample of construction noise in which the “standard deviation” was typically 5 dB; the L10 consistantly exceeded Leq by an average of 2 dB.
Effective community noise measurement and description without expensive audio-equipment or computer systems
Authors: Lane S. R.
Published: October 2005

A research program was conducted to develop rational criteria and quick and inexpensive methods for obtaining community noise indicators of sufficient accuracy and descriptiveness to provide a comparative data base for urban noise control planning. The need for expensive audio-equipment and computer systems for statistical analyses, and integration of noise-time functions to obtain LEQ values was eliminated by the development of effective measurement strategies and simple manual calculation procedures. The required data were obtained with good accuracy at a cost of $3 per noise indicator L50, LEQ, etc., for one-hour periods) as compared to the $8 to $26 cost per similar indicator for surveys by other investigators. In addition to inexpensive noise measurements, a much greater degree of public dissemination of community noise data in a simple and objective format coupled with information about the adverse socio-economic effects of noise is required. For a wide variety of data users, the most useful and meaningful community noise indicators are dBC, noise peak rate, and LEQ values. Perhaps with this type of data, individuals can decide for themselves what their noise exposure means and act accordingly on an informed basis.
Preliminary results of anechoic termination/flow diffuser tests for valve noise experiments
Authors: Karvelis A. V. , Reethof G.
Published: October 2005

The investigation of internal valve and pressure regulator noise mechanisms underway at the Noise Control Laboratory required the design of a downstream device that functioned as an anechoic termination as well as a flow diffuser. To simulate an infinite pipe, an anechoic termination was required that provided sufficient attenuation under actual flow conditions downstream of a model valve to prevent, on one hand, the setting up of standing waves in the test section downstream of the valve and, on the other hand, to assure that jet type exhaust noise from the termination itself is not propagating upstream into the test section. The results of tests of a 36-in. -long 6° half-angle conical device made up of many 3-in. -thick disks of fiberglass and a perforated sheet inner core are reported. By using a pulse generator driving a loudspeaker mounted upstream of the termination, the attenuation characteristics were obtained with no flow and with an inlet velocity up to 200 fps for frequencies from 500 to 4000 Hz.
Reversible changes in the electrical parameters of isolated frog skin induced by ultrasound
Authors: Coble Anna J. , Dunn Floyd
Published: October 2005

Isolated frog skin was irradiated with 1-MHz noncavitating focused ultrasound in the intensity range 1–100 W/cm2 in an adaptation of the Ussing chamber permitting continuous perfusion of the skin. The electrical parameters of the skin, viz., the membrane potential (MP) and the short circuit current (SCC), were measured continuously during sonication. The magnitudes of the ultrasonically induced changes in both parameters were proportional to the acoustic intensity, though the MP exhibited a decrease while the SCC increased in response to the sound. Further, the time course of the two responses differed in that the SCC increased continuously for exposures of 0.5 sec and longer, while the MP reached a maximum value within 0.5 sec of the onset of the sound and did not change further for the duration of the exposure. [Supported in part by NIH and in part by NSF.]
Acceleration of coagulation time by ultrasound
Authors: Williams A. R. , O'Brien W. D.
Published: October 2005

The clotting time of fresh human platelet-rich plasma (PRP) was decreased by 50% immediately following ultrasonic irradiation at intensities within the range 0.4–2.2 W/cm2. PRPs was obtained from adult blood anticoagulated with citrate and irradiated in a cylindrical glass chamber (diameter 1.1 cm, volume 1.5 ml) having acoustically transparent windows of Saran Wrap (®) at each end. This chamber was immersed in a water bath at 30°C and located on axis approximately 0.5 cm in front of a 2.54-cm-diam PZT4 1-MHz transducer, operating at its fundamental frequency. Coagulation times after recalcification were measured with a clinical fibrometer, and showed a marked reduction immediately after irradiation. Subsequent clotting measurements on the same irradiated sample throughout the next 20 min showed a time-dependent increase in clotting time, presumably due to uptake of clot-promoting factors by intact platelets. An increase in the intensity of the ultrasound decreased the rate at which these clotting factors were removed from the plasma, indicating that platelets are damaged during irradiation. Irradiation of PRP for 5 min at intensities greater than those above appears to cause irreversible platelet aggregation.
Consequences of parameter learning in experiments on immediate auditory memory
Authors: Watson C. S. , Wier C. C. , Wroton H. W. , Kelly W. J.
Published: October 2005

In a recent paper we described the decay of auditory memory for patterns consisting of ten 40-msec tones played in sequence. On each trial, one of 50 permutations of a ten-tone catalog was played, and then it was repeated either with or without a frequency increment (Δf) added to one of the ten tonal components. The interval between the two patterns was varied from 0.5 to 32 sec. The frequency increments became only slightly less detectible as the interstimulus interval was lengthened (just-detectible values of Δf increased by about 1/3). In a new experiment, the same highly trained listeners were able to make almost as precise discriminations without a standard present on each trial. When the patterns consisted of ten-tone sequences drawn from a 1000-tone catalog the listeners could no longer perform at better-than-chance levels in the remembered-standard conditions. The results begin to establish the limits of listeners' ability to learn the parameters of a catalog from which acoustic stimuli are sampled. [Supported by Grant No. NS03856, National Institute of Neurological Diseases and Stroke, Department of Health, Education and Welfare.]
Frequency discrimination in auditory patterns: effects of interpolated stimuli and decision activity
Authors: Wier C. C.
Published: October 2005

Frequency discrimination for the seventh tone in ten-tone sequences was examined in a same-different delayed comparison procedure with highly trained subjects. Performance for interstimulus intervals (ISIs) of 8 or 16 sec was compared for blank ISIs vs ISIs containing interpolated tonal stimuli. Unlike some recent reports, the level of performance observed for a blank ISI was not reduced by interpolated stimuli or an interpolated task. The parameters included similarity of the interpolated stimulus to the standard, presence or absence of a decision task, and placement of the interpolated stimuli and task in the ISI. The result suggests that the capacity for brief auditory memory for nonverbal material may be organized differently from that serving verbal memory. [This research was supported by NINDS Grant No. NS03856 and NIGMS Grant No. GM1900.]
Auditory attention as a major factor in the discrimination of tonal sequences
Authors: Watson C. S. , Kelly W. J. , Wroton H. W.
Published: October 2005

In two recent experiments we described the discriminability of acoustic patterns consisting of ten 40-msec tonal components played in sequence, as measured with a same-different psychophysical method. Listeners can readily discriminate two such patterns when the only difference is in the frequency of one of the ten tonal components, although values of the just-detectible Δf increase by as much as an order of magnitude for early components as compared to later ones and for lower as compared to higher frequency components. In a new procedure listeners attempted to match the patterns by controlling the frequency of a single tonal component. They were able to routinely adjust that frequency within ± 10 Hz of its value in a standard pattern (which was presented in alternating sequence with the pattern in which they controlled the frequency of one component), although ± 60- to 100-Hz values of Δf had been required for the same patterns to be discriminated in the same-different procedure. Possible interpretations of this result are that the adjustment procedure enables extremely precise focusing of auditory attention, and that the site of intrapattern interference is more centrally located than previously suspected. [Supported by Grant No. 03856, National Institute of Neurological Diseases and Stroke, Department of Health, Education and Welfare.]
Perception of “hidden” auditory figures
Authors: Pollack Irwin
Published: October 2005

Periodic pulse sequences were “hidden” amidst other periodic pulse sequences within auditory pulse trains. The generating rule was to emit a pulse for the occurrence of any one of up to ten prime counts of a 10-kHz pulse train. The absolute detection, the relative discrimination, and the identification of such sequences were examined for trains of up to 10 prime components. Two extreme hypotheses may be identified: a “masking” hypothesis, based upon the greater masking by lower auditory frequencies than by higher auditory frequencies; and an “extraction” hypothesis, in which periodic components are extracted in terms of the number of occurrences of each component. The former suggests that performance should be best with low-frequency information (large counts); the latter suggests that performance should be best with high-frequency (small counts). The data favor the latter hypothesis: performance is most closely related to the rank-ordering of the to-be-examined component within its presented series.
Binaural subjective tones and melodies without monaural familiarity cues
Authors: Kubovy M. , Cutting J. E. , McGuire R. M.
Published: October 2005

Julesz [Foundations of Cyclopean Perception (Chicago, 1971)] has produced the binocular perception of form by presenting two random-dot stereograms, neither of which contains monocular familiarity cues. Analogously, Cramer and Huggins [J. Acoust. Soc. Amer. 30, 413–417 (1958)] have produced a faint perception of pitch by presenting white noise to one ear and the same noise to the other, but with a narrow band of frequencies time-delayed. We have obtained a much stronger auditory analog of Julesz's so-called cyclopean perception by simultaneously playing an array of eight tones (which form an even-tempered scale) to both ears; when one tone is time-delayed relative to its counterpart in the opposite ear, it is clearly perceived. All 24 subjects reported hearing the sequence of tones we embedded in the noise, and most recognized it as the tune Daisy. When one input was presented to both ears, no subject reported hearing individual tones, let alone the melody. Just as Julesz's technique has helped disentangle peripheral and central processes in vision, the present technique should help unravel peripheral and central processes in audition, and serve as an efficient detector of stereophonic deficiencies.
Correlates of parallel wall detection for blinded veterans
Authors: De l'Aune W. R. , Needham W. , Scheel P. D. , Gillespie G. M.
Published: October 2005

A two-alternative forced-choice methodology was designed to assess the ability of subjects to determine in a nonvisual manner the presence of a wall on their right side. Blindfolded subjects, beginning at a variable starting point, were led down a straight corridor and were either stopped at a point with walls on both sides (“closed” condition) or at a point with an intersecting corridor at their right side (“open” condition). The subject was asked to respond “open” or “closed” and was then led back to the starting area. This procedure was repeated for 50 trials and P.C. scores were determined for 22 blinded veterans and 16 sighted high school students. P. C. scores ranged from 0.38 to 0.98, with the mean falling at 0.64. Physical acoustic analysis of the ambient noise present in the opened and closed situations through the use of an artificial head recording technique revealed differential intensity information present below 2 kHz, with no signals recordable above 4 kHz. Results from the psychophysical procedure were examined for correlation with pure-tone audiometric data, age, personality factors as measured by C. P. I. and M. M. P. I. scales, verbal IQ, and the degree of visual impairment. Training methodologies as well as implications for the older blind are discussed.
Detection of simple tunes
Authors: Creelman C. Douglas
Published: October 2005

Four-tone sequences making up simple tunes were used as signals for detection in noise, using the adaptive PEST psychophysical procedure. Thresholds (80% correct, two-alternative temporal forced-choice) showed that tunes are more difficult to detect than four repetitions of the components making up the tunes, and that transposition of a particular tune on a trial-by-trial basis to an unpredictable key adds further difficulty for detection. Trial-by-trial random selection of the sequence of notes, in another experimental condition, produced no greater difficulty in detection than a fixed order on every trial. It appears that “configuration,” or the intervals between notes in a tune, does not add information which observers are able to use for detection. [Supported by the Canadian National Research Council.]
Two-channel detection of the missing event
Authors: Gilliom J. D. , Mills W. M.
Published: October 2005

Continuing the investigation of two-channel processing mechanisms, this study explored the effect of defining a signal event as a gap in a continuous sinusoid. The two channels were differentiated spatially by earphone channel; temporally by use of contiguous, non-overlapping observation intervals; and spectrally by use of differential, channel signal frequencies. The observer's task was to make two channel-specific, yes-no decisions concerning the presence of signal events. Signal events in each channel were independently determined and occurred with a probability of 0.5. Three two-channel conditions were explored: a pulse condition where signal events were 100-msec sinusoids, a gap condition where signal events were 100-msec gaps in otherwise continuous sinusoids, and a mixed condition where signal events were defined as a gap in one channel but as a pulse in the other. Two-channel performance in each condition was evaluated relative to signal-channel performance (a task where signal events could only occur in one channel). In the gap detection conditions, approximately a 4-dB increase in level was required to equate performance with that of pulse detection conditions. With respect to signal-channel performance, two-channel decrements were greatest in the gap condition, less in the mixed condition, and least in the pulse condition. Two-channel performance in the pulse condition was similar to that obtained in equivalent conditions in earlier studies. Performance was much better in the pulse and gap conditions in the absence of a contralateral signal event. This effect was minimal or absent in the mixed condition.
Subcritical detection using computer-generated tones and noise
Authors: Rosenstein M.
Published: October 2005

Tones and variable-bandwidth noise generated by computer are used to determine the course of subcritical thresholds versus noise bandwidths. Computer usage permits control of parameters otherwise uncontrollable, namely, the mean intensity of individual noise presentations and the starting phase of tones re noise. Computer filtering also permits tests to range over noise bandwidths of 1.25 to 640 Hz, essentially encompassing the continuum from tone-on-tone discrimination to tone detection in wide-band noise. Though the two frozen noises used throughout the tests had identical presentation mean intensities, two significantly different threshold-noise bandwidth curves emerged, one fitted subcritically, by a straight line rising 3 dB per octave, the other by a curve exhibiting a pronounced hump between the 10- to 40-Hz noise bandwidth region. These results, which contradict the contention that energy ratios alone determine subcritical thresholds, were linked to two features of the noise, the differences in the concentrations of energy in the signal onset region and the presence of signal-like excursions of the envelope.
Speaker identification by long-term spectra under normal, stress, and disguise conditions
Authors: Hollien H. , Majewski W. , Hollien P.
Published: October 2005

The authors have developed a method of speaker identification based on long-term statistical measures of speech. That is, n-dimensional Euclidian distances between long-term speech spectra are calculated and used in the identification of speakers. In this experiment, the technique was evaluated in order to discover if it was resistant to the effects of variations in the mode of speech production and signal transmission. Three different speaking conditions were studied: (a) normal speech. (b) speech under stress-talkers were subjected to randomly distributed electric shocks while speaking, and (c) disguised speech-talkers were permitted to disguise their speech in any manner they chose except by whispering or the use of a foreign dialect. Moreover, the entire procedure was replicated for a restricted passband of 300–3500 Hz; this band is similar to that found in telephone transmissions. Speech samples were obtained from 25 adult American males who read Stevenson's “Apology for Idlers” under the three different experimental conditions. A portion of the tape-recorded reading was analyzed in 1/3-octave bands by means of a GR-1925 Multifilter and a GR-1926 Multichannel rms Detector; four speech samples of 32-sec duration were analyzed for each subject and experimental condition. Further processing of the spectral results (expressed in decibel levels for each frequency band) was carried out on an IBM 370/175 computer. The normalized data were used to obtain Euclidian distances; in turn, they were utilized to evaluate both intra- and interspeaker variations in the speech spectra for the different speaking and (parallel) passband conditions. The normalized mean values of the four subsamples produced by each speaker under all conditions constituted the set of reference samples, and two different sets of test samples were studied. The reference data were used to discriminate among the speakers in the normal speaking mode; they were then used in an attempt to identify the speakers in the stress and disguise conditions. The entire process was replicated for the passband condition. The results of the research relative to the normal mode procedure demonstrated a relatively high level of correct speaker identification (slightly over 90%); the correct identification level was reduced by nearly 20% for the passband mede. The identification levels for the stress condition were nearly as high as they were for the normal mode but the correct speaker identification level for disguise was little better than chance. Replication of the procedures for the passband condition resulted in a slight further degradation of the stress condition but a marked improvement with respect to disguise. It appears that this method can identify individuals from their speech reasonably well when they are speaking normally or under stress; it cannot do so, however, when they attempt to disguise their voices.
Active model for extraction of nasality and its perceptual evaluation
Authors: Takeuchi Shoji , Kasuya Hideki , Kido Ken'iti
Published: October 2005

A description of nasality in the acoustic domain and a model for the extraction of nasality are discussed from a perceptual point of view. Nasality is perceptually regarded as some kind of distortion against the vowel quality, and the acoustic correlate of nasality is assumed to be the degree of spectral deviation from the oral sound. As a functional model for the extraction of nasality, a kind of A-b-S model is assumed. In the model the trial oral spectra are synthesized internally, the input speech spectrum is compared with each of the trial spectra, and the minimum value of the matching errors is taken out as the parameter representing nasality after the best matching is obtained. The trial spectra are specified as those calculated from the equation for the spectral description of vowels given by Fant [“Acoustic Theory of Speech Production” (1960)]. The correspondences were examined between the parameters extracted in the model and the results of the identification test on nasality using the oral and nasal sounds in natural speech.
Estimation of the vocal-tract length from acoustic data
Authors: Wakita H.
Published: October 2005

There are several ways to estimate the vocal-tract length from acoustic data. The use of the higher formant frequencies is one of them, but this does not give satisfactory results. The use of the zeros and poles of the lip impedance [Paige and Zue, IEEE AU-18, 265–270 (1970)] is not quite accurate because of the approximation involved. A third choice which is the topic of this paper is to estimate the length using formant frequencies and bandwidths. In this case, the tract length is chosen such that the error criterion proposed by Paige and Zue is minimized. This was not an easy task until a simple transformation from given acoustic data into a unique tract shape was developed [Wakita, SCRL Monograph No. 9 (1972)]. Utilization of the first four formant frequencies and bandwidths resulted in errors of less than 5% in estimating the length when tested on the tract shapes obtained by Fant. The length information can be used for obtaining a more reasonable vocal-tract shape from speech sounds and for normalizing speech parameters among various speakers. [The Office of Naval Research supported this work.]
Time-domain pitch extraction of speech signals using a digital filter
Authors: Hess W. J.
Published: October 2005

An algorithm is described which derives pitch with respect to the beginning and end of individual pitch periods of speech signals in time domain using a nonrecursive digital filter. In a preliminary step, the higher formants are attenuated by a gap-function low-pass filter with a fixed first zero at about 1800 Hz. Analyzing the distribution of distances between adjacent zero crossings in this “pre-filtered” signal, the algorithm estimates the formant frequency F1. In the following step, a time-variant filter is applied whose zeros are situated around F1. The output signal of this filter shows a structure related to the waveform of the glottal voice source. Within a single pitch period, this “filtered signal” contains but one significant maximum which is marked a preliminary pitch period limitation. Since its position still depends on the structure of the original speech signal, the markers are shifted to the more reliable position of the first zero crossing which follows the significant maximum. To define the proper beginning and end of the pitch periods with correct phase, the markers are finally shifted by half a period of F1 against time. The algorithm is run on a digital computer and is being realized as a hardware device.
Procedure for time-domain segmentation of connected speech
Authors: Hess W. J.
Published: October 2005

The segmentation procedure subdivides the speech signal into discrete successive sections whose sequence permits localization of most of the spoken phonemes in connected speech. In two preliminary steps, the pauses and the voiceless sections are separated from the rest of the signal. Subsequently, the main segmentation procedure tries to describe the speed of the vocal-tract movement. Since, for an individual phoneme, the vocal tract attempts to realize the accompanying “target position” and to keep it for a certain time before approaching the next phoneme rather than moving in a constant way, sections with little change in vocal-tract position (“stationary” segments) can be separated from ones with major, faster changes (“dynamic” segments). A small number of crude speech signal parameters (such as level) have proved sufficient to run this algorithm. By analyzing the direction of change within the dynamic segments, these segments can be further subdivided into “monotonously transient” sections. Except for a slight adjustment of the parameter change thresholds used when defining a “major” change, the algorithm doesn't need an adaptation to an individual speaker.
Segmentation of speech into syllabic units
Authors: Mermelstein P. , Kuhn G. M.
Published: October 2005

An algorithm has been designed to detect boundaries between syllabic segments of the speech signal, segments that contain but one peak or syllabic phoneme. A loudness function is defined, roughly corresponding to the short-time signal energy, and this function exhibits alternating sequence of points of relatively major maxima and minima. No absolute energy criteria appear adequate to decide whether a given maximum in the function is in fact a syllabic peak. However, minima can be characterized as significant with reference to the magnitudes of neighboring maxima. The algorithm uses the difference between the magnitude of the convex hull of the loudness function and the function itself as a measure of significance and recursively subsegments to the total signal until a given threshold of difference is nowhere exceeded. The algorithm is used as the first stage of a sequential procedure for segmentation and phonetic labeling of speech for recognition applications. [Research supported by the Advanced Research Projects Agency, Department of Defense].
Syllable-mapping strategy for a continuous speech understanding system
Authors: Weeks R. V.
Published: October 2005

Assuming a speech understanding system that predicts words, a crucial part of such a system is that part which associates a given word with its acoustic manifestation. In the system implemented at the System Development Corporation, this process is called mapping. It can be viewed as a set of transformations between successive levels of representation, some conceptual and some real. Predictive mapping assumes a unit of mapping. Some systems use the obvious phone unit, while others map an entire word or phase. SDC has taken an intermediate approach; words are mapped a syllable at a time. Reasons for this choice are given, as well as a preliminary evaluation of its success.
Analysis and recognition of Korean vowels
Authors: Kim Bu-Il , Fujisaki Hiroya
Published: October 2005

As a preliminary study to the automatic recognition of spoken Korean, the eight Korean vowels, uttered in isolation by 20 native speakers, including adults and youths of both sex, are analyzed and recognized. The speech materials are sampled at 10 kHz with 11-bit accuracy and the most stationary segment of 50-msec duration is selected from each utterance and converted into the logarithmic power spectrum. Frequencies of the first four formants are then accurately determined by the method of “analysis-by-synthesis,” using a model containing five formants and a correction factor for higher formants, as well as variable factors representing source and radiation characteristics. These vowel data are then classified by an optimum decision tree using 28 linear decision functions, each determined for the optimum separation of a pair of vowels in the three-dimensional space of the first three formant frequencies, giving a recognition score of 93%.
Formant synthesis by rule in a speech understanding system
Authors: Klatt Dennis H.
Published: October 2005

A speech understanding system is a device that will accept and demonstrate an understanding of spoken English sentences within limited syntactic, semantic, and lexical domains. Naturally spoken sentences involve the recoding of phonetic segments by a process that is most easily described by a set of generative phonological rules, many of which are optional. Thus phonological processes such as assimilation, reduction, flapping, and deletion, etc., can be accounted for optimally by designing a speech understanding system component that does varification of an hypothesized word or word string. The effects of segmental coarticulation on formant motions and on other acoustic parameters can also be treated conveniently in a generative mode by incorporating a set of rules for formant synthesis into the word-varification component. A set of rules will be described that generates formant tracks from an abstract phonological description of an utterance. The output of the rules has been compared with formant data extracted from a 1000-segment paragraph read by a single speaker. The adequacy of this schema for multiple-speaker word varification is discussed.
DRAGON speech understanding system
Authors: Baker James K.
Published: October 2005

Through consistent use of a single theoretical model, the DRAGON system is implemented much more easily than comparable speech recognition systems. Speech is modeled as a hierarchy of probabilistic functions of Markov processes. Various sources of knowledge—acoustic, lexical, phonological, syntactic, and semantic—are represented in a conceptual hierarchy: each is a probabilistic function of the sources of knowledge which are deeper in the hierarchy. The knowledge is integrated into a network representing an overall Markov process. Recognition of a given utterance is then formulated as finding the path through this network which maximizes the probability of the given acoustic observations. Several efficient computational schemes exist for finding such a maximizing path.
Surface reflection of an underwater acoustic parametric source
Authors: Khosla R. C. , Eller Anthony I.
Published: October 2005

An acoustic parametric source was mounted underwater in such a way that its axis could be aimed horizontally or at some angle of elevation, heading upward to the surface above. Free-field measurement of the radiation pattern showed the typical absence of side lobes in the difference-frequency beam. However, reflection of the beam by a calm water-air surface caused severe broadening of the beam and reduction of the sound pressure level at the difference frequency, as compared with the direct path signal from source to hydrophone. Measurements of the received signal as a function of horizontal range and of azimuthal angle are presented. The parametric source was a USRD Type E8 transducer. It and the receiving hydrophone were located at the same depth of 0.47 m. The reflecting surface was located within the nearfield of the parametric source, and the beam disruption is attributed to phase reversal that occurs upon reflection. The results are consistent with results presented by Mellenbruch and Muir at the 86th Meeting of the Acoustical Society of America [J. Acoust. Soc. Am. 55, 429(A) (1974)].
Nearfield effects in a parametric transmitting array
Authors: Bartram James F. , Fugitt R. Payson
Published: October 2005

Experimental measurements were made of the secondary source level from a parametric transmitting array under the following conditions: relatively short distance from the transducer to the measuring point, low attenuation of the primary and secondary waves, and primary source level too low for shock formation. The resulting source levels were low compared to predictions based on current farfield theories. A purely geometrical theory has been derived, neglecting absorption and saturation due to shock formation. The theory is successful in two respects: it is in good agreement with the data, and it matches the Berktay-Leahy theory when the distance from the transducer to the measuring point is increased to the absorption distance.
Error analysis of the Doppler correction factor in underwater communications
Authors: Foote K. G.
Published: October 2005

One limitation to the attainment of high spectral resolution in underwater communications is the Doppler effect. In principle, this purely geometric, or kinematic, effect can be defined precisely and the attendant Doppler-generated frequency shift can be removed by suitable signal processing. In practice, however, uncertainties in knowledge of instantaneous position and velocity of transmitter and receiver always exist and constitute an instrinsic limitation to spectral resolution. An error analysis is presented here of the Doppler correction factor for underwater communication between the sonar systems of two mobile platforms whose instantaneous positions and velocities, including rotation and other accelerations, are known imprecisely. Doppler errors produced by inhomogeneities in flow structure and sound profile are ignored, but the uncertainty in knowledge of sound speed is recognized. Some simple expressions for rms and worst-case errors are derived for cases which are distinguished by the relative size of transmission time to a characteristic rotational period. An application to spectral resolution by mobile parametric receivers is mentioned.
Optimum array processing in noise fields with uncertain power
Authors: Adams S. L.
Published: October 2005

In practical array detection problems, the noise field statistics may vary spatially and temporally. In this paper, the problem of detecting a farfield signal source which generates a known signal field at an array aperture is considered. Space-time samples of the noise field are assumed to be zero-mean Gaussian random variables whose second moments are random variables. Bayes optimum receivers and their performance are derived for several different joint statistical characterizations of the second moments. The results are believed to be the first detection performance results, in terms of the ROC, for optimum array processors in uncertain noise power environments. It is shown that, for certain characterizations, the optimum processor in uncertain noise power either requires spatial processing which is not accomplished by a beamformer or requires spatial processing in addition to that provided by a beamformer. For noise power correlation which decreases rapidly as a function of spatial and temporal coordinates, the optimum nonlinearity is determined to be a limiter.
Measurements of low-velocity flow noise on pressure and pressure gradient hydrophones
Authors: Finger R. A. , Abbagnaro L. A. , Bauer B. B.
Published: October 2005

A circular flow tank facility has been developed which features extremely low acoustic and vibration ambient conditions. This facility is described, operating procedures associated with its use are presented, and limitations in the measurement procedures are mentioned. The flow facility has been used to measure the noise output of pressure and pressure gradient hydrophones during actual flow conditions and results of these tests are presented and discussed. [This work was sponsored in part by the Naval Air Development Center.]
Undersea volcano in the Marianas located by SOFAR
Authors: Northrop John
Published: October 2005

Transient underwater second signals were detected at widely spaced SOFAR depth hydrophones in the Pacific during 14 December 1973. The signals, which had sound pressure levels of up to 45 dB above background in the frequency band 10–100 Hz, were spaced from 15 sec to several minutes apart, and all came from the same source area. Preliminary SOFAR fixes obtained from selected transients plot south of Myojin volcano in the Mariana Island arc near 22°N, 145°E, and are believed due to submarine volcanic eruptions. [Supported by Office of Naval Research and Naval Ship Systems Command.]
Bandwidth and averaging time dependence of ambient noise amplitude statistics
Authors: Weinstein M. S. , Hecht R. J. , Mole L. A.
Published: October 2005

Two 914-day samples of ambient noise were processed and evaluated at 50 and 100 Hz as a function of four bandwidths and five averaging times. For 10-min averaging, the second moment of the sample population appears independent of bandwidth. For the 0.2-Hz bandwidth the noise level distribution is nearly log-normal for 10-sec averaging time, whereas for the 10-min averaging time a skewness toward the higher noise levels is apparent. Decorrelation times for 0.2-Hz bandwidth levels averaged for 1 min were observed to range from 1 to 246 min. With averaging time increased to 10 min, decorrelation times ranging from 20 to 280 min were observed. The results show that statistical parameters are a function of bandwidth, averaging time, and time. [Supported by ONR and ARPA.]