bluuebunny/crossref_metadata_2025 huggingface dataset:
Estimates of spatial coherence in multipath channels
Authors: Smith P. W.
Published: October 2005

Pure-tone response in a multipath channel, wherein by virtue of reflection or refraction many paths connect the source to any receiver point, is highly variable due to interference between the components transmitted along different paths. A statistical measure of these fluctuations is the spatial coherence—the averaged product of the complex response field evaluated at two spatially lagged points. Estimates of the coherence are readily formed by combining (i) a field description in which the average response “energy” is expressed as a density in wavenumber or propagation direction, and (ii) the hypothesis that the local properties of each component are those of a free plane wave. This procedure models the field as being locally homogeneous, but with statistical parameters that vary slowly as a consequence of refraction and attenuation. Analytical estimates of coherence have been made for several canonical problems of ocean acoustics: the isospeed channel with flat bottom; the bigradient duct with a limiting ray.
Acousto-optic moiré
Authors: Clark J. A.
Published: October 2005

Experimental observation of acousto-optic moiré phenomena is reported. The experiment employs optical phase gratings generated by high-frequency (0.5–2 MHz) acoustic tonebursts propagating in a transparent water tank. Two or more superposed gratings are observed by means of an optical, stroboscopic, scanning schlieren imaging system. Still and motion picture photographs of acoustic tonebursts and of moiré patterns associated with superposed tonebursts are presented and analyzed. Additive and subtractive moiré patterns are distinguished. It is shown that the subtractive moiré patterns propagate at speeds higher than that of the individual acoustic tonebursts, while the additive moiré patterns do not propagate. A moiré method of quantitative schlieren phase measurement of complicated acoustic fields is introduced, and an application to the visualization and analysis of scattering by underwater targets is indicated.
Rejection of sonar interference of finite bandwidth by amplitude shading of transducer arrays
Authors: Wang H. S. C.
Published: October 2005

Following the work reported earlier [H. S. C. Wang, J. Acoust. Soc. Am. 56, S18(A) (1974), The possibility of rejecting coherent interference of finite bandwidth by amplitude shading of a transducer array is investigated. Using the same transducer array immersed in the same distributed interference and ambient noise fields, it is demonstrated that, if the array shading is optimized by assuming all the interference power to be concentrated at its band center, interference of bandwidths up to 10% of its center frequency fc and intensity 70 dB above ambient noise field can be effectively rejected. This indicates that the technique reported earlier is adequate for applications in narrow-band receiving systems. If a more elaborate formulation is used in the process of shading optimization, where the array interference power is expressed in terms of certain sine integrals of “weighted arrival time differences,” the above interference-rejecting capability can be extended to a bandwidth equal to 0.4fc. At moderately strong field strength (30 dB above ambient noise field), broad-band interference of bandwidth equal to fc may be reduced to the level of ambient noise by optimum shading. These results suggest that an optimum/adaptive broad-band beamformer probably needs to be amplitude shaded in only two to five bands.
Sound propagation over a complex impedance boundary
Authors: Donato R. J.
Published: October 2005

An expression is developed for the propagation of an acoustical wave over a boundary of complex impedance, e.g., the ground. The total field produced may be considered to be the sum of the direct wave from the source, a specularly reflected component from a perfect reflector, a modification produced by the departure of the boundary from infinite impedance, and a possible surface wave. The analysis involves splitting the incident spherical wave into a set of cylindrical waves with various angles of incidence and integrating these waves together with their reflected counterparts over the complex angle plane. The encirclement of a pole determines whether a surface wave exists. Calculations of the total acoustical field for different frequencies are made ewer a wide horizontal range, and its dependence upon height above the boundary of both the source and the receiver, and the impedance of the boundary, is discussed. Height above the boundary and the impedance of the boundary also affect the existence of the surface wave. Comparison with experimental results is made.
TTS from neighborhood aircraft noise
Authors: Ward W. Dixon
Published: October 2005

A series of tests designed to determine the effect on the auditory threshold of the worst “neighborhood” exposure to aircraft noise that can reasonably be expected has been conducted. Tape-recorded overpasses, both landings and takeoffs, were reproduced at 111 dBA (peak level) in a reverberant room. Six-hour exposures to (1) one landing every 3 min, (2) one landing every 90 sec, (3) one takeoff every 3 min, and (4) one takeoff every 90 sec were given to one of two groups of five normal listeners. TTS2 (the temporary threshold shift 2 min after an overpass) at 0.5, 1, 2, 2.8, 4, and 8 kHz was determined at hourly intervals. The mean TTS2 did not reach 5 dB at any frequency in any condition; the highest value was 3.9 dB at 4 kHz after condition 2 (90-sec-interval landings). Implications of these results to the auditory hazard of airport noise are discussed; clearly such hazard is ordinarily minuscule. [Research supported by the National Institute for Occupational Safety and Health, Public Health Service.]
Construction noise studies
Authors: Reynolds Ralph J.
Published: October 2005

General-Services-Administration-sponsored field studies of construction noise and traffic noise around site produce comparisons of interest to drafters of community noise management ordinances, to public agencies establishing construction noise criteria, and to construction workers. Construction noise limits should be established in relationship with background ambient noise levels. G.S.A. controls should be expanded to include noise produced by hand activity, to differentiate between outdoor and indoor environments, and to establish more appropriate measurement criteria. Construction noise management requires preplanning to establish achievable goals by a team including representatives of all concerned disciplines and a sound consultant. Construction noise generally represents greater hazard for construction workers than for the community.
Noise control in school auditoriums, gymnasiums, and multipurpose rooms
Authors: Benasutti Roger
Published: October 2005

Most multipurpose rooms in schools do not have adequate acoustical control. These rooms are often used as auditoriums, cafeterias, and gymnasiums. Acoustical consultants and architects in the past have usually attempted to control the reverberation time and absorption in these rooms via unit wall absorbers on the side walls. This has not always been practical, and often the unit absorbers are susceptible to damage or there just is not enough wall area to control the noise. A practical and economical way of controlling the reverberation time suitable for multipurpose rooms has been developed. It incorporates an impact-resistant acoustical ceiling system for these types of rooms and other high-abuse areas. This system combines lighting, aesthetics, and acoustical control; yet allows access to the plenum area. Reverberation time can be controlled so as to provide a better acoustical environment for multipurpose rooms. This paper discusses details of the components used for making up this system and the acoustical benefits realized.
Optical processing of scattered ultrasonic fields
Authors: Greguss P.
Published: October 2005

In the study of ultrasonic scattering from inhomogeneous media, most of the efforts have been devoted to determine the angular and spectral distribution of scattered flux. Little is known about the spectra of the index of acoustic refraction fluctuations in the material itself. It was found that the ratio of the mean-square fluctuation of the index of acoustic refraction and the smallest characteristic length associated with the index of refraction variations bears upon the effective diameter of the beam after it has travelled a given distance in a random medium. The scattered field was investigated by using a recently developed real-time acoustical-to-optical conversion cell (AOCC). Since a laser can be used for visualizing the scattered ultrasonic field, conventional optical data-processing methods, including optical Fourier transforms, can be applied to evaluate the ultrasonic scattering properties of the medium in question.
Subjective response of individuals to vibration directed into the hand
Authors: Reynolds D. D. , Standlee K. , Keith R.
Published: October 2005

This paper presents the results of tests conducted to determine the subjective response of human test subjects to vibration directed into the hand by means of a small diameter vibrating handle. For these tests individuals were exposed to a reference vibration level at 100 Hz and then were asked to adjust the vibration levels at other test frequencies until the same sensation as was perceived due to the reference vibration was produced. This resulted in equal sensation curves for the reference frequency and levels that were used. The effects of the selection of reference frequency, of the area of the hand into which the vibration was directed, and of the manner in which the vibrating handle was clasped upon the resulting equal sensation curves were discussed. A possible explanation of the physiological characteristics of the sensory receptors in the hand and arm and how they interact to produce the measured equal sensation curves is presented.
Use of the visual speech training aid (VSTA) in speech training for the deaf
Authors: Larkin W. D. , Houde R. A. , Stewart L. C.
Published: October 2005

The possible roles played by instrumental speech displays in speech training were examined to determine the reasons for their apparent ineffectiveness tn this application. Individual speech training for the deaf is viewed as consisting of two components: (a) the instruction component, in which the teacher elicits an approximation to some element of speech production, and (b) the practice component, in which the student makes repeated attempts at the target production. The principal shortcoming of current practice in speech training for the deaf is its inability to provide all deaf children with a sufficient quantity of practice. The principal role of feedback instrumentation is seen to be that of serving the practice component, relatively independent of the teacher's attention. The VSTA, developed to explore this approach, is described. Assessment of its ability to provide independent practice in training in prosodic features and stop consonant production has been carried out and is reported.
Student experiments on acoustics and the physics of music
Authors: Edge R. D.
Published: October 2005

Many universities now offer a course showing the relationship of physics to music and art. Frequently, the laboratory experiments associated with the course are of the pendulum and monochord variety, inherited from the education of physics majors, and this approach does not appeal to music students. To design a series of more stimulating experiments, a two-week workshop, supported by the National Science Foundation, was held in June 1974 at the University of South Carolina. Eighteen participants and five faculty members designed a series of 24 experiments on sound. These covered a great variety of topics at different levels of sophistication. Vibrations in musical instruments such as the guitar, violin, and trumpet were employed, rather than the purely “physical” equipment generally used previously. In addition, simple experiments on sound levels and reverberation times of rooms, noise, and various psychoacoustical experiments were devised, together with more advanced experiments on resonance modes of objects such as violin plates and heavy strings. Several of the experiments have proved very successful when tested in class during the fall semester. A manual of these experiments has been written, and is available to those interested.
On the use of linear prediction error energy for speech and speaker recognition
Authors: Wakita Hisashi
Published: October 2005

The error energy of the linear prediction inverse filter seems to give an effective triter ion for speech and speaker recognition. In this task, reference sounds are stored in terms of their filter coefficients and the error energy of their optimal inverse filters. The error energy of an unknown sound from each reference filter is compared with the reference error energy (Method A). Alternately, the error energy of the optimal inverse filter for an unknown sound is compared with the error energy of the same sound from the reference filters (Method B). Method A has its advantage in requiring only the autocorrelation function of unknown sounds. The recognition task was conducted with ten American vowels in the context of “hVd” produced by nine male and seven female speakers. A single speaker system averaged 99% recognition, whereas a multispeaker system in which one of the speakers was chosen as the reference averaged 46%. A proper threshold for the criterion suggested that the method is effective for speaker recognition. In general, Method A was slightly inferior to Method B. This work is part of a study of the effectiveness of inter-speaker normalization. [The work was supported by the Office of Naval Research.]
Effects of spectral composition on the detectability of pulse-train signals
Authors: Pohlmann Lawrence D. , Sorkin Robert D.
Published: October 2005

The detectability of a pulse-train signal as a function of its spectral composition was investigated. A two-interval forced-choice detection task was employed with the level of the masking noise controlled by a PEST program. In all conditions, the signal was a pulse train composed of 60 100-μsec positive or negative polarity pulses with an interpulse period of 2 msec. Polarity manipulations and filtering were used to vary both the overall bandwidth of the signal and the distribution of energy within that bandwidth. Conditions were run in which pulse polarity was a random variable or was varied according to some fixed (periodic) pattern. A variety of low-pass and bandpass filter settings were used. Both monaural (NmSm) and binaural (N0Sm) conditions were run. Detectabilities were found to be a function of both the overall bandwidth and the spectral distribution within that bandwidth. Results indicated that although pulse-train signals are broad-band signals, it is the low-frequency regions that are most important for detection. The relative importance of particular frequency regions was found to vary with the spectral distribution and with the mode (monaural or binaural) of presentation. Relationships to previous critical band and multicomponent signal detection research are discussed.
Frequency discrimination of complex periodic tones
Authors: Feth L. L. , Wales J. G.
Published: October 2005

Pairs of complementary two-tone complex periodic signals produce distinct pitch differences if the constituent tones differ in amplitude. Previous work indicates that the discriminability of such signal pairs may be based upon a weighted average of the instantaneous frequency functions of these signals. Psychometric functions for two-tone discriminations are not monotonic. Listener performance falls toward chance beyond a critical frequency separation between the constituent tones. Using this critical separation, we have defined the resolution bandwidth of the auditory system for center frequencies from 250 to 4000 Hz.
Acoustical logging of oil wells
Authors: Rader D.
Published: October 2005

A classical technique for determining the mechanical properties of geological formations penetrated by oil-well boreholes is based upon acoustical measurements. In acoustical logging, a transmitter-receiver system is packaged in a tubular probe which is pulled up along the length of the well by means of a cable which also serves as a data transmission line. In open hole, i.e., in freshly drilled wells, the acoustical measurement offers a means for determining the compressional wave speeds in formation beds as well as other formation properties which can be obtained by computer processing of the received waveforms. In the completed well, after placement of steel casing, an acoustical log can be used to evaluate the quality of the cement bend between casing and formation or to “see through” casing for determination of formation properties when the open-hole data are not available. In the present work, an experimental model study has been performed in which the data have been acquired and processed with a minicomputer. With these facilities it has been possible, for the first time, to elucidate graphically the essential features of the borehole acoustical problem. In particular, the presence of the compressional, pseudo-Rayleigh, Stoneley, and reflected propagation modes are demonstrated and simultaneously displayed in variable-density format. The relevance of acoustical logging to the detection of gaseous and liquid hydrocarbons trapped in porous formations is also discussed.
Speech resynthesis from phoneme-related parameters
Authors: Olive Joseph P. , Spiekenagel N.
Published: October 2005

In our recent work in speech analysis and resynthesis, we have been using predictor-derived area functions [B. S. Atal and S. L. Hanauer, “Speech Analysis and Synthesis by Linear Prediction of the Speech Wave,” J. Acoust. Soc. Am. 50, 637–655 (1971)] to describe the spectrum of the acoustic signal. It has been found that these parameters are insensitive to small changes. If the acoustic signal is indeed overspecified by these parameters, then bit reduction might be possible. Usually, bit reduction is achieved by specifying the data less often, or with less accuracy; however, we have chosen to reduce the bit rate in a way which is much more closely related to the speech-like nature of the signal being encoded. We have found that if the boundaries of the steady-state portion of the phonemes are found, these, as well as the transitions between the phonemes, can be represented by straight lines. This method allows for the description of the acoustic signal with two sets of area parameters per phoneme. Numerous sentences have been encoded by this method, and the resulting sentences do not sound different from the sentences from which the data were derived.
Some acoustic characteristics of stressed and reduced schwa
Authors: Pfeifer L. L. , Oshika B. T.
Published: October 2005

Approximately five minutes of conversational speech from an interview was transcribed and digitized. In the transcription the symbol /ə/ was used to note the midcentral vowel in both stressed and reduced positions. There were 275 vowels transcribed as schwa. The digitized speech is bandlimited at 5 kHz, sampled at 10 kHz, and quantized to 12 bits per sample. Each of the vowels which was transcribed as schwa is located by visual inspection of a time-synchronized display of the waveform, spectral peaks, and the rms of the signal. Formant parameters, derived from the inverse filter method of linear prediction, are then extracted from four consecutive spectral samples representing 29.6 msec of the steady state of each vowel. The characteristics of the vowel samples are examined in various acoustic spaces to determine if there is consistent evidence to discriminate between stressed and reduced schwa. This acoustic evidence is compared with a second transcription in which the transcriber makes a distinction between stressed and reduced schwa. Some of the acoustic characteristics of the vowel samples are presented and their relationships to perception are discussed. [This research is supported by the Air Force Office of Scientific Research under contract F44620-74-C-0034.]
Observation of the tongue movement by computer-controlled x-ray microbeam
Authors: Kiritani S. , Imagawa H. , Ito K. , Fujimura O.
Published: October 2005

A flying-spot-type x-ray beam generator with computer control [O. Fujimura et al., Computer Biol. Med. 3, 371 (1973)] is now in operation at the University of Tokyo. The new system employs a 150-kV acceleration voltage and an effective current of 1 mA for the electron beam which is deflected electromagnetically. Some preliminary data, primarily on Japanese utterances, are reported, together with some performance characteristics of the system. Three pellets on the tongue of one subject so far have been tracked successfully at a rate of 100 frames/sec. A typical exposure for each space-time sample is 100 μsec. Trajectories of the three pellets in movement can be monitored on the oscilloscope surface, and the time functions of the two-dimensional coordinates can be displayed. An overall scan of the image field of interest can also be obtained, and it can be displayed in a few different formats. This mode of observation is particularly useful in the initial estimation of pellet positions before an experimental session. The articulation of /k/ has been examined in different vowel contexts, and a systematic collection of data is being started.
Low-frequency seismic waves in fluid-saturated layered rocks
Authors: White J. E. , Mihailova Natasha , Lyakhovitsky Felix
Published: October 2005

Expressions are derived for speed and attenuation of compressional waves travelling normal to the boundaries of a laminated medium which consists of alternating layers of two fluid-saturated porous solids. Fluid flow across boundaries is taken into account through approximate expressions, applicable at low frequencies, derived through extension of the approach used by Gassmann. Calculations show that when one of the saturating fluids is gas, substantial dispersion and extreme attenuation may result.
Spatial coherence of sound radiated from sea quakes near the Mid-Atlantic Ridge
Authors: Porter R. P. , Leslie H. D. , Spindel R. C.
Published: October 2005

Sonobuoy arrays located over the Mid-Atlantic Ridge have been used to measure the spatial coherence of sound radiated from sea quakes. These measurements were made with array elements separated by several kilometers and tracked to an accuracy of 20 m. Spatial coherence, defined as the normalized, cross-power spectral density, has been found to be as high as 0.7 at 14 Hz, to be typically about 0.5 between 16 and 60 Hz, and to drop sharply to zero above 60 Hz. Short-range events, distinguished by time delays of the order of 0.1 sec across the array and by a rapid onset of the pulse, are presumed to leave the bottom at sharp angles. These events are significantly more coherent than long-range events that have little time delay across the aperture and whose onset is smeared out in time. Long-range events are presumed to exit from the bottom at shallow angles. Significant processing gain is possible using high-resolution spectral analysis. Improvements over the conventional delay and sum processor of as much as five to one have been observed. The resolving power obtained with these sonobuoy arrays is consistent with measured spatial coherence.
Behavior of volume strainmeters
Authors: Evertson D. W.
Published: October 2005

Earthstrain is usually measured with extensometers installed in tunnels or mines. A new strainmeter for installation in a 6-in.-diameter borehole has been developed. It measures volume strain over the frequency range of 0–1.0 Hz. The strain amplitude range is from less than 10−11 to 10−5. The developmental units are operating in Washington, D. C., and there is a network of three volume strainmeters gathering near-earthquake data near Matsushiro, Central Hoshu, Japan. In a free field, a volume strainmeter would not respond to deformation caused by shear waves. However, at a boundary SV energy is partially reflected as P energy, and hence induces volume strain. Horizontally polarized (SH) motion such as Love waves are not recorded. Coupling analysis is outlined, and strainmeter response as a function of emergence angle is given. The response to surface waves is given, and a primary noise source, fluctuating atmospheric pressure, is treated.
Sources of geoacoustic noise in the ocean
Authors: Northrop John
Published: October 2005

Very low-frequency (<50 Hz) underwater sound measurements made from fixed hydrophone arrays have revealed several major sources of geoacoustic noise in the deep ocean. For example, sounds from undersea volcanic eruptions are detected at transoceanic distances in the deep ocean sound (SOFAR) channel [Northrop, J. Acoust. Soc. Am. 56, 837–841 (1974)]. Underwater explosions cause about 20 000 hydroacoustic signals in the Pacific per year [Spiess, Northrop, and Werner, J. Acoust. Soc. Am. 43, 640–641 (1968)], and acoustic waves radiated from ocean margin earthquakes account for another 10 000 per year. Sounds from microearthquakes on the midoceanic ridges, however, generally are not well coupled with the SOFAR channel, but account for considerable local seismic noise in the vicinity of the ridges. Recent measurements in the eastern Pacific using sonobuoys, modified to record these very low-frequency sounds, have shown an average of one microearthquake per hour on the Rivera fracture zone at 19°15′N, 108°40′W, and 41 per hour on the Galapagos spreading center at 1 °N, 86°W. These sources of ambient noise in the ocean should be incorporated in environmental acoustic models. [Work Supported by Office of Naval Research.]
Analysis of ground reflection of jet noise obtained with various microphone arrays over an asphalt surface
Authors: Miles Jeffrey Hilton
Published: October 2005

Ground reflection effects on the propagation of jet noise over an asphalt surface are discussed for data obtained using a 13-in.-diameter nozzle mounted 1234 ft above grade. One-third-octave sound-pressure levels were measured for microphone heights from 0.02 to 1234 ft and at distances from 25 to 75 ft from the nozzle. Limited ground-effect data are also presented for a 2-in.-diameter jet. Analysis of ground reflection effects is made using the concept of a reflected ray transfer function in a model. The range of values of the single complex number used to define the reflected ray transfer function was larger than that expected if the range was determined only by the asphalt surface. This may indicate that the atmosphere is affecting the propagation of the reflected signal The model used as a basis for the Computer program minimized the ground reflection effects. Also discussed is the selective placement of the ground-effect reinforcements and cancellations in the planning of an experiment in order to minimize their effects.
Effects of intravehicular noise on the braking reflex in a simulated vehicle
Authors: Mergler-Racine D. , Bhéreur P. , Giguère D.
Published: October 2005

Intravehicular noise was recorded in a moving vehicle using two condenser microphones fixed on the driver's helmet and calibrated with a sound-level signal. This noise was reproduced through a calibrated sound system to subjects in a simulated vehicle. The simulator consisted of an actual car with a functioning speedometer controlled by the accelerator and brake pedals. During the three-hour-long experiments, the subjects were exposed to continuous intravehicular noise at either one of two mean intensities: 79 dB or 87 dB. They were required to maintain a fixed speed and brake quickly when a red traffic light lit up at random intervals between 26 and 114 sec. Microswitches on the pedals were linked to three digital counters, which provided the reaction times (RTs) of different phases of the braking reflex from the moment of stimulus to: (a) accelerator release (TA); (b) pressing the brake (TB); (c) full stop, zero mph (TS). At 79 dB the RTs remained constant or increased slightly throughout the experiment, whereas at 87 dB there was a continual and significant increase in the braking reflex time. However, the degree of increase differed for the various RTs which correspond to vigilance (T0−TA). transfer time (TA−TB) , and braking force (TB−TS). These results suggest that prolonged exposure to intravehicular noise of “normal” intensity can inhibit motor performance.
Acoustical analysts, testing, and design of flow-reversing muffler chambers
Authors: Young Cheng-I James , Crocker Malcolm J.
Published: October 2005

The transmission loss characteristics of flow-reversing muffler chambers were predicted by a numerical approach based on the finite-element method. The theoretical model developed is described in this paper; its validity is established experimentally with a number of different chambers. The standing-wave method was used to measure transmission loss and measurements were conducted with and without steady air flow. Depending on the inlet and outlet configurations, the flow-reversing chambers have two completely different transmission loss characteristics: one is similar to that for two simple expansion chambers in series and the other is similar to that for a side-branch resonator. Using the theoretical model developed, transmission loss curves were calculated for a large number of different flow-reversing chamber dimensions. A summary of the transmission loss characteristics of flow-reversing chambers is given for the assistance of muffler designers. [This work is a continuation of that reported by Young and Crocker in J. Acoust. Soc. Am. 54, 331(A) (1973)].
Effect of input cross spectra in estimation of frequency response and prediction in multiple-input, single-output systems
Authors: Seybert A. F. , Crocker M. J.
Published: October 2005

It is shown that under certain conditions cross spectra between correlated inputs may be neglected when estimating frequency response of multiple-input, single-output systems. When the input time histories are similar except for specified time lags, the above conditions exist, providing appropriate frequency smoothing is used. An additional requirement is that the magnitude of the spectra of the inputs should be constant, or vary periodically over the smoothing bandwidth. When these conditions are fulfilled, the frequency-smoothed input cross spectra will be considerably less than the input auto spectra and can be neglected in the spectral matrix. Neglecting input cross spectra can save experimental and computer time, since for n inputs, n(n − 1) cross spectra need to be measured. The disadvantage is in a loss of frequency resolution due to the frequency smoothing which forms the basis of the technique. However, in most estimation procedures, some frequency smoothing is needed to reduce the random error. Two examples are presented to illustrate the effect for a two-input and a six-input system. This work has been directed at measuring the frequency responses and predicting the noise of a six-cylinder diesel engine.
Preliminaries to the automatic recognition of speech: language identification
Authors: House A. S. , Neuberg E. P. , Wohlford R. E.
Published: October 2005

The goal of automatically understanding speech usually has been approached by first simplifying the task—by studying the speech of one talker, by restricting vocabulary, by considering only high-quality signals, by analyzing one language, etc. In addition, much experimental work has been characterized by an emphasis on either the speech-analysis or the pattern-recognition/statistical aspects of the problem. A new procedure is described for automatically determining the language of a connected-speech sample. The parameters extracted from the acoustic signal are a small number of gross linguistic categories that seem to be robust and reasonably independent of talker and channel, yet carry enough information to separate languages. Successful preliminary feasibility studies using manually transcribed parameters, as well as the performance of the procedures currently in use for automatic implementation of the scheme, are described. The effects of inaccuracies in labeling on the identification of a language are considered. Implications of these studies for the implementation of schemes for word identification and speech recognition are discussed.
Acoustic phonetic data base for the study of selected English consonants, consonant clusters, and vowels
Authors: Zue Victor W.
Published: October 2005

Improved understanding of the acoustics of speech events and recent advances in speech analysis techniques, combined with the availability of large and fast computers, have enabled speech researchers to examine and interpret a large corpus of acoustic data. These reasons provided motivation for constructing an acoustic phonetic data base where the acoustic characteristics of speech sounds can be studied in a detailed and systematic way. The data base includes a substantial number of nonsense CVC utterances (embedded in a carrier sentence) produced by several speakers. The consonants and the vowel are systematically varied. Using the acoustic analysis system that has been implemented, analog recordings can be digitized and processed (via DFT or linear prediction) in real time. The waveform, spectra, and hand-marked phonetic labels are stored on digital tapes for later referencing. During the presentation of this paper, the display and data analysis capabilities of the data-base will be demonstrated. Results on the relative durations of frication and aspiration of voiceless plosives, and results on rapid spectral changes at the release of stops will be presented.
Efficient LPC vocoder
Authors: Sambur Marvin R.
Published: October 2005

Of primary interest in any method for producing synthetic speech is to minimize the number of bits per second required to generate acceptable quality speech. An efficient method for transmitting the linear prediction parameters has been obtained by using the techniques of differential PCM. Using this technique, speech transmission employing fewer than 1000 bits/sec is achieved. Further reductions in the linear prediction storage requirements can be realized at a cost of higher system complexity by the transmission of the most significant eigenvectors of the parameters and a form of optimal pre-emphasis. These techniques in combination with differential PCM can lower the storage to 600 bits per second. Tape recordings of the generated synthetic speech will be played.
Speaker identification from a minimal set of training data
Authors: Pfeifer L. L.
Published: October 2005

Two different data bases have been prepared for the purpose of performing speaker identification experiments. The first data base consists of a set of 10 male talkers, speaking a list of well-controlled sentences. The second is a set of 20 talkers, 10 male and 10 female, speaking the rainbow passage. The experiments are carried out under what could be termed a “minimal information” basis, i.e., the speech utterances are bandlimited to 3200 Hz and there is only one training sample. Identification decisions are based upon the analysis of various sound units such as the vowels and nasals, and the use of a minimum-distance classifier. Speaker identification results are presented and discussed in terms of both open and closed-choice tests. Using the combination of both data bases, results are also presented on a text-independent experiment in which a 100% identification score was obtained for seven male speakers. [This research was supported by the Air Force Systems Command, Rome Air Development Center, Griffiss Air Force Base, New York, 13441.]
Line spectrum representation of linear predictor coefficients of speech signals
Authors: Itakura F.
Published: October 2005

It has been known that the linear predictor coefficients (LPC) of speech signals can be transformed into a “pseudo” vocal-tract area function whose boundary conditions are (a) a complete opening at the lips and (b) a matching resistance termination at the glottis. If the boundary condition at the glottis is replaced by a complete opening or a complete closure, all the poles of the resulting system function will move onto the unit circle in z plane. Using this fact it is possible to describe the original LPCs by two sets of pole frequencies corresponding to the two new boundary conditions at the glottis, or a set of frequency-residue pairs corresponding to either set of poles. These representations have several important features: (1) If an original pole is narrow band, the new pole is close to the original pole; (2) the two sets of pole frequencies alternate and are ordered on the frequency axis; and (3) the problem of locating complex poles can be reduced to solving a polynomial with real roots whose order is one-half that of the original all-pole transfer function.
Overshoot suppression in Adaptive Delta Modulation
Authors: Canas Miguel , Bradshaw Franklyn
Published: October 2005

Overshoots and signal uncorrelated noise are major sources of distortion and quality degradation of speech in Adaptive Delta Modulation (ADM) systems. In this paper a new technique is prepared which eliminates the overshoots and uncorrelated noise at the decoder. The technique uses past as well as future values of the ADM bit-stream to prevent the overshoots and oscillatory portions of the output signal. It is observed that the quality of the speech produced by this ADM receiver is improved for a wide range of the adaptation parameters. Also, it is observed that the out-of-band noise is eliminated to a level where minimal filtering is required for dealiasing in ADM-PCM conversion. Quality evaluation of the computer simulation is reported.
Stability of Adaptive Delta Modulation systems with memory
Authors: Canas Miguel , Bradshaw Franklyn
Published: October 2005

It is well known that multiplicative Adaptive Delta Modulation (ADM) systems with a 1-bit memory are unstable if the adaptation parameters P and Q are such that PQ > 1 [Jayant, “Adaptive Delta Modulation with a 1-Bit Memory,” BSTJ 48 (Feb. 1969)]. Furthermore, for this class of ADM systems it has been generally accepted that the maximum signal-to-noise ratio (SNR) occurs at PQ = 1 and P = 1.5. Subsequently, Zetterberg et al. have shown (1) that such systems can be made to perform stably with PQ > 1, and (ii) that the point at which maximum SNR occurs is dependent upon the class of the input signal [Zetterberg and Uddenfelt, “Adaptive Delta Modulation with Delayed Decision,” IEEE Trans. Comm. COM-22 (Sept. 1974)]. Effectively, Zetterberg uses a limited step size to stabilize the ADM system with a 1-bit memory and PQ > 1. In this paper it is demonstrated that this form of ADM implementation can lead to the loss of low-energy signals, rendering the technique impractical for speech encoding. Alternative techniques are proposed for implementing ADM systems which are stable across a wide range of values for PQ and P. For speech, the maximum SNR is observed for values of P > 2.
Spline encoding of speech
Authors: Arcese A.
Published: October 2005

In this paper we present the theory and application of cubic spline interpolation, which restores speech samples previously eliminated to lower the transmission rate. Spline interpolation gives us piecewise polynomials that are connected in a continuous manner. Spline interpolation may be performed on speech samples with data arbitrarily eliminated or on the speech zero crossings. Audio tapes will demonstrate speech processed with these algorithms.
Role of time-varying interaural time and interaural intensitive differences in the detection of tones
Authors: Grantham D. Wesley , Robinson Donald E.
Published: October 2005

Detection of a low-frequency tonal signal in the presence of various types of masker was studied using a two-alternative temporal forced-choice paradigm. The first experiment employed a tonal masker of the same frequency as the signal (500 Hz). The phase angle between signal and masker was either 0°, 45°, 90°, or 135°. The second experiment employed a tonal masker of 500 Hz and a tonal signal of 468 or 492 Hz. The third experiment employed an amplitude-modulated masker (carrier frequency = 500 Hz) and a tonal signal of 500 Hz; the phase angle between the signal and carrier component of the masker was again either 0°, 45°, 90°, or 135°. In all experiments, signal and masker were pulsed together for 128 msec, with a 10-msec rise-decay time. Masker stimuli were presented diotically at a level of 70 dB SPL. The signal tones were either identical in the two ears or phase-reversed in one ear relative to the other. The results are discussed in terms of previous tone-on-tone data and tone-in-noise data. Specifically, some conclusions are reached concerning the role of time-varying cues in binaural detection.
Binaural detection at high frequencies with differing masker bandwidths
Authors: McFadden Dennis , Pasanen Edward G. , Moffitt C. Michael , Clark Mark H.
Published: October 2005

Detectability of a 4000-Hz tone was measured using masker bandwidths of 50, 280, 550, and 1100 Hz, all centered at 4000 Hz. As bandwidth increased, detectability remained essentially the same in the N0-Sπ conditions, but it gradually improved in the N0-S0 conditions. Thus, the size of the MLD measured at high frequencies with a tonal signal and a noise masker is critically dependent upon the latter's bandwidth, and further, the basis for the variation is detectability in the non-MLD condition. Here the MLDs ranged from about 10–15 dB with the 50-Hz masker down to about 4–6 dB at 1100 Hz, the value most cited for high-frequency MLDs. [Supported by NINDS grant.]
Binaural detection of time-delayed signals at high frequencies
Authors: McFadden Dennis , Pasanen Edward G. , Clark Mark H. , Moffitt C. Michael
Published: October 2005

Detection performance was measured at 4000 Hz using for the signal a complex waveform that yields good lateralization performance when not masked (see McFadden, Pasanen, and Saltwick, paper S5, this meeting). The masker was a wide-band noise perfectly in phase interaurally. Psychometric functions were obtained with the signal either perfectly in phase interaurally (0-μsec time delay; N0-S0) and with it time delayed (N0-Sdelay). Detection performance in the latter conditions varies in a regular and periodic way as a function of time delay (250-μsec period), Just as did the data of Langford and Jeffress [J. Acoust. Soc. Am. 36, 1455–1458 (1964)] at 500 Hz (2-msec period). The latter authors interpreted their data as evidence for detection being based on cycle-by-cycle interaural time differences; the similar finding at high frequencies makes that interpretation questionable. [Supported by NINDS grant.]
Lateralization at high frequencies
Authors: McFadden Dennis , Pasanen Edward G. , Saltwick Steven E.
Published: October 2005

The ability to use interaural time differences to lateralize high-frequency sounds was measured using two types of complex waveform-noise bands of varying width and two-tone complexes with varying frequency separation. Signals were 200 msec in duration and were gated on slowly out of the quiet. The basic outcome was that lateralization was better the more rapid the envelope fluctuations (the wider the noise band or the greater the frequency separation in the two-tone complex). For the best-lateralized waveforms, only a 20–30-μsec difference of interaural time was necessary for 80% correct performance-approximately the value cited as the limit at low frequencies. [Supported by NINDS grant.]
Effect of training in sound localization in the median plane
Authors: Yorifuji Y. , Morimoto M. , Ando Y.
Published: October 2005

It has been reported that monaural subjects can localize the sound image correctly [L.B.W. Jongkees, Acta OtoLaryngologica 465 (1957)]. It is supposed that training is an important factor in localization. The purpose of this investigation is to determine the effect of training on localization in the median plane. When normally binaural subjects are made monaural by occluding one of the ears, the localization errors increase at the beginning and decrease after temporary training. In this case, subjects can localize the sound image by discriminating the timbre according to the head-related transfer function depending on the direction. Also, it is found that binaural and permanently monaural subjects can localize the sound image correctly even at the first test. It is considered that monaural subjects have the possibility of correct sound localization as the same process of language recognition, and that the localization of binaural subjects is caused by a resultant from long training by the small interaural difference in the median plane as an easier process.
Detectability of sinusoid signal as phase between sinusoid and noise microstructure is manipulated
Authors: Bell D. W. , Schubert E. D.
Published: October 2005

The relationship between the microstructure of 24 computer-generated noise waveforms and the phase of a 500-Hz sinusoid was manipulated. The sinusoid (signal) and the wide-band noise waveforms (maskers) were both 50 msec in duration and were concurrent. The onset phase of the sinusoid relative to the zero crossing was varied systemetically through four phases: 0°, 90°, 180°, or 270°. A 2IFC paradigm was used, with the same noise waveform being presented in both intervals. The stimulus was always presented in the N0S0 configuration. Following each trial, the observers received information about which interval had contained the signal. Approximately 300 trials/waveform/signal-masker phase configuration were obtained with each of four observers. Variation in detectability of the signal as the phase relationship with the masker was altered was dramatic and reasonably consistent among observers.
Internal filter characteristics for uncertain-frequency detection
Authors: Macmillan Neil A. , Schwartz Madelyn
Published: October 2005

Observers in a 2IFC detection task were instructed to listen for “primary” tones with one of two frequencies, in the presence of a broad-band background noise. On a small proportion of trials, probe tones of other frequencies were presented instead, following the procedure of G. Z. Greenberg and W. D. Larkin [J. Acoust. Soc. Am. 44, 1513–1523 (1968)]. The function relating percent correct to signal frequency may be interpreted as an average internal filter characteristic. All six observers responded correctly on 70%–90% of trials on which primary signals were presented; performance on probe trials was poorer, falling to chance for probe frequencies sufficiently above the higher primary, sufficiently below the lower primary, and for some probes between the two primaries. The widths of the listening bands around the primaries were found to be similar to the widths estimated from one-primary control conditions. Examination of the sequential data suggests that, while the pattern of sensitivity changes from trial to trial, a model postulating a single filter shifting from one primary to the other is not adequate. Other models of uncertain-frequency detection are discussed.