bluuebunny/crossref_metadata_2025 huggingface dataset:
Design of a new auditorium using electroacoustic technology
Authors: Yamaguchi Kiminori
Published: October 2005

In recent years, pops, jazz and rock'n'roll concerts have often been presented at many auditoriums, and musical programs performed have the tendency of using mainly electric musical instruments such as electric bass, electric guitar, electric piano, and music synthesizer. Accordingly, the sound reinforcement system design becomes as more important to improve the PA sound mixing and the design of room acoustics for the auditorium. This paper describes the example of the electroacoustic systems for the World Popular Song Festival in Tokyo at the Nippon Budokan Hall with 100 000 m3 and explains the idea of design using electroacoustic technology for the Exhibition Hall. In view of the control of sound field for the Exhibition Hall, new tools have been adopted such as (1) the movable proscenium loudspeaker, (2) movable sound absorbing canvas, (3) artificial reverberation adding system, (4) multichannel public address console and delay/reverberation control console, and (5) other various types of electrical and spatial reverberation adding devices.
The architectural acoustics design of Sherwood Hall, Salinas, California
Authors: Paoletti Dennis A.
Published: October 2005

This paper describes the architectural acoustics design of a recently completed multipurpose performing arts hall. The facility, known as Sherwood Hall, is part of the Salinas Community Center in Salinas, California (located approximately 100 miles south of San Francisco). In order of priority, the program requirements called first for a space to accommodate banquets, dances, and community activities, and then for a theater and music hall. The design response to these requirements yielded a space consisting of a platform-stage area with a fixed, flat main floor and a permanent balcony. Folding theater-type seats on movable risers stored under the balcony allow the flat floor to be converted to sloped audience seating. The most difficult challenge acoustically was to control reflected sound energy for speech-oriented events occurring on the flat floor and to encourage reflected sound from the stage to the audience for events in the theater/music configuration.
Recent progress in high fidelity sound reproduction system
Authors: Kawakami Hiroshi
Published: October 2005

While musical instruments (including human voice) are the tools for creating music, audio equipments are used for the reproduction of music. There is a common feature for both, that is, they presuppose the existence of original idea of music, and aim at the genuine expression of this idea. In the design process of audio equipments, it is necessary to fit the technique of equipment to the original idea of music. In order to do this, refinement is necessary both the “software technique” of recording, PA technique of real sound field or reproduced sound field of music, and the “hardware technique” of equipment design, component parts design, and sound field design. In this paper are reviewed new technical informations about audio equipments—from the musical input of pick up (cartridge) and microphone to the output of speaker and headphone—and characteristics of sound reproducing space. Besides, here is a discussion of philosophy (or aesthetics) of music reproduction, which may be included under the category of artistic thought.
Subjectively optimal conditions of sound fields for recording and reproducing
Authors: Ando Y.
Published: October 2005

Does the high fidelity sound reproduction system itself provides desired hearing conditions in a listening room? That is only a necessary condition, because sound fields in a recording room and/or a listening room affect on subjective preference. Results of subjective preference judgements of the synthesized sound fields reveal following optimal objectives, namely “Principle of Desire” of listeners [Y. Ando, J. Acoust. Soc. Am. 62, 1436–1441 (1977) and Y. Ando, 95th Meeting of Acoust. Soc. Am., Providence, paper JJI]: (1) Sound fields with a smaller value of the interaural crosscorrelation are always judged as preferred (Incoherence at the Two Ears). In order to obtain the small value of interaural cross correlation, the preferred direction of reflections is found in the range centered on ±55° in the azimuth to the listener. (2) The discrete reflection arriving after the direct sound is preferred, so that the initial time delay gap can be determined by the coherence of autocorrelation function of source signals and the level of reflections (Renewal of the Direct Sound). For example, a preferred scale factor for the dimensions of rooms exists for a given music source. Taking these points into consideration, it is possible to design a new sound system from recording to reproducing.
Loudspeaker measurements weighted by psychoacoustic modeling
Authors: Berkovitz Robert
Published: October 2005

Modern data acquisition methods permit comprehensive measurements of prototype loudspeakers in various acoustical environments. Digitization and subsequent transformations appear to offer new information about loudspeaker spectral output as a function of time. However, problems of interpretation exist, as with classical measurement methods, especially when the details of complex displays are considered. By convolving acquired data with models of psychoacoustic processes, subjectively important features of the data are disclosed with improved contrast.
Application of digital techniques to the high fidelity sound system
Authors: Itow Takeshi
Published: October 2005

PCM magnetic tape recorder is now in practical use for manufacturing high fidelity disk records. 14 bits PCM signal is recorded on VTR magnetic tape, and almost all distortions caused by ordinal magnetic tape sound recorders are eliminated. Circuit for fade-in. fade-out, time delay, and tone control to digitalized signals have been developed so that a mixing table for digitalized signals can be constructed. Impulse response of audio apparatus is measured precisely by averaging method, and we can obtain steady-state response, transient response, and phase characteristics of the apparatus. The same method is applied to rooms to obtain the reverberation characteristics, steady-state transmission characteristics, sound pressure distribution, and the locations of images of a sound source in the room with respect to reflecting walls. Methods for reducing PCM bit number without injuring tone qualities have been developed by using some statistical properties of musical signals.
Equalization for the dummy-head system
Authors: Fukudome Kimitoshi
Published: October 2005

In the dummy-head system, consisting of a dummy-head, head-phones, and a listener, an equalization method for precise reproduction of original sound is presented. We have three objects: (A) The quantitative expression of the sound pressure at the eardrum influenced by the human acoustic elements and the headphone. (B) The necessary equalization characteristics. (C) The realization of the equalizer irrespective of the direction of incident sound. In the analysis, the human acoustic system is divided into two parts at the ear canal entrance. Equivalent circuits for the listener's ear in the original sound field and for the one with headphones are derived. An expression of the necessary equalization characteristics for the listener is obtained in terms of elements of these equivalent circuits. The validity of the present method is verified by the experiment using a spherical head. Furthermore, variations between individuals about these elements should be investigated in order to make the precise reproduction for possibly many listeners.
Statistical analysis of error correcting schemes of stationary head digital audio recorders
Authors: Doi Toshitada , Tsuchiya Yoshikazu , Fukuda Joji
Published: October 2005

In spite of the great advance in performance, digital audio recorders have some critical points in handling and reliability resulted from the code errors These code errors are caused by (1) drop outs of tapes, by (2) peak, shift, jitter or noise, and by (3) finger prints, injuring tape edge, or dusts. (1) and (3) will cause rather burst errors and (2) will cause random errors. But, in effect, the burst and random errors cannot be separated discretely, and the trend is described by one parameter “bit error correlation coefficient.” In this paper, the changes of parameters of a statistical model of errors of magnetic tapes are discussed due to the increase of packing density. That is not only the increase of error rate but also the decrease of bit error correlation coefficient. Various error correcting schemes are evaluated by means of computer simulation for various values of error rate and bit error correlation coefficient. Another important point to increase the reliability of the systems is error interpolation ability when errors exceed the ability of error correcting schemes. These simulations will be helpful to determine the tape formats for digital audio recorders with stationary head.
Remote sensing of physical processes in the ocean using high frequency acoustic backscattering systems
Authors: Orr M. H. , Hess F. R.
Published: October 2005

High frequency acoustic backscattering systems are being used to remotely sense fluid processes in the oceanic environment. The systems are being used to detect and study short period internal waves, large scale Kelvin-Helmholtz instabilities, turbulence, interleaving water masses, frontal zones and variability of natural, and man introduced particle distributions. Graphic acoustic records will be presented demonstrating some of the results obtained to date. [Work supported by NSF, NORDA, and NOAA.]
Meterological timing interactions for linear based sound ranging
Authors: Enbegos B. F. , Miller W. B.
Published: October 2005

In a typical field artillery setting, sound ranging is the only available passive target acquisition device. This involves the use of six acoustic microphones placed in a near linear array. The accuracy of the location of enemy artillery depends on many factors. Two major contributors are the timing data obtained via these microphones and the ability to estimate existing meteorological conditions. In this study the interrelationships between timing and meteorological are investigated from both an analytic and a numerical point of view. It is shown that errors in temperature act as bias errors in timing and are nearly linear functions of the distance between the source and the midpoint of the array. Errors in wind however provides a more complicated situation.
An investigation of acoustical propagation under conditions resulting in the formation of a shadow
Authors: Van Moorhem W. K. , Butterworth J. R.
Published: October 2005

A comparison of theoretical and experimental results has been carried out for propagation into an acoustic shadow zone resulting from refraction caused by a thermal gradient. The experiments were carried out on a very large, dry mud flat under no wind conditions. The existing thermal gradient was measured as part of the procedure. The theoretical results follow closedly from Pekeris [C. L. Pekeris, J. Acoust. Soc. Am. 18, 259–315 (1946)] but include the effects of finite surface impedance and a logarithmic temperature profile. An assessment of the importance of turbulent scattering under these conditions is carried out. [Work supported by NSF.]
Accuracy in outdoor ground impedance measurements
Authors: Van Moorhem W. K. , Butterworth J. R.
Published: October 2005

The acoustical impedance of ground surfaces is deduced from data on the interference pattern developed between direct and reflected acoustic waves. Two geometrically different approaches to these measurements have been used, normal incidence to the ground surface and oblique incidence. The accuracy required in the geometrical and acoustical measurements of the interference pattern indicate a significant advantage in the oblique incidence approach for required accuracy in the surface impedance. Other effects present in the uncontrolled outdoor environment tend to complicate this simple result, however. Some simple analyses have been carried out to compare these two basic methods under various adverse conditions. [Work supported by NSF.]
Experimental studies of the acoustic signature of proton beams traversing fluid media
Authors: Sulak L.
Published: October 2005

Recent experiments at Brookhaven National Laboratory and Harvard University demonstrate that a detectable sonic signal is produced by energetic proton beams traversing fluids. The novel results are in agreement with the predictions of a thermal expansion model of the generation of the acoustic wave. They are inconsistent with any significant contributions from either microbubble implosion or molecular dissociation. Frequency and amplitude distributions, radiation patterns, and temperature, pressure, and medium dependences are explored. This phenomenon may have immediate applications as a beam monitor and as a detector of energetic heavy ions. Signal thresholds may be low enough to permit detection of particle showers induced by single particles at the next generation of high energy accelerators or from high energy cosmic rays. The inexpensive transducers and long sonic transmission lengths obtainable in liquids suggest that high energy particle detectors may be feasible with masses many orders of magnitude greater than those currently in use.
The signals and noises of DUMAND, and a proposed space-time processing scheme
Authors: Parvulescu Antares
Published: October 2005

The signals from neutrinos are expected to have a spatial signature (radial as well as directional) and a temporal signature (waveform) adequate for discimination against the ambient noise with a receiving system of finite cost well within the gross national product. The author's estimate of the signal space-time signature is presented together with the rather scant knowledge available at this time about the nature of the noises that would interfere with the detection. Since the desired signals might occur 300 times a year—give or take one order of magnitude—the noise statistics of importance in the system design are the extreme-value statistics, the tails of the distribution function, and the high-order moments rather than the usual Gaussian white noise approximations. In an aside, the very assumption that the noise can be treated as random (i.e., possessing a distribution) is questioned.
Acoustic radiation by charged nuclear particles in liquids
Authors: Learned John G.
Published: October 2005

An analysis of the propagation of acoustic pulses produced by local heating of liquids due to ionization by charged particles is presented. It is shown that the wave equations with loss dominate the pulse shape after small distances and that due to the bipolar, delta-function behavior of the individual pulses, a net observed pulse is simply the time derivative of the received density of pulses from individual heating centers. Angular distributions, signal to noise ratios, and detectable volumes are discussed. One important result is that the effect of attenuation in the acoustic medium is to produce a power law rather than an exponential cutoff at large distances. For example, in the thermal-noise-limited case, the signal to noise ratio as defined herein only steepens by 1/2 power in fall off with distance due to attenuation.
Effects of visual, tactile, and olfactory modalities on the threshold and magnitude of the acoustic reflex
Authors: Robinette M. S. , Snyder K. , McCandless G. A.
Published: October 2005

The purpose of this study was to evaluate the inhibitory effects of nonauditory stimuli on the acoustic reflex. Acoustic reflex threshold and magnitude (in millimho's) at four suprathreshold intensity levels were measured under several visual conditions: eyes closed in darkness, closed in light and open while focusing on a dot, reading, and solving a maze. Auditory stimuli consisted of white noise and a 1000 Hz puretone. Acoustic reflex thresholds and magnitude were less sensitive for visual attention tasks only under the white noise stimulus. These data will be compared with similar acoustic reflex measures obtained under conditions of tactile stimulation (hand vibrator) and olfactory stimulation (ammonia and vanilla inhalants). Comments will be made regarding mechanisms of acoustic reflex inhibitions elicited by the nonauditory modalities.
Growth and recovery of temporary threshold shifts from 24 hour continuous, 48 hour continuous, and 48 hour intermittent noise exposure
Authors: Stephenson M. R. , Nixon C. W. , Johnson D. L.
Published: October 2005

The effects on human hearing of 24 and 48 hour continuous and 48 hour intermittent exposures to pink noise at an Leq of 85 dB were measured by pure tone audiometry to address two questions. First, previous studies have demonstrated an asymptotic temporary threshold shift (ATTS), i.e., growth of TTS ceases, after 8–16 hours of exposure to sound. On the assumption that these effects on hearing are the same once asymptote is reached, there should be no difference between the growth of TTS, level of ATTS and recovery from the two exposure durations. Second, if the equal energy concept is appropriate to exposure durations of 48 hours then the effects found for the interrupted conditions should also demonstrate some similarity with the effects from continuous noise exposures. For instance, results of earlier work has shown similar recovery times between 24 hour interrupted and continuous noise exposures. Present results demonstrate that (1) for the 24–48 hour continuous exposures, TTS growth patterns and ATTS levels are similar. However, recovery from the 48 hour exposure was significantly longer than for the 24 hour condition, and (2) for the 48 hour interrupted exposures, growth and recovery patterns are similar to those of continuous exposures, although the magnitude of the TTS was less and the recovery to pre-exposure hearing more rapid than for continuous exposures.
Growth of TTS as a function of the peak level of impacts noises of long duration
Authors: Hétu R. , Boudreau V.
Published: October 2005

The Concept of critical level for impulse noise [H. McRobert and W. D. Ward, J. Acoust. Soc. Am. 53, 1297–1300 (1973)] was tested using impact noises of long duration. Fifteen normal listeners were exposed to repeated series of 30 impact noises presented one every 3 s at peak levels ranging from 106 to 136 dB SPL. The duration of the impacts were approximately 500 ms each. TTS2 at 4 and 6 kHz in both ears was measured following each series of 30 impacts and the peak level was increased by 3 dB steps until a TTS2 of 15 dB at any frequency was observed. The peak noise level producing the 15 dB TTS was identified as the critical noise level (CL). Results indicated that growth of TTS was not linearly related to the peak level of the noise. For the 13 subjects who reached the target amount of TTS, the CL's ranged from 118 to 136 dB SPL, In 8 out of 13 subjects, the target amount of TTS could be observed following a single series of 30 impacts presented at individuals CL. However, exposure to 120 impacts at 6 dB below the CL also induced the critical amount of TTS2, suggesting that the equal energy assumption may be partially correct for establishing a trading relationship between peak level and number of impacts.
Studies on sound diffusion in reverberation rooms in Japan
Authors: Koyasu Masaru
Published: October 2005

During the 1960's, round-robin tests on the measurements of sound absorption coefficient in reverberation rooms had been carried out extending over four times in Japan. One of the main objectives of these investigations was to find a sensitive and useful measure of sound diffusion in reverberation room. Two kinds of indices, the ratio of horizontal to vertical mean energy during decay process and the directional distribution of spatial correlation coefficient, were adopted for the evaluation of the degree of diffuseness. Most of reverberation rooms in Japan have nonparallel walls in order to obtain diffuse sound field. Moreover, suspended diffusing plates are often used to improve the degree of diffuseness. Compared with the case of rectangular room, relatively small number of diffusers would be sufficient to obtain the same order of diffuseness. An essentially important factor would be the volume of the room, and for measurements at low frequencies, extremely large volume would become necessary. Thus, it would be desirable to use two or three reverberation rooms with different volumes in order to cover the whole frequency range. In parallel with these investigations, theoretical studies on the sound field in rectangular reverberation rooms have been carried out extensively.
A review of experimental techniques in modal testing
Authors: Fowler J. R.
Published: October 2005

Digital vibration control and analysis equipment has had a dramatic impact on structural dynamic testing. Previously, many tests or analyses were impractical if not impossible. This paper reviews current applications of digital equipment to the identification of structural dynamic characteristics including modal testing. The applications presented will cover the following input excitations; pure random, pseudorandom, periodic random, impact (hammer), swept sine, and shaped transient. The resulting transfer functions will be discussed. The effects of using different measurements for the reference signal will be discussed and illustrated. Most of the structures used in the examples will be communication satellites built by the Space and Communications Group of Hughes Aircraft Company.
Methods for the improvement of convergence of multichannel time-history control algorithms
Authors: Smith Strether , Stroud Richard C. , Hamma George A.
Published: October 2005

Two methods for increasing the probability of convergence of the control algorithm for multiple channel structural dynamic testing systems are analytically and experimentally investigated. The techniques are applicable when the testing system/specimen is characterized by a combination of low damping and nonlinearity that produces divergence of the classical linear algorithms. The first appraoch is to introduce “artificial damping” into the control algorithm to reduce phase sensitivity. This produces slower but more stable convergence to the desired time histories. The second procedure uses an over-defined system (more desired response histories than control inputs) to determine a set of comments that will produce a “best fit” to the test requirement. This method reduces the mathematical sensitivity of the algorithm to produce a more stable convergence to an approximation of the desired history.
Temporal patterns of articulatory and phonatory controls
Authors: Sawashima M.
Published: October 2005

This report is a brief review of recent works on the temporal aspects of speech production, mainly focusing on the results and interpretations of physiological observations of the phonatory and articulatory controls of the speech organs. Kinesiological aspects of speech organs, such as speed of movement and relative timing of different muscles involved in the elementary speech gestures, are problems to be studied in more detail. Relative timing of the laryngeal (phonatory) and supraglottal (articulatory) gestures is responsible for the realization of some segmental features and also prosodic features, especially word accent. The phenomenon of coarticulation has been studied by many investigators, their discussions being concerned with the temporal organization of articulatory gestures and the proposal of various speech production models. The effect of speaking rate, as well as that of stress, on articulatory movements are also discussed in this report. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research, No. 31707.]
Correlation analysis of the temporal patterns of articulatory movement and EMG
Authors: Kiritani S. , Hirose H.
Published: October 2005

The movements of pellets on the velum, jaw, etc., were recorded by an X-ray microbeam system simultaneously with the EMG activity of the pertinent muscles, and the relationship between the temporal patterns of motor control signals and articulatory movements were analyzed. The magnitude of EMG activity at a given time instant was assumed to be related to the displacement, velocity, and acceleration of the movement of the articulator, and a functional formula which approximated the relationship between these variables and the EMG signal was estimated. For the velar movement and the EMG activity of the levator palatini muscle, the velocity dependent component in the EMG signal had nearly the same amplitude of variation as that of the displacement dependent component. Quantitative analysis revealed that the patterns of EMG activity did not reflect the difference in the velocity of velum lowering between /m/ and /N/ (the syllable-final nasal segment in Japanese). It is concluded that the movement of the tongue dorsum in the production of /N/ results in an enhancement of the velum lowering. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research No. 310707.]
Temporal characteristics of spoken words in Japanese
Authors: Sato Hirokazu
Published: October 2005

Temporal characteristics of the so-called “syllable-timed rhythm” of spoken Japanese are investigated with a view to their application to speech synthesis by rule. In order to find the way in which the timing of speech is controlled, both nonsense and meaningful words were uttered by a speaker while trying to maintain synchronization with externally given periodic tone bursts. A comparison of the utterances with the tone bursts shows that the timing points of speech are located near the vowel onset. An analysis of natural utterances, produced without the timing signal, also indicates that isochronism is maintained in the interval between these timing points rather than in the interval between consonantal onsets. The influence of inherent consonantal duration upon the duration of the timing interval was further investigated using 60 nonsense words in which various VC syllables were embedded in the same phonetic context. A partial compensation of durational differences among consonants was found to occur within the immediately neighboring vowels; about 60% of the durational increments were compensated for by the preceding vowel in the context of a word, but the rate drops to 30% in the context of a sentence. The significance of isochronism was also confirmed by a perceptual experiment using synthetic speech, indicating that complementary changes in the duration of a vowel and the immediately following consonant produce a rather small deterioration in naturalness. These results suggest that the vowel-consonant unit plays an important role in the rhythmic structure of spoken Japanese.
On the nature of normalization functions for durations in speech
Authors: Liberman Mark Y.
Published: October 2005

What is the nature of the function which maps the duration of a given phonetic element into the duration of a different one, across a set of contexts in which either one can appear? Let us call such a function a “normalization function.” We observe that a number of models for predicting duration as a function of context (e.g., those of Lindblom, Klatt, and Likerman, among others) agree in predicting linear normalization functions, although they differ greatly in the properties they attribute to the duration-predicting function itself. Of course, certain other proposed duration models, as well as many imaginable ones, do not allow the extraction of simple normalization functions. Linearity of normalization functions (or indeed any other uniform, context-independent way of computing such functions) would have a number of important practical and theoretical consequences. The normalization functions associated with different sorts of duration-predicting functions will be explained, and the empirical adequacy of a linear normalization hypothesis will be explored.
Word accent in Japanese and English: A comparative study of acoustic characteristics in disyllabic words
Authors: Hirose Keikichi , Fujisaki Hiroya , Sugitō Miyoka
Published: October 2005

In order to determine both the universal and language-specific characteristics for word accent, its acoustic manifestations are analyzed and compared in the disyllabic words “áme/amè” of the Osaka dialect of Japanese and in “pérmit/permít” of American and British English. An analysis of the pitch contours of these words indicates that in both languages pitch contour characteristics can be represented well using parameters, such as onset and offset of the accent command, extracted from the pitch contour on the basis of a functional model of pitch contour generation [H. Fujisaki and H. Sudō, J. Acoust. Soc. Japan 27, 445–453 (1971); Proc 7th ICA 3, 133–136 (1971)]. While a general similarity exists between the pitch contour characteristics of “áme/amè” and “pérmit/permít,” respectively, individual differences were found to be much greater in the timing of accent command for “pérmit” than in the other three cases. On the other hand, an analysis of the segmental and syllabic durations indicates a marked difference between Japanese and English: In Japanese, accentual changes are only found tn the duration of the second syllable, while in English they are found to be complementary in the first and second syllables. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research No. 310707.]
Developmental use of vowel duration in perception and production
Authors: Krause S. E. , Fisher H. B. , Wightman F. L.
Published: October 2005

This study was designed to investigate developmental use of vowel duration as a cue to postvocalic consonant voicing in both perception and production. The subject sample consisted of 10 three-year-olds, ten six-year-olds, and ten adults, with normal articulation and language skills and normal hearing sensitivity. For the perception study, steady-state vowel duration was varied in each of the three synthetic continua BIP-BIB, POT-POD, BACK-BAG. The percentage of responses judged as voiced was calculated by subject for each synthetic stimulus value and plotted against vowel duration, yielding 90 sets of identification data. The logistic function was fit to these data by maximum likelihood techniques. Two parameters, crossover and slope, were obtained from each fitted function and served as dependent measures for statistical analyses. For the production study, total vowel duration was measured in ten trials of each of the six test words, BIP, BIB, POT, POD, BACK, and BAG, for each speaker. Comparative analyses between the perception and production data revealed a refinement in the use of vowel duration, with an increase in age.
Acoustic consequences and perceptual indicators of stress
Authors: Streeter Lynn A. , Krauss Robert M. , Apple William , Macdonald Nina H.
Published: October 2005

All telephone conversations of Consolidated Edison's system operator beginning an hour before the 1977 New York blackout were studied for indications of emotional stress. The system operator had primary responsibility for monitoring and switching power loads in the New York area. The operator's utterances were analyzed in terms of speech rate, utterance length, fundamental frequency, and amplitude parameters as well as utterance type (e.g., questions, orders, and statements). Average fundamental frequency and maximum amplitude reliably decreased with the imminence of the blackout. In a perceptual study all utterances were randomized and presented to listeners, who rated the perceived situational stress of the operator. In contrast to the production data, there were significant positive correlations between perceived stress and: fundamental frequency parameters, maximum amplitude, and time from the blackout. These results are discussed with respect to the type of utterance and the status relationship between the operator and the other party involved in the conversation.
Variations of voice pitch at vowel onset after various consonants
Authors: Umeda N.
Published: October 2005

The change in voice pitch (Fo) at the onset of a stressed vowel after various consonants was studied from readings of an essay by two male speakers. Measurements were made from narrowband spectrograms using an expanded frequency scale. Speaker 1 maintains a relatively flat Fo, using a high Fo only for new ideas or very emphatic words. Speaker 2 alters the Fo almost constantly. Speaker 1 uses the downward Fo change at the vowel onset more frequently than speaker 2. Both speakers show a very similar pattern in relation to different consonants. Downward changes occur the greatest percentage of the time for voiceless stops; voiceless fricative second; and the voiced stops third (about 50%). Nasals, liquids and semivowels show upward changes 80% of the time. Dental consonants show the highest percent of downward change and labials the lowest. For speaker 2, for voiceless consonants, the use of downward changes increases when the Fo peak within the vowel decreases. Speaker 1 is less consistent. The difference between the two speakers will be discussed. Results suggest that a proper assignment of the position of Fo peak in stressed vowels according to their preceding consonant should eliminate a Swedish accent in synthesized speech.
Shifted band beamformer: Theory and concept
Authors: Pridham Roger G.
Published: October 2005

This paper introduces a fundamentally different beamformer structure which can be used to process bandpass sensor signals.The beamformer operates directly on frequency translated versions of the input signals to compute a frequency translated version of the beam output. This new technique, which is referred to as the shifted band beamformer, is functionally a time domain beamformer but it combines attributes of both time and frequency domain beamforming. It is especially compatible with systems where the sensor data is initially basebanded to facilitate the use of bandwidth sampling procedures. Beamformer vernier delay and throughput requirements are relaxed since the delay quantization and beam output rate depends on the highest frequency component of the basebanded signal rather than that of the bandpass signal. The shifted band beamformer can also use interpolation [R. G. Pridham and R. A. Mucci, “A Novel Approach to Digital Beamforming,” J. Acoust. Soc. Am. 63, 425–434 (1978)] to achieve further hardware savings. Relationship between the new structure and frequency domain beamforming is also discussed.
Digital equalization of underwater transducers for the projection of broadband acoustic signals
Authors: Au Whitlow W. L. , Floyd Robert W.
Published: October 2005

There is a need to project undistorted broadband sonar signals in certain approaches to the development of a bionic sonar and other broadband sonar systems. Most transducers by themselves cannot project undistorted broadband pulse signals having 3-dB bandwidths on the order of an octave and higher. A digital technique which considered both the magnitude and phase of the transfer function of a transducer was used to derive a digital equalization filter. The digital equalization filter was used to predistort the original signals to compensate for the nonuniform transmission response of a transducer. A bionic signal with an octave bandwidth could be projected with a 0.99 cross correlation coefficient between the projected and desired signal using an Edo Western 6122 transducer. The fidelity of the reproduced sounds was determined as a function of the center frequency and the bandwidth of the signal. For a 1-octave bandwidth signal, it was found that the center frequency could also be varied over an octave and still achieve cross correlation coefficients of 0.97 and greater.
Measuring instrument for piloting ship for docking
Authors: Yamamoto M.
Published: October 2005

A very large vessel must be brought into contact with the jetty with a speed low enough to ensure that no damage is caused to the vessel or the jetty structure. Generally speaking, the maximum allowable speed is 15 cm/s, and the approach angle is six degrees from parallel. A measuring instrument, using underwater echo ranging technique, has been developed as an aid for use by ship's master and berthing pilot. Its function is to assist them to secure safety in the berthing operation of large vessels by accurate measurement of bow and stern approach velocities and distance-off. The transducer, fixed on the dolphine pile, has two 100 kHz PZT disks so as to gain ±15° horizontal directivity, and its position is about 3 m deep. The measure range of the instrument is over 200 m and the speed resolution is ±1 cm/s. Docking records show that the range of the contact speed is 3–5 cm/s, and the angle is 5°–6° with the bow touching the dolphine earlier than the stern.
A study on the improvement of the picture of side looking sonar—Part 1
Authors: Shishido M.
Published: October 2005

A preliminary study for the realization of the synthetic aperture side looking sonar was done. The quantization effect of the picture was estimated by using usual side looking sonar's record and digital image processing techniques. The optimum sampling rate of the sonar data in order not to lose the information included but to save the memory was given. Ratio of length and breadth of the record was corrected by assuming the vehicle's speed. The effect of random phase error on the synthetic array was computed. The maximum allowable motion fluctuation of the vehicle in order to conserve the quality of the synthetic image would be given.
Quadratic multiplexing—a new method for secure communication
Authors: Hinich Melvin J.
Published: October 2005

A secure discrete-time multiplexing technique is presented which uses a pseudorandom process with a fundamental cycle length N. The method can handle D ≪ N signal sequences of length M ≪ N. If all the signals have the same bandwidth, the multiplexed signal will have that bandwidth. When N is large the SNR's of the demultiplexed signals is high, even if the additive noise power is large. The demultiplexer must compute the same pseudorandom sequence that is used to encode the signals. Demultiplexing requires 3 DM multiplications and NDM additions. [This work was supported by the Office of Naval Research under contract.]
Standards? A sceptical look at acoustical standards defining“quantities” and units, with particular attention to so-called logarithmic quantities, notable “levels” and “losses.”
Authors: Maples Louis C.
Published: October 2005

Speak acronymic—read logarithmic. The spoken and written languages of levels, losses, ratios, reference quantities, and “nonlinear arithmetic” are reviewed in the context of “standard” acoustical terminology. The roles of the standards organizations, the sponsor, and the writing group are discussed, with comments on responsibility and authority. A different approach is proposed, based simply on the application of physical laws and scientific principles, with special attention given to dimensions and clear, unambiguous language. Current practice utilizes “representative” committees which seem to function not unlike congressional subcommittees, often influenced by political, parochial, and “party” considerations. To achieve real significance and appropriate recognition, a scientific/engineering standard should be the ultimate arbiter, legally “the authority,” and should therefore be unassailably valid, accurate, precise—and limited to matters of substance. The fate of the dictionary, which has become simply a reporter of usage, improper as well as proper, must be avoided.
Recent progress in free-field sound power estimates
Authors: Holmer Curtis I.
Published: October 2005

Recent work on improved understanding of the sound field in the vicinity of large sound sources has led to significant simplifications in viewpoint regarding these sources. The initial thrust of this work arose in Europe and led to the development of several ISO standards for estimating sound power from sound pressure level using only measurements taken on a surface surrounding the source which conforms to the shape of the source. This approach is novel to the extent that measurements are taken at distances surprisingly close to the source by conventional standards. An upper-bound sound-power estimate is formed from the integral of the mean-square sound pressure over such a surface, or experimental approximations to that ideal. Experimental and analytic work that supports this approach will be reviewed to provide a coherent picture of the current state of the art. Directions for further application of this work in related areas such as enclosure performance prediction and estimating sound fields in large spaces are discussed. Directions for further research in this topic are suggested.
Wall and ceiling structures for efficient sound diffusion over wide frequency bands
Authors: Schroeder M. R. , Gerlach R. E. , Gottlob D.
Published: October 2005

Work on efficient sound diffusors [M. R. Schroeder, “Diffuse Sound Reflections by Maximum-Length Sequences,” J. Acoust, Soc. Am. 57, 149–150 (1975)] has progressed to the point where we can now construct surfaces with high sound diffusion over many octaves for both normally and obliquely incident sound. These surface structures hold great promise for enhancing laterally traveling sound, thus increasing subjective preference in halls for music. In this paper we describe the design principles of wide-band diffusors and report experimental results obtained with such diffusors.
Visualizing reflections in rooms with computer graphics
Authors: Noson Dennis
Published: October 2005

A computational technique has been developed to calculate and display, in two dimensions, wave fronts from a point source reflected in rooms of variable shape. The reflections are generated in a series of time frames, generally at 10-ms intervals, for both a plan and section of the room. The technique calculates reflections from re-entrant surfaces such as balconies and suspended ceiling panels, and it includes variation in sound intensity and surface absorption. Rapid changes may be made in room shape and source location. Visual estimates can be quickly made of intensity distribution and arrival time of early reflections. The method has been used to examine some acoustical qualities of existing auditoriums, to evaluate the effects of remodeling room shapes, and to demonstrate acoustic defects.
A design principle for the sound reinforcement system of an open-field stage
Authors: Konishi K. , Morimoto M. , Maekawa Z.
Published: October 2005

If loudspeakers radiate sound from images which lie behind the imaginary walls of an open-field auditorium, the audience can hear sounds in a simulated indoor auditorium. This principle was applied to an open-field stage in Hyogo Prefecture, Japan. In order to supply simulated reflections over all of the audience seats, four positions for the loudspeakers were selected, two on each side of the stage symmetrically. Two poles 18 m in height were built close to the stage, and two other 12 m high poles were kept at some distance from the stage. Two loudspeakers were installed at each position, one for the seats near the stage, and the other for the more distant seats so that everyone in the audience would be able to receive at least one simulated reflection within a suitable delay-time after arrival of the direct sound from the stage. In order to arrange delay-time for each loudspeaker, four channels of digital delay-machines were used. The suitable delay-times were determined by psychoacoustical experiments according to the sound-pressure level, delay-time, and direction of the reflected sound in relation to the direct sound.
Enclosure of outdoor theatre for electroacoustically amplified pop music
Authors: Meyer Norman J.
Published: October 2005

To minimize neighboring community complaints due to concerts at an outdoor theatre while providing the opportunity for year-round use of the facility, the adding of a cover was considered. The desire to retain an outdoor theatre environment led to unusual interior surface acoustical treatment concepts. The resulting facility will be essentially acoustically passive to be compatible with the high quality electroacoustically amplified pop music. The reverberation times were measured and calculated for the open theatre, and calculated for the recommended treatment. The resulting reverberation times will deviate most from the existing open-air theatre in the lower frequency bands, as expected. To significantly reduce the low frequency reverberation times below those for the recommended treatment, a much more costly treatment would be required. Some of the benefits of the combination of an acoustically passive shell with a high quality sound system are briefly considered.
Effect of random shading, phasing errors, and elements failure on beampattern and directivity of a linear and planar array
Authors: Quazi Azizul H. , Porter David T.
Published: October 2005

The effects of random shading, phasing errors, time quantization and elements failure on the beampattern and directivity of a linear (31 elements) and planar (405 elements) array at 6 kHz are investigated. The results indicate that the effect of shading errors on the mainbeam is minimal, however, it effects the “far off” sidelobe critically. The elements positional uncertainty and time quantization effect the beampattern and directivity dramatically compared to the effects of shading errors. In case of planar array, the effects of these phase errors are pronounced near broadside and endfire. With time quantization of 20 μs (about 4 times the Nyquist rate) and shading error of 5% of ideal condition, the sidelobe level increases up to 10 dB which depends on the steering direction of the beam. The combined effects of shading and phasing errors on directivity is about 3 dB. The elements failure (6%) increases the sidelobe level about 10 dB for a linear array, whereas it has minimal effect on planar array.
Acoustic radiation from prolate spheroidal lenses
Authors: Valenzuela F. M.
Published: October 2005

This study deals with the mathematical analysis of acoustic radiation from multiple prolate spheroidal surfaces. It forms the basis for developing a better model to predict a submarine's sonar performance. The sonar array and its dome were modeled as a prolate spheroidal array confocal to a prolate spheroidal dome. The dome was represented by a mass only 0.25-in.-thick steel shell, with a major axis equal to twice the length of the bow sonar cavity and a minor axis dimension typical for a submarine. Four loading conditions were imposed on the array: (a) free-field radiation, (b) acoustic lens, (c) mass-loaded acoustic lens, and (d) mass-loaded non-lens. The influence of the dome mass on the radiated pressure of a free-field array was found to be significant and must be included in analyses of sonar systems. The influences of the wall mass on the performance of the acoustic lens was not significant. Significant focusing by the lens was not observed. This was due to the low frequency of the array and the internal acoustic scattering by the array itself. [Doctorial dissertation.]
A new procedure for simultaneous comparison of the annoyance of multiple noise sources
Authors: Fidell S. , Horonjeff R. , Pearsons K. , Green D.
Published: October 2005

An adaptive free response technique has been implemented to make simultaneous determinations of the annoyance of multiple noise sources. Individual subjects may terminate exposure to a single noise source at any time, in favor of exposure to one of five other noise sources at random. If exposure to the current source is not terminated by the subject, it is terminated after an unpredictable time period in any event. Signal presentation levels are adjusted after a block of trials to maintain equal total energy, in proportion to the mean exposure duration. The size of the adjustments is halved after a reversal of direction of adjustment. Data collection ceases at the point of indifference among the signals (when the subject's distribution of termination responses is equally likely to be random as based on true preferences, or when a minimal step size criterion is satisfied). The new procedure was first applied to a comparison of the annoyance of a number of noise sources of disparate spectral shapes, whose relative annoyance had previously been determined by conventional paired comparison testing.