bluuebunny/crossref_metadata_2025 huggingface dataset:
In search of the shruti
Authors: Burns Edward M.
Published: October 2005

Earlier experiments indicate that the phenomenon of “categorical perception” is present in the perception of musical intervals. The present experiment was undertaken to determine whether musicians from musical systems having more categories in their musical pitch scales would exhibit finer discrimination in categorical perception experiments. The subjects were 13 musicians trained in Hindusthani (North Indian classical) music, a system which theoretically has 22 degrees per octave in its musical pitch scale. However, these subjects were not able to identify musical intervals with any more precision than the Western musicians in our previous experiment. That is, they were not able to do more than identify the interval as one of the notes of the 12-note chromatic scale and were unable to consistently identify the “shrutis” or microtonal variations. In addition, they had much greater difficulty adapting to the moving reference pitch in the experiment. In fact, only one of the 13 was able to decide with any accuracy which of two intervals (defined by two successive pairs of tones, none of which had the same frequency) was the larger. Possible intercultural differences in the perception of musical pitch are discussed. [Research supported, in part, by a Fulbright-Hayes grant and performed at the Phonetics Laboratory, Deccan College, Pune 411006, India.]
The bowed string: an integral equation formulation
Authors: Schumacher R. T. , Amick C. J. , Croke C. B.
Published: October 2005

The problem of the bowed string has been cast in the form of an nonlinear integral equation of the Hammerstein (Fredholm) type, and some solutions of the fixed-point problem resulting therefrom have been obtained both analytically and numerically on a digital computer. In the equation v(τ) = ∫01K(τ,τ′)F[v(τ′)]dτ′, the velocity at the bowing point, v(τ), a function of the reduced period τ, enters the integrand via the nonlinear force F(v) between bow and string. In the model we have solved, an essential assumption is that the bow presents a real, finite mechanical impedance to the string. The kernel K involves the string admittances at the bowing point. Realistic admittances are used which include the effects of string stiffness and complex terminating impedance at the bridge. The equation, and the techniques of solution we have used, should apply to a large class of multimode self-sustained oscillators, such as wind instruments. [Supported by a grant from the National Science Foundation.]
Automatic recognition of speech using optimal search
Authors: Baker J. K.
Published: October 2005

The Dragon speech recognition system models speech as a probabilistic function of a Markov process. The degree of match between a hypothesized sequence of states W and an observed sequence of acoustic parameter values A is the a posteriori probability of the sequence W and gives the language L, that is, P(W|A,L). Some experiments have been done with an implementation of this model using an optional search strategy based on dynamic programming. The algorithm finds a sequence W0 such that P(W0|A,L) is the maximum achievable over all possible sequences in the language L. The system has been tested on several languages, each representing a specialized interactive computer task, with vocabulary sizes ranging from 24 words to 195 words. In a series of experiments, each consisting of about 20 sentences from a given language, the system correctly recognized between 85% and 95% of the words.
Speech recognition through spectrogram matching
Authors: Ingemann Frances , Mermelstein Paul
Published: October 2005

In order to assess human analysis of acoustic data before attempting such analysis by machine, a series of experiments was conducted in which subjects were asked to match spectrograms of continuous speech to reference spectrograms of the same words. Although error rates varied with sentence difficulty and size of vocabulary, comparison of the matches shows greater agreement in phoneme segments than other experiments have obtained in phonetic transcriptions of unknown utterances without semantic or syntactic processing. Accuracy in matching can be further improved by feedback in the form of spectrographic representation of a sequence of tentative matches spoken as if they made up the unknown utterance. Automatic matching of word- or syllable-sized acoustic patterns may provide a more accurate phonemic input to the syntatic-semantic component of a speech recognition system than other methods so far attempted. [Research supported by the Advanced Research Projects Agency, Department of Defense.]
An overview of the Lincoln Laboratory speech recognition system
Authors: Forgie J. W. , Hall D. E. , Wiesen R. A.
Published: October 2005

The Lincoln speech recognition system is capable of recognizing spoken sentences made up of words drawn from a limited vocabulary and constrained to conform to a context-free grammar. Speech input is taken from a close-talking, noise-cancelling microphone in a relatively noisy computer room. The speech is digitized and subjected to a detailed acoustic-phonetic analysis [C. J. Weinstein et al., Proc. IEEE Symposium on Speech Recognition (April 1974), pp. 89–100] which produces a string of acoustic phonetic elements (APELs). The APEL string is scanned by a linguistic module which attempts to find and score candidate sentences which satisfy the syntactic and semantic constraints of the grammar and which are composed of words having acceptable matches between phonemic dictionary spellings and APEL representations. Typical sentences of 3- to 4-sec duration require the order of one minute of computer processing. Tests have involved vocabularies of 125 to 500 words, grammar of varying complexity, many speakers (both male and female), and several hundred test sentences. [This work was sponsored by the Advanced Research Projects Agency of the Department of Defense.]
Speech data base for testing components of speech understanding systems
Authors: Lea Wayne A.
Published: October 2005

A set of over 700 sentences was designed to test acoustic effects due to various minimal contrasts in linguistic structures. One subset of sentences includes a comprehensive variety of phonetic sequences, to allow efficiently testing phonetic effects on prosodies, accuracy of phone recognition algorithms, and reliability of parameterization procedures such as ferment tracking. Another subset of sentences tests how acoustic correlates of stress and syntactic boundary locations (as determined by valleys in fundamental frequency contours) are altered as the position of stress within a constituent is moved. To study intonation patterns in isolation from the local fundamental frequency variations that accompany obstruents, many sentences have only sonorant sounds in them. Other sentences include only vowels and unvoiced consonants, to provide easy syllabication with energy measures. Sentences vary in complexity from simple declarative structures to sentences which introduce auxiliary verbs, prenominal adjectives, quantifiers, possessives, adverbs in various positions, there insertion, passives, yes/no and WH-questions, commands, and embedded structures such as restrictive and nonrestrictive relative clauses, complement structures, and conjuncts. The sentences were recorded by five male and three female talkers, with three repetitions spaced at least one week apart.
Comparison of methods of formant analysis
Authors: Curtis J. F. , Allen E. L. , Paige A.
Published: October 2005

The use of inverse filter computer programs may now allow a more viable approach to frequency and bandwidth assessments of speech samples. The purpose of this investigation was to assess the efficacy of automatic spectral assessments. Two male subjects produced samples of the vowels /a/, /i/, and /o/, both in isolation and in /hVd/ and /hVn/ contexts. Samples were analysed both with Kay Sonagraph cross sections and with INFIL2, and LPC inverse filter program developed by Paige. This program essentially picks spectral peaks according to bandwidth specification of 300 Hz or less. An acoustic spectrum (amplitude vs frequency plot) is also generated by the program. Two inexperienced observers and one experienced observer made formant estimations from the program plots. In addition, the experienced observer made formant estimates from the Sonagraph amplitude cross sections. Spectral features which were identified mad the ranges of assigned spectral frequencies will be compared.
Sound absorption in a mixture of O2 and N2
Authors: Bass H. E. , Keeton Roy G.
Published: October 2005

The attenuation of a 1-MHz ultrasonic signal has been measured in 72% N2/28% O2 at temperatures of 300°, 500°, and 700 °K and pressures from 1 atm to 1/100 atm. The measured attenuation was corrected for classical effects to obtain the attenuation due to rotational relaxation. In addition, the total absorption has been compared to computed values based on a linear combination of viscous, thermal conduction, diffusion, and relaxation to check the constants used for each of these terms. [Work supported by Army Research Office, Durham, North Carolina.]
On the excess attenuation of sound in the atmosphere
Authors: DeLoach Richard
Published: October 2005

The attenuation suffered by a sound wave propagating from an elevated source to the ground, in excess of spherical spreading losses and classical and molecular absorption effects, is studied. Reported discrepancies between attenuation measurements made in the field and the predictions of theories based only on absorption effects are discussed. It is concluded that atmospheric dynamics (turbulence) is a major contributor to the attenuation of sound waves propagating vertically in the atmosphere. Earlier theories on the attenuation of sound due to scattering have led to predictions of a strong (square-law) frequency dependence and relatively large magnitudes which are not supported by the available field data. The present theory predicts a square-law frequency dependence only for the special case of a homogeneous, isotropic medium. For a medium such as the atmosphere, with an inhomogeneous, anisotropic outer scale, a much milder frequency dependence is predicted which approaches a cube-root dependence for the case of an outer scale of turbulence large compared to half an acoustic wavelength (the general case for audio frequencies in the atmosphere). This result is supported by field data, as are the excess attenuation magnitudes predicted by the present theory.
Saturation effects associated with wave propagation in a random medium
Authors: Wenzel Alan R.
Published: October 2005

A theoretical analysis of the propagation of spherical time-harmonic waves in a random medium is presented. The smoothing method is used to study the incoherent (or randomly fluctuating) component of the wave field. With the aid of the Fresnel approximation, an expression for the second moment of the incoherent wave is obtained which is valid for the case of high-frequency waves propagating in a medium which is statistically homogeneous and isotropic. This expression shows that the rms fluctuations of the wave field increase initially as the square root of the propagation distance, but that at larger distances (and higher frequencies) the fluctuations tend to saturate. These results agree with observations of waves propagating in real media.
Measurement of sound power and energy densities
Authors: Olson Harry F.
Published: October 2005

The sound power density (intensity) of a sound field in a specified direction at a point is the sound power transmitted through a unit area normal to this direction. The sound energy density is the sound energy per unit volume. The conventional measurement of the sound pressure alone, except in rare instances, will not provide either the sound power density or the sound energy density. The means and uses for the measurement of sound power density and sound energy density are described. The measurements include sound absorption of room boundaries, machine noise, and the sound power density and sound energy density of sound-reproducing systems.
Sound amplification in muffler expansion chamber
Authors: Fricke Fergus R. , Crocker Malcolm J.
Published: October 2005

Expansion chambers have long been used in automotive exhaust systems to achieve sound attenuation. However, in some instances, within certain limitations of frequency, flow velocity, sound intensity, and duct geometry it has been found that sound amplification can occur. The phenomenon is due to sound and flow interactions. Results showing this phenomenon, together with an explanation of why it occurs, are presented in this paper.
Fog signal baffle design
Authors: Malme Charles I.
Published: October 2005

As part of its automation program, the U. S. Coast Guard has been replacing multifrequency air-driven sound signals with electric air oscillators. The pure-tone character of this signal has been a cause of noise complaints by residents near the installation. For example, an unbaffled ELG 300/02 dual emitter is capable of producing levels of the order of 80 dBA 1/2 mile to the rear of the signal. Thus, to be effective, a baffle design must provide at least 30–40 dB of attenuation in its dark sector. A program was conducted to develop high attenuation baffles for ELG 300-Hz emitters using the principles of diffraction source interference, acoustically soft baffle edges, and end-fired array lenses. The program utilized 1/8-scale models to test the effectiveness of experimental configurations. The most effective baffle designs were then verified by full-scale field tests. [This research was supported by the Coast Guard under Contract DOT-CG-30200-A.]
Magnetomechanical coupling in highly magnetostrictive rare earth—Fe2 compounds
Authors: Savage H. T. , Clark A. E. , Powers J.
Published: October 2005

It was recently discovered that certain rare earth—Fe2 alloys exhibited huge static magnetostrictions (∼0.2%) at room temperature. Here we report the first magnetomechanical coupling measurements on Tb0.3Dy0.7Fe2. A 4-in. thin bar was placed in an external field consisting of a 2-Oe ac component plus a dc bias. A small pickup winding was wound tightly around the sample to measure the magnetic flux density inside the sample. The “admittance” (the 2-Oe exciting current divided by the pickup coil voltage) data was analyzed by the usual method. A maximum coupling coefficient exceeding 0.6, a Q of 26, and a relative permeability of 4.55, were observed at a bias of 150 Oe. The coupling coefficient decreased to 0.10 with a Q of 155 and a relative permeability of 1.64 at a 4350-Oe bias. Over this same dc bias range the frequency of resonance, fp, increased by 57%, exhibiting the largest “ΔE effect” known. We discuss application of these alloys to sonic transducers, variable-frequency resonators, and variable delay lines. [Supported by the Naval Sea Systems Command.]
An all-purpose fluid for filling sonar transducers
Authors: Green C. E.
Published: October 2005

A fluid suitable for use in all liquid-filled transducers has long been needed. Historically, castor oil has been the most satisfactory; however, it will no longer meet all requirements. Substitute fluids have been used in special cases, but these have serious limitations even under the best of conditions. Studies have been made on silicone fluids and polyglycols as well as castor oil to determine their eligibility as a sound energy transfer medium in sonar transducers. Acceptability criteria covering acoustical, chemical, physical, and environmental properties are discussed. A modified polyalkylene glycol has been found to possess compatible features with transducer materials and to meet the criteria for a fluid to use in all sonar transducers.
Statistical pattern recognition approach to speech segmentation
Authors: Ganesan K. , Lin Wen C.
Published: October 2005

It is believed by many speech researchers that there is abundant information contained in the transition segments [J. L. Flanagan, Speech Analysis, Synthesis and Perception (1972), 2nd ed.]. This paper describes an algorithm to recognize different segments of connected speech, e.g., voiced, unvoiced, silence, and transition. While others are obvious, the transition segment of a speech utterance can be any of the following: (1) voiced-unvoiced, (2) unvoiced-voiced, and (3) voiced-voiced. The features used in the segmentation procedure are (a) total per-unit energy (dB⋅msec), (b) per-unit glottal energy (dB/msec) (obtained after filtering the original signal by a 300-Hz FIR low-pass filter), and (c) zero-crossing rate (number of crossings/msec). These features are obtained from a 25.6-msec segment of the speech signal. The algorithm makes use of a set of thresholds which are derived by means of maximum-likeihood principle, so that the probability of classification error p(e) is minimum [G. Sebestyen, Decision-Making Processes in Pattern Recognition (McMillan, New York 1962)]. p(e) = p(m) + p(f), where p(m) and p(f) are the probabilities of missed detection and false alarm, respectively. The probability of classification error is found by checking against the results of manual segmentation and labelling on the speech waveform in both time and frequency domains. The data base consists of isolated words and sentences (both phonetically balanced) spoken by a male speaker with a General American accent. The recordings were done in a relatively noise-free environment on two different days with an interval of 40 days apart. The speech was sampled at 10-kHz rate and quantized to 10-bit accuracy. An 80-dB/octave dealiasing filter with a cutoff at 4 kHz was used. The overall error rate using this approach is 0.1%. Information content of recognized segments will be investigated in the near future.
Automatic time-domain techniques for segmentation of connected speech
Authors: Baker J. M.
Published: October 2005

Connected speech, spoken by males and females, has been intensively studied in the time-domain. The parameters chiefly used included cycle-frequency, cycle-amplitude, and cycle-microstructure, which were computed for each cycle in the waveform, sampled at 20 kHz. Patterns of these parameters were discerned for the following acoustic states: silence, unvoiced release-aspiration, unvoiced frication, nasalization, transition (actually two states, one for very brief transitions and another for much longer transitions), vowel, voiced release-aspiration, and voiced frication. These patterns were embedded in an automatic segmentation program. To determine the acoustic state of each waveform cycle, this program evaluated some precontext and postcontext information and then applied appropriate tests establishing whether the present cycle marked the beginning of a new acoustic state and which type. After utterance segmentation, this output was run through an editing program which concatenated certain kinds of segments and deleted segment markers of brief noiselike regions. The final result was a sequence of times at which segment boundaries were found. The results of this program (without speaker training) compared favorably in all respects, and superiorly in certain respects, to segmentation results of four other groups of the speech community run on the same data set (five speakers recorded under different conditions).
The elastic syllable: an acoustic view of the stress-intonation link
Authors: Gaitenby J. H.
Published: October 2005

That fundamental frequency, duration, and intensity interact in signalling relative syllable stress is well known. Yet both prosodic description and analysis Of connected speech remain quite rare in the published literature. This report provides acoustic measurements of selected aspects of sequential syllables (as read separately by four talkers of American English) and describes one method of handling such syllabic data in order to reveal stress patterns in polysyllabic words and phrases. The method of relative stress measurement used—a simple addition of prosodic values within each successive syllable—also produces curves that resemble so-called “intonation contours.” [This research was supported by the Prosthetic and Sensory Aids Service, Veterans Administration, under Contract V101 (134) P71.]
Estimating the magnitude of linguistic stress
Authors: Cheung J. Y. , Minifie F. D.
Published: October 2005

The perception of linguistic stress has been known to correlate highly with fundamental frequency, syllable duration, and intensity. In this paper, the estimation of the magnitudes of perceived linguistic stress is demonstrated feasible and consistent through the use of a computer program that automatically extracts and analyzes the three parameters. Several talkers are asked to read various passages, including parts of the rainbow passage. Their readings are recorded and presented to trained listeners. The magnitude of linguistic stress for each syllable is then perceptually rated on a continuous scale. These ratings, in turn, served as standards in developing an accurate and efficient computer program. The results show that close approximations of the magnitudes of linguistic stress can be obtained from the analysis of the three acoustic correlates above. However, some of the variance can only be accounted for by additional features, such as intonational inflections toward the end of the syllable, the position of the syllable in the utterance (the alternate stress rule), and idiosynchratic strategies of individual talkers. It is seen that computer-analyzed values are more consistent for comparisons, less time consuming to obtain than the actual perceptual ratings, and can be used directly for other applications, such as speech recognition and analysis of stress patterns.
What determines the phonetic durations of phonemically long and short vowels?
Authors: Allen George D.
Published: October 2005

If differences in segment durations are to be a phonemically effective cue, they must be within the abilities of all speakers in the community to produce and perceive. Recent studies [Allen and Cooper, J. Acoust. Soc. Am. 53, 379(A) (1973)] suggest that different normal speakers may vary widely in their speech timing abilities and that our estimate of a performance criterion for “normal” language should therefore be revised upwards to accommodate the more variable speakers. Differences in segment duration will be examined, contrasting four theoretical models relating the measured lengths of short and long vowels: (1) constant difference [e.g., 100 msec; A.S. House, J. Acoust. Soc. Am. 33, 1174–1178 (1961)]; (2) constant ratio [e.g., 2:3; G. Peterson and I. Lehiste, J. Acoust. Soc. Am. 32, 693–703 (1960)]; (3) general linear relationship [Lindblom and Rapp, PILUS No. 21 (1973)]; (4) constant relative variance [Allen and Cooper, J. Acoust. Soc. Am. 53, 379(A) (1973)]. The usefulness of duration as a cue will be discussed, in relation to both theories of time perception and phonological rules for segment length. [Supported by NSF Grant GS-41863.]
Boundary placement in connected speech
Authors: Umeda N. , Harris M. O.
Published: October 2005

Pitch is closely correlated with stress and boundary (pause and pseudopause). Stress and pause are not mutually independent; they often form an auditory unit in speech. Such units, we believe, must help listeners to grasp the grammatical and semantic framework of the message. Therefore, before we study fundamental frequency at the acoustic level, we chose to investigate the relation between sentence structure and the placement of boundaries. Two people, whose backgrounds include linguistics and some musical training, listened carefully to an essay read by four speakers, indicating boundariess where they perceived them. The listeners were concentrating more on overall perception of boundary than on its different acoustic characteristics. Analysis of preliminary data shows the listeners to be in 90% agreement with each other about the location of boundary reponses. Boundaries were perceived between two successive words in these conditions: on 10% of all boundaries within the same phrase; on 50% of grammatical-unit boundaries in normal structural order, such as subject-verb; and on 70% of boundaries in reversed structural order, such as interruptions. Eighty percent of the boundary events noted by the listeners were heard as occurring in all four speakers.
Exterior truck noise measurements on turbocharger retrofit
Authors: Pisarski John J.
Published: October 2005

Exterior noise-level measurements were conducted on a typical in-service diesel engine heavy-duty truck to determine the noise-level reduction resulting from a turbocharger retrofit. The tests were conducted in accordance with procedures recommended by the Society of Automotive Engineers and the United States Department of Transportation. Overall noise-level measurements and a narrow-band spectrum analysis indicate that a significant noise-level reduction is obtained from the installation of a turbocharger kit. The lower noise level is the result of a decrease in the exciting forces due to combustion. The narrow-band analysis shows a significant amplitude reduction at the engine firing frequency and an overall spectrum reduction.
Propagation of aircraft noise near airports: effects of hillsides, inversions, and source directionality
Authors: Embleton T. F. W. , Piercy J. E.
Published: October 2005

At ranges of one to three miles especially, the noise levels due to operations on the ground (e.g., acceleration, thrust reversal, engine run-ups) are strongly dependent on topography and weather. Listeners up a hillside lose some or all of the noise reduction that would otherwise be provided over flat land by ground shielding. The effects of weather are variable, but under conditions of temperature inversion (which occur frequently at night for many major airports) there is a reduction of ground shielding compared with a neutral atmosphere. Because ray paths in an inversion are curved we calculate that the effects of hillside and inversion are different functions of frequency. When the plane is more than 5°–10° above the horizon, in the air, these effects disappear. Here, however, the change in directionality of a jet engine with frequency, coupled with highly frequency-dependent molecular absorption, can introduce substantial errors into methods of calculating noise exposure, both level and time duration, for distances greater than about 2000 ft, particularly in sideline areas adjacent to the airport.
Detectability of inserted and deleted tones: effects of tone frequency, masker intensity, and observer training
Authors: Bourbon W. T. , Conkright R. K.
Published: October 2005

The detectability of the insertion of tones into masking noise was compared to the detectability of tones deleted from a combination of tone and masking noise. In a 2AFC paradigm, psychometric curves were determined for one group of 16 naive observers and one group of two highly trained observers. Each observer was presented a counterbalanced sequence of four stimulus conditions involving tones of 500 or 1000 Hz and masking noises with spectral levels of 29 or 40 dB SPL. The conditions of the present study provide a comparison of conditions used by Schulman [Perception Psychophys. 9, 496–498 (1971)] and by Bourbon and Lembo [J. Acoust. Soc. Am. 53, 376(A) (1973)]. The following results were obtained: (1) for all stimulus conditions, experienced observers were more sensitive than naive observers by approximately 4 dB; (2) for both groups of observers and for all combinations of frequency of tone and intensity of noise, inserted tones were approximately 4 dB more detectable than were deleted tones.
Detection of a noise signal: the effects of duration and masker level
Authors: Viemeister Neal F.
Published: October 2005

Monaural thresholds for a wide-band noise signal in the presence of a pulsed or continuous wide-band noise were measured in a 2IFC task as a function of signal duration (0.8–1600 msec) and masker level. For masker spectrum levels of −5, 15, and 45 dB DPL: (1) performance is better with a pulse masker than with a continuous masker for durations less than 70 msec, and for longer durations performance is slightly worse with a pulsed masker than with a continuous masker; (2) the improvement in performance with increasing duration is less with a pulsed masker (−3.6 dB/decade) than with a continuous masker (−6.7 dB/decade); (3) performance with both maskers is independent of duration for durations greater than 200 msec; (4) with pulsed maskers, the form of the psychometric function is dependent on duration; (5) at each duration, Weber's law describes the data for both pulsed and continuous maskers. With no masker, an intensity-time trade of −7.1 dB/decade is obtained; similar results are obtained at short durations with a masker level of −25 dB. These data are compared to predictions based upon the ideal (energy detector) and the ideal degraded by temporal uncertainty.
Audibility of infrasound
Authors: Johnson Daniel L. , von Gierke H.
Published: October 2005

The distortion of pure tones (1–16 Hz) caused by the nonlinearities of the middle ear was calculated. It is shown that the slope of the audibility curves for infrasound of Yeowart and Evans could be predicted, thus implying that infrasound might not be heard in the normal sense, but only heard as distortion after being transduced through the middle ear. To verify this result, subjects were exposed simultaneously with the 1–10-Hz stimuli to a low-frequency masking noise (10–100 Hz). This noise was shown to mask pure tones of infrasound of 1–10 Hz even when the SPLs of these tones were 15–25 dB above the masking noise overall sound-pressure level. Clearly, this result implies that the pure tones of infrasound below 10 Hz are not heard in the same manner as tones above 16 Hz. The implications of these results to the importance of the infrasound components of any broad-band noise and to the auditory effects of infrasound are discussed.
Masking of tone by tone as a function of duration
Authors: Jeffress L. A. , Sharpley Alan
Published: October 2005

A 500-Hz tonal signal was masked by a continuous tone, both from the same oscillator. A two-interval-forced-choice procedure was used to determine the level required for 80% correct. The masker level was 50 dB SPL, and the signal was always in phase with the masker. Signal durations from 20 msec to 2 sec were employed. The results showed a slope of approximately 3 dB per doubling over the entire range from 20 msec to 1 sec. Detection at 2 sec tended to be worse than at 1 sec. The slope of 3 dB per doubling is much steeper than has commonly been found with noise maskers, especially at the longer durations. Data from an electrical model are discussed.
Masking of narrow-band noise by pure tone
Authors: Young I. M. , Wenner C. H.
Published: October 2005

Threshold measurements of narrow-band noise with Zwicker's critical bandwidth were made in the presence of various pure tones in subjects with normal hearing. For a given masking level, the masking effect on the narrow-band noise by the center frequency decreased as the center frequency increased. This finding agrees with the masking of pure tone by pure tone in that the slope of masking versus level of the masker was inversely related to the frequency of the masking tone. Spread of masking was greater by frequencies below the center frequency than above.
Loudness contours and growth functions derived from difference limen data
Authors: Corliss Edith L. R.
Published: October 2005

In 1933, R. R. Riesz pointed out that his data on difference limen for intensity (1928) paralleled several contours for equal loudness measured by Fletcher and Munson (1933). Another extensive set of difference limen data was measured by Zwicker and Kaiser in 1955. They agreed so closely with the Riesz data that both sets could be combined. In 1972, Schneider, Wright, Edelheit, Hock, and Humphrey published results of a loudness magnitude estimation study. These data make it possible to test Riesz's hypothesis: Over more than two orders of magnitude there is a one-to-one correspondence between loudness and the difference limen for intensity. Thresholds determined from the Riesz, and Zwicker and Kaiser data, considering the differential intensity sensitivity of the ear to stem from a “least count” in the hearing mechanism are in close agreement (Corliss, 1967) with the observed free-field threshold of hearing for frontal incidence of sound. By scaling up from threshold, equal-loudness contours have been estimated for pure tones, working from difference limen data. As would be expected, they fit the Schneideret al. loudness data for high levels, and—as has been the experience in the past—agree with some but not all other contours.
Pumphrey and Gold revisited, again
Authors: Green D. M. , Wier C. C. , Wightman F. L.
Published: October 2005

In 1948 Pumphrey and Gold investigated the ability to distinguish between a periodic sequence of identical sinusoidal pulses and a second sequence in which alternate pulses were inverted. The ability to detect differences between these sequences led them to calculate the Q of an assumed resonance system representing the auditory filter or critical band. They concluded that the Q was of the order of 100 for the conditions of these experiments. A detailed analysis of the spectrum of the two pulse trains indicates the critical cue for the discrimination is a pitch cue as described by Hiesey and Schubert. This pitch cue is the result of a ripple in the power spectrum of the stimulus and can be related to the pitch of delayed and added noise, as studied by Bilsen and others. [This research was supported by the National Institutes of Health.]
Measurement of reaction time and absolute auditory threshold in cat
Authors: Gerken G. M. , Sandlin D.
Published: October 2005

A self-paced operant threshold procedure developed for monkeys [Stebbins, Animal Psychophysics (1970)] was adapted for use with cat. In the modified procedure, the cat (a) initiates a trial by insertion of its paw into a slot operandum, (b) maintains paw placement during a variable delay of 1 to 9 sec following which a brief tone is presented, and (c) removes its paw to activate a dispenser of liquid reinforcement. Reaction time, the interval between stimulus onset and paw withdrawal, was generally between 250 and 500 msec for supraliminal stimuli. Absolute thresholds were measured with a tracking procedure using 4-dB steps. Threshold and reaction time data have been obtained from two animals at octave frequencies between 250 Hz and 8 kHz. In addition, the effects on threshold of other test variables were evaluated. It was concluded that paw withdrawal is a preferred response because of its rapidity. The self-paced operant procedure presents advantages over other positive reinforcement methods and avoidance procedures. [Supported by a grant from NINDS.]
Operant conditioning of the guinea pig
Authors: Walloch Richard A.
Published: October 2005

Guinea pigs have been trained to respond on a chained VR-20 crf schedule of reinforcement for a food reward. The guinea pig had to press the bar until a 1-kHz tone at −4 dB re dyn/cm2 was presented. The number of bar presses needed to turn on the tone varied randomly and the average number was 20. Once the tone was introduced, the guinea pig had to cross to the opposite side of the cage to receive a food reward. A premature crossing of the cage before the tone was introduced was punished by a 15-sec time out during which the bar was withdrawn. Discriminative behavior developed. In a typical session (1) 77% of the cage crossings occurred after the tone was introduced, while (2) 23% of the crossings occurred before the tone was introduced. This discriminative auditory behavior may serve as a behavioral method by which many aspects of hearing in guinea pigs may be evaluated. This paper is the first report of a food-rewarded response under the control of an auditory stimulus in the guinea pig. [This work was supported by a grant from the Deafness Research Foundation.]
Quantitative effects of noise on sonar performance in bats
Authors: Simmons J. A. , Lavender W. A. , Lavender B. A.
Published: October 2005

Precise measurements were made of the effects of random noise on target detection and range discrimination in two-alternative, forced-choice situations by echolocating bats (Eptesicus fuscus). Targets were simulated electronically to provide good stimulus control and accurate specification of echo signal-to-noise ratios. Acoustic data were digitized at a sampling rate of 512 kHz and analyzed with a PDP-11/40 computer to estimate echo and noise power. The results show that bat sonar operates efficiently very close to ideal limits on performance, and they identify acoustic conditions which evoke increases in the bat's transmitted signal power.
Energy spectrum analysis: a model of echolocation processing
Authors: Johnson R. A. , Titlebaum E. L.
Published: October 2005

Several analyses of the abilities of echolocating animals have suggested that nearly ideal processing—e.g., correlation or matched filtering—is used by the animals. A review of the details required for time-domain coherent processing reveals many difficulties when considering the possible biological implementations. An alternate processing mechanism, energy spectrum analysis of the combined signal and echoes, is proposed. It is equivalent to autocorrelation processing, yet overcomes most difficulties when considered for biological implementation. Strongest evidence for this processing is found in the psychoacoustic phenomenon usually termed time separation pitch. In fact, this coherent echolocation analysis could be considered as an alternate derivation of the pattern transformation model of pitch recently published by Wightman (J. Acoust. Soc. Am. 54, 407–416 (1973)]. Observations of the performance of an auditory spectrum analyzer help to explain many of the phenomena observed in echolocation. The importance of this proposed model lies in its universal application to all echolocating animals, including bats, dolphins, and echolocating birds. [Supported by a Naval Undersea Center contract to the University of Rochester and Air Force contract number AFOSR 71-2201C to Yale University.]
Mating call recognition in the green treefrog (Hyla cinerea)
Authors: Gerhardt H. Carl
Published: October 2005

Synthetic calls with a waveform periodicity of 300/sec and a bimodal spectrum attracted female treefrogs as effectively as natural calls. Reducing the relative amplitude of the lower peak by 10 dB or the upper peak by 15 dB rendered a synthetic call less attractive. Addition of two components between the peaks had the same effect. The optimal band far the lower peak was 900–1100 Hz and for the upper peak, about 2700–3300 Hz. When a bimodal stimulus was unavailable, most females responded to a call having only a lower peak of 900 Hz. Many females responded to a call having only high-frequency energy with a periodicity of 300/sec (2700 + 3000 Hz), but very few responded to a 3000-Hz call and then only when played back at 100–105-dB peak SPL (14–19 dB above the usual level). Results of this study indicate some pertinent features used for mating call discrimination and relate to neurophysiological studies of the treefrog auditory system. [Supported by NSF GB-41209.]
New time-domain analysis for complex animal vocalizations
Authors: Baker J. M.
Published: October 2005

The time-domain analysis introduced here incorporates a transformation of the acoustic waveform, which is functionally analagous to the physiological operation of phase-locking in the peripheral auditory nervous system. The acoustic signals are sampled and digitized at 20 kHz. Then the time of up-crossing is determined for each cycle in the waveform. Each cycle is individually evaluated with regard to three classes of time-domain parameters, namely, cycle frequency, cycle amplitude, and cycle microstructure. Some of this information is graphically presented by a Log Inverse Period (LIP) plot, a modification of the “instantaneous frequency” plot used by neurophysiologists. This plot is particularly useful for cleanly delineating the fine temporal structure and acoustic transients embedded in complex waveforms. The chief advantage of applying these time-domain techniques is their perfect time resolution. In contrast to the classical frequency-domain analyses, these time-domain analyses have no inherent bandwidth limitations. This fact is significant when the acoustic signal to be analyzed contains rapidly changing acoustic states believed to bear information. Especially where the auditory temporal resolution of humans may not be equal to that of the animals which communicate with complex vocalizations, precise temporal resolution is desirable. Analysis results are shown with examples drawn largely from avian vocalizations.
Behavioral evaluation of electrolytic lesions in the cochlea
Authors: Ryan Allen , Ferraro John
Published: October 2005

Pure-tone thresholds of normal chinchillas were determined by the shock-avoidance procedure. Under aseptic conditions, differential electrodes were placed in a given turn of the cochlea microphonic (CM) response was measured. The cochlea was then electrolytically lesioned with the method described by Ferraro and Dallos [J. Acoust. Soc. Am. 55, S78(A) (1974)]. Holes in the cochlear bone and the bulla were sealed, the contralateral cochlea destroyed, and the animal allowed to recover. Pure-tone thresholds were determined daily, until the observed shift to threshold had stabilized. Each animal was then anaesthetized and further electrophysiological data collected. Finally, the animal was sacrificed and the ear prepared for histological examination. Lesions required several days to stabilize. Correlations between behavioral, electrophysiological, and histological measures were good. [This research was supported by grants from NINDS.]
Discrimination of the intensities of tones by the parakeet (Melopsittacus undulatus)
Authors: Dooling R. J. , Saunders J. C.
Published: October 2005

Using avoidance conditioning and a modified method of limits, we measured the intensity difference limens of three parakeets at six sensation levels of a 2.86-kHz tone. At sensation levels above 50 dB, the average value of the relative difference limen is about 3.0 dB. Below a sensation level of 50 dB, the relative difference limen increases as the intensity of the tone is decreased. Intensity difference limens for the parakeet are 2–3 dB larger than similar measures for man and other mammals and suggest that the avian ear may not be as sensitive to intensity changes as are the ears of many mammals. [Supported by NIH Grant No. NS 03856 to the Central Institute for the Deaf.]
Studies of stereophonic hearing in the albino rat
Authors: Kelly Jack B.
Published: October 2005

The behavioral response of rats to paired sound sources has been determined using the conditioned suppression technique. The rats were reinforced for steadily licking a water spout and were then conditioned to withdraw from the spout when a stimulus change was detected. They were first trained to detect a change in the location of a single click from a speaker on the left to a speaker on the right of a test cage. They were then given further training with paired clicks from both speakers simultaneously, but with an intensity difference between left and right clicks. Thus the animals detected a change from “right speaker attenuated” to “left speaker attenuated.” The method of constant stimuli was then used to determine the least attenuation which could be detected. Estimates of threshold ranged from 2.8 to 4.2 dB for six animals. These results are compared with previous studies in which time differences rather than intensity differences were employed. Attempts to disrupt discriminations based on time differences by attenuation of the leading click (0 to 20 dB) proved unsuccessful. [Supported by the National Research Council.]
Wave properties of the human external ear and various physical models of the ear
Authors: Shaw E. A. G.
Published: October 2005

The wave properties of ten ears under blocked-meatus conditions have been determined using the progressive wave source described earlier (Denver, 1971). Physical models of the ear with four or more modes matching the average ear can be readily constructed provided that the crus helias and fossa are properly represented. These and other anatomical structures of the human ear are clearly associated with specific acoustic parameters. For example, the presence of the crus helias substantially reduces the frequency of the second mode (∼7.1 kHz) without affecting the fourth mode (∼ 12.1 kHz), which is largely dependent on the breadth of the concha. The third mode at 9.6 kHz, overlooked in earlier work, owes its existence to the fossa. Differences in coupling between the fossa and the cymba account for some of the intersubject differences observed at 10 kHz. Physical models of the complete human ear, useable up to 13 kHz, now appear feasible.
Pressure variation about the ear
Authors: Madaffari P. L.
Published: October 2005

The pressure variation in the region of the ear near the surface of the head was measured on KEMAR, a manikin especially constructed to duplicate the acoustic performance of the human head and body. The purpose of the experiment was to compare the response in various locations so that a single curve could be selected as being representative and used as the free field to human-head correction curve. Twenty-five curves on a 5 × 5 matrix with a 2-cm separation were recorded. Results indicated a variation with both frequency and spatial location. An attempt was made, therefore, to mathematically represent the data in a single equation using frequency and spatial location as variables This would serve as both a means for interpolation between curves and as a mathematical model for the acoustic interactions involved.
Simulating the behavior of the eardrum by the finite-element method
Authors: Funnell W. R. J. , Laszlo Charles A.
Published: October 2005

The eardrum has been modelled as a plane, isotropic membrane. We have included the effects of the embedded malleus, the frequency-dependent reactive and resistive forces exerted on the malleus by the rest of the ossicles, and the closed middle-ear air cavity. The variations in the form of the drum and malleus among different species have also been considered. The simulation is done using an adaptation of the finite-element method, in which the two-dimensional continuum is divided into a number of arbitrarily shaped triangles. The results obtained are compared to recent servations of eardrum vibration in the cat and guinea pig, as well as to middle-ear impedance data. [Supported by the Medical Research Council of Canada, the Quebec Department of Education, and the McConnell Foundation.]
Middle-ear transmission: effect of tympanic-membrane perforations
Authors: Kruger B. , Tonndorf J.
Published: October 2005

A small perforation (diam 1 mm) was placed in the postero-superior quadrant of the feline tympanic membrane (TM). Together with the voltages across the transducer, sound pressures (amplitudes and phases) required for a 10-μV CM response (round window) were measured over a range from 200 to 4000 Hz (a) separately in front and behind the TM, (b) in open and closed sound systems, and (c) before and after perforation. Sound-pressure changes in front of the TM after perforation revealed low-frequency losses, identical in shape and magnitude for open and closed systems: 12 dB/octave slopes below 1600 Hz. The voltage changes paralleled the sound-pressure changes only in the open system. Considerably larger voltages were required in the closed system, especially between 630 and 2000 Hz (maximally 18 dB at 1250 Hz). Furthermore, from the sound pressures measured on both sides of the TM, the net force acting upon the TM was calculated. The change in net force after perforation followed an approximate slope of only 6 dB/octave. Implications of these findings will be discussed. [Supported by NIH and HEW grants.]