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Aug 1971

Volume 50, Issue 2B, pp. 459-714

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Monaural Temporal Interactions

Harvey Babkoff and Samuel Sutton

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 459-465 (1971); (7 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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The monaural temporal interaction of transients was investigated by using a “mirror‐image” paradigm. A pair of unequal monaural clicks in which the less intense click precedes the more intense click by a given interpulse interval (backward masking) was discriminated from an identical pair of clicks in which the more intense click precedes the less intense click by the same interval (forward masking). The functions obtained show an initial rise in discrimination as a function of increasing interpulse interval (Δt) followed by a leveling of the curve at a high discrimination level for Δt's of up to approximately 8–10 msec, followed by a decrease in discrimination to approximately 12–15 msec, followed by a second increase in discrimination at Δt's greater than 15 msec. Comparing the discrimination of mirror‐image click pairs to forward masking and to backward masking alone indicates that monaural temporal interactions exist at intermediate interpulse intervals even though forward and backward masking are no longer evident. At these intervals, the discrimination is unrelated to the perception of temporal order. The shape of the functions and the Δt range over which it is evident are dependent upon the relative intensities of the two clicks comprising the pair. At longer intervals (greater than 15 msec), perception of temporal order is the dominant cue. Possible mechanisms are discussed.

Localization and the Law of the First Wavefront in the Median Plane

Jens Blauert

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 466-470 (1971); (5 pages) | Cited 8 times

Online Publication Date: 11 Aug 2005

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Test persons having their heads held in a fixed position were stimulated simultaneously from the front and the rear with identical broad‐band (music and noise) signals. Between the front and rear signal, a time delay of maximum ±880 μsec could be set. It was found that the direction of the sound sensation coincided with the angle of incidence of the first wavefront for delay times greater than about ±550 μsec. For smaller delay times, however, the direction of the sound sensation depends on the spectrum of the ear signals and can be predicted by applying the concept of directional bands.

Two‐ versus Four‐Tone Masking at 1000 Hz

Julius A. Canahl, Jr.

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 471-474 (1971); (4 pages)

Online Publication Date: 11 Aug 2005

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The threshold of a 1000‐Hz sinusoidal signal masked by two or four tones placed symmetrically around the signal was investigated as a function of the frequency difference (ΔF) between consecutive maskers as well as masker level. The results concur with previous evidence that (1) the masked threshold decreases linearly as ΔF increases for a two‐tone masker, and (2) the difference between masked thresholds is much greater than would be predicted on the basis of a simple summation of masking power or intensity. For small ΔF, the size of the difference (6–8.5 dB) did not vary greatly as level changed. At larger frequency separations, the highest levels produced more difference than lower levels. It was concluded that the difference between two‐ and four‐tone masking was (1) dependent upon the level of the maskers and the frequency separation between the maskers, and (2) independent of the masking contributed by the individual maskers.

Two‐Frequency Stimulation of a Cutaneous Mechanoreceptor

R. W. Cholewiak and J. F. Hahn

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 475-482 (1971); (8 pages)

Online Publication Date: 11 Aug 2005

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First‐order afferent single‐unit responses were obtained to sinusoidal one‐ and two‐frequency stimulation of the touch corpuscle in the cat. With two‐frequency stimulation, the amount of “masking” varied with the frequency and with the amplitude of the stimuli. In some instances, the addition of a second frequency decreased unit's rate of firing below the rate at which it was responding to a single frequency. The “masked” thresholds in the two‐frequency stimulation were predictable from the single‐frequency rate‐intensity functions for the unit, as were some of the decreased firing rates occurring during two‐frequency stimulation.

Effect of Unilateral Masking on the Lateralization of Binaural Pulses

Bruce E. Dunn

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 483-489 (1971); (7 pages)

Online Publication Date: 11 Aug 2005

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Pulsed tones were presented binaurally to subjects. Pure‐tone frequencies were 250, 1000, 4000, and 8000 Hz. Loudness was varied in each ear. One ear was continuously stimulated with an octave band of noise. The subject attempted to lateralize the pulsed tones in either the masked or unmasked ear. Results indicate that the pulsed tone in the masked ear had little or no effect on the lateralization until it exceeded masked threshold. Comparison of the loudness levels in the unmasked ear, required for lateralization in that ear with curves for loudness in noise, indicated that there may well have been centrally mediated loudness enhancement in the unmasked ear produced by the masking stimulus.

Cortical Responses of Awake Cat to Narrow‐Band FM Noise Stimuli

Edmund M. Glaser

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 490-501 (1971); (12 pages)

Online Publication Date: 11 Aug 2005

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Unanesthetized cats with chronically implanted epidural electrodes in the region of primary auditory cortex were stimulated with frequency‐modulated (FM) noises of varying bandwidths. The average evoked responses to these stimuli were compared with responses to tone bursts of the same center frequency and intensity. Two different noise stimuli were used: (a) bursts with rise/fall times the same as the tone burst; (b) transitions from tone to noise and back with transition times equal to tone‐burst rise time. It was found that: (1) the magnitude of the early response components increases with the bandwidth of the modulating noise, the relationship being fitted well by a power function; (2) there is a smaller power‐law type of increase in response amplitude with rms rate, noise bandwidth being held constant; (3) responses to transitions from tone to noise were quite marked, often exceeding burst responses, while responses to transitions from noise to tone were only rarely observed. These results are discussed in terms of the activity of single units in auditory cortex. A simple neuronal model is proposed to explain and unify the findings. The results are also compared with psychological loudness summation studies.

Aural Combination Tones and Auditory Masking

Donald D. Greenwood

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 502-543 (1971); (42 pages) | Cited 11 times

Online Publication Date: 11 Aug 2005

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In experiments mapping pure‐tone masked audiograms at low to moderate masker intensities, in which the signal is either a pure tone or a narrow band of noise, a notch appears on the high‐frequency side of the masker with a low point at a frequency one critical bandwidth from the masker frequency. The evidence indicates that this notch is caused by the detection, on the lower‐frequency side of the masker, of the cubic combination tones that are produced by the addition of masker and signal. At higher masker intensities, the difference tone may also be involved in accordance with the conclusions of Wegel and Lane [Phys. Rev. 23, 266–285 (1924)]. Similar results are obtained when a very narrow band of noise masks a pure‐tone signal. The implication that the same process that creates combination tones creates combination bands further supports other evidence as to the mechanical nature of combination tones. The immediate further implications that combination bands or tones arising from complex maskers should produce low‐frequency masking effects in the same way as external stimuli also has been tested and confirmed over a wide range of masker intensities and spectra. The maskers consisted of pairs of tones, tones plus narrow bands of noise, pairs of narrow bands of noise, and single bands of noise. Some of the implications of these results for cochlear physics and critical bandwidth are outlined.

Some Observations on Underwater Hearing

Donald A. Norman, Robert Phelps, and Fred Wightman

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 544-548 (1971); (5 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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We examine the relative role of bone conduction and tympanic conduction in underwater localization and detection by humans. Basically, we placed divers underwater and covered up various parts of the skull and outer ear with neoprene foam (an insulator of sound) and measured thresholds and localization accuracy. Sound conduction through the ear canal (tympanic conduction) appears to play only a minor role in the detection of sounds 1000 Hz and higher. Yet conduction through the canal appears to be very important for localization.

Depth of Sequential Auditory Information Processing: III

Irwin Pollack

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 549-554 (1971); (6 pages)

Online Publication Date: 11 Aug 2005

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The auditory discrimination of quasiperiodic finite‐state sequences was examined for four alternative methods of encoding binary states: pulse polarity, the occurrence or nonoccurrence of a pulse, the ear to which a pulse was delivered, and interpulse interval. Except for relatively poor sensitivity for pulse polarity within low‐frequency pulse trains, all four methods of encoding led to nearly comparable discrimination. The specific method of encoding finite‐state information is apparently not critical for the auditory discrimination of quasiperiodic sequences. For each method of encoding, best performance is achieved for pulse trains of many short interpulse intervals. The detection of sequential constraints is apparently carried out primarily by means of information within high‐frequency auditory channels. Under selected conditions, the three methods based upon binary coding of periodic clock intervals are more sensitive than that based upon pulse interval coding. The auditory analysis for partial periodicities is apparently carried out over relatively constant temporal sampling intervals.

Spectral Basis of Auditory “Jitter” Detection

Irwin Pollack

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 555-558 (1971); (4 pages)

Online Publication Date: 11 Aug 2005

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The contribution of high‐frequency and of low‐frequency spectral information to the detection of temporal variability or “jitter” of auditory pulse trains was examined. Pulse trains were subjected to low‐pass, to bandpass, and to high‐pass filtering. Jitter discrimination for low‐frequency pulse trains is substantially poorer than for high‐frequency pulse trains. The reason is not that the low spectral frequencies within low‐frequency pulse trains interfere with jitter discrimination, nor that low‐frequency pulse trains contain insufficient high‐frequency spectral information. Rather, it appears that the spacing between high‐frequency spectral components of low‐frequency pulse trains is too close to permit effective resolution.

Recognition of Phase Changes in Octave Complexes

Carolyn A. Raiford and Earl D. Schubert

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 559-567 (1971); (9 pages)

Online Publication Date: 11 Aug 2005

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It is generally accepted that even trained listeners have great difficulty perceiving static phase differences in two‐tone stimuli. However, if the waveform with the changed phase is “inserted” into a longer signal, most observers can reliably detect whether the change has been made. This study used this “insert” method to compare a series of waveforms which differed from a standard waveform by 15°, 30°, 45°, 60°, 75°, 90°, 102°, 150°, and 180°. The standard waveform was A sin2π(250)t+B sin[2π(500)t+90]. By the time the comparison waveform differed from the standard by 60°, observers could perceive the phase change correctly on over 75% of the trials. An attempt is made to identify the cues which permit the system to recognize such nonenvelope phase changes, but the data do not permit positive identification of these cues.

The Effect of Interaural Signal‐Frequency Disparity on Signal Detectability

Donald E. Robinson

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 568-571 (1971); (4 pages) | Cited 3 times

Online Publication Date: 11 Aug 2005

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The function relating interaural signal‐frequency difference (Δf) to masking‐level difference (MLD) was obtained using a two‐alternative forced‐choice procedure. The masking noise was presented diotically (N0) at a spectral level of 45 dB. The signals to the two ears were gated simultaneously and each was 1 sec in duration. For all Δf conditions, the signal to one ear was 400 Hz. In separate sessions, signal frequencies of 406, 420, 440, 460, 500, 550, 600, and 700 Hz were presented to the other ear. The MLD was found to decrease from about 10 dB at Δf = 6 Hz to about 7 dB at Δf = 150, 200, and 300 Hz. This limiting value of the MLD is also the MLD for the 400‐Hz signal presented monaurally (N0‐SM).

Intracochlear Potential Recorded with Micropipets. I. Correlations with Micropipet Location

H. S. Sohmer, W. T. Peake, and T. F. Weiss

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 572-586 (1971); (15 pages)

Online Publication Date: 11 Aug 2005

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Glass micropipets (filled with 2M KCl) were inserted through the round window of anesthetized cats. As the micropipet was advanced through the cochlea, we recorded (1) depth of penetration of the micropipet, (2) resistance of the micropipet, (3) dc potential, (4) magnitude and phase of the fundamental component of the response to tones, and (5) magnitude of response from a stationary electrode. At the end of the experiment, the micropipet was cemented to the temporal bone and kept in place during histological preparation. From examination of 54 sectioned cochleas, we determined which structures in the cochlea had been penetrated by the micropipet. The inaccuracies involved in determining the location of the tip of the micropipet with this technique made it impossible, however, to associate recorded electric events with structures as small as individual cells. The average dc potentials (referred to scala tympani) were scala media, +103 mV; scala vestibuli, −2 mV. Occurrences of large negative dc potential (−50 to −100 mV) were frequently recorded between scala tympani and scala media; these potentials were generally stable for a short time only (<10 sec).

Intracochlear Potential Recorded with Micropipets. II. Responses in the Cochlear Scalae to Tones

T. F. Weiss, W. T. Peake, and H. S. Sohmer

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 587-601 (1971); (15 pages)

Online Publication Date: 11 Aug 2005

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Cochlear potential response to tones was recorded with glass micropipets in all three scalae of the basal region of the cochlea of 32 anesthetized cats. The fundamental component of the potential (referred to an electrode on the chin) is represented by complex amplitudes EV, EM, ET, where the subscripts denote the scala in which the potential was recorded. EM/EV is relatively independent of stimulus level and frequency and has a magnitude of approximately 1.0–1.8 and an angle of approximately zero. EM/ET is dependent on stimulus level and frequency. Its magnitude is as large as 10 at frequencies below 300 Hz and decreases for higher frequencies. The angle of the ratio increases from approximately zero at 100 Hz to a value between +90° and +180° above 1000 Hz. Similar results were obtained from two cochleas with a severed auditory nerve, which implies that these results apply to the cochlear microphonic (CM) potential response to tones. An electric‐network model of the spatial distribution of CM is analyzed.

Intracochlear Potential Recorded with Micropipets. III. Relation of Cochlear Microphonic Potential to Stapes Velocity

T. F. Weiss, W. T. Peake, and H. S. Sohmer

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 602-615 (1971); (14 pages) | Cited 3 times

Online Publication Date: 11 Aug 2005

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Cochlear potential response to tones of known sound pressure was recorded with micropipets from the three scalae of the basal region of the cat's cochlea. By using an average transfer function for the middle ear of the cat, we calculated the complex ratio of cochlear potential response to stapes velocity, E/Ẋs. ∣E/Ẋs∣ is approximately constant for a decade or two of frequency in the range 100 Hz–10 kHz. As frequency decreases below 100 Hz, ∣E/Ẋs∣ decreases and ∠ (E/Ẋs) becomes positive. As frequency increases above a few kilohertz, ∣E/Ẋs∣ decreases and ∠ (E/Ẋs) becomes negative. The dependence of E/Ẋs on frequency for potential in scala media is somewhat different from that in scala tympani. To interpret our measurements, we postulate a model that relates cochlear microphonic potential, V(x, f), to stapes displacement, Xs(f), in terms of three cascaded transformations: mechanical, H(x, f); transduction, T(f); and electric, W(x,f). According to the model, the transfer function can be expressed as V(x,f)/Xs(f)  =  H(x,f) ⊗[T(f)W(x,f)], where ⊗ denotes convolution on x (distance from stapes). Special cases of this relation are discussed and compared with experimental data.

Auditory Critical Bandwidth for Short‐Duration Signals

R. Srinivasan

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 616-622 (1971); (7 pages)

Online Publication Date: 11 Aug 2005

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A new experimental technique is proposed to estimate the bandwidth of the auditory‐filter mechanism when detecting short‐duration sinusoidal signals masked by wide‐band noise. The experiment requires the use of only wide‐band noise and two signals of the same frequency, power, and duration, but having different energy density spectra. The special feature of this technique is that the subjective impression made by the masking noise remains the same throughout the experiment unlike experiments that employ progressively changing bandwidths of the masking noise. This avoids the possibility of any change in the parameters of the detection mechanism due to changes in the subjective impression made by the noise. The analysis makes use of the energy detection model consisting of a bandpass filter, square‐law rectifier, and an integrator to describe the auditory detection system. A rectangular filter and an LCR single tuned filter have been considered in the analysis. From the experimental results it is found that the rectangular filter provides a better description of the auditory‐filter mechanism than the single tuned filter. The bandwidth estimates show an increase with signal frequency. The results also support the idea of a minimum auditory‐filter bandwidth for a continuous or long‐duration signal and increased values for short‐duration signals depending on the signal duration.

Detection of Binaural Tones as a Function of Masker Bandwidth

Frederic L. Wightman

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 623-636 (1971); (14 pages) | Cited 8 times

Online Publication Date: 11 Aug 2005

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Several experiments have shown that the difference in masked threshold between signals in the S0M0 and SπM0 conditions, the masking‐level difference (MLD), increases markedly with reduction in masker bandwidth. However, there are two inconsistencies this result. First, apparently similar experiments conducted by different laboratories have produced quite different results. Second, the MLD‐bandwidth relation appears to be nonmonotonic; with very narrow‐band maskers (e.g., a pure tone), the MLD is often small. The results of the experiments reported here suggest that these problems arise from anomalies in the S0M0 data alone. It appears that in some narrow‐band masking conditions the SπM0 threshold is spuriously low, resulting in deceptively small MLDs. The proposed explanation is that observers detect signal energy which falls outside the masker band, for when this energy is attenuated by filtering the signal, the S0M0 threshold rises dramatically. The SπM0 threshold is unaffected by this manipulation. With specially filtered signals, the nonmonotonicity of the MLD‐bandwidth relation is considerably reduced, and MLDs with continuous pure‐tone maskers are observed which are nearly as large as the largest narrow‐band noise MLDs ever reported (28 dB).

Speech Analysis and Synthesis by Linear Prediction of the Speech Wave

B. S. Atal and Suzanne L. Hanauer

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 637-655 (1971); (19 pages) | Cited 19 times

Online Publication Date: 11 Aug 2005

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We describe a procedure for efficient encoding of the speech wave by representing it in terms of time‐varying parameters related to the transfer function of the vocal tract and the characteristics of the excitation. The speech wave, sampled at 10 kHz, is analyzed by predicting the present speech sample as a linear combination of the 12 previous samples. The 12 predictor coefficients are determined by minimizing the mean‐squared error between the actual and the predicted values of the speech samples. Fifteen parameters—namely, the 12 predictor coefficients, the pitch period, a binary parameter indicating whether the speech is voiced or unvoiced, and the rms value of the speech samples—are derived by analysis of the speech wave, encoded and transmitted to the synthesizer. The speech wave is synthesized as the output of a linear recursive filter excited by either a sequence of quasiperiodic pulses or a white‐noise source. Application of this method for efficient transmission and storage of speech signals as well as procedures for determining other speech characteristics, such as formant frequencies and bandwidths, the spectral envelope, and the autocorrelation function, are discussed.

Signal Processing for a Cocktail Party Effect

O. M. Mracek Mitchell, Carolyn A. Ross, and G. H. Yates

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 656-660 (1971); (5 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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A binaural listener has the ability to concentrate on speech from a particular location while suppressing speech from other locations (binaural “cocktail party” effect). In some communication situations where sounds are picked up by a microphone system for transmission on a single path to a remote listener, it would be desirable to preprocess the signals to achieve a similar effect. We describe a class of nonlinear processes for the outputs of an array of microphones which emphasize speech coming from a particular (on‐center) location in a background of other sounds. These processes completely eliminate any off‐center impulsive noise which is nonoverlapping at the four microphones. Results of processing outputs of real and computer‐simulated microphone arrays for speech and noise signals are described. Under anechoic conditions, the processing results in reproduction of the on‐center speech without change, and in distortion and attenuation of an off‐center speech source. The distortion produced by the processing appears to be an important factor in subjective suppression of the off‐center source.

Automatic Formant Tracking by a Newton‐Raphson Technique

J. P. Olive

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 661-670 (1971); (10 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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A wide range of speech investigation requires natural‐sounding synthetic speech whose individual acoustic features—such as pitch, formants, and duration—can be varied independently. This study describes an algorithm for obtaining such synthetic speech by automatically analyzing natural speech to obtain data for the control of a formant synthesizer. The lowest three formants of natural speech were determined by finding simultaneous solutions of the least‐square‐fit equations by means of a Newton‐Raphson technique. The data obtained from this formant analysis, together with pitch data which were also obtained automatically, were used for controlling a computer‐simulated formant synthesizer. This algorithm was tested for various sentences and speakers, and the synthesized speech was found to be of good quality.

Estimate of the Inherent Channel Capacity of the Ear

Edith L. R. Corliss

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 671-677 (1971); (7 pages)

Online Publication Date: 11 Aug 2005

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The growth of intelligibility of speech stimuli as a function of level above hearing threshold can be computed from the “circuit parameters” of the hearing mechanism by applying Shannon's concepts of channel capacity, equivocation, and “bits.” In the ear, the unit of response is an effective “least count,” derived from experimental data on hearing by means of the equations for a model resembling a frequency‐selective circuit [E. Corliss, J. Acoust. Soc. Amer. 41, 1500–1516 (1967)]. The model predicts that the number of least counts available rises as the one‐fourth power of the signal intensity above threshold. Experimentally, this growth rate is observed for the intensity‐resolving power of the ear. Approximately the same power law is observed for the sensation of loudness. The model ascribes both effects to the same mechanism. From the observed integration time of the ear, the model predicts the rate at which transitions of single counts can be detected. From the counting rate and the integration time, the channel capacity available at the ear and its increase with level above threshold can be computed. The information content of speech as a source function is evaluated from the rate at which single “distinctive features” of speech phonemes are produced. Intelligibility scores can be predicted from the ratio between the rate at which information is being produced by the source and the rate at which the receptor can accept the source material. The scores predicted agree fairly closely with experimental data on random‐word and random‐syllable intelligibilities. This agreement shows that the listener need recognize no more than a single distinctive feature of each phoneme to display the recognition functions that have been observed. From a theorem of C. Shannon [Inform. and Control 1, 6–25 (1957)] relating code length and error probability, one can show that the channel capacity required for polysyllabic words is lower than the channel capacity required for monosyllabic words because the duration of correlated utterance may be taken as a code length. Evidently, contextual effects are not prominent in the intelligibility of random‐word lists; the hearing process involved is primarily recognition of groups of sounds; meaning is secondary. The results also lead to the inference that a direct relation may exist between channel capacity and perceived loudness when speech is transmitted over a broad‐band system, and suggest that loudness functions for impaired ears might prove to be correlated with intelligibility functions.

Investigation of the Timing of Velar Movements during Speech

Kenneth L. Moll and Raymond G. Daniloff

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 678-684 (1971); (7 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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High‐speed cinefluorographic films were taken of four normal subjects speaking English sentences containing various combinations of nasal consonants (N), consonants (C), and vowels (V) at normal speaking levels and rates of production. Word and syllable boundaries were designed to fall across the sequences in various ways. From frame‐by‐frame tracings, measures of velar movement and velopharyngeal opening were made. Results indicate extensive anticipatory coarticulation of velar movement toward velopharyngeal opening in CVN and CVVN sequences such that velar movement toward opening began during the approach to the initial vowel in all cases and some velopharyngeal opening was observed on all vowels. For NC and NCN sequences, velar movement toward closure for the consonant usually began during the preceding nasal such that some velar closure was observed during all plosive and fricative consonants used. These results directly contradict the hypothesis that a CV‐type syllable is the minimal unit of coarticulation and production. The data are consistent with the predictions of a model which assumes phone‐sized input unit and which incorporates a “look ahead” mechanism whereby an articulatory feature can be systematically anticipated prior to the occurrence of the phone with which that feature is associated.

Jaw Movements under Delayed Auditory Feedback

Harvey M. Sussman and Karl U. Smith

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 685-691 (1971); (7 pages)

Online Publication Date: 11 Aug 2005

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The spatial and temporal movement characteristics of the jaw while subjects spoke under delayed auditory feedback was investigated. An analog‐digital‐analog computer system was used to instrument the delayed speech signal and a strain‐gauge transducer was used to monitor the movements of the jaw. The speech sample consisted of the high, middle, and low front vowels /i/, /ɛ/, and /æ/ embedded in the carrier phrase: “That's a CVC a month.” Measurements were made of maximal jaw opening distances during the vowel segment, jaw activity duration throughout the utterance, and jaw velocity during the opening and closing phases of the vowel gesture. The disruptive effects caused by the temporally delayed speech signal were found to be specific to the: (1) magnitude of the delay interval; (2) vowel context; and (3) space‐time dependent variable being measured. Increased jaw‐opening excursions were found, depending upon the vowel context, and a positional target overshoot was noted at the 0.3‐sec delay interval, especially for the midvowel /ɛ/. The time period of active jaw articulation was considerably lengthened at the 0.1‐sec delay interval, especially for the high vowel /i/. The jaw velocity measures showed increased movement rates for both opening and closing vowel gestures as a function of vowel openness.

Mechanism of Absorption of Ultrasound in Liver Tissue

H. Pauly and H. P. Schwan

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 692-699 (1971); (8 pages) | Cited 1 time

Online Publication Date: 11 Aug 2005

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The dominant part of the acoustic absorption of liver tissue and its components results from macromolecular relaxation processes. The absorption has been investigated over the frequency range 1–10 MHz and the following results have been obtained: (1) About two‐thirds of the total absorption arises at the macromolecular level, with the remainder caused by macroscopic structure. (2) The specific absorption of tissue macromolecules, as expressed in absorption per weight percent, varies considerably from one biopolymer to another. (3) The absorption is related to the structure of the biological macromolecule or its hydration and changes with heat denaturation and pH. (4) A similar frequency dependence results for all materials investigated. This dependence is to be expected if one assumes that the molecular processes of absorption are characterized by a broad spectrum of relaxational time constants and activation energies extending over a range of at least 1:7.

Auditory Sensitivity and Song Spectrum of the Common Canary (Serinus canarius)

Robert J. Dooling, James A. Mulligan, and James D. Miller

J. Acoust. Soc. Am. Volume 50, Issue 2B, pp. 700-709 (1971); (10 pages)

Online Publication Date: 11 Aug 2005

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The auditory sensitivity of one strain (Belgian Waterschlager) of common canary (Serinus canarius) was measured by behavioral audiometry. Four birds were trained by instrumental avoidance conditioning in a double‐grille cage, and their thresholds for pure tones (0.25–9.0 kHz) were measured. Auditory sensitivity is greatest between 2.0 and 4.0 kHz with a possible maximum at 2.8 kHz, declines about 15 dB/oct for frequencies below 2.0 kHz, declines about 25 dB between 4.0 and 8.0 kHz, and 13 dB between 8.0 and 9.0 kHz. The acoustic power in the songs and calls of the Belgian Waterschlager falls primarily in the range 1.8–4.5 kHz as do the critical frequencies of a substantial proportion of the neural units in the cochlear nucleus of the canary. Thus, the auditory sensitivity and the neural machinery of the peripheral auditory system appear to be matched to the long‐term‐average power spectrum of the songs. In addition, these facts are compared to those for other birds and mammals, and speculations as to some of the selective pressures that influenced the evolution of hearing are presented. Certain relevant problems of the biophysics of hearing are also discussed.
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