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A two-part study was conducted to determine the sources of variation in nasalance scores derived from the Nasometer. In Study #1, a function generator was used as a signal source to calibrate and input sine and square waves directly into the Nasometer. Ten stimuli ranging from 105 to 330 Hz in 25 Hz increments were evaluated. In Study #2, the same signal source and an amplified loudspeaker were used to calibrate and present square waves to the Nasometer via five different sets of microphones. The sound pressure level of all stimuli was maintained at 88 dB. Each microphone set was calibrated using the 105 Hz signals. Results from Study #1 indicated consistent nasalance scores across all frequencies (i.e., all scores were within 2% of calibration). Results from Study #2 demonstrated deviations greater than 2% from calibration as a function of frequency for all five sets of microphones. The smallest deviation was 5%, whereas the largest deviation was 14%. We suggest that the variation in nasalance as a function of stimulus frequency may be due to a mismatch in the sensitivity of microphones (i.e., different frequency response characteristics). It is further suggested (a) that individual investigators determine the response characteristics of their microphones and (b) that relatively small variations in nasalance scores (i.e., 5–14%) either within or across speakers be interpreted with caution.

A hearing aid with multiple frequency responses was simulated by programming an equalizer to produce spectral tilt factors of -6, 0, and +6 dB/octave over the frequency range from 0.25 kHz to 4 kHz. Listeners with normal hearing matched the loudness of signals (speech and white noise) that were shaped by these different equalizer settings and delivered via an insert earphone. All signals with spectra that were tilted, either negatively or positively, were perceived as louder than untilted signals. The general pattern of loudness matching was similar across subjects, and intrasubject judgments were found to be highly transitive. A measure of signal power was found to account only moderately well for the individual data. Preliminary evidence from a follow-up study using tilt factors less severe than those used in the main experiment suggests that loudness differences are roughly proportional to the degree of spectral tilt. The incorporation of level corrections approximating those necessary to achieve equal loudness is recommended in the fitting of programmable hearing aids.

Recent investigations (Cranford, Boose, & Moore, 1990a,b; Moore, Cranford, & Rahn, 1990) studied the ability of normal adult subjects to localize sounds under conditions that elicit the Precedence Effect. In different tests, subjects were required either to report the perceived location of a stationary fused auditory image (FAI) or track the apparent motion of a "moving" FAI. Movement of the FAI was simulated by incrementally varying the delay between pairs of clicks presented, one each, from two matched loudspeakers placed on opposite sides of the listener. In the present study, groups of normally developing children, ranging in age from 6 to 12 years of age, were tested with these two procedures. Although subjects performed at normal adult levels with the stationary FAI test, a significant age-related trend was observed with the moving FAI test. The younger children exhibited poorer tracking performances than did the older children. These results provide evidence that significant changes in binaural temporal processing abilities may occur in the early childhood years.

This study examined the attenuation characteristics of five FM system sound delivery options for a group of 10 adults and 15 children (5–13 years). Sound delivery options included a tube-fitting, lightweight headphones, a CROS earmold with tubing, a CROS earmold with a snap-ring, and a standard snap-ring earmold with a vent. Attenuation was defined as the difference between probe-tube microphone measures of the ear canal resonance and the SPL in the ear canal with each sound delivery option in place. A statistically significant but clinically inconsequential difference in attenuation for the CROS earmold with tubing was noted between adults and children. No significant differences in attenuation for any of the other sound delivery options were noted between adults and children. An investigation of the relationship between magnitude of attenuation and percentage of the ear canal occluded suggests that degree of occlusion is a major factor in determining degree of attenuation provided by a particular sound delivery option. Results also indicate that significant attenuation of high-frequency signals can occur with earmolds commonly considered nonoccluding. Caution should be used in fitting hearing aids or FM systems to individuals with normal high-frequency hearing sensitivity to prevent attenuation of unamplified high-frequency speech information.

Clinical measurements of the loudness discomfort level (LDL) are generally performed while the subject listens to a particular stimulus presented from an audiometer through headphones (AUD-HP). The assumption in clinical practice has been that the sound pressure level (SPL) corresponding to the sensation of loudness discomfort under AUD-HP conditions will be the same as that corresponding to LDL with the hearing aid. This assumption ignores the fact that the distortion produced by a saturating hearing aid could have an influence on the sensation of loudness. To examine these issues, 5 hearing-impaired subjects were each fit with four linear hearing aids, each having a different saturation sound pressure level (SSPL90). Probe-tube microphone measurements of ear canal SPL at LDL were made while the subjects listened to continuous discourse in quiet under aided and AUD-HP conditions. Also using continuous discourse, real-ear coherence measures were made at various output sound pressure levels near LDL. All four hearing aid types produced mean LDLs that were lower than those obtained under AUD-HP conditions. Those hearing aids with higher SSPL90 produced significantly higher LDLs than hearing aids with lower SSPL90. A significant negative correlation was found between real-ear SPL and real-ear coherence. Quality judgments made at LDL indicated that sound quality of hearing aids with higher SSPL90 was preferred to that of hearing aids with lower SSPL90. Possible fitting implications regarding the setting of SSPL90 from AUD-HP LDL measures are discussed.

In the absence of national or international electroacoustic standards for the evaluation of Frequency Modulated (FM) amplification systems, it becomes important to know the variability one may expect across similar models. Evaluation of thirty FM systems of the same model obtained from three different educational sites was performed to determine the variability that may occur as a result of the receiver, lapel microphone, or neckloop. There was a range as great as 20 dB in high frequency average saturation sound pressure level and equivalent input noise across receivers, lapel microphones, and neckloops. These results highlight the need for regular electroacoustic monitoring of not only the FM transmitter and receiver, but also the individual components, such as the lapel microphone and the neckloop.

In recent years, a number of commercially available systems have been developed to analyze the electroacoustic characteristics of hearing aids. In addition to pure-tone signals, these systems often use a wide variety of complex signals such as broadband noise, clicks, and multitonal complexes. In this paper, a number of practical and theoretical issues concerning the use of pure-tone and complex signals in the evaluation of hearing-aid characteristics are described. The circumstances under which discrepancies in estimated gain and maximum output might occur using these two types of signals are described and the clinical implications of these differences are discussed.

The monosyllabic word discrimination of normal and hearing-impaired children was evaluated in situations selected to simulate acoustical conditions in current educational environments. All listeners were tested in a high-fidelity (loudspeaker-aided) condition under 12 combinations of reverberation and noise. Hearing-impaired subjects were also evaluated in the same 12 conditions while listening through a monaural hearing aid. Performance of the normal-hearing group was superior to the hearing-impaired listeners in all environments. Results suggest that classroom acoustics should be considered a critical variable in the educational achievement of children.

The chest wall has been treated as a two-part kinematic system comprised of the rib cage and diaphragm-abdomen in parallel, and wherein the volume displaced by each part is linearly related to the motions of points within it. Using measurements of changes in anteroposterior diameters of the rib cage and abdomen, we studied subjects in upright and supine postures during several respiratory maneuvers and utterance tasks. Results are charted in relative motion diagrams (rib cage vs abdomen), which include the relaxed configuration of the chest wall and departures therefrom during utterances. For conversation, reading, and singing, lung volume events were restricted to the midvolume range and were dependent upon body posture and utterance loudness. Relative volume contributions of the two parts differed for subjects and utterances and ranged from all rib cage displacement to all abdominal displacement. During utterances, the chest wall was distorted from its relaxed configuration, and differently so in the two postures studied. Potential mechanisms responsible for these distortions are discussed. We conclude that the distortions observed constitute a “volume platform” or posturing of the chest wall, off of which the speaker produces speech but does not significantly further distort the system in providing the changes in driving pressure required for typical utterances.

This experiment examined the effects of continuous and contingent white noise masking upon the speech of 10 stutterers. The methodology equated the frequency of masking opportunities during (1) noise onset made contingent upon phonation, and (2) noise cessation made contingent upon phonation. A continuous noise condition and a no-noise control condition were also included. All noise conditions produced significantly less stuttering than the no-noise control condition. The three masking conditions yielded approximately the same reductions in the frequency of stuttered responses. Fluency enhancement by the various masking conditions could possibly be explained by reflex functions of the middle ear muscles.

EMG activity from the inferior constrictor and cricopharyngeus muscles and the voice signal were obtained from 18 laryngectomized male subjects as they produced isolated vowels using alaryngeal phonation. To inflate the esophagus prior to phonation, all subjects but one demonstrated a similar muscle pattern: either one or both muscles studied showed a burst of activity at the moment of inflation. The remaining subject had a muscle pattern during inflation that was identical to a post-laryngectomy swallowing pattern. No typical or modal muscle patterns were found for subjects during the phonatory portion of the alaryngeal voice task. Consistency of pattern within each subject was extremely high during a given procedure and on repeated procedures. The findings suggested that poor talkers had less control of differential muscle contraction than did the adequate talkers and that each laryngectomized talker adopts a phonatory method that is unique to him and consistent with his postoperative anatomy and physiology.

A series of five experiments was conducted to investigate the effects of spatial separation of speakers on the intelligibility of spondaic and PB words in noise and the identification of synthetic sentences in noise and competing message. Conditions in which the spatial location of the speakers produced interaural time differences ranked highest in intelligibility. The rank order of other conditions was dependent on the S/N ratio at the monaural near ear. Separations of only 10° between the speech and noise sources resulted in measurable changes in intelligibility. The binaural intelligibility scores were enhanced substantially over the monaural near ear results during conditions where an interaural time difference was present. This result was observed more effectively when spondaic words or sentences were used rather than PB words. The implications of this result were related to the interaural time difference and the frequency range of the critical information in the primary message. Although the initial experiments were facilitated by recording through an artificial head, almost identical results were obtained in the final experiment when subjects were tested in the sound field.

An electronic integrator, for use with an airflow meter, was designed to overcome the disadvantages of respirometers when used for research in speech breathing physiology, such as the inherent inertia which may produce an undesirable impedance to the expiratory airstream. An effort was also made to overcome certain limitations of integrators which restrict their use. The experimental instrument has minimal drift and a nearly infinite discharge time constant. Also, it may provide for the simultaneous recording of partitions of lung volume and airflow rate.

This manuscript examines the issue of potential decline in hearing sensitivity due to hearing aid usage through an analysis of data obtained from the temporary threshold shift (TTS) paradigm. Following a critique of the traditional measures of TTS, the concept of integrated TTS (ITTS) is reviewed and data on hearing-aid-induced ITTS are presented. In addition, a series of equations relating permanent threshold shift (PTS) to a recently developed measure of noise dose (Dn) is derived and predictions for hearing-aid-induced PTS are made. Recommended gain settings established to protect the hearing of a person wearing a hearing aid from further decline following various durations of hearing aid usage are also provided.

This experiment determined the effects of amplitude compression on speech intelligibility when both a target speech signal and a competing message were whitened and amplitude compressed. The target CNC discrimination words were electrically mixed with a competing message composed of five talkers. This composite signal was presented to normal hearing subjects in four ways: unmodified, whitened, whitened plus 3:1 amplitude compression and whitened plus 10:1 amplitude compression. Discrimination functions were obtained for the CNC material by varying the signal-to-competition ratio. The unmodified and whitened speech yielded comparable discrimination functions, but reduced discrimination scores were obtained with the whitened plus compressed speech. However, the reduction in speech discrimination for the whitened plus compressed speech was slight and was most evident when the target signal and the competing background were at the same intensity.

The aim of this study was to assess the effect of low-frequency amplification on speech recognition performance by hearing-impaired listeners. Consonant identification performance by subjects with flat hearing losses and high-frequency hearing losses was assessed in three different hearing aid conditions, in quiet and noise. The experimental hearing aids all provided extra high-frequency amplification but differed in the amount of low-frequency amplification. The results showed that listeners with flat hearing losses benefited by low-frequency amplification, whereas subjects with high-frequency hearing losses exhibited deteriorating scores in conditions with greatest low-frequency amplification. Analyses of phonetic feature perception and individual consonant recognition scores revealed subtle interactions between hearing loss configuration and amplification contour.

In the present study, 24 stutterers and 24 matched nonstutterers spoke spontaneously under conditions of quiet, noise (60, 80, 100 dB SPL re .0002 dynes/cm 2 ), and amplified sidetone (0, + 10, + 20 dB SPL re .0002 dynes/cm 2 ). All 24 stutterers and 23 of the 24 nonstutterers experienced a decrease in vocal intensity from quiet to 20 dB amplified sidetone. All the nonstutterers and 23 of the stutterers experienced an increase in vocal intensity, from quiet to 100 dB noise. As a group, the stutterers experienced a significant decrease in stuttering in the noise conditions; however, with regard to the amplified sidetone conditions, only those stutterers who experienced noise first showed a significant reduction in stuttering during amplified sidetone.

Nine subjects with prelingually acquired, sensorineural, hearing loss were given a three-interval, forced-choice, test of speech pattern contrast perception under two amplification conditions. The first involved adjustment of the low and high frequency outputs of a two-channel Master Hearing Aid to each subject's highest comfortable level, but without compression of the short-term dynamic range of the signal. The second involved the additional compression of a 30 dB input range into the subject's dynamic range of hearing, as measured by the difference between speech awareness threshold and highest comfortable level, in each of the two channels. One of the subjects performed much better with compression than without. Among the other eight, however, there was a small but significant reduction of performance when compression was introduced. It is proposed that the one positive result is due to the increased audibility of speech cues made possible by amplitude compression. It is further proposed that the negative results are due mainly to the distortions of time-intensity cues introduced by amplitude compression. The results suggest that, in terms of potential access to meaningful speech cues, the addition of amplitude compression, to an otherwise optimize signal, is unnecessary, or even detrimental, for most profoundly deaf subjects, but could be beneficial for some.

An essential building block for any high-fidelity hearing aid is an amplifier-transducer-coupling combination that does not audibly degrade the sound, that is, provides high-fidelity sound reproduction as judged by someone with normal hearing. To demonstrate that such a combination is possible, two binaural pairs of hearing aids were assembled using available hearing aid transducers and electronic components, one pair of Over-The-Ear hearing aids with 8-kHz bandwidth and one pair of In-The-Ear hearing aids with 16-kHz bandwidth. Objective insertion-gain measurements on these aids, obtained with a KEMAR manikin in a diffuse sound field, revealed a frequency-response accuracy comparable to that available in expensive high-fidelity loudspeakers. Subjective fidelity ratings obtained from three groups of listeners judging prerecorded A-B-A comparisons (made from equalized eardrum-position microphones in a KEMAR manikin) produced a similar conclusion. We conclude that the important question for hearing aid research is no longer "What can a hearing aid be designed to do?" but "What should a hearing aid be designed to do for the hearing impaired?"

Two hearing aids and three telephone receivers were used to compare electroacoustic and word intelligibility data for hearing aid microphone reception versus telephone/telephone and telephone/telecoil reception. Measurements included test reference position gain and frequency response characteristics (Experiment 1) and word intelligibility scores (Experiment 2). The results indicated that performance was similar for hearing aid microphone reception and telephone/telephone reception. Telephone/ telecoil performance varied, depending on which telephone receiver was involved, but was poorer than hearing aid microphone performance. Recommendations were made concerning the selection of hearing aids and the management of hearing aid users relative to telephone/telecoil communication.

Input-output (I/O) functions of hearing aids were measured in response to a 2000-Hz tone burst, having 0.5 ms rise/fall time and 10 ms duration. I/O functions, measured with a hearing-aid analyzer, served as reference conditions. Heating-aid outputs at onset and during the steady-state portion of the waveform differed; these differences often depended upon stimulus rate. The relation between onset and steady-state estimates of output were not always predictable from hearing-aid attack and release times. These findings indicate that the steady-state output limitation characteristics of hearing aids cannot be estimated from their onset responses. In turn, this suggests that ABR measurements may not provide accurate estimates of the compressive characteristics of hearing aids.

The Fairbanks Rhyme Test was presented to 12 hearing-impaired children, ages 10–16, to obtain information on the intelligibility of initial consonants in monosyllabic words. All five forms of the rhyme test were presented randomly at five sensation levels (10, 15, 20, 30, and 40 dB) to generate intelligibility gain functions. The degree of amplification produced differential effects on the intelligibility of the 14 initial phonemes studied. Increases in gain substantially improved the intelligibility of the phonemes /d, b, k, p, t/. The phoneme /g/ showed a substantial intelligibility gain but not in a similar fashion to the rest of the stop phonemes. Although the nasal consonants /m/ and /n/ showed equivalent intelligibility gains to that of the stops, they achieved their maximum intelligibility level within the intensity range studied. The phoneme /l/ showed an intelligibility gain function similar to that of the nasal phonemes. The phonemes /s/ and /r/, while highly intelligible throughout all conditions, showed only slight improvement as a function of intensity. Finally, intensity had no systematic effect on the intelligibility of /f/, /h/, and /w/.

Thirty adult subjects with high frequency hearing loss were given speech discrimination tests under conditions of flat and selective amplification. It was hypothesized that performance for the two amplifying conditions is related to amount of distortion in the auditory system as defined by selected clinical tests. An assumption was made that excessively distorted auditory systems are more easily overdriven by high frequency emphasis amplification than auditory systems with less distortion. Therefore, subjects with least distortion were expected to discriminate speech better through a high-pass filtered amplifying system which, within limits, restored frequency-intensity balance. Results indicated slightly better speech discrimination performance for the selective amplifying condition. There was some evidence that the extent of the difference favoring selective amplification may be influenced by the amount of distortion in the auditory system.

Proceeding from prior experimental evidence that better speech-sound identification most often occurs with right-ear presentation, an experiment was conducted to test for differences in speech production with right-ear and left-ear auditory feedback of one’s own speech. A hybrid-computer system and techniques of experimental programing were employed to control the intervals of aural delay. Presentation of delayed auditory feedback to the right ear during speech, with white noise masking the left ear, resulted in a significantly greater number of articulatory errors than did delayed feedback to the left ear with white noise masking the right ear. With a measure of total speaking time, however, similar differences between ears during delayed hearing were not found. The findings were interpreted as an indication of differences in aural function during auditory feedback control of speech. Such differences are consistent with aural laterality differences reported with speech identification.

Differences in speech intelligibility and identification between binaural, monaural near ear, and monaural far ear conditions were studied in sound field conditions. Scores from listeners with normal hearing and with sensorineural losses were evalated in sound field conditions (unaided) and under conditions of hearing aid amplification (aided). For both conditions listeners with sensorineural hearing loss obtained a binaural advantage similar to that found for normal listeners. The binaural advantage could be demonstrated only when the primary and/or competing signal sources were located at an azimuth which resulted in interaural time differences for at least one of the signals. When the signals arrived simultaneously at the ears from the same loudspeaker, no binaural advantage was obtained. Differences in intelligibility and identification scores between monaural near ear and far ear conditions (6.0 dB) were almost twice as large as those found between binaural listening and monaural near ear listening (3.3 dB).