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In 1975, the threshold hearing levels (HLs) of 83 Swedish and British pop/rock musicians were examined with pure tone audiometry. At that time we found 13% of the musicians with a hearing loss > 20 dB HL at a high frequency pure tone average (3, 4, 6, and 8 kHz). The aim of the present investigation was to conduct an audiometric follow-up study of these musicians 16 years after the first examination. Fifty-three of the 83 pop/rock musicians were retested in 1991 to 1992. They also filled in a questionnaire concerning occupational noise activities, leisure noise activities, general health problems and subjective symptoms related to loud music exposure. The median pure tone audiogram was within 20 dB HL at all test frequencies. On an individual basis, 63% had normal high high frequency pure tone average scores and another 15% had a very limited high frequency hearing loss (< or = 25 dB HL). It seems surprising that pop/rock musicians after performing for 26 years have such well-preserved hearing. There might be a protective effect by the generally positive attitude from the musicians toward their performance and their audience. Regarding the loud and more or less continuous sound levels over 85 dBA there might also be a protective effect from a continuous contraction of the stapedius muscle.

The mismatch negativity (MMN) recordings provide an objective measure of the preattentive, automatic auditory discrimination function. This article deals with issues central to the recording of the MMN and the interpretation of the results for clinical and electrodiagnostic purposes. The methods of acquiring as pure an MMN response as possible, i.e., one not contaminated by auditory cortical responses reflecting other functions, are discussed first. Second, other technical questions associated with the recording are reported on, e.g., what MMN parameters should be recorded and how, what is the smallest recordable MMN response, and what is the repeatability of the MMN recordings. Then, the effect of various physiological factors on the MMN (age, alertness, gender, topographic distribution of the MMN) is considered. The correlation between auditory discrimination performance and the MMN amplitude, observed in normal population, is dealt with. Finally, there is a short concluding overview on clinical findings of MMN recordings and discussion on their electrodiagnostic applications.

This study assessed the efficacy of an adaptive frequency response hearing aid (AFR) for improving speech perception ability in noise among a group of elderly hearing-impaired listeners. A speech recognition task, self-assessed speech intelligibility task, and qualitative judgment task were administered to examine subtle differences in the effects of the AFR "signal processing" versus linear amplification. Group scores showed statistically significant improvement with AFR processing on the speech recognition task involving high-predictability sentences, but not on any other measures. However, there was a trend toward improved scores with AFR processing for low-predictability sentences as well. These results suggest that AFR circuitry may be most useful for enhancing recognition of speech in high-cue contexts. Wide individual subject variability was observed on all measures. This demonstrates the importance of evaluating the effectiveness of noise reduction hearing aids on an individual basis and with more than one task.

The goals of this investigation were to determine whether hearing aid benefit improved significantly over the first 10 weeks of hearing aid use and whether time-related changes in benefit (if any) were affected by the type of benefit measurement (i.e., objective or subjective). A total of 17 hearing-impaired subjects participated, with different subjects completing different phases of the study. Benefit was measured soon after the hearing aid fitting and again after 10 weeks of adjustment to hearing aid use. Objective benefit data were determined using the Connected Speech Test. No significant changes in objective benefit were noted in noisy or reverberant listening environments when visual cues were available. However, in a low-noise setting and in a noisy setting without visual cues, improvements in objective benefit were seen over time. Subjective benefit data were derived from responses to the Profile of Hearing Aid Benefit. These data indicated significant benefit improvement over time in all five types of daily life situations assessed, although the improvement was small in reverberant and noisy environments. Significant, but modest, correlations were found between objective and subjective data for low-noise and reverberant listening environments. Comparison of experienced and novice hearing aid wearers suggested that although experienced wearers obtain more benefit than novice wearers, they evidence similar time-related changes in benefit during the first 10 weeks of new hearing aid use.

Several methods have been proposed to estimate the sensation level (SL) at which children receive amplified speech from their hearing aids. The present study compared the SL estimates obtained with two such methods: (1) a sound field aided audiogram approach, and (2) an electroacoustic approach that incorporated the use of a probe tube microphone system (Seewald, Ross, & Stelmachowicz, 1987). Sound field aided thresholds were obtained for 13 hearing-impaired subjects at eight audiometric frequencies. For the electroacoustic approach, in situ thresholds were obtained using a button-type hearing aid receiver attached to a custom earmold. Real ear aided responses were measured using a 70 dB RMS speech-weighted composite noise signal (Frye, 1986). A comparison of the frequency-specific SL estimates derived from the two different methods revealed that the sound field aided audiogram approach yielded higher SL estimates for 74% of the individual comparisons. A detailed analysis of the findings obtained from two subjects suggested that when the results of the two methods did not agree, the differences were due to an interaction between signal level and the unique input/output characteristics of the subjects' hearing aids. A precautionary measure is suggested for those who wish to use sound field aided threshold data to estimate the SLs at which children receive amplified conversational speech.

The use of FM systems as primary amplification for children with severe and profound hearing loss is discussed. The FM advantage for improving speech perception and for reducing the detrimental effects of distance, noise, and reverberation is described. Both hypothetical and actual cases are used to demonstrate the FM advantage. An evaluation method for selecting an appropriate FM system is described.

Realistically evaluating the effectiveness of hearing protectors continues to be a major problem in hearing conservation. The purpose of this study was to examine a laboratory-based fitting procedure (User Fit) that was designed to yield hearing protector attenuation values similar to that derived from field studies. Ten subjects who were naive to hearing protectors were used in a repeated measures design that measured real ear attenuation at threshold for two types of plugs. Each subject was tested in two fitting conditions that varied based on the type and degree of assistance given to the subjects by the experimenter. The results showed significant differences in attenuation based on the fitting procedure used, with the User Fit best approximating field data. In addition, a generalized learning effect was noted. The results suggest that any experience with earplugs leads to subsequent improvement in attenuation despite the type of earplug used. Further testing is planned with greater numbers of subjects and additional types of hearing protectors.

Two experiments were conducted to determine the effects of various upper cutoff frequencies on performance of listeners with steeply sloping, high-frequency hearing loss. The upper cutoff frequency of individually prescribed responses was systematically adjusted to 0.71, 1.79, and 6.0 kHz. Measures of nonsense syllable recognition and subjective ratings of speech intelligibility and speech quality were obtained at three overall gain settings for each frequency response. Experiment 1 assessed performance on tests in quiet and Experiment 2 assessed performance on tests in noise. Nonsense syllable recognition improved as the upper cutoff frequency increased, particularly for tests in noise. In contrast, the subjects gave significantly poorer quality ratings for the shaped response with the highest cutoff frequency (6.0 kHz). Few significant effects of cutoff frequency were found for ratings of intelligibility in either quiet or noise.

Objectives: Literature reviews of binaural loudness summation assume nearly perfect summation (i.e., a tone presented binaurally is assumed to be twice as loud as the same tone presented monaurally). However, some recent data and classroom demonstrations of this phenomenon using speech stimuli from a familiar visually present talker yield much less summation. Therefore, the following two hypotheses were tested using a preliminary procedure that controlled some, but not all, variables. First, the amount of binaural loudness summation is less for speech from a visually present talker than for recorded speech or tones. Second, the amount of binaural loudness summation is less when sounds are presented via loudspeakers than when sounds are presented via earphones. Design: Three types of stimuli (monitored live-voice [MLV] spondees, recorded spondees, and tones) were presented monaurally and binaurally across a wide range of levels. The same stimuli were presented via earphones and loudspeakers in an audiometric test booth. Eight young listeners with normal hearing judged the loudness of the stimuli using magnitude estimation. Results: The amount of binaural loudness summation was significantly less for MLV spondees than for tones or recorded spondees. Binaural loudness summation was also significantly less for loudspeaker presentation than for earphone presentation. Binaural loudness summation was found to be less than perfect (i.e., a sound presented binaurally is less than twice as loud as the same sound presented monaurally) for all conditions. The amount of binaural loudness summation was the least for MLV spondees presented via loudspeakers. Conclusions: The present results support both hypotheses and indicate that binaural loudness summation in the loudspeaker conditions is significantly less than binaural loudness summation in typical laboratory test conditions using earphones. There may be a subjective effect resulting from expectations about loudness of a familiar, visually present talker, termed here as "Binaural Loudness Constancy."

Objectives: In noisy, reverberant environments, older adults often find it difficult to process acoustic signals, possibly because their ability to differentiate reflected waves that belong to a source from those generated by other sources diminishes with age. Therefore, older adults may be less efficient than younger adults at parsing the auditory scene into its component sound sources. To parse the auditory scene into its component sources the listener has to be able to group correlated waves coming from different directions (the direct wave and its reflections off of environmental surfaces). Because detecting a change in correlation is an important component of scene parsing, this study examined whether there is an age-related deficit in detecting break in correlation (BIC) between the noises presented over left and right headphones or over left and right loudspeakers, where a BIC refers to a change in interaural correlation from 1 to zero and then a return to 1. Design: In experiment 1, we determined the shortest BIC duration at which 10 younger and 10 older adults could detect the BIC in the middle of identical noises (bandwidth = 10 kHz; duration = 1 sec) presented simultaneously to the left and right ears over headphones or played simultaneously over loudspeakers positioned 45 degrees to the left and right of the listeners. In experiment 2, we determined the longest delay between the left-side noise and the right-side noise, at which a 100 ms BIC presented in the middle of the noise could be detected in 10 younger and 8 older adults. Results: The results of experiment 1 show that younger participants could detect significantly shorter BICs than older participants independent of whether the noises were presented over headphones or loudspeakers. The results of experiment 2 show that younger participants could detect the 100 ms BIC at significantly longer interaural delays than older participants. Also, for both age groups, detecting the BIC was easier under the loudspeaker-stimulation condition than the headphone-stimulation condition. Moreover, in the loudspeaker condition, the spectral cues arising from interactions between correlated sound sources seemed to be of greater benefit to younger than to older participants. Conclusions: The age-related decrease in sensitivity to a BIC indicates that older adults are less able than younger adults to detect a change in correlation in an ongoing sound. The inability of older adults to detect the 100 ms BIC as readily as younger adults, when the noise arriving at one ear is delayed relative to the noise arriving at the other ear, suggests that the representation of aspects of the sound's waveform decays more rapidly in older adult than in younger adult listeners. Moreover, these age-related deficits are not related to listeners' audiograms. In addition, younger adults seem to be much better than older adults at using the spectral cue provided by comb filtering to detect the BIC when there is a delay between the noises presented over loudspeakers. The more rapid decay of waveform details, combined with the lesser sensitivity to change in correlation and to spectral cues, suggest that older adults may not be as capable as younger adults in parsing auditory scenes.

A theoretical criterion for preliminary prescription of hearing aid gain and frequency response was formulated on the basis of Zwicker's loudness model, modified for impaired auditory frequency resolution in cochlear hearing losses. The procedure was designed to restore normal relative loudness contributions from each frequency band of input speech. Prescribed frequency responses had less steep bass cut than recommended by either the Prescription of Gain/Output or the National Acoustic Laboratories' procedures, regardless of various model assumptions. Prescribed overall gain depended on the loudness growth assumed in the model. In cases where auditory filters are wider than normal and loudness recruitment is complete, the procedure yielded a nonlinear relation between hearing aid gain and hearing loss: Required insertion gain (in dB) was 25 to 30% of hearing threshold loss (in dB HL) for mild to moderate losses, but this percentage increased for more severe impairments. For cases with incomplete loudness recruitment, the model prescribed that insertion gain should be 40 to 50% of hearing threshold loss for a wide range of impairments.

Four different prescriptions of hearing aid insertion gain versus frequency were validated with a group of 26 moderately hearing-impaired, elderly hearing aid users. Three prescriptions were based on calculating the loudness and articulation index (AI) for aided speech, and ranged from a frequency response with moderate high-frequency emphasis, restoring normal loudness for speech peaks, to a response with the greatest high-frequency emphasis, maximizing the AI. The fourth prescription was a well-established formula of the half-gain type. The frequency responses were evaluated by paired comparison ratings of the pleasantness and intelligibility of speech in noise, and by speech identification tests in noise. The subjects rated the flattest response as significantly more pleasant than the other responses, and significantly more intelligible than the prescription with maximal high-frequency emphasis. There were no detectable differences in signal/noise ratios required for 50% speech identification. These results indicate that a prescription that restores normal loudness for speech peaks in each critical band is probably more easily accepted than either a procedure which is intended to make all speech bands equally loud, or a prescription which maximizes the AI.

Recent advances in electrode design have helped to rekindle clinical interests in noninvasive electrocochleography--to enhance auditory brain stem response (ABR) recordings (namely, wave I) and to screen Meniere's disease. The salient feature of the response in suspected cases of endolymphatic hydrops is an unusually large summating potential (SP). Since the SP is a DC potential, conventional wisdom suggests the use of relatively low cutoff frequencies (i.e., long-time constants) for recording, but this may degrade the recording due to the presence of excessive low-frequency noise. However, the click-elicited SP also is a transient of relatively short duration, considering the characteristics of the click. This may permit more liberal high-pass filtering with acceptable wave-form distortion. This is a report of findings obtained using an analog model for SP generation and recordings from normal subjects using an electrode placed directly on the tympanic membrane. Responses were obtained using various low-frequency cutoffs. Cutoffs up to 30 Hz caused little distortion. Even in the face of considerable distortion at the highest cutoffs (100 and 300 Hz), the SP remained detectable and the SP:AP essentially unchanged. Therefore, higher filter settings may be used in recording the SP than commonly assumed, although the decreased amplitudes of the component potentials may not be tolerable under all clinical test conditions.

This study demonstrates degradation of a modulated speech-like signal in a typical listening environment as measured by the new Bruel and Kjaer Rapid Speech Transmission Index (RASTI) System. Thirty-four adults were seated in a typical college classroom. Measurements of the RASTI signal were made at 17 different seating locations. Results showed that the speech-like signal was substantially degraded, even at the front-row center seat. Loss of critical speech information was greater at less favorable locations. These results suggest that because hearing aids cannot amplify nonexistent speech information, the effects of distance must be considered when fitting amplification systems.

In this investigation, the relation between various directivity measures and subject performance with directional microphone hearing aids was determined.Test devices included first- and second-order directional microphones. Recordings of sentences and noise (Hearing in Noise Test, HINT) were made through each test device in simple, complex, and anisotropic background noise conditions. Twenty-six subjects, with normal hearing, were administered the HINT test recordings and directional benefit was computed. These measures were correlated to theoretical, free-field, and Knowles Electronic Manikin for Acoustic Research (KEMAR) directivity index (DI) values, as well as front-to-back ratios (FBR), in situ signal-to-noise ratios (SNR), and a newly proposed Db SNR, wherein a predictive value of the SNR improvement is calculated as a function of the noise source incidence.The different predictive scores showed high correlation to the measured directional benefit scores in the complex (diffuse-like) background noise condition (r = 0.89 to 0.97, p0.05) but not across all background noise conditions (r = 0.45 to 0.97, p0.05). The Db SNR approach and the in situ SNR measures provided excellent prediction of subject performance in all background noise conditions (0.85 to 0.97, p0.05). None of the predictive measures could account for the effects of reverberation on the speech signal (r = 0.35 to 0.40, p0.05).For environments that included a discrete number of noise sources, the in situ SNR and Db SNR estimates were most predictive of subject performance. No predictive approach was indicative of the directional benefit achieved when the speech was also subjected to reverberation (temporal distortion). This finding has implications for real-world estimates of directional benefit.

This paper evaluates the benefit of the two-microphone adaptive beamformer BEAM in the Nucleus Freedom cochlear implant (CI) system for speech understanding in background noise by CI users.A double-blind evaluation of the two-microphone adaptive beamformer BEAM and a hardware directional microphone was carried out with five adult Nucleus CI users. The test procedure consisted of a pre- and post-test in the lab and a 2-wk trial period at home. In the pre- and post-test, the speech reception threshold (SRT) with sentences and the percentage correct phoneme scores for CVC words were measured in quiet and background noise at different signal-to-noise ratios. Performance was assessed for two different noise configurations (with a single noise source and with three noise sources) and two different noise materials (stationary speech-weighted noise and multitalker babble). During the 2-wk trial period at home, the CI users evaluated the noise reduction performance in different listening conditions by means of the SSQ questionnaire. In addition to the perceptual evaluation, the noise reduction performance of the beamformer was measured physically as a function of the direction of the noise source.Significant improvements of both the SRT in noise (average improvement of 5-16 dB) and the percentage correct phoneme scores (average improvement of 10-41%) were observed with BEAM compared to the standard hardware directional microphone. In addition, the SSQ questionnaire and subjective evaluation in controlled and real-life scenarios suggested a possible preference for the beamformer in noisy environments.The evaluation demonstrates that the adaptive noise reduction algorithm BEAM in the Nucleus Freedom CI-system may significantly increase the speech perception by cochlear implantees in noisy listening conditions. This is the first monolateral (adaptive) noise reduction strategy actually implemented in a mainstream commercial CI.

OBJECTIVES: To determine whether digital signal processing (DSP) hearing aids produce conducted radio frequency interference that can affect the use of personal FM systems, to quantify the nature of any such interference, and to discuss practical remedies. DESIGN: Sixteen DSP hearing aids were used. Measurements were made of the spectral characteristics of any conducted radio frequency interference produced by each aid with FM shoe and 40 cm direct audio input (DAI) lead when the DAI facility was enabled. Measurements were made with the aid, shoe, and lead inside an electrically screened chamber. The effect of DAI lead length was also investigated with one of the hearing aids. Finally, some subjective listening tests were carried out by using different FM systems coupled to a number of the aids. RESULTS: All but four of the DSP hearing aids tested produced readily measurable interference, with some much worse than others. Levels of interference were high enough with some hearing aids to be likely to significantly impede signal perception when the radio frequency of the interference coincided with the radio frequency of the FM system. This usually occurs intermittently as a result of the processor design of most DSP hearing aids. The listening tests suggested that when personal FM systems are in use with some DSP hearing aids, the interference would be audible, unpleasant, and detrimental to audio quality. CONCLUSIONS: DSP hearing aids without low electromagnetic interference processors should not be fitted to clients if personal FM systems are expected to be used. Manufacturers of DSP aids should be encouraged to use low electromagnetic interference processors in their DSP hearing aid design. Meanwhile, FM systems should be used with DSP hearing aids in such a way as to ensure high received radio signal levels, and FM receivers should be switched off when not in use.

To measure the sound levels generated by the headphones of commercially available portable compact disc players and provide hearing healthcare providers with safety guidelines based on a theoretical noise dose model.Using a Knowles Electronics Manikin for Acoustical Research and a personal computer, output levels across volume control settings were recorded from headphones driven by a standard signal (white noise) and compared with output levels from music samples of eight different genres. Many commercially available models from different manufacturers were investigated. Several different styles of headphones (insert, supra-aural, vertical, and circumaural) were used to determine if style of headphone influenced output level.Free-field equivalent sound pressure levels measured at maximum volume control setting ranged from 91 dBA to 121 dBA. Output levels varied across manufacturers and style of headphone, although generally the smaller the headphone, the higher the sound level for a given volume control setting. Specifically, in one manufacturer, insert earphones increased output level 7-9 dB, relative to the output from stock headphones included in the purchase of the CD player. In a few headphone-CD player combinations, peak sound pressure levels exceeded 130 dB SPL.Based on measured sound pressure levels across systems and the noise dose model recommended by National Institute for Occupational Safety and Health for protecting the occupational worker, a maximum permissible noise dose would typically be reached within 1 hr of listening with the volume control set to 70% of maximum gain using supra-aural headphones. Using headphones that resulted in boosting the output level (e.g., insert earphones used in this study) would significantly decrease the maximum safe volume control setting; this effect was unpredictable from one manufacturer to another. In the interest of providing a straightforward recommendation that should protect the hearing of the majority of consumers, reasonable guidelines would include a recommendation to limit headphone use to 1 hr or less per day if using supra-aural style headphones at a gain control setting of 60% of maximum.

Objective The feasibility and practicalities of performing probe-tube microphone measures with infants is addressed, as well as two aspects of acoustic functioning of infant ears: the real ear unaided response (REUR) and the real ear to coupler difference (RECD). Design Part 1 is a longitudinal study involving 12 infants. Serial measures of REUR were obtained over an 18-mo period. Infants were 53 mo for the first test and 521 mo at the last test visit. Practicalities of probe-tube microphone testing of unsedated infants and changes in the position (i.e., frequency) of the primary REUR peak were addressed. Design For Part 2 of the project, 33 infants under 12 mo of age took part. A comparison of real ear hearing aid gain versus coupler gain was made. Test-retest differences for real ear aided response were estimated. Results Part 1 results indicate that probe-tube microphone measures in unsedated infants are feasible and show good within-subject repeatability. REUR measures for the more alert and mobile older subjects, as for the younger infants, showed an acceptably small degree of intersubject variability. The frequency of the primary REUR peak decreased during the first year of life, with a group mean of 2932 Hz reached at the end of the first year. However, thereafter, instead of stabilizing at around this value there was considerable fluctuation in the frequency of the REUR peak. Results For Part 2, results a) confirmed the large RECD value for infants in the first year of life and b) showed a high degree of intersubject variability. Test retest measures of real ear aided response (REAR) gave values small enough to indicate the clear potential of probe-microhone use with infants during the hearing aid selection and fitting process. Conclusion These findings should encourage attempts to carry out individual probe-tube microphone measurements with very young infants. They point to the need for infant hearing aid fitting procedures which involve such measurements to secure appropriate amplification.

OBJECTIVE Several commercial otoacoustic emission probes (Virtual 330, Etymotic Research ER-10 and ER-10B, and Otodynamics ILO88 and ILO92) were tested for internal coupling (cross-talk) between the sound source and probe microphone. DESIGN The response of the probe under test was measured, in a cavity that simulates the human ear canal, first in its normal configuration and then with either the sound source tube or the microphone inlet blocked with modelling clay. Cross-talk was indicated if the pressure failed to drop substantially in the "blocked" condition. RESULTS Cross-talk in the Virtual 330 and Etymotic ER-10B was found to be severe enough to compromise the accuracy of "in the ear" acoustic calibrations. In both cases the cause was the radiation of sound through the walls of the flexible silicone tubing used to conduct the stimuli through the body of the probe. Substituting harder-walled sound source tubing reduced cross-talk to levels acceptable for routine application. The probes used in the otodynamics ILO88 and ILO92 systems and the original Etymotic ER-10 probes all have acceptably low cross-talk as supplied from the manufacturers. CONCLUSIONS For the two probes where excessive cross-talk was found, most of the cross-talk was acoustic and/or mechanical, rather than electromagnetic. Artifacts due to cross-talk have undoubtedly contaminated data collected with these units. Manufacturers of emission probes are urged to adopt a standard for specifying cross-talk levels and include this specification in product literature.

Transient acoustic stimuli are used for generating auditory evoked responses (AERs). Common transducers do not accurately reproduce transient stimuli, since the transducer diaphragm oscillates at its resonant frequency at the abrupt initiation or termination of a stimulus. These oscillations are called "acoustic ringing." This paper discusses methods for reducing and canceling acoustic ringing, and describes one method, a systems approach, in detail. It also delineates one method for enhancing acoustic ringing. The methods described provide effective techniques for empirical study of the effects of acoustic ringing on AERs or for any application requiring accurate transduction of a transient stimulus.

Earphone attenuation values were determined for 17 children (6-14 years old) using supra-aural (TDH-49P/Model 51 cushion) and insert earphones (E-A-Rtone 3A) terminated by an E-A-Rlink 3A (for normal size ear canals) or E-A-Rlink 3B (for small size ear canals) foam eartips, and for 10 adults having small ear canals using insert earphones and E-A-Rlink 3B foam eartips. The test signals were 1/3-octave bands of noise presented in a diffuse sound field (re: ANSI S12.6-1984). The supra-aural earphone attenuation values for the children were slightly higher (more attenuation) or similar to reported adult values, and always lower (less attenuation) compared with insert earphone/E-A-Rlink 3A (IE/3A) or 3B (IE/3B) values for both children and adults. The IE/3B attenuation values were similar between the children and adults and provided slightly more attenuation than the IE/3A. Overall, the results indicated that the ANSI S3.1-1991 maximum permissible ambient noise levels allowed in a test room for ears covered testing with a supra-aural earphone, which were determined using adult values, are appropriate for testing children. Future revisions of ANSI S3.1-1991 may include maximum permissible ambient noise levels for testing with insert earphones. The IE/3A and IE/3B attenuation values could be used for that purpose. In the meantime, because more attenuation was provided by the IE/3A and IE/3B, they can be used for testing both children and adults in higher ambient noise levels than specified in ANSI S3.1-1991.

The purpose of this project was to determine if the hearing-impaired population reports satisfaction with their ability to converse over the telephone in both aided and unaided situations. For this study, we surveyed 104 hearing-impaired persons using a 43 item questionnaire that investigates problems that hearing-impaired people encounter when using the telephone. The questionnaire also probes the solutions that hearing-impaired people use for overcoming these problems. Of the 91 respondents who wear hearing aids, 55% use their aids while operating the telephone. However, 70% of these respondents reported that coupling the hearing aid to the telephone is problematic. Use of a telephone amplifier was reported by 73% of the sampled population. The majority of the subjects, 75%, indicated an interest in improvements in telephone communications for hearing-impaired people.

This article describes an evaluation of an in the ear hearing aid, which applies fast-acting full dynamic range compression independently in two frequency bands. This can compensate for the loudness recruitment typically associated with sensorineural hearing loss. The crossover frequency between the two bands and the gain and compression ratio in each band are programmable to suit the individual patient. Twenty subjects with moderate sensorineural hearing loss were tested in a counterbalanced order using the aid programmed as a linear amplifier (condition L) and as a two-band compressor (condition C). All subjects were fitted binaurally. Subjects were also tested without hearing aids (condition U) and using the hearing aids that they normally wore (condition Own). Speech intelligibility was measured in quiet at three sound levels (50, 65, and 80 dB SPL), and speech reception thresholds (SRTs) in 12-talker babble were measured under monaurally and binaurally aided conditions, with the speech and babble both coincident and spatially separated. In condition C, speech intelligibility in quiet was high at all sound levels. Speech intelligibility at the two lower levels decreased in condition L, and decreased still further in conditions Own and U. Condition C gave, on average, better speech intelligibility in babble (lower SRTs) than conditions L, Own, or U. The advantage of condition C over condition L varied across subjects and was correlated with the dynamic range for tones at high frequencies; small dynamic ranges were associated with greater benefit from compression. A significant advantage for binaural aiding was found both when the speech and noise were spatially separated and when they were coincident. The binaural advantage was similar for the C and L conditions, indicating that the independent compression at the two ears did not adversely affect the use of binaural cues. Questionnaires on the subjects' experiences with the aids in everyday life indicated that they generally preferred condition C over condition L.

Several recent studies have reported large intersubject variability of high-frequency thresholds measured with circumaural earphones. In the present study, high-frequency thresholds of 10 subjects were measured with circumaural (Sennheiser HD-250) and insert (Etymotic ER-1) earphones at 10, 12, 14, and 16 kHz. Overall results show significantly smaller variability of the threshold data obtained with insert earphones than with circumaural earphones. The above data indicate that insert earphones may be more suitable for high-frequency testing than circumaural earphones.

Attenuation measurements were made using the ANSI S12.6-1984 protocol on a standard Telephonics headset with TDH-50P earphones and Model 51 cushions, Amplivox Audiocups headphone enclosures, and Peltor AudioMate headphone enclosures. Each of the enclosures housed Telephonics TDH-50P earphones with Model 51 cushions. The mean attenuation values obtained were compared with those previously reported, and reasons for discrepancies were analyzed. Pure-tone threshold shifts in background noise complying with ANSI S3.1-1977 and Occupational Safety and Health Administration (1983) maximum permissible ambient noise level standards were estimated on the basis of the attenuation values for each headphone device, and the adequacy of these current standards for accurate pure-tone threshold assessment was considered. The results indicated that Model 51 cushions alone are insufficient to attenuate the ambient noise levels permitted under ANSI S3.1-1977, and even the utilization of noise-excluding headphone enclosures does not reduce the background noise levels permitted under the Occupational Safety and Health Administration (1983) to a sufficient degree to permit testing down to 0 dB HL.

In probe tube measures, the method used to equalize the sound field will determine the reference value used in calculating real ear insertion gain (REIG) and real ear aided gain (REAG). Several equalization methods are available with current probe tube systems. This study determined the effects of three equalization methods (substitution, modified comparison, and ipsilateral comparison) on REIG and REAG measures using open and closed earmolds for ear canals varying in circumference. Twelve subjects were selected representing a wide range of ear canal circumferences. Results revealed significant differences for REIG and REAG among the three equalization methods. These differences appear to be due to the different acoustic components included or excluded for each method. The results of this study suggest that the sound field equalization method used for probe tube measures needs to be considered when interpreting the hearing aid gain reported among various probe microphone units. Also, canal circumference may be a quick method for categorizing canal cavity size and appears to influence measured hearing aid gain.

Hearing aid fitting involves a two-phase process of preselection and evaluation (Seewald RC and Ross M. Amplification for the Hearing Impaired 1988:213-271). The purpose of the present study was to examine alternative procedures that clinicians might use in the evaluation phase to verify the adequacy of hearing aid preselection decisions for severely and profoundly hearing-impaired listeners. Bekesy tracking, loudness rating, and conventional bracketing procedures were used to determine threshold, most comfortable listening level, and uncomfortable listening level for 10 hearing-impaired young adults. Stimuli were pulsed pure tones of 500, 1000, and 2000 Hz and filtered words. Means and standard deviations of most comfortable listening levels and uncomfortable listening levels derived from loudness judgments of the 10 subjects showed only nominal differences across procedures. However, correlation analysis (Pearson r) indicated that individuals responded to the three procedures in varying ways, producing different loudness judgments and overall dynamic ranges. Thus, test procedure may influence the clinician's final evaluation of a preselected hearing aid. Initial work suggests that closed-set response categories such as loudness rating can limit measurement variability and potentially guide the clinician's evaluation of hearing aid preselection decisions.

Patient preference for monaural versus binaural hearing aids was studied to evaluate the subjective differences noted between these two types of fittings. Subjects were 30 patients with bilateral hearing impairment who were being fit with amplification for the first time. Subjects wore both monaural and binaural hearing aids for controlled periods of 1 hr each and 2 days each. This was followed by an additional 3 days during which the fittings were alternated at the subjects' own discretion. Subjects reported preferences and subjective differences after each experimental phase. Interviews were conducted 3 months after participation in the study to determine actual hearing aid use. Binaural amplification was preferred by 90% of the subjects for consistently similar reasons. Results also suggest that consistency of preferences throughout an initial trial period may be predictive of successful adjustment to and use of amplification. Findings indicate that the routine practice of fitting monaural hearing aids may not provide optimum fittings for many patients.

Binaural advantage was measured for 12 normally hearing subjects in a typical rectangular audiometric test room with a loudspeaker located in each corner. Four different loudspeaker configurations for presentation of signal and competition were evaluated. The results indicated that a configuration in which uncorrelated competition was delivered from both sides of the subject while the signal was presented from a 0 degree azimuth was the most resistant to interaural asymmetries resulting from the room geometry and produced the most consistent binaural advantages. Binaural advantage was then measured using this loudspeaker configuration for 15 hearing-impaired individuals wearing hearing aids. Although the group results indicate a mean unaided binaural advantage only slightly smaller than seen in the normal hearers, when hearing aids were worn an interaural asymmetry in a signal-to-competition ratio developed that reduced the clinical usefulness of the data for individuals. This outcome emphasizes that a valid clinical demonstration of binaural advantage is critically dependent on interaural symmetry in signal-to-competition ratios for both aided and unaided tests. Such symmetry may be difficult to achieve in a typical audiometric test room.