Simulating the open-loop transfer function as a means for understanding acoustic feedback in hearing aids

Suppressing unstable acoustic feedback in hearing aids will first require knowledge of the open-loop transfer functions of such systems. Reported herein is a mathematical technique for simulating the open-loop transfer function of an in situ eyeglass-type hearing aid. In particular, a computer program was developed that characterized the hearing aid as a serial connection of two-port blocks, each representing one individual component of a hearing aid. Included, for example, were two-port blocks representing the microphone, amplifier, receiver, sound tubes leading to the eardrum (including the ear canal itself), earmold vent, and external pathway from the vent outlet back to the microphone. The computer program was validated by replicating laboratory data derived from an experiment involving a nonstandard manikin fitted with a nonstandard artificial ear. Next, the open-loop transfer function of an eyeglass-type hearing aid in situ on the manikin was simulated via the computer program. Unfortunately, those computer-generated data were not replicated in the laboratory due to the difficulty encountered in actually measuring the open-loop transfer function. Nevertheless, investigators were able to utilize those data to predict, within +/- 25 Hz, the "squeal" frequency of unstable acoustic feedback.     

A computer simulation of hearing aid response and the effects of ear canal size

The response of a hearing aid is affected by many factors which include the head and outer ear, the microphone, amplifier, and receiver used in the hearing aid, the properties of the ear canal and the eardrum, and acoustic feedback through the vent. This article presents a computer simulation of an in-the-ear (ITE) hearing aid that includes all of the above factors. The simulation predicts the pressure at the eardrum for a frontal free-field sound source. The computer model was then used to determine the effects on the hearing aid response due to variations in the size of the ear canal. The simulation indicates that, for an unvented hearing aid, changes in the size of the ear canal shift the overall sound-pressure level at the eardrum but have only small effects on the shape of the frequency response. The situation is more complicated when a vent is present, however, since changes in the size of the ear canal that cause apparently small perturbations in the acoustic feedback signal may, nonetheless, have large effects on the overall system response.     

Transverse pressure distributions in a simple model ear canal occluded by a hearing aid test fixture

The sound field in a model ear canal with a hearing aid test fixture has been investigated experimentally and theoretically. Large transverse variations of sound pressure level, as much as 20 dB at 8 kHz, were found across the inner face of the hearing aid. Variations are greatest near the outlet port of the receiver and the vent port. Deeper into the canal, the transverse variations are less significant and, at depths greater than 4 mm, only a longitudinal variation remains. The model canal was cylindrical, 7.5 mm diameter, and terminated with a Zwislocki coupler to represent absorption by the human middle ear. The outer end of the canal was driven by the receiver in the hearing aid test fixture, with the acoustic output entering the canal through a 1 mm port. The hearing aid was provided with a 20-mm-long vent, either 1 or 2 mm in diameter. The sound field inside the canal was measured using a specially designed 0.2-mm-diam probe microphone [Daigle and Stinson, J. Acoust. Soc. Am. 116, 2618 (2004)]. In parallel, calculations of the interior sound field were performed using a boundary element technique and found to agree well with measurements.     

A technique for simulating the amplifier-to-eardrum transfer function of an in situ hearing aid

There are numerous articles wherein mathematical models of various parts of an in situ hearing aid have been reported. Such parts include, for example, the microphone, receiver, cylindrical tubes carrying sound to the eardrum and out through the earmold vent, and the external path from the vent back to the microphone. This article extends these earlier works to include the hearing-aid amplifier. In particular, a mathematical technique for characterizing the amplifier in combination with the receiver is reported. Cascade parameters of a two-port model of one particular amplifier/receiver combination are obtained by this method. The cascade-parameter data and the method of obtaining this data are verified by two different experimental procedures. One procedure involves both computing and measuring the input driving-point impedance of the amplifier/receiver combination. In the second procedure, the amplifier-to-eardrum transfer function of a hearing aid incorporating this same amplifier/receiver combination and mounted on an artificial ear is both computed and measured. Experimental and computed values of this transfer function for three different earmold geometries are in reasonably close agreement. The amplifier/receiver model reported herein will be used in future studies of acoustic feedback in hearing aids.     

The hearing aid feedback path: mathematical simulations and experimental verification

Acoustic feedback in hearing aids has received little attention in the literature. Feedback occurs when stability conditions of the open-loop transfer function of an in situ hearing aid are violated. Solving the feedback problem will first require knowledge of the open-loop transfer function. Included in the open-loop transfer function is the acoustical path by which sound emanating from the earmold vent returns to the microphone (i.e., the feedback path). Reported herein are two different mathematical procedures for simulating transfer functions of the feedback path of an eyeglass-type hearing aid. In one procedure the vent exit was modeled as a point source of sound located on a flat plane, while it was treated as a point source on a sphere in the other. Results of laboratory experiments indicate that the mathematical models accurately predict those acoustic phenomena for which they were intended: point sources on plane and spherical baffles. Results of manikin experiments showed both models to be less accurate for simulating the feedback path around the human head. The maximum difference between experiment and theory was 6 dB at one frequency. Surprisingly, the flat-baffle model produced better agreement with experimental results than did the sphere model.     

Variations in the feedback of hearing aids

Variations in the loop response of hearing aids caused by jaw movements, variations in acoustics outside the ear, and variations of vent size have been identified. Behind The Ear (BTE) and In The Ear Canal (ITEC) hearing aids were considered. The largest variations among the variations of the acoustics outside the ear, except when the hearing aid was partly removed, were found with the ITEC when a telephone set was placed by the ear. The variations of the loop response caused by changes in vent size were compared with the variations of a theoretical model of the feedback path. The theoretical model was also used to compare the feedback of different designs of the vent that gives the same acoustic impedance at low frequencies. The calculated feedback was less with the short vents (12 mm) than the long vents (24 mm).     

Room reverberation effects in hearing aid feedback cancellation

Room reverberation can affect feedback cancellation in hearing aids, with the strength of the effects depending on the acoustical conditions. These effects were studied using a behind the ear (BTE) hearing aid mounted on a dummy head and coupled to the ear canal via an open fitting. The hearing aid impulse response was measured for the dummy head placed at eight closely spaced locations in a typical office. The feedback cancellation in the hearing aid used a set of filter coefficients that were initialized for one location within the room, and then allowed to adapt to the feedback path measured at the same or to a different location. The maximum stable gain for the hearing aid was then estimated without feedback cancellation, for the initial set of feedback cancellation filter coefficients prior to adaptation, and for the feedback cancellation filter after adaptation. A low-order ARMA model combining a fixed set of poles with an adaptive FIR filter is shown to be effective in representing the feedback path exclusive of reverberation. Increasing the adaptive filter length has only a small benefit in improving the feedback cancellation performance due to the inability of the system to model the room reverberation. The mismatch between the modeled and actual feedback paths limits the headroom increase that can be achieved when using feedback cancellation, and varies with the location within the room.     

Constrained adaptation for feedback cancellation in hearing aids.

In feedback cancellation in hearing aids, an adaptive filter is used to model the feedback path. The output of the adaptive filter is subtracted from the microphone signal to cancel the acoustic and mechanical feedback picked up by the microphone, thus allowing more gain in the hearing aid. In general, the feedback-cancellation filter adapts on the hearing-aid input signal, and signal cancellation and coloration artifacts can occur for a narrow-band input. In this paper, two procedures for LMS adaptation with a constraint on the magnitude of the adaptive weight vector are derived. The constraints greatly reduce the probability that the adaptive filter will cancel a narrow-band input. Simulation results are used to demonstrate the efficacy of the constrained adaptation.     

Speech-reception threshold in noise with one and two hearing aids

The binaural free-field speech-reception threshold (SRT) in 70-dBA noise was measured with conversational sentences for 24 hearing-impaired subjects without hearing aids, with a hearing aid left, right, and left plus right, respectively. The sentences were always presented in front of the listener and the interfering noise, with a spectrum equal to the long-term average spectrum of the sentences, was presented either frontally, from the right, or from the left side. For subjects with only moderate hearing loss, PTA (average air-conduction hearing level at 500, 1000, and 2000 Hz) less than 50 dB, the SRT in 70-dBA noise in both ears is determined by the signal-to-noise ratio even if only one hearing aid is used. For larger hearing losses the SRT appears to be partly determined by the absolute threshold. In conditions with a high noise level relative to the absolute threshold, in which case for both ears the SRT is determined by the signal-to-noise ratio, a second hearing aid, just as a monaural hearing aid, generally does not improve the SRT. However, in the case of a high hearing level, or a low noise level, in which a monaural hearing aid is profitable, the use of two hearing aids is even more profitable. In a separate experiment, acoustic head shadow was measured at the entrance of the ear canal and at the microphone location of a hearing aid. It appeared that, for a lateral noise source and speech frontal, the microphone position of behind-the-ear hearing aids has a negative effect on the signal-to-noise ratio of 2-3 dB.     

Verifying the attenuation of earplugs in situ: method validation using artificial head and numerical simulations

The use of in situ measurements of hearing protectors' (HPD's) attenuation following the microphone in real ear (MIRE) protocol is increasing. The attenuation is hereby calculated from the difference in sound levels outside the ear and inside the ear canal behind the HPD. Custom-made earplugs have been designed with an inner bore that allows inserting a miniature microphone. A thorough understanding of the difference, henceforth called transfer function, between the sound pressure of interest at the eardrum and the one measured at the inner bore of the HPD is indispensable for optimizing the MIRE technique and extending its field of application. This issue was addressed by measurements on a head-and-torso-simulator and finite difference time domain numerical simulations of the outer ear canal occluded by an earplug. Both approaches are in good agreement and reveal a clear distinction between the sound pressure at the MIRE microphone and at eardrum, but the measured transfer functions appear to be stable and reproducible. Moreover, the most striking features of the transfer functions can be traced down to the geometrical and morphological characteristics of the earplug and ear canal.     

High-frequency audiometric assessment of a young adult population

The hearing thresholds of 37 young adults (18-26 years) were measured at 13 frequencies (8, 9,10,...,20 kHz) using a newly developed high-frequency audiometer. All subjects were screened at 15 dB HL at the low audiometric frequencies, had tympanometry within normal limits, and had no history of significant hearing problems. The audiometer delivers sound from a driver unit to the ear canal through a lossy tube and earpiece providing a source impedance essentially equal to the characteristic impedance of the tube. A small microphone located within the earpiece is used to measure the response of the ear canal when an impulse is applied at the driver unit. From this response, a gain function is calculated relating the equivalent sound-pressure level of the source to the SPL at the medial end of the ear canal. For the subjects tested, this gain function showed a gradual increase from 2 to 12 dB over the frequency range. The standard deviation of the gain function was about 2.5 dB across subjects in the lower frequency region (8-14 kHz) and about 4 dB at the higher frequencies. Cross modes and poor fit of the earpiece to the ear canal prevented accurate calibration for some subjects at the highest frequencies. The average SPL at threshold was 23 dB at 8 kHz, 30 dB at 12 kHz, and 87 dB at 18 kHz. Despite the homogeneous nature of the sample, the younger subjects in the sample had reliably better thresholds than the older subjects. Repeated measurements of threshold over an interval as long as 1 month showed a standard deviation of 2.5 dB at the lower frequencies (8-14 kHz) and 4.5 dB at the higher frequencies.     

Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.     

Calibration of ear canals for audiometry at high frequencies

A procedure is described for determining the absolute sound pressure at the inner end of the ear canal when a sound source is coupled to the ear, for frequencies in the range 8-20 kHz. The transducer that generates the sound is coupled to the ear canal through a lossy tube, yielding a source impedance that is approximately matched to the characteristic impedance of the ear canal. A small microphone is located in the coupling tube close to the entrance to the ear canal. Calibration is carried out by measuring the response at this microphone when an impulse is applied at the transducer. To estimate the sound pressure at the medial end of the ear canal, the Fourier transform of this impulse response is corrected by an all-pole function in which the poles are estimated from the minima in this Fourier transform. Data on individual ear canals are presented in terms of gain functions relating the sound pressure at the medial end of the ear canal to the sound pressure when the coupling tube is blocked. The average gain function for a group of adult ears increases from 2 to 12 dB over the frequency range 8-20 kHz, in rough agreement with data from ear-canal models. Possible sources of error in the calibration procedure are discussed.     

Active cancellation of occlusion: an electronic vent for hearing aids and hearing protectors

The occlusion effect is commonly described as an unnatural and mostly annoying quality of the voice of a person wearing hearing aids or hearing protectors. As a result, it is often reported by hearing aid users as a deterrent to wearing hearing aids. This paper presents an investigation into active occlusion cancellation. Measured transducer responses combined with models of an active feedback scheme are first examined in order to predict the effectiveness of occlusion reduction. The simulations predict 18 dB of occlusion reduction in completely blocked ear canals. Simulations incorporating a 1 mm vent (providing passive occlusion reduction) predict a combined active and passive occlusion reduction of 20 dB. A prototype occlusion canceling system was constructed. Averaged across 12 listeners with normal hearing, it provided 15 dB of occlusion reduction. Ten of the subjects reported a more natural own voice quality and an appreciable increase in comfort with the cancellation active, and 11 out of the 12 preferred the active system over the passive system.     

Isolating the auditory system from acoustic noise during functional magnetic resonance imaging: examination of noise conduction through the ear canal, head, and body

Approaches were examined for reducing acoustic noise levels heard by subjects during functional magnetic resonance imaging (fMRI), a technique for localizing brain activation in humans. Specifically, it was examined whether a device for isolating the head and ear canal from sound (a "helmet") could add to the isolation provided by conventional hearing protection devices (i.e., earmuffs and earplugs). Both subjective attenuation (the difference in hearing threshold with versus without isolation devices in place) and objective attenuation (difference in ear-canal sound pressure) were measured. In the frequency range of the most intense fMRI noise (1-1.4 kHz), a helmet, earmuffs, and earplugs used together attenuated perceived sound by 55-63 dB, whereas the attenuation provided by the conventional devices alone was substantially less: 30-37 dB for earmuffs, 25-28 dB for earplugs, and 39-41 dB for earmuffs and earplugs used together. The data enabled the clarification of the relative importance of ear canal, head, and body conduction routes to the cochlea under different conditions: At low frequencies (< or =500 Hz), the ear canal was the dominant route of sound conduction to the cochlea for all of the device combinations considered. At higher frequencies (>500 Hz), the ear canal was the dominant route when either earmuffs or earplugs were worn. However, the dominant route of sound conduction was through the head when both earmuffs and earplugs were worn, through both ear canal and body when a helmet and earmuffs were worn, and through the body when a helmet, earmuffs, and earplugs were worn. It is estimated that a helmet, earmuffs, and earplugs together will reduce the most intense fMRI noise levels experienced by a subject to 60-65 dB SPL. Even greater reductions in noise should be achievable by isolating the body from the surrounding noise field.     

Occluded-ear simulator with variable acoustic properties.

Ear simulators were designed to replicate acoustical characteristics of the average adult ear. Due to variability of ear-canal geometry and eardrum impedance among individuals, the possibility of any one person exhibiting such "average" characteristics--especially if that person is a child and/or has a conductive pathology--is remote. Thus, ear simulators have been of only peripheral value when prescribing a hearing aid (a high output impedance device) to fit the acoustical requirements of a particular patient. Reported herein is development of a programmable artificial ear (PAE) that can account for individual differences in ear-canal geometry and eardrum impedance. It consists of a 2.0-cc coupler, microphone, amplifier, computer, PAE code, and a computer card and/or software for digitization and Fourier transformation. Required input data includes ear-canal dimensions, eardrum impedance, and output impedance of the hearing aid being tested. Sound-pressure recordings produced in the 2.0-cc coupler by the hearing aid are adjusted by the computer to what they would have been had the recordings been made at the eardrum of a particular patient wearing the same hearing aid. Good agreement was observed between experiment and theory for one test case involving a totally occluding miniature earphone.     

虚拟传声器在有源抗噪声护听器的应用

考察了虚拟传声器应用于有源抗噪声护听器的稳定性和降噪性能。理论上分析了虚拟传声器算法的稳定性,并通过引入反馈控制器和设计次级通道模型使得算法在不同护听器佩戴方式下都保持稳定,获得高噪声衰减和高稳定性。实验结果证实了虚拟传声器的应用可以在耳道口获得更高的窄带噪声衰减,而且所采取的措施可以保证护听器在不同佩戴状态下都能稳定。      

Toward bias minimization in acoustic feedback cancellation systems

A novel technique for bias suppression within acoustic feedback cancellation systems is proposed. This is achieved based on the use of all-pass filters in the forward part of the hearing aid. The poles of these filters are made time-varying, which results in a frequency response with constant magnitude and varying phase. This is a desired feature of the proposed approach, since the results from human psychoacoustics show that the human ear is not sensitive to moderate phase perturbations. The derivation of the proposed algorithms for the time variation of the location of the poles of all pass filters is based on a rigorous analysis of the phenomenon of bias in acoustic systems. Practical issues, such as the dependence of the steady-state error on the order of the all-pass filter, the number of varying poles, and their standard deviation are examined and strategies for the variation of the poles are introduced. Results obtained from a simulated hearing aid are provided to support the analysis. The quality of the processed audio signals is evaluated through subjective tests.     

Horizontal localization with bilateral hearing aids: without is better than with

This paper studies the effect of bilateral hearing aids on directional hearing in the frontal horizontal plane. Localization tests evaluated bilateral hearing aid users using different stimuli and different noise scenarios. Normal hearing subjects were used as a reference. The main research questions raised in this paper are: (i) How do bilateral hearing aid users perform on a localization task, relative to normal hearing subjects? (ii) Do bilateral hearing aids preserve localization cues, and (iii) Is there an influence of state of the art noise reduction algorithms, more in particular an adaptive directional microphone configuration, on localization performance? The hearing aid users were tested without and with their hearing aids, using both a standard omnidirectional microphone configuration and an adaptive directional microphone configuration. The following main conclusions are drawn. (i) Bilateral hearing aid users perform worse than normal hearing subjects in a localization task, although more than one-half of the subjects reach normal hearing performance when tested unaided. For both groups, localization performance drops significantly when acoustical scenarios become more complex. (ii) Bilateral, i.e., independently operating hearing aids do not preserve localization cues. (iii) Overall, adaptive directional noise reduction can have an additional and significant negative impact on localization performance.