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.     

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.     

The constant-volume-velocity nature of hearing aids: conclusions based on computer simulations

In the literature there are several references which imply that various parts of a hearing aid are sources of constant volume velocity. Reported herein are the findings of an investigation of the validity of such statements. A computer scheme, referenced elsewhere, for modeling in situ hearing aids was utilized to test the constant-volume-velocity hypothesis. In particular, capabilities of the receiver, ear hook, and earmold tip to deliver constant volume velocity were investigated via a computer. To facilitate such an investigation, a universal receiver/earmold model was created. This model was broken down into "source" and "load"at three locations: the receive output, output of the ear hook, and medial tip of the earmold. At each location comparisons were made between computed values of source and load impedance. The constant-volume-velocity hypothesis was assumed to be valid for those cases where source impedance was much, much greater than load impedance. Plots of such impedances show that, for the cases investigated, this rarely occurred, except over certain frequency bands. With the exception of in-the-ear hearing aids, these results appear to contradict inferences made in the literature about the constant-volume-velocity nature of hearing aids.     

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.     

System identification of feedback in hearing aids.

The feedback problems of behind the ear (BTE), in the ear (ITE), and in the ear canal (ITEC) hearing aid categories have been investigated. All possible feedback paths (acoustical via vent, via tubing wall, mechanical, etc.) were converted to a single transfer function from the ear canal to the hearing aid microphone, here called the acoustic feedback equivalent (AFE). The attenuation of the AFE represents the maximum gain that can be used without the hearing aid starting to howl. Magnitude and phase responses of the AFE were identified on ten human subjects and on a Knowles ear manikin (KEMAR). The acoustic feedback via vent and leak between earmould and ear canal dominated the AFE. The transfer function from a reference point under the ear to the position of microphone of the different hearing aid categories was identified and used together with the AFE to calculate the maximum real ear aided gain (REAG) for the hearing aid categories. A model of the AFE, consisting of a fourth-order filter together with a delay, showed good agreement with the measured data.     

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.     

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.     

Experimental determination of cascade parameters of a hearing-aid microphone via the two-load method

Presented in this article is a computer-aided experimental method for obtaining the cascade parameters of the two-port model of a miniature hearing-aid microphone. The method is an adaptation of the "two-load" method [D.P. Egolf and R.G. Leonard, J. Acoust. Soc. Am. 62, 1013-1023 (1977)] to acoustoelectric, rather than electroacoustic, transducers. The cascade parameters of a particular microphone, determined by this method, were within 2.5 dB of the manufacturer's published open-circuit sensitivity data. In an attempt to further verify the numerical cascade-parameter data, a two-port model of the microphone was used to simulate experimental voltages developed across two different complex electrical load impedances attached to the microphone. The results showed experimental/simulation differences of no greater than 3.0 dB at any frequency. The two-port microphone model and associated cascade parameters are currently being incorporated into a computer-based plan for mathematical simulation of an entire in situ hearing aid.     

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.     

Mathematical predictions of electroacoustic frequency response of in situ hearing aids

The amplitude spectrum of an acoustic signal presented to the microphone of a hearing aid is altered drastically before it finally reaches the user's eardrum. A major part of this alteration is due to the interaction of various mechanical and acoustic resonances which are characteristic of the hearing-aid receiver and the sound transmission system linking the receiver with the eardrum. Because of the complexity of this phenomenon, there is yet no means for predicting, a priori, the true shape of the sound spectrum that will occur at the user's eardrum. This paper reports on the development and testing of just such a scheme. The accuracy of this scheme--a computer-aided mathematical technique--is measured in the laboratory on real and artificial ears. The results of those measurements show good agreement between experimental and computer-generated data below 5000 Hz.     

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.     

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.     

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.     

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.     

Noise in miniature microphones

The internal noise spectrum in miniature electret microphones of the type used in the manufacture of hearing aids is measured. An analogous circuit model of the microphone is empirically fit to the measured data and used to determine the important sources of noise within the microphone. The dominant noise source is found to depend on the frequency. Below 40 Hz and above 9 kHz, the dominant source is electrical noise from the amplifier circuit needed to buffer the electrical signal from the microphone diaphragm. Between approximately 40 Hz and 1 kHz, the dominant source is thermal noise originating in the acoustic flow resistance of the small hole pierced in the diaphragm to equalize barometric pressure. Between approximately 1 kHz and 9 kHz, the noise originates in the acoustic flow resistances of sound entering the microphone and propagating to the diaphragm. To further reduce the microphone internal noise in the audio band requires attacking these sources. A prototype microphone having reduced acoustical noise is measured and discussed.     

Boundary element modeling of the external human auditory system

In this paper the response of the external auditory system to acoustical waves of varying frequencies and angles of incidence is computed using a boundary element method. The resonance patterns of both the ear canal and the concha are computed and compared with experimental data. Specialized numerical algorithms are developed that allow for the efficient computation of the eardrum pressures. In contrast to previous results in the literature that consider only the "blocked meatus" configuration, in this work the simulations are conducted on a boundary element mesh that includes both the external head/ear geometry, as well as the ear canal and eardrum. The simulation technology developed in this work is intended to demonstrate the utility of numerical analysis in studying physical phenomena related to the external auditory system. Later work could extend this towards simulating in situ hearing aids, and possibly using the simulations as a tool for optimizing hearing aid technologies for particular individuals.     

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).     

Design of full-duplex and multifunction bidirectional CMOS transceiver for optical interconnect applications

This work presents a full-duplex and multifunction bidirectional transceiver for optical interconnect application. The transceiver utilizes a common limiting amplifier/gain stage, thus reducing total chip area and total power consumption. While providing a full-duplex bidirectional transmission with the aid of a hybrid circuit between the electrical input/output (I/O) and the optoelectronic signals from the transmitter and receiver circuits, it also allows for a half-duplex operation with the aid of a switch between the transimpedance amplifier signals and the transmitter electrical input from the I/O port. The multifunction bidirectional CMOS transceiver is designed in a 0.13 µm Si-CMOS technology, with power dissipation of 79 and 54.4 mW for the transmitter and receiver, respectively. It shows a 3-dB bandwidth of 5.58 and 5.69 GHz for the transmitter and the receiver respectively and with a 3-dB gain of 66.14 and 69.6 dB, in full-duplex mode. The transceiver operates up to 7 Gb/s in full-duplex mode.     

Extracting the invariant model from the feedback paths of digital hearing aids

Feedback whistling is a severe problem with hearing aids. A typical acoustical feedback path represents a wave propagation path from the receiver to the microphone and includes many complicated effects among which some are invariant or nearly invariant for all users and in all acoustical environments given a specific type of hearing aids. Based on this observation, a feedback path model that consists of an invariant model and a variant model is proposed. A common-acoustical-pole and zero model-based approach and an iterative least-square search-based approach are used to extract the invariant model from a set of impulse responses of the feedback paths. A hybrid approach combining the two methods is also proposed. The general properties of the three methods are studied using artificial datasets, and the methods are cross-validated using the measured feedback paths. The results show that the proposed hybrid method gives the best overall performance, and the extracted invariant model is effective in modeling the feedback path.