#2014 The Acoustical Society of Japan
Acoust. Sci. & Tech. 35, 6 (2014)
Occlusion reduction system for hearing aids using active noise control technique Masahiro Sunohara, Keisuke Watanuki and Makoto Tateno Rion Co., Ltd., Higashimotomachi 3–20–41, Kokubunji, 185–8533 Japan (Received 21 April 2014, Accepted for publication 9 June 2014) Keywords: Hearing aids, Occlusion effect, Adaptive filter, Feedback active noise control, Fast H1 filter PACS number: 43.50.Hg, 43.50.Ki [doi:10.1250/ast.35.318] 1.
Introduction Many people who wear hearing aids have complained about the discomfort of their hearing aid for their own voice and/or the mastication sound [1–3]. Such discomfort is caused by the increased sound pressure at low frequencies when the ear canal is blocked by the hearing aid itself [3,4]. This is often referred to as the ‘‘occlusion effect’’ and is one of the critical issues for hearing aid users. There has been an attempt to reduce the effect by installing a small pipe in the hearing aid and venting the internal air to the outside. However, there is a risk of inducing the oscillation of sound due to acoustic feedback if the internal diameter of the pipe is too large [5]. The aim of this paper is to develop an occlusion reduction system for hearing aids, which can be regarded as an electrical embodiment of an acoustically transparent state in the ear canal completely blocked by the hearing aid. This study should be applicable to any type of in-ear headphone or hearing protector. Mejia et al. proposed a system for producing an occlusion reduction of 20 dB at 200 Hz using a fixed electrical circuit [4], but this amount of reduction is insufficient to resolve severe cases. In this paper, an occlusion reduction system using the feedback active noise control (ANC) technique with two types of adaptive filter is proposed. In addition to the conventional Filtered-x least-mean-squares (LMS) algorithm [6], the fast H1 filter (FHF) algorithm [7,8] is employed to improve the optimization for nonstationary signals such as speech. 2.
Measurement of the occlusion effect The increase in sound pressure in the ear canal was measured when a subject uttered a vowel while wearing an inthe-ear (ITE)-type hearing aid. The shell of the ITE hearing aid was designed by the standard ear impression technique to fit in each subject’s ear [9]. The right ear canal of each subject was filled with the ITE hearing aid while the left ear was free. Eight Japanese subjects were asked to utter the vowel /i:/ continuously for about 5 s. Two electret condenser microphones were used to measure the sound pressure in both ear canals. Then, the difference in the power spectrum between the left and right ears at the fundamental frequency of the vowel and its three harmonics was calculated. The result of the measurement is shown in Fig. 1. The average increase in the sound pressure was about 15 dB from
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100 to 300 Hz, although it varied considerably depending on the subject. One of the subjects experienced an increase in the sound pressure of about 30 dB. 3.
Proposed system Figure 2 depicts the configuration of the proposed occlusion reduction system incorporated in an ITE hearing aid. The system consists of a microphone (external), a receiver and a digital signal processor for the general purpose of a hearing aid, and a microphone (internal) for the purpose of picking up occluded sound. Figure 3 shows the signal block diagram for the proposed occlusion reduction system, which is based on the feedback ANC technique [10]. An occluded sound can be produced by many kinds of action in the body; however, the only sound considered in this paper is the increase in one’s own voice dðnÞ. The adaptive filter CðzÞ is updated to reduce the error signal eðnÞ obtained by the internal microphone. Reducing eðnÞ is equivalent to estimating the inverted signal of dðnÞ to be radiated from the receiver to the ear canal. Noted that the hearing aid signal sðnÞ is not affected by the occlusion reduction system while the increase in one’s own voice dðnÞ is reduced by the system. The Filtered-x LMS and FHF algorithms are used to update the coefficients of the adaptive filter CðzÞ. The Filtered-x LMS algorithm can be performed by the following equation [6]: cðn þ 1Þ ¼ cðnÞ þ xp ðnÞeðnÞ; ð1Þ kxp ðnÞk2 where cðnÞ ¼ ½c0 ; c1 ; . . . ; cN1 T denotes the coefficient vector of the adaptive filter CðzÞ, N is the tap length, n is the discrete sampling index and xp ðnÞ ¼ ½xp ðnÞ; xp ðn 1Þ; . . . ; xp ðn N þ 1ÞT is the vector of the signal input to the adaptive filter ^ filtered by the path model PðzÞ. On the other hand, the FHF algorithm can be recursively calculated through several steps [8]. The main symbols used in the equations are described as follows: Kn is the gain matrix, K n is the auxiliary gain matrix, Ks;n is the filter gain used to update the filter coefficients, Xn is the observation matrix, which has a shifting property [8] such T that Xn ¼ ½ xp ðnÞT , and An , Sn and Dn are auxiliary variables. xp ðnÞ
The FHF algorithm is performed through steps 0 to 6 as follows: [step 0] Set the initial conditions of K0 ¼ 0N2 , A1 ¼ 0N1 , S1 ¼ 1="0 , D1 ¼ 0N1 and cð0Þ ¼ 0N1 , where 0NM
M. SUNOHARA et al.: OCCLUSION REDUCTION SYSTEM FOR HEARING AIDS
Fig. 2 Configuration of the proposed system.
Fig. 1 Increase in sound pressure in the ear canal due to the occlusion effect. Filled circles denote averaged pressures and error bars denote 1 SD.
Fig. 3 Signal block diagram for the proposed occlusion reduction system.
denotes an N M zero matrix and "0 is a sufficiently large positive number. [step 1] Determine An and Sn recursively as e~n ¼ x~n þ Xn An1 ; An ¼ An1 Kn W n e~n ;
K~nþ1 ðiÞ ¼ Knþ1 ði; 1Þ; i ¼ 1; . . . ; N Ks;nþ1 ¼
ð2Þ
K~ nþ1 : þ 2 x~nþ1 K~ nþ1
ð11Þ ð12Þ
[step 5] Update the filter coefficients cðn þ 1Þ as
ð3Þ
en ¼ x~n þ Xn An ;
ð4Þ
cðn þ 1Þ ¼ cðnÞ þ Ks;nþ1 eðnÞ:
Sn ¼ Sn1 þ eTn W n e~n ;
ð5Þ
[step 6] Increment the discrete sampling index n and return to step 1.
where W n ¼ ½ 10 02 , x~n is the first row of Xnþ1 , is the admissible error and is the forgetting factor ð ¼ 1 2 Þ. [step 2] Calculate K n as " # T S1 n en Kn ¼ ð6Þ 2
