IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO. 5, MAY 2015 923
Design of Low Complexity Adjustable Filter Bank
for Personalized Hearing Aid Solutions
Ying Wei, Member, IEEE, and Yinfeng Wang
Abstract—The emerging demand for personalized hearing aids
requires the filter bank of a hearing aid system to be capable of de-
composing the sound waves in accordance with the characteristic
of the patient’s hearing loss. In this paper, an efficient adjustable
filter bank is proposed to achieve this goal. By careful design, the
number of the subbands as well as the location of the subbands can
be easily adjusted by changing a 4-bit control signal. The proposed
filter bank has extremely low complexity due to the adoption of
fractional interpolation and the technique of symmetric and com-
plementary filters. Only one prototype filter is needed for each of
the stages, the multiple passbands generation stage and masking
stage. We show, by means of examples, that the proposed filter
bank can meet different needs of hearing loss cases with accept-
able delay.
Index Terms—Adjustable, filter bank, fractional interpolation,
hearing aids.
I. INTRODUCTION
T
HE auditory system is a very sensitive and complex net-
work. Diseases, drugs, noise, trauma and aging may have
resulted in varying degrees of hearing loss, which makes hearing
impairments one of the most common sensory disturbances in
the world. The most effective way to compensate hearing loss
is to employ a hearing aid system which is an integration of
voice amplification, noise reduction, feedback suppression, au-
tomatic program switching, environmental adaptation, and etc.
The basic function of a hearing aid system is to amplify sounds
selectively and then transfer the processed signal to the ear [1].
A schematic diagram for digital hearing aids is shown in Fig. 1.
After the analog sound signal is transformed into the digital
signal by an A/D converter, the digital signal is divided into sub-
band signals within different frequency bands by a filter bank
[2][3]. Each subband has its own amplification coefficient. The
amplified subband signals are then synthesized and fed into the
D/A converter. Filterbank-based algorithms permit an easy ad-
justment of speech amplification. Within the considered speech
Manuscript received February 09, 2014; revised October 08, 2014; accepted
February 13, 2015. Date of publication March 06, 2015; date of current version
April 09, 2015. This work was supported in part by the Shandong Province Sci-
ence and Technology Development Plan (No. 2013GGX10103), in part by the
National Natural Science Foundation (No. 61201372), in part by the Promotive
Research Foundation for Excellent Young and Middle-Aged Scientists of Shan-
dong Province (No. BS2013DX042), and in part by the Taishan Scholar Foun-
dation Project (No. 1170082963013). The associate editor coordinating the re-
view of this manuscript and approving it for publication was Prof. Søren Jensen.
The authors are with the School of Information Science and Engineering,
Shandong University, Jinan 250010, China (e-mail: eleweiy@sdu.edu.cn;
673645473@qq.com).
Color versions of one or more of the figures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identifier 10.1109/TASLP.2015.2409774
Fig. 1. A schematic diagram for digital hearing aid.
spectrum, the adjustment is fully programmable and is able to
suit patients’ comfort.
Much study has been invested into the design of digital filter
banks for selective amplification. Most of the current studies
focus on fixed (cannot be reconfigured) filter banks. Uniform
filter banks are the first and most widely used filter banks in
practice. Over the past decade, researchers have done much
work to reduce the complexity of uniform filter banks. Lattice
wave digital filter bank (LWDFB) was employed for hearing
aids [4]. LWDFB has lower complexity than the FIR filter
bank and is not sensitive to the coefficients. Then, a DFT filter
bank with a multi-dimensional logarithmic number system
(MDLNS) was reported to obtain reduced complexity [5].
Later, some well-known simple methods for critically sampled
filter banks were extended to the over-sampled case [6]. Its
efficiency comes from the flexibility to generate multiple pro-
totype filters by one method. The complexity of hearing aid
system was further reduced by using a joint stereo filter bank
to satisfy the requirements of both the audio coding and the
hearing aid application [7].
Dividing the frequency range uniformly is straightforward
yet does not consider the unique characteristic of human
hearing. Therefore, non-uniform filter banks that mimic the
resolution characteristic of human hearing have gained the
attention of hearing-aid researchers. A tree-structured filter
bank based on all-pass compensatory filters and elliptic min-
imal Q-factor (EMQF) filters was used as the analysis filter
bank in [8]. An 8-band filter bank based on frequency response
masking (FRM) was proposed for hearing compensation in [9].
Both the designs lower the complexity at the cost of delay. A
critical band-like spaced filter bank was used in [10]. It can
obtain satisfactory hearing compensation, yet the irregularity
of the subbands increases the difficulty of the design and
implementation. A 1/3 octave filter bank was realized to cover
the hearing frequency range in [11]. It was based on IIR struc-
ture thus could not provide linear phase response. In general,
non-uniform filter banks have better performance in hearing
compensation compared to the uniform filter banks. However,
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