18th Annual International Conference of the IEEE Engineering in Medicine and Biology Society, Amsterdam 1996$
2.4.2: Aids for the Hearing Impaired
HEARING IMPAIRMENT SIMULATION FOR THE
PERFORMANCE EVALUATION OF HEARING AID SYSTEM
Dong-Ook Chung', Won Doh', Dae-Hee Youn'
Jae-Yeon Choi**, Hyo-Chang Woo**,Dong- Wook Kim***,Won-Ki Kim**"
* Dept. of Electronic Eng., Yonsei Univ. 134 Shinchondong Sudaemunku, Seoul, Korea
** Samsung Medical Center, 50 Ilwondong Kangnamku, Seoul, Korea
*** Samsung Biomedical Research Institute, 50 Ilwondong Kangnamku, Seoul, Korea
ABSTRACT
With the advent of high speed digital signal processing chips,
many digital techniques have been introduced to hearing aids. As
new techniques are developed, subject-based test is needed to
verif). the performance of each of them which requires much
time and cost. In this paper, we proposed an indirect way to
evaluate and predict the performance of the hearing aid system
without the help of such subject-based test. I'he system is
developed based on the model constructed from the auditory test
results of sensorineural hearing impaired persons. To verify the
proposed system, processed signal was heard to normal persons
and their response were compared with those of real impaired
person. .4s results, only a slight difference was found between
them. In addition, three kinds of currently available digital
hearing aid algorithm were applied to the system and evaluated.
1. INTRODUCTION
Hearing loss can be classified into three categories: conductive,
sensorineural, and mixture of both. The conductive loss is caused
by R problem in the outer ear and/or middle ear so that the
power of input acoustic energy is leveled down on its way from
tympanic membrane to cochlea. In this case, the linear
amplification could be a good remedy which can be easily
implemented using an analog circuitry. The sensorineual loss is a
result of a difficulty at the level of the cochlea or a damage
along the neural pathways to the brain. However, in this case,
although linear gain hearing aids do offer help, significant
degradation in speech intelligibility is often complaint.
'
'1'0 compensate for the hearing loss of this kind, many
researchers have proposed novel hearing aid systems for best
fitting. Among those, multichannel filterbank systems are most
common. But the filterbank system often exhibits undesirable
nonlinear spectrum distortion. To overcome this problem, Asano
et a1.[31 proposed a system that uses only one FIR filter to
minimize the distortion and to reduce the complexity of the
system.
However, although many researchers have developed various
types of aid systems so far, there still exist many problems to
solve to meet the urgent requirement of the hearing impaired.
The most important thing in development stage of hearing aid
system would be how to effectively consider the "feeling" that
is experienced by a listener. The only and the most
straightforward way to evaluate the performance would be a
direct test for impaired person. However, as the majority of
0-7803-3811-1/97/$10.00 OIEEE
impaired listeners are of old age, it is hard to perform a lot of
various kinds of experiments on them, and even worse. their
response are not reliable from time to time. So, it is strongly
required to develop a way to evaluate and predict the
performance of the hearing aid system without the help of such
subject-based test. Similar consideration have been done by
Rutledgel41. Rutledge modeled impaired listener's transfer function
using neural network and developed an objective measure from
it. And Chabries[fl's approach involves the modeling of impaired
person's auditory parameters based on filterbank system when
developing hearing aid. But in case of Rutledge's neural network,
it is hard to manipulate auditory parameters. and Chabries' s
filterbank seems to be too much complex when it comes to the
problem of 'impairment simulation'.
'I'he proposed method in this paper simulates the main
characteristics of sensorineural hearing loss -reduced dynamic
range and gain as a function of frequency and input signal level-
so that it could be used to give help when developing hearing
aid. The experience of pseudo impairment gave u s impressive
feeling that is beyond imagination.
2. PROCESSING ALGORITHM
,.
I he purpose of the simulation system is to provide normal
listener with a way to experience the 'feeling' of impaired
listener. This can be accomplished by mapping the auditory
parameters of normal listener to those of impaired listener. As
the main characteristic of sensorineural hearing loss can be
characterized as the increment of hearing level, and the
compression of dynamic range, it could be feasibly modeled using
signal processing technology.
Processing steps are as follows. First, perform Am conversion
with the sampling frequency of 12kHz. For the spectrum
measurement, 128 digitized samples(l0.67msec) are grouped into a
block with 50% overlap, multiplied by blackman window, and
short time FFT is performed. Blackman window was chosen
because it shows lower side-lobe level than other windows.
Larger side-lobe level may cause inaccuracy in spectrum
measurement. Then, from the measured frequency spectrum.
hearing levels in 19 critical bands are calculated. This critical
band processing is consistent with human psychoacoustic model
and effect of large main-lobe bandwidth of blackman window
may be compensated. The hearing loss gain of each critical batid
is calculated using HLF(Hearing Loss Function). IiLF is B
function which transforms the dynamic range of normal person
to that of impaired person, i.e.. it defines a relation between input
415
 18th Annual International Conference of the EEE Engineering in Medicine and Biology Society, Amsterdam 1996
2.4.2: Aids for the Hearing Impaired
signal level and output signal level at each frequency. HLF could
be obtained from the audiometric data of the normal and the
impaired person at 6 frequency points(250, 500,lk, 2k, 4k, 8M).
From this 6 HLFs, 19 HLFs at the center frequency of each
critical band are obtained using linear interpolation. With these,
the hearing loss gains at 19 critical bands can be calculated. And
finally, a frequency sampling filter is designed in such a way
that the frequency response of the filter should be the desired
gain calculated in above step. Then, time domain filtering is
performed and output speech is generated. Fig. 1 shows the
block diagram of proposed system.
-~ -
Fig, 1. Block diagram of proposed system
3. EVALUATION EXPERIMENT
Two experiments have been performed. First, expenments for
the verification of the performance of developed HIS(Hearinp
Impairment Simulator) system was done. And then, three kinds
of currently available digital hearing aid algorithm are applied to
the system and evaluated.
For the first experiment, auditory parameters, such as absolute
hearing threshold level, most comfortable level, uncomfortable
level, and etc., of a real sensorineural hearing impaired person
were used to process speech signal that is used for audiometry.
And then, with this processed speech test data, audiometry for
normal person is done to get auditory characteristics. If the
system works as desired, two audiometric results of original
impaired person and simulated person should be the same, or at
least, of little difference. The auditory test to get parameters of
listener is composed of a) measurement of absolute hearing
threshold, b) speech reception test(SRT), and c) speech
discrimination test(SDT). The second experiment is for simple
comparison among three well-known digital hearing aid
algorithmshnplitude compression filterbank, linear gain filterbank,
CLAIDHA) using verified HIS. As long as the HIS works fine,
normal person would perceive distortion just like impaired pason
does when h e 4receives speech signal processed with HIS. To
compensate for this, hearing aid may be applied prior to HIS. As,
in theory, the function of HIS and hearing aid is inverse to each
other, distortion should be removed. For this test, speech
discrimination test with HA+HIS configuration were performed.
The higher the score the better fitting is assumed.
All clinical experiment were performed on 2 male persons and
2 female persons with normal hearing level at the audiology lab
in Samsung medical center. The test word sets for SRT test
were composed of spondeed words which are pronounced with
equal stress on both syllables, and for discrimination test,
phonetically balanced word set were used. These word sets were
recorded on DAT recorder using audiometer. The system was
implemented on IBM PC with 16bit Sound Blaster card as VO
interface and the earphone used as output transducer was
TDH-49.
Table 1 shows absoluxe hearing threshold level and SRT test
results for simulated listeners and original impaired listener. The
value in parenthesis is the hearing level at normal condition, i.e.,
results with original speech test data set. As can be seen, the
difference between two listeners falls within 5dB except cases in
which normal HTL is greater than 1CdB. It can be seen that
SRT results are better than expected except the case of subject
D, whose normal SRT score is slightly higher than others.
However, it is not clear if this high normal SRT score is a
reason that caused that amount of error. We believe more works
should be done regarding this matter.
Table 2 shows the evaluation result using SDT for three
hearing aid algorithms. CLADHA seems to be better than others.
Table 1. Threshold table and SRT.
Table 2. SDT score for each hearinrr aid svstems.
DersonM HIS HA-A HA-B HA-C
subA 80% 94% 88% 88%
subB 80% 96% 90% 86%
subC W% 96% 88% 86%
1 sub D I 82% 1 94% I 88% I 86% I
* HA-A: CLAIDHA, HA-B: Amplitude compression
filterbank, HA-C: Linear gain filterbank
4. CONCLUSION
In this paper, an indirect way to evaluate and predict the
performance of the hearing aid system without the help of
subject-based test is proposed. The system was designed based
on a real auditory test data of an impaired person. Experiments
to veri% the performance of the system was performed, and the
result showed little difference between real impaired person and
simulated person. The real time implementation using Morotola's
DSP 96002, and Samsung's SSP1605 is under working.
5. References
111 E. Vichur, "Signal processing to imDrove speech intelligihility in
perceptive deafness," 1.Awnst. SOC Am, vol. 53, pp.1646-1657,
Jun, 19'73.
[21 R. P. Rippman, L. D. Braida, and N.I. Durlach, "Study of
mdtichannel amplitude compression and linear amplification for
persons with sensorineural hearing loss," J. Acoust. Soc. Am., vol.
69, pp.524-534, Feb 1981.
[31 F. Asano et al. "A digital hearing aid that compensates, loudness
for sensorineural listeners," Proc IEEE ICdSSP, pp.3625-3628,
1991.
[41 J. C. Rutledge et al. "Objective measures for hearing loss
compensation techniques," Proc IEEE ICASSP, pp.141-144, 1993.
[51 D. M. Chabries et al. "Application of a human audibrv model to
loudness pxception and hearing compensation," Proc. IEEE
~
ICASSP, pp.3527-3530,1%
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