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Configurable Digital Hearing Aid System with Reduction of Noise for Speech
Enhancement Using Spectral Subtraction Method and Frequency Dependent
Amplification

Conference Paper · February 2019

DOI: 10.1109/TENCON.2018.8650450

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 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)

Configurable Digital Hearing Aid System with Reduction of

Noise for Speech Enhancement Using Spectral Subtraction
Method and Frequency Dependent Amplification.
Biswajit Saha1, Shakil Khan1, Celia Shahnaz, Shaikh Anowarul Fattah, Mohammad Tariqul Islam, Asir Intisar Khan
Department of Electrical and Electronic Engineering
Bangladesh University of Engineering and Technology, Dhaka-1205
Email:celia.shahnaz@gmail.com; asir@eee.buet.ac.bd
1
These authors contributed equally to this work

Abstract—The greatest challenge for hearing impaired people is
the realization of speech in a noisy environment. For the
compensation of hearing losses, hearing aid devices are used by
hearing impaired people because of being programmable to
regulate gain values with different frequencies, reduction of noise
for speech enhancement and improved signal to noise ratio (SNR)
which have made it better in performance than analog hearing
aid. The speech signal is always deteriorated by background
noise in a noisy environment which degrades the original speech.
To develop the speech quality, perceptibility and degree of
listener’s exhaustion, speech enhancement is done using various
methods. In this paper, we propose a configurable digital hearing
aid system with reduction of noise for speech enhancement using
spectral subtraction method using MATLAB. In this work,
configurable digital hearing aid includes a combination of
frequency dependent amplification of speech and noise reduction
filter for background noise reduction which will provide more
flexibility to hearing impaired people along with improvement of
quality of the speech signal.
Keywords— Hearing aid, Spectral Subtraction, Noise
Reduction, Speech Enhancement, Frequency Dependent
Amplification;
I. INTRODUCTION
Hearing loss is defined as partial or complete incapability to
hear. Diminishing sensitivity to the sounds which can be heard
normally interprets the existence of hearing loss. The term,
hearing impaired, is usually used for those people who have
lack of sensitivity to sound which can be normally heard in
speech frequencies. The intensity of a hearing loss is sorted by
the rise in volume above the normal level necessary before the
detection of the listener. Hearing loss is usually measured by
the difference between auditory threshold of a person and that
of a person with normal auditory sensitivity [1]. Hearing aid
devices are designed for the improvement of hearing by
making sound audible to hearing impaired persons [2].The
purpose of hearing aid is to amplify audio signal for making it
audible to the persons with hearing loss. Different amount of
hearing loss for a person [1] have been shown in Table I.
Due to wide application prospects, much research has been
carried out on designing process of configurable Digital
Hearing Aids. A digital hearing aid is proposed in [3] with
noise reduction using wavelet coefficients thresh-holding. A
study of hearing aid with different frequency shaping function
TABLE I. Different amount of hearing loss
Level Degree of hearing loss Range of hearing
loss(dB)
1 Normal -10 to 26
2 Mild 27 to 40
3 Moderate 41 to 70
4 Severe 71 to 90
5 Profound 91+
is proposed in [4]. The model of a digital hearing aid system
with digital infinite impulse response (IIR) filter was proposed
in paper [5] where the coefficients of the filter were attained
from the genetic algorithm (GA) optimization process. With
the parallelization of variable digital low pass filters, band
pass filters and high pass filters leading to a 3 channel VFB
method has been proposed in [6] for the application of digital
hearing aid system. Nowadays, for being smaller in size,
digital hearing aids can be kept hidden inside the ear and can
reproduce the sound perfectly. Presently, a small
programmable computer which is capable of amplifying
millions of different sound signals has been installed in the
hearing aid devices, thus developing the hearing capability of
the people with hearing loss [8]. Much research has been done
also on the noise reduction for speech enhancement. The
objective of speech enhancement is to develop the quality of
corrupted speech. Speech enhancement algorithms are
techniques used for the suppression of noise, and hence
develop the perceived speech quality and speech intelligibility
[6-7]. During the suppression of background noise, the crucial
part is not to distort the speech signal[9]. Different types of
noise reduction algorithms have been developed but most of
the algorithms are based on transform domain approach,
adaptive filtering techniques, and different model-based
methods. Amongst various speech enhancement methods,
DFT-based transforms domain methods have been widely
used in the form of spectral subtraction [10].
In this context, we have developed a hearing aid system with
noise reduction by spectral subtraction method for speech
enhancement. The noise reduction method using wavelets
proposed in [3] has the drawback that the chosen threshold
may not match the specific distribution of signal and noise

978-1-5386-5457-6/18/$31.00 ©2018 IEEE 0735

 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)
components at different scales and orientations. The reason
behind choosing spectral subtraction for noise reduction is its
simplicity of implementation for stationary noise and if a
reasonably limited loss of speech intelligibility is acceptable,
the spectral subtraction technique seems to allow us to remove
a fair amount of noise, including cancelling almost completely
musical noise. Our hearing aid system is configurable and it
can be used for patients with different hearing losses by just
changing the frequency adjustment filter which gives the
flexibility to the hearing impaired persons to meet the
specification for increasing the gain of the audio signal. In our
proposed work, filters are designed to reduce noise by spectral
subtraction method to increase the gain at specific frequencies
that patients have difficulty to hear and amplitude adjustment
have been done to keep the signal level within the maximum
and minimum value. This system can be effective to the
hearing impaired patients to improve the quality of the speech
[11] audible for them.
II. METHODOLOGY:
The MATLAB implementation of Configurable Digital
Hearing Aid System is shown by the block diagram in fig. 1.
The input signal was the voice of a person recorded in a noisy
environment. The input recorded signal will pass through
several filters i.e. noise reduction by spectral subtraction filter,
frequency adjustment filter and amplitude adjustment filter
before constructing an output speech signal which is audible to
the people with hearing loss.
Input Recorded Signal
Reduction of Noise
Frequency Adjustment
Amplitude Adjustment
Output Speech Signal
Fig. 1. Block Diagram of Configurable hearing aid system
A. Reduction of Noise by Spectral Subtraction Method
An independent noise suppression technique is used for the
reduction of the spectral effects of noise in speech. Digital
speech processors operating in practical environments may
require suppression of noise from the digital waveform for
effective performance. A computationally efficient and
processor independent way to effective digital speech analysis
is offered by spectral subtraction method. The method,
requiring about the same computation as high-speed
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convolution, removes stationary noise from a speech by
subtracting the spectral noise bias calculated during non-
speech activity.
The background noise is spontaneously mixed with the
speech. The characteristic of background noise is that it
remains locally static to the degree that its spectral magnitude
anticipated value just before speech activity equals its
anticipated value during speech activity. If the surrounding
changes to a new static condition, there exist enough time
(about 300 ms) for the estimation of a new background noise
spectral magnitude anticipated value before speech activity
begins. For the gradually varying no static noise environment,
the algorithm necessitates a speech activity detector to signal
the program that speech has been stopped and a new noise bias
can be estimated. Finally, it is thought that effective reduction
of noise is feasible by eliminating the effect of noise from just
the magnitude spectrum. Speech, appropriately low-pass
filtered and digitized, is analyzed by windowing data from
half-overlapped input data buffers. The magnitude spectra of
the windowed data are calculated and the spectral noise bias
calculated during non-speech activity is subtracted off.
This spectral subtraction method is an effective method of
noise reduction. An average signal spectrum and average noise
spectrum are evaluated in different parts of the recorded audio
signal and subtracted from each other, so that average signal-
to-noise ratio (SNR) is improved. Signal may be distorted by a
wide-band, static, additive noise, the noise estimation is the
same during the analysis and the restoration and the phase is
the same in the original and restored signal. For the
measurement of SNR, we have used “Decision-Directed”
Estimation Approach from [12].
The recorded signal 𝑦(𝑚) is a total of the desirable signal
𝑥(𝑚) and the noise 𝑛(𝑚):
𝑦(𝑚) = 𝑥(𝑚) + 𝑛(𝑚)
In the frequency domain, this can be expressed as:
𝑌(𝑗𝜔) = 𝑋(𝑗𝜔) + 𝑁(𝑗𝜔)
=> 𝑋(𝑗𝜔) = 𝑌(𝑗𝜔) − 𝑁(𝑗𝜔)
Where 𝑌(𝑗𝜔), 𝑋(𝑗𝜔), 𝑁(𝑗𝜔) are Fourier transforms of
𝑦(𝑚), 𝑥(𝑚), 𝑛(𝑚) respectively.
As the statistic parameters of the noise are unknown, the noise
and the speech signal are replaced by their estimates:
𝑋̂(𝑗𝜔) = 𝑌(𝑗𝜔) − 𝑁 ̂ (𝑗𝜔)
The noise spectrum estimation 𝑁 ̂ (𝑗𝜔) is related to the
expected noise spectrum 𝑬[|𝑁(𝑗𝜔)|] which is usually
calculated using the time-averaged noise spectrum 𝑁 ̅(𝑗𝜔)
taken from parts of the recorded signal where only noise is
available. The noise estimation is given by:
 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)

𝑘−1
1 Where P(x) is:
̂ (𝑗𝜔) = 𝑬[|𝑁(𝑗𝜔)|] ≅ |𝑁
𝑁 ̅(𝑗𝜔)| = ∑|𝑁𝑖 (𝑗𝜔)|
𝑘 𝑥; 𝑥 ≥ 0
𝑖=0 𝑝(𝑥) = {
0; 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
Where |𝑁𝑖 (𝑗𝜔)| is the amplitude spectrum of the 𝑖 𝑡ℎ of the K
frames of noise. Noise estimate in 𝑘 𝑡ℎ frame may be obtained 𝑁𝑘−1 (𝑗𝜔) is the variance of the noise spectrum in the
by filtering the noise using first-order low-pass filter: previous frame, 𝑋̂(𝑗𝜔) is estimate of the restored signal and
η is constant (0.9 < η < 0.98). The variance is usually replaced
̂𝑘 (𝑗𝜔) = |𝑁
𝑁 ̃𝑘 (𝑗𝜔)| = 𝜆𝑛 . |𝑁
̃𝑘−1 (𝑗𝜔)| + (1 − 𝜆𝑛 ). |𝑁𝑘 (𝑗𝜔) by spectral power of noise estimate:

Where 𝑁 ̃𝑘 (𝑗𝜔), the smoothed noise estimate in 𝑖 𝑡ℎ frame, 𝜆𝑛 ̅̅̅̅̅̅

𝑆𝑁𝑅𝑘
𝑎 𝑝𝑟𝑖𝑜𝑟𝑖
(𝜔) = (1 − 𝜂). 𝑃(𝑆𝑁𝑅𝑘𝑎 𝑝𝑜𝑠𝑡𝑒𝑟𝑖𝑜𝑟𝑖 (𝜔) − 1)
is the filtering coefficient (0.5 ≤ 𝜆𝑛 ≤ 0.9). To obtain the noise 2
|𝑋̂𝑘−1 (𝑗𝜔)|
estimate, the part of the recording containing only noise that + 𝜂.
|𝑁̂𝑘 (𝑗𝜔)|2
precedes the part containing speech signal should be analyzed
(the length of the analyzed fragment should be at least 300 |𝑌𝑘 (𝑗𝜔)|2
𝑎 𝑝𝑜𝑠𝑡𝑒𝑟𝑖𝑜𝑟𝑖
𝑚𝑠). To achieve this, additional speech detector has to be 𝑆𝑁𝑅𝑘 (𝜔) =
used. ̂𝑘 (𝑗𝜔)|2
|𝑁

The spectral subtraction error may be defined as:

B. Frequency Adjustment
𝜖 ≝ 𝑋̂ (𝑗𝜔) − 𝑋(𝑗𝜔)
The frequency adjustment filter is designed to apply different
This error decreases the quality of the signal and introduces gains at different frequencies for the correction of hearing loss
distortion known as residual noise or musical noise. The error at definite frequencies. As input parameter, the filter takes in a
is a function of expected 𝐸[|𝑁(𝑗𝜔|] or average 𝑁 ̅(𝑗𝜔) noise vector of frequencies which is configurable. In our system, the
spectrum estimate: input recorded audio signal (.wav file) with noise is taken
̅(𝑗𝜔)| − 𝐸[|𝑁(𝑗𝜔)|] using the ‘audioread’ function of MATLAB. The signal is
𝜖 = 𝑁(𝑗𝜔) − 𝐸[|𝑁(𝑗𝜔)|] ≅ |𝑁
sampled at 44100 Hz using this built-in function. The
Therefore, the longer noise section is used in analysis, the frequency vector is initialized for the range 0 to 22050 Hz.
more accurate the noise estimate is.
The frequency adjustment filter divide this whole range into
The signal-to-noise ratio may be defined in frequency domain small regions as per the values of input vector of frequencies
as SNR a priori (for clean signal) or SNR a posteriori (for and apply different gain for different region as per the
noisy signal). SNR in 𝑘 𝑡ℎ frame is given by: necessity of hearing impaired person. According to the
Specification of Hearing Aid Characteristics [13] from The
𝑎 𝑝𝑟𝑖𝑜𝑟𝑖 |𝑋𝑘 (𝑗𝜔)| 2 American National Standards Institute (ANSI), the shape of
𝑆𝑁𝑅𝑘 (𝜔) ≝
|𝑁𝑘 (𝑗𝜔)| 2 the curve is taken. For example, a hearing impaired person
with moderate hearing loss at the frequency range 4000-5000
𝑎 𝑝𝑜𝑠𝑡𝑒𝑟𝑖𝑜𝑟𝑖 |𝑌𝑘 (𝑗𝜔)| 2 Hz, the frequency adjustment curve is shown at Fig. 2.
𝑆𝑁𝑅𝑘 (𝜔) ≝
|𝑁𝑘 (𝑗𝜔)| 2
For this curve, the input frequency vector [250 1000 4000
During the restoration process, the clean signal is not known, 5000].The regions will be 0 to 250Hz; 250 to 1000Hz; 1000 to
hence the SNR a priori value has to be estimated. Using the 4000Hz; 4000 to 5000Hz and 5000 to 22050 Hz
Gaussian model, optimal SNR in 𝑘 𝑡ℎ frame may be defined as:
For 0-250 Hz region, a linearly increasing function is
generated which is
𝑎 𝑝𝑟𝑖𝑜𝑟𝑖
̅̅̅̅̅̅
𝑆𝑁𝑅𝑘 (𝜔) = (1 − 𝜂). 𝑃(𝑆𝑁𝑅𝑘𝑎 𝑝𝑜𝑠𝑡𝑒𝑟𝑖𝑜𝑟𝑖 (𝜔) − 1)
2 . 25 ∗ (𝑔 − 1)
|𝑋̂𝑘−1 (𝑗𝜔)| 𝑓𝑖𝑟𝑠𝑡𝐶 = ;
+ 𝜂. 𝑓𝑖𝑟𝑠𝑡
𝑣𝑎𝑟(𝑁𝑘−1 (𝑗𝜔))
𝑘
𝑎 𝑝𝑜𝑠𝑡𝑒𝑟𝑖𝑜𝑟𝑖 |𝑌𝑘 (𝑗𝜔)|2 𝑔𝑎𝑖𝑛 = 𝑓𝑖𝑟𝑠𝑡𝐶 ∗ ;
𝑆𝑁𝑅𝑘 (𝜔) = 𝑁𝑇
𝑣𝑎𝑟(𝑁𝑘 (𝑗𝜔))
Where first=250; g is the highest gain value which is an input
parameter; k is used as pointer for numbers of samples for
modification; N is the total number of samples and T is
sampling period.

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 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)

For 250-1000 Hz region, an exponentially rising curve is removes a considerable amount of noise in the low power
generated using the following equations levels.
𝑠𝑒𝑐𝑜𝑛𝑑𝐶 = 𝑓𝑖𝑟𝑠𝑡𝐶 ∗ 𝑓𝑖𝑟𝑠𝑡;
𝑠𝑒𝑐𝑜𝑛𝑑 − 𝑓𝑖𝑟𝑠𝑡 III. RESULT ANALYSIS & DISCUSSION
𝑠𝑒𝑐𝑜𝑛𝑑𝐶2 = ;
10 The above mentioned filters were applied to the recorded
𝑔𝑎𝑖𝑛(𝑘 + 1) = 𝑔 + (𝑠𝑒𝑐𝑜𝑛𝑑𝐶 − 𝑔) speech signal and the performance was assessed through the
𝑘 simulations in MATLAB. In this paper, two different hearing
( ) − 𝑓𝑖𝑟𝑠𝑡
∗ 𝑒𝑥𝑝 (− 𝑁 ∗𝑇 ); losses of different patients at two different noisy environment
𝑠𝑒𝑐𝑜𝑛𝑑𝐶2
has been taken into account to solve these problems by using
Where second = 1000 our different filters to get a final sound audible to patients.

For 1000-4000 Hz region, a constant gain is necessary. So, the Case I: Our first patient has a moderate hearing loss at a
equation will be following: medium frequency range of 4000 - 5000Hz. The suitable gain
𝑠𝑒𝑐𝑜𝑛𝑑 is 45 dB on the above range of frequency. For this patient, the
𝑡ℎ𝑖𝑟𝑑𝐶 = 1 + 𝑠𝑒𝑐𝑜𝑛𝑑𝐶 ∗ 𝑒𝑥𝑝 (− );
𝑠𝑒𝑐𝑜𝑛𝑑𝐶2 frequency adjustment function is shown in Fig.2
𝑡ℎ𝑖𝑟𝑑 − 𝑠𝑒𝑐𝑜𝑛𝑑
𝑡ℎ𝑖𝑟𝑑𝐶2 = ;
5
𝑔𝑎𝑖𝑛 = 𝑔;
Where third = 4000

For 4000-5000 Hz region, a linearly decreasing function is

generated which is
. 6 ∗ (𝑔 − 1)
𝑓𝑜𝑢𝑟𝑡ℎ𝐶 = ;
𝑓𝑜𝑢𝑟𝑡ℎ
𝑘
𝑔𝑎𝑖𝑛 = 𝑔 − 𝑓𝑜𝑢𝑟𝑡ℎ𝐶 ∗ (( ) − 𝑡ℎ𝑖𝑟𝑑)
𝑁∗𝑇
Where fourth = 5000

For 5000-22050 Hz region, the curve is exponentially

decaying with the following equations
𝑘
𝑓𝑖𝑓𝑡ℎ𝐶 = 𝑔 − 𝑓𝑜𝑢𝑟𝑡ℎ𝐶 ∗ ( − 𝑡ℎ𝑖𝑟𝑑 ) ;
𝑁∗𝑇
𝑓𝑠
− 𝑓𝑜𝑢𝑟𝑡ℎ
𝑓𝑖𝑓𝑡ℎ𝐶2 = 2 ;
5
𝑘
( ) − 𝑓𝑜𝑢𝑟𝑡ℎ
𝑔𝑎𝑖𝑛 = 1 + (𝑓𝑖𝑓𝑡ℎ𝐶 − 1) ∗ 𝑒𝑥𝑝 (− 𝑁 ∗𝑇 )
𝑓𝑖𝑓𝑡ℎ𝐶2 Fig. 2.Frequency adjustment function 1

The regions can be changed by changing the values in input
frequency vector and the equations of gain functions for any
regions can be changed accordingly which give more
flexibility to adjust the frequency for hearing impaired persons
with different hearing losses.
C. Amplitude Adjustment
The recorded signal will pass through an amplitude adjustment
filter after noise reduction and frequency adjustment filter.
This amplitude adjustment filter will check the amplitude level
in terms of output power of the signal and output level of each
sample will be compared with a higher and lower threshold
level. If the signal level exceeds the upper threshold value,𝑃𝑠𝑎𝑡
then it is reduced to 𝑃𝑠𝑎𝑡 .And if the signal level is lower than a
lower threshold value, 𝑃𝑙𝑜𝑤 it is reduced to zero. The filter also
The signal is recorded in the classroom where Excessive
background noise or reverberation is present which can
decrease the intelligibility of speech. In Fig. 3, two graphs
have been plotted which are the magnitude response for input
as well as the modified output signal for the hearing impaired
person who is suffering from the above frequency losses. The
red portion of Fig. 3 indicates the noise present in the speech
which has been defined as an input of the noise reduction
filter.
A spectrogram is a visual representation of the spectrum of
frequencies of sound with the variation of time. Spectrogram
for first patient has been shown in Fig 4. From the Fig. 3 and
Fig. 4, it is understood that the noise reduces by a significant
amount and the amplitude of the signal also changes as per the
necessity of first patient.

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 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)
Fig. 3.Amplitude vs. Time plot for first recorded audio signal
(with noise) and hearing aid output signal (without noise).
Fig. 4.Spectrogram of original speech and hearing aid output
for first audio signal.
Case II: The second patient has a severe hearing loss at a
medium frequency range of 5000 - 9000Hz.The required gain
is 60db on this frequency range. So the frequency adjustment
function is shown in Fig. 5. The original signal was recorded
in a street where noise from vehicles, people etc. were present.
With the aid of frequency adjustment function shown in Fig.
5, the gain of speech signal has been altered on the particular
frequency range. Fig. 6 shows the magnitude response of input
speech signal and modified signal without noise.
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Fig. 5.Frequency adjustment function 2
Fig. 6.Amplitude vs. Time plot for second recorded audio
signal (with noise) and hearing aid output signal (without
noise).
Spectrogram for second patient has been shown in Fig. 7. It is
seen after the comparison of spectrograms of the original input
signal and the output signal that the amount of noise has been
noticeably reduced.
So, it is seen from our results that our hearing aid system is
not only capable of adjusting gain according to the need of a
hearing impaired person by changing the frequency
adjustment function; it can also reduce the background noise
by spectral subtraction method for the improvement of quality
of the recorded speech signal.
 Proceedings of TENCON 2018 - 2018 IEEE Region 10 Conference (Jeju, Korea, 28-31 October 2018)
Fig. 7.Spectrogram of original speech and hearing aid output
for second audio signal.
IV. HUMANITARIAN IMPACT
According to World Health Organization (WHO), more than
5% of the world’s population has disabling hearing loss (432
million adults and 34 million children). It is estimated that by
2050, more than 900 million people will have disabling
hearing loss [14]. But only a small percentage of them use
hearing aids and the problems associated with wearing a
hearing aid, customer dissatisfaction with the performance of
hearing aid and the expense of premium technology hearing
aids with acoustical noise processing features lead to the lower
use of hearing aid devices. Current hearing aids yield to the
limitation of inadequate spectral adjustment and partial noise
reduction leading to sub-standard clarity and audibility
restoration, as well as sub-standard speech perception in noisy
environments. In this context, our proposed hearing aid system
can be effective in solving the above problems and it will
provide more benefit to the people than the available hearing
aids as this system has combined frequency dependent gain
adjustment, noise reduction for speech enhancement and
amplitude adjustment which will provide greater convenience
and flexibility to the user of this hearing aid.
V. CONCLUSION
We have developed an efficient digital hearing aid system
which is capable of both modifying the gain according to the
necessity of the patient and reducing of background noise for
improving the quality of the sound audible to the patient. By
using the frequency adjustment filter, amplification can be
done only at the frequencies where the hearing impaired
person have difficulty to hear, and thus eliminating the
problem with typical amplifier which amplify the whole signal
including the noise. The use of different filters like noise
0740
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reduction, frequency adjustment and amplitude adjustment
filters can make the signal more satisfactory to the hearing
impaired persons with less complexity and great flexibility.
REFERENCES
[1] O. O. Khalifa, M. H. Makhtar, and M. S. Baharom, “Hearing Aids
System for Impaired Peoples,” Inf. Sci. (Ny)., vol. 2, no. 1, pp. 23–26,
2004.
[2] https://en.wikipedia.org/wiki/Hearing_aid.
[3]https://www.clear.rice.edu/elec301/Projects01/dig_hear_aid/index.
html.
[4]N. Kaur and H. S. Ryait, “Study Of Digital Hearing Aid Using
Frequency Shaping Function,” vol. 2, no. 5, pp. 71–77, 2013.
[5]P. Srisangngam, S. Chivapreecha, and K. Dejhan, “A Design of
IIR Based Digital Hearing Aids Using Genetic Algorithm Md ( J ,) =
Md ( J ,) , if J , = J , J ,. o Xl Xo = [ XOI X02 X03 X04 SOL SOB
SOH ] Nb Pm ’,” pp. 967–970, 2011.
[6]T. L. Kumar and K. S. Rajan, “Speech Enhancement Using
Adaptive Filters 1,” vol. 2, no. 2, pp. 92–99, 2012.
[7]C. Juang, C. Lin, and S. Member, “Noisy Speech Processing by
Recurrently Adaptive,” Control, vol. 9, no. 1, pp. 139–152, 2001.
[8]C. Science and S. Engineering, “Speech Processing for
Sensorineural Hearing Impairment : A Review,” vol. 3, no. 3, pp.
710–712, 2013.
[9]S. Kamath and P. Loizou, “A multi-band spectral subtraction
method for enhancing speech corrupted by colored noise,” ICASSP,
IEEE Int. Conf. Acoust. Speech Signal Process. - Proc., vol. 4, no. 2,
p. 4164, 2002.
[10]S. Boll, “Suppression of acoustic noise in speech using spectral
subtraction,” IEEE Trans. Acoust. Speech Signal Process., vol. 27,
no. 2, pp. 113–120, 1979.
[11]T. Islam, C. Shahnaz, and S. A. Fattah, “Speech Enhancement
Based on a Modified Spectral Subtraction Method,” no. 3, pp. 1085–
1088, 2014.
[12]Y. Ephraim and D. Malah, “Speech Enhancement Using a-
Minimum Mean- Square Error Short-Time Spectral Amplitude
Estimator,” IEEE Trans. Acoust. Speech Signal Process., no. 6, pp.
1109–1122, 1984.
[13]https://law.resource.org/pub/us/cfr/ibr/002/ansi.s3.22.2003.html
[14]http://www.who.int/news-room/fact-sheets/detail/deafness-and-
hearing-loss