A Hybrid MFWT Technique for Denoising
Audio Signals
2022 2nd International Conference on Innovative Sustainable Computational Technologies (CISCT) | 978-1-6654-7416-0/22/$31.00 ©2022 IEEE | DOI: 10.1109/CISCT55310.2022.10046509
Uma Rani V
Department of Computer Science and Engineering,
Saveetha Engineering College,
Chennai, Tamil Nadu, India
umaranibharathy@gmail.com
Kaladevi Ramar
Department of CSE,
Saveetha Engineering College, Chennai, India
Kalaramar26@gmail.com
Abstract—Any unwanted background sound is considered
noise. The perception of a sound at the same decibel level is
determined by whether it is considered pleasant music or irritating
noise. The removal of such noise is necessary to increase the
quality of the audio signal as it travels through the communication
system.A microphone is used in this study to collect real-time
audio signals. A real audio signal is collected and combined with
salt, pepper and white Gaussian noise for this experimental study.
To remove disturbances from an audio signal, this study integrates
linear and nonlinear spatial filtering mechanisms such as the
median low pass filter, finite impulse response low pass filter, and
wavelet transform approach. The suggested hybrid filter
technique is compared to individual filtering methods, and its
performance is measured using fundamental metrics such as mean
square error and signal-to-noise ratio. When compared to other
individual nonlinear filtering approaches, the hybrid method has
a high signal to noise ratio (SNR - 88.04%) and a lower mean
square error (MSE - 0.003).
Keywords—De-noising audio signal, Median filter , FIR
filter,wavelet transform
I. INTRODUCTION
Today, a wide range of industries use digital image
processing [1], including satellites, CT image recognition, face
recognition, pattern matching, dashboard image processing,
obstacle processing, machine or robot vision and acoustic signal
analysis, among others. The audio or noise signals were
gathered from multiple sources but were tainted by noise. Data
transmission mistakes, noise reduction techniques like
amplifiers and different quantization methods, or thermally
generated electrons at sensor sites are the main causes of this
noise [2]. To better the analysis of the audio signal, this
distorted noise is first eliminated.
Audio de-noising [3,4] is a technique for preserving original
voices after noise or distortions from noisy backgrounds have
been removed. Researchers [5,6] have developed a variety of
audio de-noising methods to eliminate noise from audio
sources, including spatial filtering, transform domain filtering,
and wavelet thresholding. In this scenario, spatial filtering is an
effective method for removing additive noise from audio
signals. The two types of spatial filtering approaches are linear
filters and nonlinear filters for image or audio or video
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Jebamalar Tamil Selvi J
Department of Computer Science,
SRM Institute of Science and Technology,
Ramapuram,Chennai, Tamil Nadu, India,
jebamalj@srmist.edu.in
Hariharan Shanmugasundaram
Department of Computer Science and Engineering,
Vardhaman College of Engineering, Hyderabad, India
mailtos.hariharan@gmail.com
denoising. But they have some advantages and disadvantages;
linear filter is suitable for salt and pepper noise removal.
nonlinear filter is suitable to multiplicative or function-based
noise removal. But both are failed to remove noise in sharp
edges and lines, linear and nonlinear filters [7] perform poorly.
To remove audio sounds in transform domain filtering, low
pass or high pass filters with a cut-off frequency and a
frequency domain filter are utilized. However, they impair the
originality of the audio signal. Wavelet Thresholding [8]
reduces picture noise by using threshold-based computations.
Thresholds can be adaptive or non-adaptive, but if the best
threshold is not chosen, the original image blurs. As a result, a
hybrid filter is required to reduce noise while maintaining signal
quality.
Several studies are being carried out to perform denoising
on images, audio, and videos. This work is divided into several
sections. Section II examines various approaches developed by
researchers to denoising audio signals. Section III discusses the
proposed hybrid methodology for removing noise from audio
streams. Section IV discusses the experimental results of
proposed methodology after using filters in each level and
compares its performance to that of individual filters.
II. LITERATURE REVIEW
Many studies on text or image denoising, audio and video
denoising from diverse sources have been undertaken. Several
denoising algorithms and noise reduction approaches have been
created to eliminate various types of noise from the original
image, audio, or video, however they fail to retain their
originality. The noise linked with the texture and edge of a
picture, as well as the same frequency of audio and video, is
difficult to remove using conventional methods. This section
summarises some of the research that has been done by various
research groups.
Gaetan Frusque et al [9] created a wavelet packet transform
approach to improve audio signal resolution. They use the
DCASE08 data collection for experimental purposes. Julius et
al. [10] developed a new noise estimator based on the wavelet
mechanism and combined it with a log spectral subtraction to
moderate the noise level in windmill sound.
 Brusa et al [11] have utilized a wavelet transform and
thresholding technique to detect local defeats in rolling bearing
audio signals. They employ principal component analysis to
extract features in the temporal domain. For channel estimation,
Palanisamy et al [12] employed an integrated discrete fourier
transform, least square Wiener, and minimal mean square error
technique. When compared to the traditional discrete fourier
transform technique, it has a lower symbol error rate, a lower
bit error rate, and a higher SNR.
Ihza et al [13] created a method for de-noising heart sound
audio that is integrated. Three algorithms are used: discrete
wavelet transform, low pass filter, and short term fourier
transform. To decrease noise in the decomposition process,
Huijie et al [14] employed a threshold shrinkage technique. Fig.1 Flow of audio noise removal using HMFWT Technique

Y.Cho et al. [15] have developed a denoising-based driver

assistance system for off-road teleoperation. In their study, HMFWT has following stages
Jaroslavceva et al [16] employed the labanotation template 1. Audio signal collection and noisy signal generation
subtraction method to reduce noise for human-robot 2. Median Low Pass filtering
communication. J Hao et al [17] used an FFT to extract 3. FIR Low Pass filtering
frequency and a neural network to decrease signal defects. 4. DWT transformation
Panah et al [18] have created an audio processing pipeline to
improve cardiac sound quality. For signal analysis, Leng et al. A. Audio signal Collection and noisy signal generation
[19] have used a conditional diffusion model. They emphasise
the importance of wavelet transformation in signal denoising. The first step in denoising is to gather real-time audio
Bai et al. [20] have employed a multichannel Wiener filter to signals. To generate the noise signal, add AWGN after
remove noise. gathering the audio signal.

The main objective of proposed methodology is B. Median low Pass Filtering

a) Developed a hybrid model which combines the
advantages of three filtering methods, Median filter, FIR filter,
and wavelet transform approach to eliminate noise from audio
signals.
b) It remove noise from distorted audio signals without
reducing original quality.
c) Compare the performance of hybrid model at each
stages and provide basic quality metric analysis.
MLPF is a type of nonlinear filter that successfully removes
speckle and salt pepper noise from a signal. It also retains all
signal regions by employing a dynamic windowing mechanism.
C. FIR low pass Filtering
FIRLPF is an intrinsically stable linear phase symmetric
filter. It does not require feedback or a reaction to any input
signal of finite length. Fig.2 depicts the basic block diagram of
FIRLPF.

III. PROPOSED METHODOLOGY

The audio signal is evaluated using noise theory and audio
theory basics. The audio signal is concatenated with AWGN to
form a noisy signal. The Fast Fourier Transform technique is
used to convert this noisy signal from time domain to frequency
domain (FFT). The proposed filter HMFWT would require
creating linear and nonlinear spatial filters such as the Median
Low Pass Filter (MLPF), finite impulse response Low pass
(FIR-LPF) filter, and Discrete Wavelet Transform (DWT)
Fig.2 Block diagram of FIR filter
while taking various factors of filter design into account such
as system order and cutoff frequency. The HMFWT filtering
FIR low pass filter produce output signal Y[n] from input
method includes signal normalization, a decomposition
signal X[n] by convolving input signal x and its impulse
mechanism, and the reconstruction process itself. During the
response  . This is defined by equation (1).
decomposition stage, the audio signal is rebuilt and inverse FFT
is employed to transform the frequency domain to the time [] =  0  × [ − ] (1)

domain. Fig. 1 depicts the flow of audio noise removal using Here  is a unit delay operator in Z- transform notation.
HMFWT technique.Table 1 describe the notations and TABLE 1 NOTATIONS USED
abbreviation used in proposed methodology.

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 Abbreviation and Meaning Step 2:Using equation (2) and (3) ,Calculate SNR and
Notations RMSE of noisy signal.
FFT Fast Fourier Transform 
 =  log0   | ( −  )  (2)
AWGN Additive White Gaussian Noise
DWT Discrete Wavelet Transform 
1(  )
2

SNR Signal to Noise Ratio RMSE = =  (3)


RMSE Root Mean Square Error
IFFT Inverse Fast Fourier Transfer
MF Median Filter Step 3:Using the wavelet transform method, establish a
X[n] ,Y[n] Input, Output signal threshold value for the decomposition stage.The threshold
  Unit delay point Tnewis derived from number of samples N and Noise
intensity  . This is described by equation (4)
N Number of samples
 New Threshold  =  log  (4)
WT Wavelet Transform
AFFT Average Fast Fourier Transform
PR Phase Response Step 4: Decompose received signal into wavelet
MLPF Median Low Pass Filter coefficients using filter bank.
SP Salt and Pepper Step 5: Apply a threshold value to the wavelet coefficients
FIRLPF Finite Impulse Response Low Pass that have been decomposed. It employs the Soft threshold,
Filter which is defined by an equation (5).
HMFWT Hybrid Median,FIR filter Wavelet
Transform r = sign(s)(|s| -Tnew) (5)

D. DWT transformation Step 6: Rebuild the signal with threshold wavelet

DWT transformation [19,20,21] is a useful technique for coefficients and a filter.
minimizing noise in digital communications. The DWT
algorithm uses signal threshold value based on number of
samples and density. It decomposes the incoming signal into
wavelet coefficients using the associated analytical filter banks. 2. RMSE: The Root Mean Square Error is a statistic that is
It processes each sub-band separately in a loop. Across all used to calculate the level of error in statistical models. It
scales and sub-bands, apply the global threshold values to the calculates the average squared difference between observed
decomposed wavelet coefficients. It uses a soft threshold for and predicted visual output.
wavelet denoising. Finally, using threshold wavelet coefficients
and a filter bank, the audio signal is recreated. This process is 
illustrated in DWT algorithm (presented in Table 2).  = √ × ∑ () (7)



Initially, an audio signal is captured by an audio recorder with

IV. RESULTS AND PERFORMANCE ANALYSIS
Matlab is used to run the denoising audio signal tests, and the
filter's performance [21] is measured using SNR and RMSE, as
indicated by equations (6) and (7), respectively.
1. SNR : The Signal to Noise Ratio (SNR) is the ratio of
required output signal strength to distorting noise power
that decreases input signal quality..
   
 =  log dB (6)
    
Fs = 8000, which records real-time audio for up to 5 seconds
and saves the resulting audio file (minimum 78.1kb file size) in
.wav format. Fig. 3 depicts the frequency of the input audio
signal, while Fig. 4 depicts the noise analysis of the input audio
signal. The Averaged FFT Spectrum of the input audio signal is
as shown in Fig.5. After adding noise (SP and white Gaussian
noise), the MLPF, FIRLPF and DWT transform are applied to
noisy input signal. Fig. 6 depicts the output of the MLPF.

TABLE 2 DWT ALGORITHM FOR AUDIO DENOISING

Algorithm 1: DWT algorithm for audio denoising

Input : Noisy audio signal
Output : Denoising audio signal
Step 1: Read an input signal s(i) and noisy signal r(i).

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 Fig.6 Output of MLPF
Fig. 3 Audio signal after AWGN
The output of FIR LPF is as shown in Fig.7.

Fig.4 Noise in Input Signal

Fig.7 Output of FIR LPF
The output of DWT low pass filter is as shown in Fig.8.

Fig.8 Output of DWT

Fig.5 Input Average FFT Spectrum
The magnitude and phase response of HMFWT as shown in
Fig.9.

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 TABLE 3 PERFORMANCE ANALYSIS OF FILTERS
S.No Metrics MLPF FIRLPF DWFT HMFWT
filter
1 SNR 51.08 67.01 77.2 88.08
2 RMSE 6.02 3.98 0.83 0.003

The output Signal of the final audio signal is shown in Fig.10.

The AFFT spectrum signal of the output audio signal is as
shown in Fig.11.The Spectrum analysis of the output audio
Fig.12.

Fig.11 AFFT Spectrum of output signal

Fig.9 Magnitude(dB), PR (degree) of HMFWT

Fig.12 Output spectrum

V. CONCLUSION

Denoising audio signal is one of the major problems occur in

Fig.10 Recovered audio signal using HMFWT
real-time signal processing. Throughout the real-time audio
acquisition method, the genuine signal is tainted by random
noise. Because of the random noise, the transmission quality is
altered, and the original audio stream is diminished. This study
emphasized the significance of denoising techniques in audio
signals. The HMFWT filter described here incorporates linear
and nonlinear filters, as well as the wavelet transform
technique. HMFWT beats individual filtering algorithms in the
presence of strong background noise, with a high SNR ratio
(88.08) and a reduced RMSE score (0.003). In the future, the
current work will be enhanced using a machine learning
technique. The audio signal after stereo mixing or the medical
domain is explored.

Table 3 displays the performance of each filter. According to

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