Denoising of real-time audio signal using matlab filter techniques
Denoising of real-time audio signal using matlab filter techniques
techniques
Jayashree K C Karthik M Bharamappa Manoj Kumar P
School of ECE School of ECE School of ECE
REVA UNIVERSITY REVA UNIVERSITY REVA UNIVERSITY
Bangalore , India Bangalore , India Bangalore , India
[email protected] [email protected] [email protected]
Abstract— In this paper we analyze real time audio signals and creates major problem to the audio signals. Electrostatic
try to reduce the noise associated with the message signal under noise which is generated due to the presence of the voltages
consideration. The main drawback of noise being present in an during the design implementation and other random noises
audio signal, is that it reduces the quality of the signal that is which gets added to the signal. This proposed research will
being transmitted within the communication system. For
solve the drawbacks of various filtering techniques which
analysis purpose, white gaussian noise(awgn) is concatenated
with the audio signal under consideration and the resulting
also provides unique Knowledge to the reader. The project
noisy audio signal is subjected to the different filtering defines a comparison between three different filtering
techniques like IIR Filter, FIR Filter, Wavelet transform techniques i.e. IIR filtering, FIR filtering and wavelets
techniques. The noisy audio signal is analyzed with respect to transforms based on real time audio signals.
the different filter responses obtained on applying the foresaid
methods. A comparative study is done between these
techniques to arrive at a technique which would be the most II. BACKGROUND AND RELATED WORK
efficient one for audio signal denoising. Graps[1] came up with new analysis named Fourier
Keywords-Audio processing, Denoising, FIR filters, IIR
transform which could analyze the periodic function by
filters, wavelets, Daubechies. creating mathematical structures that vary in scale. But the
proper analysis cannot be analyzed using frequency
I. INTRODUCTION response.
Radhika Bhagat[2] has made an attempt of audio filtering
The importance of noise reduction in real-time audio signals
using extended filters like lowpass and highpass filter using
is said to be having high significance in communication.
FIR and IIR filtering techniques. They have designed
Due to this factor the noise weakens signal quality, and the
different formulas and difference equations for efficient
recognition of audio signals becomes difficult and cause
implementation of time varying filter applications.
serious difficulties for the users of electronic hearing aids. A
Er. Harpal Singh[3] has used fast Fourier transform
well-established method is filtering of the signal in the
technique for performing time domain and frequency
frequency domain or in the simplest way is analyzing the
domain analysis of the signal. Mannu Singla[4] has used
signal using different filter techniques like low pass, high
Butterworth filter and Chebyshev filters to reduce noise
pass and bandpass filters. As the voice or speech signals are
from signals with different frequency and ripple factors.
not periodic these filters distort the signal more than they
Seema rani[5] has proved more facts about FIR and IIR
reduce the noise. To attenuate noise, we need more
filters in their paper. The paper tells FIR is more stable than
advanced methods of filtering. As it requires high advanced
IIR. From the above research we can conclude that the error
methods of filtering, we go for the different filtering
of FIR filter is less compared to IIR filters that means the
techniques like Fourier transforms, wavelet transforms, and
output of FIR filter is very close to the desired value and
other methods are used in denoising the audio signals. A
FIR filter is more stable than IIR filters.
Fourier transform of a signal gives us the frequency
C Mohan Rao[6] has presented a new algorithm that is the
composition of the audio signal. The disadvantage of
Least Mean Square (LMS) in which the awgn is added to
Fourier transform is it is only valid within a certain Region
message signal and the denoising is done to reduce the noise
of Convergence (ROC). So, we go for short time Fourier
with minimum or no error efficiently. But it is sensitive to
transform (STFT), but this method used the window analysis
the scaling of its input.
approach of defined size. IIR and FIR algorithms uses the
B. Jai Shankar[7] has proposed the use wavelet
Fast Fourier transform (FFT) technique for analysis of
transformation technique to denoise audio signals by
frequency spectra and signal responses. Wavelet analysis
dividing the signal into blocks. This technique protects each
provide more detailed analysis about the signal compared to
and every unique and vital features of every individual block
other filtering approaches.
and exposes the finest detail contributed by the grouped set
This paper tells the readers about denoising of the audio
of blocks. The authors Ola Ratelli, Palle, Jorgensen (2013)
signal. As all the audio signals are continuously affected by
[8] in their book have proposed the about Discrete wavelet
the different types of real-time noise such as electrostatic
transform, its benefits and its functionalities. Priyanka
noise, thermal noise, channel noise, awng and etc. which
Additive
White
Gaussian
noise
Figure 3. FIR Filtering method block diagram
N−1
Scaling function ∅ (t )= ∑ h [n]√ 2 ∅ (2 t−n) (4)
n=0
N−1
Wavelet function φ ( t )= ∑ g[n] √ 2∅ (2 t−n) (5)
n=0
Approximation coefficients:
1
W ∅ [ j0 , k ]= ∑ f [ n ] ∅ j , k [n] (6)
√M n 0
Figure.5 – Audio signal.
1
W φ [ j , k ]= ∑ f [ n ] φ j , k [n]
√M n
j≥j0 (7)
T
Noisy Approx. H
Input Signal Low Coeffici R
Audio Pass ents E
A0, A1, S
L-Filter …An H
Additive
White O
Gaussian High Detailed L
Noise Pass coefficie D
nts
D0, D1… S
Dn /
H
Figure.7b pole/zero plot of IIR filter
Figure.6 Noisy audio signal
The output of IIR lowpass filter is as shown in figure 8.
The noisy signal is passed through IIR low pass filter having
magnitude & phase response is as shown in figure.7a.1 &
figure.7a.2 respectively and pole & zeroes plot is as shown
in figure.7b.
Figure.7a.1 Magnitude and phase response of IIR filter Figure8.IIR low pass filter output (denoised output)
V. DISCUSSION
An analog system is said to be stable if all its poles lie in the
left half of the ‘s’ plane, but IIR filter has a both poles and
zeros as we can see in figure 7b. So IIR is said to be
unstable. IIR filters are difficult to implement as it has
delays and distortions due to large number of poles. IIR
filters are better in lower orders as IIR filters may become
unstable with higher orders. As IIR filters are unstable they
cannot have a linear phase as shown in the phase plot in
figure.7a. Theoretically we know that IIR filters depends on
present input and previous output. By the waveforms in
figure.8, we observe that noise introduced is denoised to
Figure.9b pole/zero plot of FIR filter certain limit but the output audio signal will also get
damped.
The output of FIR lowpass filter is as shown in figure.10 Due to these drawbacks we would prefer FIR over IIR as
FIR filters do not have delays and distortions. Therefore,
FIR filters are stable than IIR as FIR filters have only zeros
on the unit circle in the s plane and one pole at origin as
shown in figure.9b. FIR filters are better for higher orders
and they maintain stability. FIR filters have linear phase
characteristics as shown in phase plot of figure.9a and it’s
output depends only on the present inputs. In order to avoid
damping of the output audio signal we have tried various
FIR filters using different windowing techniques like
hamming, Kaiser and rectangular. These windows were
compared to get the Filtered audio output but Hamming
window was preferred as it had linear phase. FIR filters had
denoised to limit much greater than IIR but the audio signal
was also damped.
Due to the drawbacks we would prefer wavelets over FIR to
remove noise and get the filtered signal. The wavelets used
here is Daubechies wavelets have highest number N of
Figure.10 Output of FIR lowpass filter vanishing moments with the support width 2N-1. db wave
solves the problem of signal discontinuities and is applicable
When the same noisy signal is denoised using wavelets for continuous and discrete wavelet transforms.
transform technique, the output is as plotted in figure.11
Process: n Efficient Audio Denoising Technique” , European Journal
of
Scientific Research, Vol.48 No.1, pp.16-28. ISSN 1450-216X.
[8] Ola Ratelli , Palle Jorgensen , ” Wavelets through a looking glass :
The world of the spectrum ” , Springer science & Business media ,
Illustrator : B. Treadway , 2013.
[9] Priya Khattar1, Dr. Amrita Rai2, Mr. Subodh Tripathi, “Audio De-
noising using Wavelet Transform”, 2016
[10] Adri E. Villanueva- Luna, Alberto Jaramillo-Nunez, Daniel Sanchez-
Lucero, Carlos M. Ortiz-Lima, “De-Noising Audio Signals Using
MATLAB Wavelets Toolbox”, Mexico.
REFERENCES
[1] A. GRAPS, “AN INTRODUCTION TO WAVELETS” ,22724 MAJESTIC OAK
WAY, CUPERTINO, CA, USA,1995.
[2] Radhika Bhagat, Ramandeep Kaur, “Improved Audio Filtering Using
Extended High Pass Filters”,2013
[3] Er. Mannu Singla, Er. Harpal Singh, “Review Paper on Frequency
Based Audio Noise Reduction Using Different Filters”, International
Journal of Science, 2015.
[4]Er Mannu Singla, Er Harpal Singh, “Paper on frequency based audio
noise reduction using Butterworth, Chebyshev and Elliptical filters”,
International Journal of Science, 2015.
[5] Seema rani, Amanpreet Kaur, J S Ubhi, “Comparative study of FIR and
IIR filters for the removal of Baseline noises from ECG signal”,
International Journal of Computer Science and Information Technologies,
Vol. 2 (3), 2011, 1105-1108.
[6] C Mohan Rao, Dr. B Stephen Charles, Dr. M N Giri Prasad, “A
Variation of LMS Algorithm for Noise Cancellation”, International Journal
of Advanced Research in Computer and Communication Engineering Vol.
2, Issue 7, July 2013.
[7] Shanar B. J & Duraiswamy K. (2010). “Wavelet-Based Block
Matching