COMBINED FEATURE EXTRACTION TECHNIQUES AND NAIVE BAYES CLASSIFIER FOR SPEECH RECOGNITION

Sundarapandian et al. (Eds) : ACITY, AIAA, CNSA, DPPR, NeCoM, WeST, DMS, P2PTM, VLSI - 2013
pp. 155–163, 2013. © CS & IT-CSCP 2013 DOI : 10.5121/csit.2013.3416
COMBINED FEATURE EXTRACTION
TECHNIQUES AND NAIVE BAYES
CLASSIFIER FOR SPEECH RECOGNITION
Sonia Sunny1
, David Peter S2
, K. Poulose Jacob1
1
Dept. of Computer Science,
Cochin University of Science & Technology, Kochi, India
2
School of Engineering,
Cochin University of Science & Technology, Kochi, India
sonia.deepak@yahoo.co.in, davidpeter@cusat.ac.in,
kpj@cusat.ac.in
ABSTRACT
Speech processing and consequent recognition are important areas of Digital Signal Processing
since speech allows people to communicate more natu-rally and efficiently. In this work, a
speech recognition system is developed for re-cognizing digits in Malayalam. For recognizing
speech, features are to be ex-tracted from speech and hence feature extraction method plays an
important role in speech recognition. Here, front end processing for extracting the features is
per-formed using two wavelet based methods namely Discrete Wavelet Transforms (DWT) and
Wavelet Packet Decomposition (WPD). Naive Bayes classifier is used for classification purpose.
After classification using Naive Bayes classifier, DWT produced a recognition accuracy of
83.5% and WPD produced an accuracy of 80.7%. This paper is intended to devise a new
feature extraction method which produces improvements in the recognition accuracy. So, a new
method called Dis-crete Wavelet Packet Decomposition (DWPD) is introduced which utilizes
the hy-brid features of both DWT and WPD. The performance of this new approach is evaluated
and it produced an improved recognition accuracy of 86.2% along with Naive Bayes classifier.
KEYWORDS
Speech Recognition; Soft Thresholding; Discrete Wavelet Transforms; Wavelet Packet
Decomposition; Naive Bayes Classifier.
1. INTRODUCTION
Spoken digits recognition has great importance in the field of speech recognition [1]. In our day-
to-day life, we encounter many applications that require recognition of spoken digits that uses
numbers as input. Some of the applications include automated banking system, airline
reservations, voice dialing telephone, automatic data entry, command and control etc [2]. This
work uses digits from Malayalam which is one of the four major Dravidian languages of southern
India and the official language of the people of Kerala. Speech recognition is one of the core
disciplines of pattern recognition which also includes a number of technologies and research

156 Computer Science & Information Technology (CS & IT)
areas like Signal Processing, Natural Language Processing, Statistics etc [3]. Speech signals are
non stationary in nature. There are many factors that make recognition of speech a complex task.
This is due to the fact that when people speak, there is difference in gender, emotional state,
accent, pronunciation, articulation, nasality, pitch, volume and speed variability [4]. Background
noise, additive noise and different types of disturbances may also affect the performance of a
speech recognition system.
Usually the performance of a speech recognition system is measured in terms of recognition
accuracy. There has been a lot of research in the area of speech recognition in different languages
like English, Chinese, Arabic, Bengali, Tamil, Hindi etc. But only few works have been reported
in Malayalam. When compared to speaker dependent speech recognition system which involves a
single speaker, developing an efficient speaker independent system is a difficult task since it
involves the recognition of the speech patterns from a group of people. As speech recognition
involves different stages like pre-processing, feature extraction and pattern classification, the
performance of the overall speech recognition system depends on the techniques selected for
these stages. Speech signals are usually affected by different additive as well as background
noise. So, pre-processing has to be done to remove the noise from the signals. Here, pre-
processing is done using wavelet denoising method based on soft thresholding [5]. Feature
extraction is done using two wavelet based feature extraction methods namely DWT and WPD
due to the good time and frequency resolution properties of the wavelets. The performance of
both these methods in classifying the digits is evaluated using Naive Bayes classifier and a new
hybrid algorithm is developed by combining the features obtained from both the methods.
The paper is organized as follows. Section 2 gives a brief description of the problem definition
and the methodology used. The digits database is explained in section 3. The method used for
preprocessing is illustrated in section 4. Section 5 elaborates the various feature extraction
techniques used in this work followed by the classification technique in section 6. A detailed
evaluation of the experiments and the results obtained are explained in section 7. Conclusion is
given in the last section.
2. STATEMENT OF THE PROBLEM AND ARCHITECTURE OF THE
SYSTEM
ASR enables people to communicate more naturally and effectively without the use of an
interface in between. In this work, a speech recognition system is designed for recognizing
speaker independent spoken digits in Malayalam, since research in Malayalam is in its infancy
stage. The speech signals captured are to be converted to a set of parameters. Feature extraction is
the front end processing and most important part of speech recognition since it plays an important
role in separating one speech from other. So developing new techniques for feature extraction has
been an important area of research for many years. Most of the speech-based studies are based on
spectral analysis of speech signals using Mel-Frequency Cepstral Coefficients (MFCCs), Linear
predictive Coding (LPCs) etc . Results from most of the works reveal that the dimensions and
computational complexity of the feature vectors obtained are higher in these methods. In wavelet
transforms, the size of the feature vector is less compared to other methods. This reduces the
computational complexity. Now, more and more studies are being done on wavelets.
The main objective of this work is to design a more efficient speech recognition system which
performs better than the existing methods. Since wavelets are found to be good in speech

Computer Science & Information Technology (CS & IT) 157
recognition, here two wavelet based feature extraction methods namely DWT and WPD are used
and the performance of both these in recognizing the digits database created are compared and
analyzed. Naive Bayes classifiers are used in order to obtain the classification performance of the
features. Both the methods are found to be good in recognizing speech. But since feature
extraction is the front end processing part of a speech recognition system, designing of a new
algorithm with improved feature vectors is a challenging research problem. So a new feature
extraction method is developed by utilizing the features obtained from both WPD and DWT to
improve the recognition rate.
In this work, first the digits database is created since there is no standard database available in
Malayalam. The captured speech signals are then pre-processed using wavelet denoising
techniques to tune the signals for extracting features by removing the noise from it. After
denoising, the signals are presented to feature extraction techniques namely DWT, WPD and the
proposed DWPD. The extracted features are then given for pattern classification using Naive
Bayes classifier. Finally these techniques are compared and the performance of these techniques
is evaluated based on recognition accuracy.
3. DIGITS DATABASE
A new database is created for spoken digits in Malayalam language using 200 speakers of age
between 6 and 70 uttering 10 Malayalam digits. 200 speakers including 80 male speakers, 80
female speakers and 40 children were entrusted with the task of recording the speech samples.
Male and female speech differ in pitch, frequency, phonetics and many other factors due to the
difference in physiological as well as psychological factors. A high quality studio-recording
microphone at a sampling rate of 8 KHz (4 KHz band limited) is used for recording the speech
samples. Ten Malayalam digits from 0 to 9 are used to create the database under the same
configuration. Thus the database consists of a total of 2000 utterances of the digits. The spoken
digits are numbered and stored in the appropriate classes in the database. The spoken digits and
their International Phonetic Alphabet (IPA) format are shown in Table 1.
Table 1. Numbers Stored in the Database and their IPA Format
Number digit Words in Malayalam IPA format
English Translation
0 /pu:dƷıəṁ/ Zero
1 /onnə/ One
2 /r˄ndə/ Two
3 /mu:nnə/ Three
4 /na:lə/ Four
5 /˄ndƷe/ Five
6 /aːrə/ Six
7 /eiƷə/ Seven
8 /ettə/ Eight
9 /onp˄θə/ Nine

4. PRE-PROCESSING USING WAVELET DENOISING
Speech signals are often degraded by the presence of additive or convolution noise present in the
background. So, in order to remove the background noise, these signals are to be denoised so that
the noise present in it are suppressed during pre-processing stage before extracting the features.
Different techniques are available for speech enhancement. In this work, we have used wavelet
denoising algorithms for reducing the noise present in the signal. Wavelet denoising uses
thresholding functions which are used to limit the values of the elements. The most popular
thresholding functions that are widely used in wavelet denoising method are the hard and the soft
thresholding functions [6]. Hard thresholding sets to zero any element whose absolute value is
lower than the threshold. In soft thresholding, the absolute values of the elements that are lower
than the threshold are first set to zero. Then it shrinks the nonzero coefficients towards 0. Hard
and soft thresholding can be defined as
Where X denotes the wavelet coefficients and ι represents the threshold value. Here we have used
soft thresholding technique. Threshold value can be obtained using different methods. The
universal threshold derived by Donoho and Johnstone [7] for the white Gaussian noise under a
mean square error criterion is used in this work which is represented as
( )2log Nτ σ= (3)
where σ denotes the standard deviation and N denotes the length of the signal.
The algorithm used for denoising mainly consists of 3 steps.
• Apply wavelet transform up to 8 levels to the noisy signal to produce the noisy wavelet
coefficients.
• Select an appropriate threshold limit. Apply soft thresholding to the detail wavelet coefficients.
• Apply inverse discrete wavelet transform to the wavelet coefficients that are obtained from the
previous step. This produces the denoised signal.
5. FEATURE EXTRACTION TECHNIQUES USED
Every speech signal has different individual characteristics embedded in it and these characteristics can be
extracted using a wide range of feature extraction techniques. A brief description of the feature extraction
techniques used in this work namely DWT, WPD and the new proposed hybrid algorithm DWPD is
explained below.
5.1 Discrete Wavelet Transforms
DWT is a more recent, popular and computationally efficient technique which is used in different
areas like signal processing, image processing etc. One of the main characteristics of the wavelet
| |
0 | | (1){X if X
Hard if XX τ
τ
>
≤=
( ) (| | | |) | |
0 | | (2){
sign X X if X
Soft if XX
τ τ
τ
− >
≤=

transforms is that it uses windows of different sizes. It adopts broad windows at low frequencies
and narrow windows at high frequencies, thus leading to an optimal time–frequency resolution in
all frequency ranges [8]. DWT and WPD use digital filtering techniques to obtain a time-scale
representation of the signals. DWT is defined by the following equation.
Where Ψ (t) is the basic analyzing function called the mother wavelet. In DWT, the one
dimensional speech signal passes through two discrete-time low and high pass quadrature mirror
filters which produces the corresponding wavelet coefficients. It produces two signals, called
approximation coefficients and detail coefficients [9]. In speech signals, low frequency
components h[n] are of greater importance than high frequency components g[n] as the low
frequency components characterize a signal more than its high frequency components [10]. The
successive high pass and low pass filtering of the signal is given by
Where Yhigh (detail coefficients) and Ylow (approximation coefficients) are the outputs of the
filters obtained by sub sampling by 2. According to Mallat algorithm, the filtering process is
continued until the desired level is reached [11].
5.2 Wavelet Packet Decomposition
DWT performs a one sided dyadic tree decomposition of signals. But, WPD allows any dyadic
tree structure analysis where the approximation as well as detail coefficients are decomposed
iteratively up to a certain level chosen [12]. Thus WPD is a more flexible and detailed method
than DWT and it also produces good time and frequency resolutions. In WPD, more detailed
time-scale analysis can be achieved. The main characteristic of WPD is that it uses long-time
interval for low-frequency information and short-time interval for high-frequency information
[I3].
5.3 Proposed Hybrid algorithm
Wavelets are a powerful and extremely useful method for speech recognition. A wavelet
transform decomposes a signal into sub-bands with low frequency components which contain the
characteristics of a signal and high frequency components which are related with noise and
disturbance in a signal [14]. The features of the signals can be retained if the high frequency
contents are removed. This reduces the noise in the signal [15]. But sometimes the high frequency
components may contain useful features of the signal. The main drawback of DWT is that it
cannot decompose the high frequency band into more partitions. Although WPD can achieve this
decomposition, it is also applied to low frequency band signals, which mainly includes the
desired signals. So this causes unnecessary computational complexity. To overcome the
limitations of DWT and WPD, we propose a new algorithm for speech enhancement by

combining the features of both DWT and WPD. The outline of the proposed DWPD algorithm is
given below.
• The speech signal is decomposed into a low frequency band signal and a high frequency band
signal. That is one level of decomposition is performed.
• 7 scales of DWT are then applied on the low frequency components and 7 scales of WPD are
applied on the high frequency components.
• The features obtained from both decomposition are combined together to form the new feature
vector set.
The wavelet decomposition trees of DWT, WPD and DWPD up to 3 levels are shown in figure 1.
Fig. 1. DWT decomposition WPD decomposition DWPD decomposition
The proposed new hybrid algorithm DWPD has the following advantages.
• Decomposes high frequency band into more partitions.
• Saves computational complexities.
• Improves recognition rate.
6. CLASSIFICATION USING NAIVE BAYES CLASSIFIER
Since speech recognition is a multiclass classification problem, Naive Bayes classifiers are used
because it can handle multiclass classification problems. Naive Bayes classifier is based on the
Bayesian theory which is a simple and efficient probability classification method based on
supervised classification technique. For each class value it estimates that a given instance belongs
to that class [16]. The feature items in one class are assumed to be independent of other attribute
values called class conditional independence [17]. Naive Bayes classifier needs only small
amount of training set to estimate the parameters for classification. The classifier is stated as
P(A|B) = P (B|A) * P (A)/P(B) (7)
Where P(A) is the prior probability or marginal probability of A, P(A|B) is the conditional
probability of A, given B called the posterior probability, P(B|A) is the conditional probability of
B given A and P(B) is the prior or marginal probability of B which acts as a normalizing constant.
This provides the mathematical representation of the way in which the conditional probability of
event A given B can be related to the conditional probability of B given A. The probability value
of the winning class dominates over that of the others [18].

7. EXPERIMENTS AND PERFORMANCE EVALUATION
In this work, we have used wavelets for feature extraction. There are different wavelets available
for signal analysis. So the first step is to select the suitable wavelet family and hence the wavelet.
In this work, the Daubechies wavelets are used which are found to be efficient in signal
processing applications. Among the Daubechies family of wavelets, the db4 type of mother
wavelet is used for feature extraction since it gives better results [19]. The speech samples after
pre-processing are successively decomposed into approximation and detailed coefficients.
Another consideration is the level up to which decomposition is to be performed. For DWT, less
frequency components from level 8 is used to create the feature vectors. In WPD, both the high
frequency and low frequency components are decomposed up to level 8. The number of features
obtained using DWT is 16 and using WPD is 20. In the proposed DWPD, the features from both
DWT and WPD are combined. Classification task using Naive Bayes classifier involves
separating data into training and test sets. In all the three methods, we have divided the database
into three. 70% of the data for training, 15% for validation and 15% for testing.
In this work, better results are obtained at level 8 during decomposition. The original signal and
the 8th
level decomposition coefficients of spoken word Poojyam (Zero) using DWT is given in
figure 2.
Fig. 2. Decomposition of digit poojyam at 8th
level using DWT
The original signal and the 8th level decomposition coefficients of spoken digit Poojyam (zero)
using WPD is given in figure 3.
Fig. 3. Decomposition of digit poojyam at 8th
level using WPD
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 10 20 30 40 50 60
-1.5
-1
-0.5
0
0.5
1
Word -zero Approximation coefficients Level 8
0 10 20 30 40 50 60
-1.5
-1
-0.5
0
0.5
1
1.5
Detail coefficients Level 8
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
1
Original Signal -zero
0 2 4 6 8 10 12 14
-0.2
-0.1
0
0.1
0.2
0.3
coefficient values using wpd -zero

All these methods produced good recognition accuracies after classification. The overall
recognition accuracies obtained using these methods are shown in table 2.
Table 2. Comparison of results
8. CONCLUSION
In this work, we have exploited the powerful features of wavelets in developing a novel speech
recognition system for speaker independent digits recognition in Malayalam. Here, two major
wavelet based feature extraction techniques such as DWT and WPD are used for feature
extraction and the performance of these methods are evaluated. Since speech recognition is a
pattern recognition problem, Naive Bayes classifiers are used for classification. Both the methods
produced an accuracy of 83.5% and 80.7% respectively. The ultimate aim of a speech recognition
system is to attain maximum recognition rate. Though both the methods produced good results, a
more enhanced method called DWPD is developed by combining the features of DWT and WPD.
An accuracy of 86.2% is obtained using this hybrid structure. Soft thesholding technique is
applied to the signal for reducing the noise before feature extraction. The experimental results
show that this hybrid architecture using DWT and WPD along with Naive Bayes classifier could
effectively extract the features from the speech signal. As an extension of this study, the
performance of this proposed method can be analyzed using other classifiers like Support Vector
Machines, Artificial Neural Networks etc.
REFERENCES
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[6] Yasser Ghanbari, Mohammad Reza Karami (2006) A new Approach for Speech Enhancement based
on the Adaptive Thresholding of the Wavelet Packets. Speech Communication 48:927–940
[7] D.L. Donoho (1995) De-noising by Soft Thresholding. IEEE transactions on Information Theory
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[8] Elif Derya Ubeyil (2009) Combined Neural Network model Employing Wavelet Coefficients for
ECG Signals Classification. Digital Signal Processing 19:297-308
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Wavelets and Artificial Neural Networks. Proc. of Int. Symposium on Intelligent Multimedia, Video
and Speech Processing 413-416
[10] S. Kadambe, P. Srinivasan (1994) Application of Adaptive Wavelets for Speech. Optical Engineering
33:2204-2211
Feature Extraction Method Recognition Accuracy (%)
DWT 83.50
WPD 80.70
DWPD 86.20

[11] S .G. Mallat (1989) A Theory for Multiresolution Signal Decomposition: The Wavelet Re-
presentation. IEEE Transactions on Pattern Analysis and Machine Intelligence 11:674-693
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Packet Decomposition for Brain Computer Interface. Measurement 41:618-625
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Beijing, China
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[19] Sonia Sunny, David Peter S, K Poulose Jacob (2013) Performance Analysis of Different Wavelet
Families in Recognizing Speech. International Journal of Engineering Trends and Technology 4:512-
517

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