Performance of different classifiers in speech recognition

IJRET: International Journal of Research in Engineering and Technology ISSN: 2319-1163
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Volume: 02 Issue: 04 | Apr-2013, Available @ http://www.ijret.org 590
PERFORMANCE OF DIFFERENT CLASSIFIERS IN SPEECH
RECOGNITION
Sonia Suuny1
, David Peter S2
, K. Poulose Jacob3
1
Associate Professor, 2, 3
Professor,
1
Dept. of Computer Science, Prajyoti Niketan College, Kerala, India,
2, 3
Professor., Dept. of Computer Science, Cochin University of Science & Technology, Kerala, India,
sonia.deepak@yahoo.co.in, davidpeter@cusat.ac.in, kpj@cusat.ac.in
Abstract
Speech is the most natural means of communication among human beings and speech processing and recognition are intensive areas
of research for the last five decades. Since speech recognition is a pattern recognition problem, classification is an important part of
any speech recognition system. In this work, a speech recognition system is developed for recognizing speaker independent spoken
digits in Malayalam. Voice signals are sampled directly from the microphone. The proposed method is implemented for 1000 speakers
uttering 10 digits each. Since the speech signals are affected by background noise, the signals are tuned by removing the noise from it
using wavelet denoising method based on Soft Thresholding. Here, the features from the signals are extracted using Discrete Wavelet
Transforms (DWT) because they are well suitable for processing non-stationary signals like speech. This is due to their multi-
resolutional, multi-scale analysis characteristics. Speech recognition is a multiclass classification problem. So, the feature vector set
obtained are classified using three classifiers namely, Artificial Neural Networks (ANN), Support Vector Machines (SVM) and Naive
Bayes classifiers which are capable of handling multiclasses. During classification stage, the input feature vector data is trained using
information relating to known patterns and then they are tested using the test data set. The performances of all these classifiers are
evaluated based on recognition accuracy. All the three methods produced good recognition accuracy. DWT and ANN produced a
recognition accuracy of 89%, SVM and DWT combination produced an accuracy of 86.6% and Naive Bayes and DWT combination
produced an accuracy of 83.5%. ANN is found to be better among the three methods.
Index Terms: Speech Recognition, Soft Thresholding, Discrete Wavelet Transforms, Artificial Neural Networks, Support
Vector Machines and Naive Bayes Classifier.
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1. INTRODUCTION
Automatic Speech recognition (ASR) is one of the intensive
areas of research since it helps people to communicate in a
more natural and effective way[1]. Speech recognition
systems can be characterized by many parameters. The
commonly used method to measure the performance of a
speech recognition system is the recognition accuracy. Many
parameters affect the accuracy of the speech recognition
system. The accuracy and acceptance of speech recognition
has improved a lot in the last few years. Though automatic
speech recognition systems have improved a lot these years
and are now extensively used, their accuracy continues to lag
behind human performance, particularly in adverse conditions
[2]. In spite of the advances in ASR, there is still a
considerable gap between human and machine performance
[3]. Speech is a multi-component signal with time varying
frequency and amplitude. Due to this variability, transitions
may occur at different times in different frequency bands.
Automatic recognition of spoken digits is one of the
challenging tasks in the field of speech recognition [4]. A
spoken digit recognition process is needed in many
applications that need numbers as input such as automated
banking system, airline reservations, voice dialing telephone,
automatic data entry, command and control etc [5]. ASR is
basically a pattern recognition problem which also involves a
number of technologies and research areas like Signal
Processing, Natural Language Processing, Statistics etc [6].
Speech signals are non stationary in nature. Speech
recognition is a complex task due to the differences in gender,
emotional state, accent, pronunciation, articulation, nasality,
pitch, volume, and speed variability in people speak [7].
Presence of background noise and other types of disturbances
also affect the performance of a speech recognition system.
Speaker independence is difficult to achieve because these
models recognize the speech patterns of a large group of
people. The paper is organized as follows. Section 2 gives a
brief description of the problem definition. The methodology
used in the design is explained in section 3. The creation of the
digits database is illustrated in section 4. Section 5 describes
the method used for preprocessing. Section 6 elaborates the
feature extraction technique used in this work followed by the
classification techniques in section 7. Section 8 presents a

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detailed analysis of the experiments done and the results
obtained. Conclusion is given in the last section.
2. PROBLEM DEFINITION
The main objective of a speech recognition system is to
recognize speech with maximum accuracy. Here, digits in
Malayalam, one of the four major Dravidian languages of
southern India and the official language of the people of
Kerala is chosen for recognition. Research in Malayalam is in
its infancy stage and only very few works have been reported
in Malayalam. The performance of the overall speech
recognition system depends on the techniques used for pre-
processing, feature extraction technique selected and the
classifiers used. In this work, wavelet denoising method based
on soft thresholding is used for pre-processing the captured
signals [8]. Wavelet based feature extraction method namely
DWT is selected for feature extraction due to the good time
and frequency resolution properties of the wavelets. For
classification, three classifiers are selected namely ANN,
SVM and Naive Bayes classifier. The performance of these
classification methods in classifying the digits is evaluated
along with DWT.
3. ARCHITECTURE OF THE SYSTEM
In this work, the speech recognition system is divided into 4
modules. First the spoken digits database is created since there
is no built-in standard database available in Malayalam. The
speech samples in the database are often corrupted by additive
noises like background noise. So these speech signals are pre-
processed using wavelet denoising techniques to suppress the
noise in it. For processing speech, the speech signal has to be
represented in the form of parameters. So, after denoising the
signals, they are presented to feature extraction stage where
the features are extracted using a wavelet based feature
extraction technique called Discrete Wavelet Transforms. The
extracted features are then classified into appropriate classes
during classification phase. Here, three classifiers namely
ANN, SVM and Naive Bayes classifier are used and the
performance of these techniques are evaluated based on
recognition accuracy. The four stages of this work are
explained below.
4. CREATION OF THE SPEECH DATABASE
In Malayalam, since there is no standard database available, a
spoken digits database is created using 200 speakers of age
between 6 and 70 uttering 10 Malayalam digits. We have used
80 male speakers, 80 female speakers and 40 children for
creating the database. Male and female speech differ in pitch,
frequency, phonetics and many other factors due to the
difference in physiological as well as psychological factors.
The samples stored in the database are recorded by using a
high quality studio-recording microphone at a sampling rate of
8 KHz (4 KHz band limited). Recognition has been made on
the ten Malayalam digits from 0 to 9 under the same
configuration. Our database consists of a total of 2000
utterances of the digits. The spoken digits are preprocessed,
numbered and stored in the appropriate classes in the database.
The number digit, spoken digits in Malayalam, their
International Phonetic Alphabet (IPA) format and English
translation are shown in Table 1.
Table -1: Malayalam digits database
Number
digit
Digits in
Malayalam
IPA format
English
Translation
0 /puː//d͡ ʒya/ /ṁ/ Zero
1 /o/ /nn/ One
2 /ɾa/ /ṇṭ/ Two
3 /muː/ /nn/ Three
4 /n̪ aː/ /l/ Four
5 /a/ /ñc/ Five
6 /aː/ /r/ Six
7 /eː/ /ɻ/ Seven
8 /e/ /ṭṭ/ Eight
9 /o//na//pa/ /t̪/ Nine
5. PRE-PROCESSIG SIGNALS
Noises from background cause degradation in the speech
signals. So, these signals are tuned so that the noise present in
it is removed before extracting the features. There are a
number of techniques available for speech enhancement. Since
we are using wavelets for feature extraction, wavelet
denoising algorithms are used for reducing the noise in the
signal. The two popular thresholding functions used in wavelet
denoising method are the hard and the soft thresholding
functions [9]. In both the methods, a threshold value is
selected. In hard thresholding, if the absolute value of an
element is lower than the threshold, then these values are set
to zero. Soft thresholding is an extension of hard thresholding.
Here, the elements whose absolute values are lower than the
threshold are first set to zero and then the nonzero elements
are shrinked towards 0. Hard and soft thresholding can be
expressed as

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Where X represents the wavelet coefficients and ι is the
threshold value. In this work, soft thresholding technique is
used. There are different standard threshold values. In this
work, we have used the universal threshold derived by
Donoho and Johnstone [10] for the white Gaussian noise
under a mean square error criterion which is defined as
ι = (3)
Where σ is the standard deviation and N is the length of the
signal. Standard deviation σ can be calculated as σ =
MAD/0.6745, where MAD is the median of the absolute value
of the wavelet coefficients. The outline of the algorithm used
for denoising mainly consists of 3 steps.
 Apply wavelet transform to the noisy signal to
produce the noisy wavelet coefficients up to 8 levels.
 Detail wavelet coefficients are then shrinked using
soft thresholding technique by selecting an
appropriate threshold limit.
 The inverse discrete wavelet transform of the
threshold wavelet coefficients is computed which
produces the denoised signal.
6. FEATURE EXTRACTION
Feature Extraction is a major part of the speech recognition
system since it plays an important role to separate one speech
from other and this has been an important area of research for
many years. Selection of the feature extraction technique plays
an important role in the recognition accuracy, which is the
main criterion for a good speech recognition system. Here,
DWT is used for extracting features and a brief description of
this is given below.
6.1 Discrete Wavelet Transforms
DWT is a relatively recent and computationally efficient
technique for extracting information from non-stationary
signals like audio. The main advantage of the wavelet
transforms is that it has a varying window size, being broad at
low frequencies and narrow at high frequencies, thus leading
to an optimal time–frequency resolution in all frequency
ranges [11]. DWT uses digital filtering techniques to obtain a
time-scale representation of the signals. DWT is defined by
(4)
Where Ψ (t) is the basic analyzing function called the mother
wavelet. In DWT, the original signal passes through a low-
pass filter and a high-pass filter and emerges as two signals,
called approximation coefficients and detail coefficients [12].
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 [13]. The successive high pass
and low pass filtering of the signal is given by
(5)
(6)
Where Yhigh (detail coefficients) and Ylow (approximation
coefficients) are the outputs of the high pass and low pass
filters obtained by sub sampling by 2. The filtering process is
continued until the desired level is reached according to Mallat
algorithm [14]. The discrete wavelet decomposition tree is
shown in figure 1.
Fig -1: DWT Decomposition over 3 levels
7. CLASSIFICATION
Classification is another important part of a speech recognition
system since the patterns are classified into different classes
during this stage. Pattern recognition is becoming increasingly
important in the age of automation and information handling
and retrieval. During classification stage, decisions are made
based on the similarity measures from training patterns using
information relating to known patterns. Then they are tested
using the unknown patterns. Since there are a number of
different classes in a speech recognition problem, a multiclass
classification technique is needed.
In almost all classification methods, the data is separated into
training and test sets. Each instance in the training set contains
a target value which represents the corresponding class and a
( ) (| | | |) | |
0 | | (2){
sign X X if X
Soft if XX
 

 

| |
0 | | (1){X if X
Hard if XX 



/2
( , ) ( )2 (2 )
j j
j k
W j K X k n k
 
 
[ ] [ ] [2 ]high
n
Y k x n g k n 
[ ] [ ] [2 ]low
n
Y k x n h k n 

__________________________________________________________________________________________
set of attributes. The test data do not contain a target value.
The objective of the classifier is to produce a model from the
training data which predicts the target values of the test data.
A brief description of the various classifiers used in this work
like ANN, SVM and Naive Bayes classifier are given below.
7.1 Artificial Neural Networks
Nowadays, ANNs are utilized in many applications due to
their parallel distributed processing, distributed memories,
error stability, and pattern learning and distinguishing ability.
ANN is an information processing paradigm consisting of a
number of simple processing units or nodes called neurons.
Each neuron accepts a weighted set of inputs and produces an
output [15]. Algorithms based on ANN are well suitable for
addressing speech recognition tasks. Inspired by the human
brain, neural network models use a number of characteristics
such as learning, generalization, adaptively, fault tolerance etc.
[16].
In this work, we are using the architecture of the MLP
network, which consists of an input layer, one or more hidden
layers, and an output layer. The algorithm used is the back
propagation training algorithm. In this type of network, the
input is presented to the network and moves through the
weights and nonlinear activation functions towards the output
layer, and the error is corrected in a backward direction using
the well-known error back propagation correction algorithm.
After extensive training, the network eventually establishes
the input-output relationships through the adjusted weights on
the network. After training the network, it is tested with the
dataset used for testing.
7.2 Support Vector Machines
SVM is a very useful technique used for classification. It is a
classifier which performs classification methods by
constructing hyper planes in a multidimensional space that
separates different class labels based on statistical learning
theory [17][18]. Though SVM is inherently a binary nonlinear
classifier, we can extend it to multiclass classification since
ASR is a multiclass problem. There are two major strategies
for multiclass classification namely One-against-All [17] and
One-against-One or pair wise classification [19]. The
conventional way is to decompose the M-class problem into a
series of two-class problems and construct several binary
classifiers. In this work, we have used One-against-One
method in which there is one binary SVM for each pair of
classes to separate members of one class from members of the
other. This method allows us to train all the system, with a
maximum number of different samples for each class, with a
limited computer memory [20].
7.3 Naive Bayes Classifier
Since speech recognition is a multiclass classification problem
and Naive Bayes classifiers can handle multiclass
classification problems, it is also used here for classifying the
digits. Naive Bayes classifier is based on the Bayesian theory
which is a simple and effective probability classification
method. This is a 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 of 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. The probability value
of the winning class dominates over that of the others [18].
8. EXPERIMENTS
Using DWT, the signals are decomposed into approximation
and detail coefficients. The low frequency components are
decomposed upto 10th level. The number of features obtained
for a digit at 10th level is 16. The decomposition of digits
zero to nine using DWT at 10th level is shown below.
Fig -2: Decomposition of digit zero at 10th level
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
Original Signal
0 5 10 15 20
-0.05
0
0.05
0.1
0.15
Word -zero Approximation coefficients Level 8
0 5 10 15 20
-0.2
-0.1
0
0.1
0.2
Detail coefficients Level 8

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Fig- 3: Decomposition of digit one at 10th level
Fig- 4: Decomposition of digit two at 10th level
Fig- 5: Decomposition of digit three at 10th level
Fig- 6: Decomposition of digit five at 10th level
Fig- 7: Decomposition of digit four at 10th level
Fig- 8: Decomposition of digit six at 10th level
0 0.5 1 1.5 2 2.5 3
x 10
4
-0.4
-0.2
0
0.2
0.4
Original Signal
0 10 20 30
-0.1
-0.05
0
0.05
0.1
Word -four Approximation coefficients Level 8
0 10 20 30
-0.05
0
0.05
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
x 10
4
-0.5
0
0.5
Original Signal
0 5 10 15 20
-0.04
-0.02
0
0.02
Word -one Approximation coefficients Level 8
0 5 10 15 20
-0.04
-0.02
0
0.02
0.04
0 0.5 1 1.5 2 2.5 3 3.5
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 10 20 30
-1.5
-1
-0.5
0
0.5
Word -two Approximation coefficients Level 8
0 10 20 30
-0.2
-0.1
0
0.1
0.2
0 0.5 1 1.5 2 2.5
x 10
4
-0.5
0
0.5
1
Original Signal
0 5 10 15 20
-0.05
0
0.05
Word -three Approximation coefficients Level 8
0 5 10 15 20
-0.1
-0.05
0
0.05
0.1
0 0.5 1 1.5 2 2.5 3
x 10
4
-0.4
-0.2
0
0.2
0.4
Original Signal
0 10 20 30
-0.2
-0.1
0
0.1
0.2
Word -five Approximation coefficients Level 8
0 10 20 30
-1
-0.5
0
0.5
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 5 10 15 20
-1
-0.5
0
0.5
Word -six Approximation coefficients Level 8
0 5 10 15 20
-2
-1
0
1

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Fig- 9: Decomposition of digit seven at 10th level
Fig- 10: Decomposition of digit eight at 10th level
Fig- 11: Decomposition of digit nine at 10th level
For classification using ANN, SVM and Naive Bayes
classifier, the feature set data is divided into training,
validation and test set. 70% data is used for training, 15% for
validation and 15% for testing. That is, 7000 data is used for
training, 1500 data for validation and 1500 for testing.
8.1 Performance Evaluation of Classifiers
For classification using ANN, one input layer, one hidden
layer and one output layer is used. Back propagation algorithm
is used for error correction.
For classification using SVM, since one-against-one SVM
classification is used, we construct a binary SVM for each pair
of classes. The number of binary SVMs built is defined as C*
(C-1)/2, where C is the number of classes. So a total of
10*(10-1)/2 = 45 binary SVMs are built. The table given
below shows the comparison of various classifiers.
Table -2: Performance Evaluation of Classifiers
Classifier Recognition Accuracy %
ANN 89
SVM 86.6
Naive Bayes 83.5
From the results it is clear that all these classifiers are good in
recognizing speech. But better results are obtained using
ANN.
CONCLUSIONS
In this work, a speech recognition system is designed for
speaker independent spoken digits in Malayalam. The speech
samples in the database are applied to wavelet denoising using
soft thresholding for removing noise. DWT is used for feature
extraction. Three classifiers namely ANN, SVM and Naïve
Bayes classifier are used for classifying the digits into proper
classes. A comparative study of these three classifiers is
performed. The performance of all these techniques are tested
and evaluated. All techniques are found to be efficient in
recognizing speech. The accuracy rate obtained by using DWT
and ANN combination is 89% which is found to be better
than that of the other two methods. The experiment results
show that this hybrid architecture using discrete wavelet
transforms and neural networks could effectively extract the
features from the speech signal and classify it for efficient
recognition of digits. For future work, other feature extraction
techniques like Linear Predictive Coding (LPC), Mel
Frequency Cepstral Coefficients (MFCC), Wavelet packet
Decomposition (WPD) can also be used and classified using
these classifiers.
ACKNOWLEDGEMENTS
I wish to express my sincere gratitude to all who have
contributed throughout the course of this work.
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 5 10 15 20
-0.04
-0.02
0
0.02
0.04
Word -seven Approximation coefficients Level 8
0 5 10 15 20
-0.1
-0.05
0
0.05
0.1
0 0.5 1 1.5 2 2.5 3
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 5 10 15 20
-0.6
-0.4
-0.2
0
0.2
Word -eight Approximation coefficients Level 8
0 5 10 15 20
-0.6
-0.4
-0.2
0
0.2
0 0.5 1 1.5 2 2.5
x 10
4
-1
-0.5
0
0.5
1
Original Signal
0 5 10 15 20
-0.02
0
0.02
0.04
Word -nine Approximation coefficients Level 8
0 5 10 15 20
-0.1
-0.05
0
0.05
0.1

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BIOGRAPHIES:
Sonia Sunny is working as Associate
professor in the department of Computer
Science, Prajyoti Niketan College,
Pudukad, Thrissur, Kerala State, India.
Currently she is doing research in the
area of Speech Recognition at Cochin
University of Science and Technology,
Cochin, Kerala State, India. In 1995, she
received her M.Sc in Computer Science
from Bharathidasan University, Thiruchirappalli and M.Phil in
Computer Science from Bharathiar University, Coimbatore in
2008. Her research interest includes Speech Processing,
Artificial Intelligence and Pattern Recognition.
David Peter S is presently working as
professor in Computer Science at School
of Engineering, Cochin University of
Science and Technology, Cochin,
Kerala, India. He did his Post-
graduation in Computer Science at IIT
Madras and PhD at Cochin University.
He has presented research papers in
several International Conferences and
has published many articles in International Journals. His
areas of interest includes Artificial Intelligence, Natural
Language Processing, Information Systems, Engineering, etc.
K. Poulose Jacob, Professor of Computer
Science at Cochin University of Science
and Technology (CUSAT) since 1994, is
currently Director of the School of
Computer Science Studies. He holds
additional charge as the honorary Director
of CUSAT Planning & Development. He
has presented research papers in several

__________________________________________________________________________________________
International Conferences in Europe, USA, UK, Australia and
other countries. He has delivered invited talks at several
national and international events. Dr. Jacob is a Professional
member of the ACM (Association for Computing Machinery)
and a Life Member of the Computer Society of India. Till
now twelve candidates have obtained PhD degrees in
Computer Science & Engineering under his supervision. He
has been PhD Theses examiner for several Universities. He
has more than 75 research publications to his credit. His
research interests are in Information Systems Engineering,
Intelligent Architectures and Networks.

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