Int. J. Computational Systems Engineering, Vol. 1, No.

1, 2012 25

A Hindi speech recognition system for connected

words using HTK

Kuldeep Kumar* and R.K. Aggarwal

Department of Computer Engineering,
National Institute of Technology,
Kurukshetra-136119, Haryana, India
E-mail: kuldeepgargkkr@gmail.com
E-mail: rka15969@gmail.com
*Corresponding author

Ankita Jain
Department of Electronics and Communication Engineering,
National Institute of Technology,
Kurukshetra-136119, Haryana, India
E-mail: ankitajain.08@gmail.com
Abstract: A speech recognition system converts the speech-sound into the corresponding text.
The uttered speech is first understood by the machine and then the corresponding text is
displayed. This paper aims to build a connected-words speech recognition system for Hindi
language. The system has been developed using hidden Markov model toolkit (HTK) that uses
hidden Markov models (HMMs) for recognition. The system has been trained to recognise any
sequence of words selected from the vocabulary of 102 words. Initially, Mel frequency cepstral
coefficients (MFCCs) have been used to extract the features from the speech-files. Then, the
system has been trained to estimate the HMM parameters using word level acoustic models. The
training data has been collected from 12 speakers including both males and females. The test-data
used for evaluating the system-performance has been collected from the five speakers. The
experiments have also been performed on the system. The experimental results show that the
presented system provides the overall word-accuracy of 87.01%, word-error rate of 12.99%, and
word-correction rate of 90.93% respectively. The work has been evaluated by performing the
comparative analysis with the existing similar works and the betterment has been reported.
Keywords: hidden Markov models; HMMs; speech recognition system; automatic speech
recognition; ASR; hidden Markov model toolkit; HTK; Hindi; connected words; Mel frequency
cepstral coefficient; MFCC.
Reference to this paper should be made as follows: Kumar, K., Aggarwal, R.K. and Jain, A.
(2012) ‘A Hindi speech recognition system for connected words using HTK’, Int. J.
Computational Systems Engineering, Vol. 1, No. 1, pp.25–32.
Biographical notes: Kuldeep Kumar is with the Department of Computer Engineering, National
Institute of Technology (NIT), Kurukshetra, Haryana, India. He received his BTech in Computer
Engineering (with honours) from University Institute of Engineering and Technology (UIET),
Kurukshetra University, Kurukshetra, India in 2009. He has also worked as a summer
trainee in the Department of Computer Engineering, Institute of Technology, Banaras Hindu
University (IT-BHU), Varanasi, India. He has several publications in reputed international
journals/conferences. His current areas of interest include speech processing, semantic web,
software engineering, and statistical models.
R.K. Aggarwal is an Associate Professor in the Department of Computer Engineering at National
Institute of Technology (NIT), Kurukshetra, Haryana, India. He has published more than
30 papers in various conferences and journals. He has delivered several invited talks, keynote
addresses and chaired the sessions in reputed conferences. His research interests include speech
processing, soft computing, statistical modelling, and spirituality. He is a life member of
Computer Society of India (CSI) and Indian Society for Technical Education (ISTE). He has
been involved in various academic, administrative and social affairs of many organisations
having more than 20 years of experience in this field.
Ankita Jain is an Assistant Professor in the Department of Electronics and Communication
Engineering, National Institute of Technology (NIT), Kurukshetra, Haryana, India. She is
involved in various academic, research and social affairs. She has many papers in
national/international journals and conferences. Her areas of interest include speech processing
and communication systems.

Copyright © 2012 Inderscience Enterprises Ltd.

26 K. Kumar et al.
1 Introduction
Speech interfacing provides a convenient and user friendly
way of man-machine communication. Their absence makes
the man-machine interaction obsolete. Unadventurously,
transfer of information between man and machine is carried
out via keyboards, pointing devices like mouse, touchpad
for input and visual display units, monitors, plotters or
printers for output. However, it is not convenient for a
common man to use these devices, as they require
certain amount of skills and are time consuming too. On the
other hand, speech interfacing offers high bandwidth
information and relative ease of use. It also permits the
user’s hands and eyes to be busy with a task, which is
particularly valuable when the user is in motion or in
natural field settings. One can speak hastily instead of
typing. Similarly speech output is more impressive and
understandable than the text output. Speech interfacing
involves two distinct areas, speech synthesis and
automatic speech recognition (ASR). Speech synthesis is
the process of converting the text input into the
corresponding speech output, i.e., it acts as a text to speech
converter. Conversely, speech recognition is the way of
converting the spoken sounds into the text similar to the
information being conveyed by these sounds. Among
these two tasks, speech recognition is more difficult but it
has variety of applications such as interactive voice
response system, applications for physically challenged
persons and others (Aggarwal and Dave, 2011). There are
many public domain software tools available for the
research work in the field of speech recognition such as
Sphinx from Carnegie Mellon University (SPHINX, 2011),
hidden Markov model toolkit (HTK, 2011) and large
vocabulary continuous speech recognition (LVCSR)
engine Julius from Japan (Julius, 2011). This paper aims
to develop and implement speech recognition system
for Hindi language using the HTK open source
toolkit.
1.1 Motivation
While speech interfacing is the expedient way of
communication, it is fruitful only when the common
man will be able to reap its benefits. Most of the
international organisations like Microsoft, SAPI, IBM, and
Dragon-Naturally-Speech as well as research groups
working on this field are concentrating on the European
languages especially English. This restricts their usage to
small fraction of the population having literacy in such
languages. However, some of the highly populated countries
like India have their own native languages spoken by large
portion of the population. A person having no proficiency in
European languages cannot use these technologies. Local
relevance and lack of effective Hindi speech recognition
system has inspired the authors to develop such system to
narrow the gap.
1.2 Paper contribution
This paper presents a Hindi speech recognition system
developed for connected words. To extract the features from
the speech sound, Mel frequency cepstral coefficients
(MFCCs) are used. Hidden Markov models (HMMs) are
used to train and recognise the speech. Hidden Markov
model toolkit (HTK) developed in 1989 at the Speech
Vision and Robotics Group of the Cambridge University
Engineering Department (CUED) is used to accomplish
this. HTK provides system tools for both training and
testing phase. Initially, HTK training tools are used to train
HMMs using training utterances from a speech database.
Features are extracted from these training utterances and
then, these features are used to model the system. Finally,
HTK recognition tools are used to transcribe the unknown
utterances. They use the system model generated during
training phase to test the system.
Apart from introduction in Section 1, the paper is
organised as follows. Some of the related works are
presented in Section 2. Section 3 presents the architecture
and functioning of the developed speech recognition
system. Section 4 deals with implementation work.
Experimental results are given in Section 5. Section 6
compares the presented work with the previous works.
Finally, the system has been concluded in Section 7.
2 Related works
This section presents some of the reported works available
in the literature that are similar to the presented work.
Among others, some of the works providing ASR system
for South-Asian languages are (Al-Qatab and Ainon, 2010;
Gupta, 2006; Pruthi et al., 2000).
Pruthi et al. (2000) have developed a speaker-dependent,
real-time, isolated word recogniser for Hindi. Linear
predictive cepstral coefficients (LPCCs) were used for
feature extraction and recognition was carried out using
discrete HMM. System was designed for two male speakers.
The recognition vocabulary consists of Hindi digits
(0, pronounced as ‘shoonya’ to 9, pronounced as ‘nau’).
An isolated word speech recognition system for Hindi
language has been designed by Gupta (2006). System uses
continuous density hidden Markov model (CDHMM) and
consists of word-based acoustic units. Again the system
vocabulary contains Hindi digits. Recogniser gives good
results when tested for sounds used for training the model.
For other sounds too, the results are satisfactory.
The work in Al-Qatab and Ainon (2010) discusses the
development and implementation of an Arabic speech
system using HTK. System can recognise both continuous
speech as well as isolated words. System uses an
Arabic dictionary built manually by the speech-sounds of
13 speakers. MFCCs were used to extract the speech feature
vectors. The vocabulary consists of 33 words.
 A Hindi speech recognition system for connected words using HTK 27

This paper shows the design and implementation of 3.1 Pre-processing

Hindi speech system. The system uses a vocabulary of
102 words. The developed system is giving good
recognition results for both speaker dependent and speaker
independent environments.
3 System architecture
The developed speech system mainly consists of three
modules, acoustic analysis module, training module and
testing module. Initially, data preparation is carried out.
Multiple occurrences of words available in the vocabulary
are recorded. These speech-signals captured by transducer
are pre-processed to convert them into digital forms.
However, these acoustic signals cannot be directly
processed by speech recognition system. These have to be
represented in a more compact and efficient form which is
achieved using acoustic analysis. Training module is used to
generate the system model which is used by the testing
module during the system testing. The architecture of the
developed speech recognition system is shown in Figure 1.
Analog speech signals captured by a transducer such as
microphone or telephone must be digitised according to the
Nyquist theorem. According to this theorem, signal must be
sampled more than twice the rate of the highest frequency
required in the analysis. The frequency equal to twice the
highest frequency is called the Nyquist rate. In general, a
sampling rate between 8 KHz and 20 KHz is used for
speech recognition application. For telephonic channel,
8 KHz sampling rate is recommended while sampling
rate of 16 KHz is used for normal microphones. For
quantisation, 16 bit, 24 bit or 32 bit float are used depending
upon applications.
3.2 Acoustic analysis module
Speech recognition system cannot be used to process the
digital waveforms. These have to be represented in a more
compact and efficient way. For this, initially, digitised input
is flattened using filters and then essential features having
acoustic correlation with the speech input are extracted
using feature extraction.

Figure 1 Developed speech system architecture

Testing module
O
W

Acoustic analysis Language Acoustic

model models P(O|W)
Preprocessing Pre-emphasis Feature Pronunciation
Spoken extraction Parameterised corresponding
word waveforms (O) to sound
Pronunciation Speech
Transcription corresponding to sound transcription
dictionary Recognition
component
P(W)

Generated
system model

Training module
Model
generation
Acoustic analysis
HMM
Preprocessing Pre-emphasis Feature re-estimation
Spoken extraction Parameterised
word waveforms(O)
Acoustic model Pronunciation Language model
generation dictionary generation

Corpus
28 K. Kumar et al.
3.2.1 Pre-emphasis
Pre-emphasis ensures that in the frequency domain all the
formants of the speech signal have similar amplitude so that
they get equal importance in subsequent processing stages.
Typically, the speech signal produced by human beings has
a spectral slope of approximately –6dB/octave for voiced
sounds due to following reasons:
• the slope of glottis introduces a slope of –12dB/octave
• the lip radiation introduces a slope of +6dB/octave.
Therefore, the resultant slope of approximately –6dB/octave
exists in the recorded voiced speech sounds. Due to this
reason, high frequency formants have smaller amplitude
with respect to the lower frequency formants. Pre-emphasis
is performed to remove this slope of –6dB/octave
and to make the signal spectrally flatten (Deng and
O’Shaughnessy, 2003).
3.2.2 Feature extraction
Feature extraction is the way of finding a set of properties
of an utterance having acoustic correlations with the speech
signal. Such parameters are termed as features. A feature
extractor is expected to discard irrelevant information
from the task while keeping the useful one. To do this,
successively some portion of the speech signal is considered
for processing, called window size. Data acquired in a
window is called as a frame. Typically, frame size ranges
from 10 to 25 milliseconds with an overlap of about
50%–70% between consecutive frames. The data in this
analysis interval is multiplied with a windowing function.
Different types of windows like Rectangular, Hamming,
Hanning, Bartlett, Blackman or Gaussian can be used. After
that features are extracted on frame by frame basis.
There are several ways to extract features from each
frame such as LPCC (Markel and Gray, 1976), MFCC
(Davis and Mermelstein, 1980), perceptual linear prediction
(PLP) (Hermansky, 1990), wavelet (Sharma et al., 2008),
temporal patterns (TRAPs) (Hermansky and Sharma,
1999), RASTA (relative spectral transform) processing
(Hermansky and Morgan, 1994) and others. HTK 3.4
supports only LPCC, MFCC and PLP.
3.3 System training module
Training module generates the system model. The designed
system architecture uses HMMs to generate the acoustic and
language models. The system model generated during the
training is used during the testing phase to recognise an
unknown utterance. The various phases used in system
training are:
3.3.1 Acoustic model generation
To recognise the unknown utterances, some reference
models are needed to compare with them. These reference
models are known as acoustic models. Using these models,
most probable sounds are identified that are produced when
the unknown words are spoken. The two kinds of acoustic
models are the word model and phoneme models. In the
word model, words are modelled as a whole. This model is
used for small vocabulary system. If a new word is added to
the vocabulary, system has to be trained for the new word
also. In phoneme model, on the other hand, parts of word
called phones are modelled. This model is used for large
vocabulary systems. In phoneme model-based system,
adding new word to the vocabulary is manageable as sounds
corresponding to the phone sequence of newly added word
may be already known to the system.
An acoustic model can be implemented using various
approaches such as HMM (Rabiner and Huang, 1986),
artificial neural networks (ANN) (Wilinski et al., 1998),
dynamic Bayesian networks (DBN) (Deng, 2006), support
vector machine (SVM) (Guo and Li, 2003), hybrid methods
(i.e., combination of two or more approaches) and others.
HMM has been used in some form or another in virtually
every state-of-the-art speech and speaker recognition
systems.
3.3.2 Language model generation
Language model predicts the probability of a word
occurring in the context. In some cases, it may be possible
that there are words which are phonetically similar but
having different meaning called homophones such as जल
(water) or ज़ल (related to burning), हं स (name of a bird) or
हँ स (laugh), आिद (start) or आदी (addicted). Handling
such homophones is a critical issue for any ASR, as
they normally increase acoustic confusability. Different
languages have different number of homophone words. For
example, French language admits a large number of
homophones; hence in this particular case automatic
transcription is a very challenging task. Homophones are
also common is English language (like I or eye, week or
weak, principal or principle) but rare in Hindi. Speech
system recognises these words based on the context in
which they occur. Language model provides this context to
the speech recognition system. There are mainly four
approaches to language modelling viz. grammar-based
approach, stochastic approach, uniform modelling and finite
state approach (Madhav, 2005).
3.4 System testing module
Testing module is used to recognise the unknown
utterances. Initially, unknown speech input is converted into
a form that can be processed easily by the system. To
achieve this, acoustic analysis of the speech input is carried
out. The parameterised features thus obtained are used by
the recognition component to give the transcription
corresponding to the sounds.
3.4.1 Recognition component
Recognition component recognises the test samples based
on the acoustic properties of the words (Young et al., 2009).
 A Hindi speech recognition system for connected words using HTK 29

Considering that each word is represented as a sequence of
speech vectors or observations O defined as:
O = o1o2 o3 … oT (1)
where oi is the speech vector observed at time i. The
recogniser finds the most probable sequence of words W
given the speech vectors O, i.e.,
Wˆ = arg max P (W | O) (2)
W∈ L
The probability P(W | O) is computed using Bayes’ rule
given as:
P(O | W ) P(W )
P (W | O) = (3)
P(O)
Given an acoustic observation sequence O, recogniser finds
the sequence W of words which maximises the probability
P(O | W)P(W). The term P(W) is the prior probability of the
word which is estimated by the language model. P(O | W) is
the observation likelihood estimated using the acoustic
model. The term P(O) is constant and hence can be ignored.
4 Implementation
This section describes the implementation of the speech
system developed based upon the system architecture
presented in the previous section.
was asked to utter each word of the vocabulary four times.
Thus giving a total of 4,896 (12*4*102) speech-files. Data
was recorded at the room environment.
To prepare test data, the voices of five people (three
male and two females) of age group between 17–25 years
were recorded with the same specification used for
training. Out of five speakers, three were those used for
training the system. From these five speakers, total of
190 speech-samples were recorded.
4.3 Acoustic analysis
Once the data was recorded and digitised, each speech file
was labelled with corresponding word. The developed
system uses manual labelling. Wavesurfer has been used for
labelling the speech files. Once the files were labelled, the
training data was parameterised into a sequence of features.
For this purpose, HTK tool HCopy was used. The system
uses MFCCs which were derived from fast Fourier
transform (FFT)-based log spectra. The input speech was
processed at the frame rate of 10 ms with a Hamming
window of 25 ms. The reason for choosing the Hamming
window is that the side lobes of this window are lower than
the others windows which avoids leakage, i.e., does not
swap relevant energy from distant frequencies. Acoustic
parameters were 39 MFCCs with 12 Mel cepstrum plus log
energy and their delta (first order derivative) and
acceleration (second order derivatives) coefficients. Table 1
shows the properties of speech-file and values of various
parameters used for acoustic analysis.

4.1 System description Table 1 Values of various parameters used for acoustic
analysis
The Hindi speech recognition system based on MFCCs has
been developed using HTK. HTK v3.4 is used. The system S. no. Parameter Value
has been developed in the Ubuntu 10.04 operating
1 Input file format .wav
environment which is a Linux platform. The system has
been designed to recognise a set of connected words from 2 Sampling rate 16,000 Hz
the set of 102 unique words. Initially, HMM learning has 3 Bit rate (bits per 16
been performed using the HTK training tools. Training sample)
utterances and their associated transcriptions were used for 4 Type of channel used mono
HMM learning. Then, unknown utterances were transcribed 5 Window size 250,000.0 (25 msec.)
using the HTK recognition tools. 6 Frame periodicity 100,000.0 (10 msec.)
7 Window used Hamming
4.2 Database preparation
8 Number of filter-bank 26
For training and testing of the speech system, data channels
needs to be collected from certain numbers of speakers. A 9 Target kind MFCC_0_D_A (MFCC
unidirectional microphone was used for recording keeping a with energy, delta (Δ) and
distance of approximately 5–10 cm between mouth of the acceleration (ΔΔ) coefficients.
speaker and the microphone. 10 Numbers of MFCC 12
System was trained using the voices of 12 people (seven coefficients
male and five female) of age group between 18–23 years. 11 Pre-emphasis 0.97
Recording of the speech sounds were done using the Linux coefficient
command brec. Sounds were recorded at sampling rate of 12 Length of cepstral 22
16,000 Hz on the mono channel. 16 bits per sample were liftering
used which divides the element position of a sample into
13 Energy normalisation true
65,536 (216) possible values. Speech files were stored in
.wav format. By using these specifications, each speaker
30 K. Kumar et al.
4.4 System training
This section describes the methodology used for training the
implemented system.
4.4.1 Language modelling – task definition
The developed speech system uses grammar-based
approach for language modelling. In grammar-based
approach, structure of the language is defined in term of
grammar rules. The grammar is specified using extended
Backus-Naur form (EBNF). From the grammar, we can
generate a task network where the nodes will be the words
in the dictionary. A network describes the sequences of
words that can be recognised by the system. HTK tool
HParse has been used to obtain the task network. A
dictionary describing the pronunciations of the words was
also made. The information regarding the correspondence of
each HMM with a particular grammar variable was also
kept in the dictionary.
4.4.2 Acoustic modelling
HMMs have been used for acoustic modelling. For acoustic
modelling of each word, a prototype HMM model was
specified. HMM prototype specifies the model topology,
number of states, the transition parameters, and the output
distribution parameters. These HMM prototypes were then
re-estimated using the training data. Apart from the models
of vocabulary words, model for silent (sil) must be included.
HMM prototype uses different numbers of states for each
word depending upon the number of phone units, duration
of the words. This system uses 5–16 state HMMs in which
the first and last are non-emitting states. The prototype
models were initialised using the HTK tool HInit which
allows a fast and precise convergence of the training
algorithm. Then HMM parameters were re-estimated using
the HTK tool HRest which estimates the optimum values of
HMM parameters. Parameters were re-estimated repeatedly
using training data until re-estimation converges. Within the
each HRest iteration, convergence is indicated by the
change measure (convergence factor). This process was
repeated until absolute value of convergence factor does not
decrease from one HRest iteration to another. A system
vocabulary word uses approximately two to five iterations
to converge.
4.5 System testing
Once the system models have been generated, system can
be used to recognise an unknown utterance. It is called
testing. During testing, initially, unknown utterance spoken
by the speaker is converted into the series of acoustic
vectors using HTK tool HCopy. These acoustic vectors are
then processed by Viterbi algorithm. HTK tool HVite is
used to perform Viterbi-based speech recognition which
uses the token passing algorithm (Young et al., 1989). HVite
takes as input a network describing the allowable word
sequences generated by using HTK tool HParse, a
dictionary describing how each word is pronounced and
acoustic model generated during HMM training.
4.6 Performance analysis
In order to analyse the system performance, HTK provides a
tool HResult. It is used to compute the accuracy of the
system. It compares the machine transcription of the test
utterances with the corresponding reference transcription
files.
The performance of speech system is evaluated as:
N −D−S H
%correct = × 100 = × 100 (4)
N N
where N is the number of words in test set, D is the number
of deletions, S is number of substitutions and H is the
number of correct labels. %correct gives the percentage of
word correctly recognised.
The accuracy is computed as:
N −D−S −I H −I
%accuracy = × 100 = × 100 (5)
N N
where I is the number of insertions. The performance of
speech recognition system can be evaluated by measuring
the word error rate (WER) defined as:
S+I+D
word error rate (WER) = × 100
N (6)
= 100 − %accuracy
5 Experimental results
The system was tested using the test data prepared
separately by a set of five speakers. Each speaker was asked
to utter some words of the vocabulary at least once a time.
Also, some test data was collected from the training data.
Out of the five speakers used for testing, three were those
used for collecting the training data. Thus, test data
contains three types of sounds – sounds used for training the
system, sounds spoken by the speaker whose other sound
files were used for training the system, sounds of a speaker
that does not participate in training. Test data was recorded
at the room environment. The recognition results are
shown in Table 2. Overall word-accuracy and word-error
rate of the system is 87.01% and 12.99% respectively.
Word-correction rate is 90.93%.
5.1 Experiment with different noisy environment
Experiments have been performed in various noise
environments: in open space, in lab room, in room
environment, in class room, in market. For each of
environment except the room environment, testing was done
using a data set of 50 words. The results are given in
Figure 2. Figure shows that as the noise level increases,
recognition performance of the system gets degraded.
 A Hindi speech recognition system for connected words using HTK 31

Table 2 Recognition results

No. of No. of
Types of No. of No. of No. of % word % word Word error
Speaker ID spoken recognised
sound deletion (D) insertion (I) substitution (S) correct accuracy rate (WER)
words (N) word (H)
Speaker 1 Seen 17 16 0 0 1 94.1 94.1 5.9
Speaker 2 16 16 0 1 0 100.0 93.8 6.2
Speaker 3 15 14 0 1 1 93.3 86.7 13.3
For seen sound seen speaker, average performance: 95.8% 91.6% 8.4%
Speaker 1 Unseen 26 23 1 1 2 88.5 84.6 15.4
Speaker 2 24 22 0 1 2 91.7 87.5 12.5
Speaker 3 31 27 0 1 4 87.1 83.9 16.1
For unseen sound seen speaker, average performance: 89.1% 85.4% 14.6%
Speaker 4 Unseen 41 36 2 3 3 87.8 80.5 19.5
Speaker 5 20 17 1 0 2 85.0 85.0 15.0
For unseen sound unseen speaker, average performance: 86.4% 82.7% 17.3%
Overall system performance: 90.93% 87.01% 12.99%

Figure 2 Performance versus environment acoustic units. Again the word vocabulary contains Hindi
digits. Recogniser gives good results when tested for sound
100
90
room environment used for training the model. For other sounds too, the results
80
open space are satisfactory. System is highly efficient but it is an
70
60 isolated word recogniser. Also vocabulary size is too small.
50
40
lab room The developed system recognises the connected word with
30
20 class room
larger vocabulary. Table 3 shows the tabular comparison of
10 the developed system with the reported works.
0
%correct %accuracy WER market environment
Table 3 Tabular comparison of performance

Al-Qatab
Pruthi et al. Gupta Our
6 Comparative analysis Parameter and Ainon
(2000) (2006) work
(2010)
In this section, the presented work has been compared with Vocabulary 10 33 10 102
the existing similar works. In paper (Pruthi et al., 2000), size
they have developed a speaker-dependent, real-time,
Speaker No Yes Yes Yes
isolated word recogniser for Hindi. To extract the features independency
from the speech data, LPCCs were used. Recognition has
Acoustic Word Phone Word Word
been carried out using HMM. System was designed for two model
male speakers. The recognition vocabulary consists of Hindi
Connected No Yes No Yes
digits (0, pronounced as ‘shoonya’ to 9, pronounced as word
‘nau’). However the system is giving good performance, but recognition
the design is speaker specific and uses very small
% accuracy 84.38% 97.99% - 94.34%
vocabulary. Also it is an isolated word recogniser. The
presented system is speaker independent connected word Word error 15.62% 2.01% - 5.66%
rate
system and uses larger vocabulary.
Paper (Al-Qatab and Ainon, 2010) discusses the
development and implementation of an Arabic speech
system using HTK. System can recognise both continuous 7 Concluding remarks and future work
speech and isolated words. System uses an Arabic This paper presents a speech recognition system for
dictionary built manually by 13 speaker sounds. MFCCs Hindi language. The presented system recognises the
were used to extract the speech feature vectors. The connected-words using acoustic word model. The system
vocabulary consists of 33 words. Although the system has been trained using a vocabulary size of 102 words. HTK
performance is good, the vocabulary size is small. The has been used to develop the system that uses HMMs for
presented system uses larger vocabulary than this one. recognition. Acoustic features are extracted using MFCCs.
Gupta (2006) describes an isolated word speech For HMM modelling, different numbers of states are
recognition system for Hindi language. System uses selected depending upon the number of phone units,
continuous density HMM and consists of word-based duration of the words. Training data has been collected from
32 K. Kumar et al.
12 different speakers. To evaluate the system performance,
system has also been tested in the room environment, open
space, lab room, class room environment as well as market
environments using a set of five speakers. It has been
observed from the performed experiments that the accuracy
and WER of the proposed system is 87.01% and 12.99%
respectively. Word correction rate was found to be 90.93%.
Based on the comparative analysis performed in the paper,
it has been found that the system is performing well with
more vocabulary-size compared to the other reported similar
works. The future works involves the development of
system for more vocabulary size and to improve the
accuracy of the system using the noise compensation/speech
enhancement techniques.
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