International Journal of Faculty Electrical Engineering ISSN 0001-0002 Volume 1 Number 1 (2017) pp.

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© Telkom University Publications. http://www.telkomuniversity.ac.id

A Review: Speech to Text by Machine

Using Discrete Method Hidden Markov Models on Smart House

Deyan Havith Dailamy

Research Scholar, Computer Engineering,
Faculty of Electrical Engineering,Telkom University,
Jl. Telekomunikasi No.1, Dayeuhkolot, Bandung, Jawa Barat, Indonesia.
Orcid Id: 0000-0001-9817-3185

Abstract review paper project the classification process used is the

Speech recognition is a technology used to recognize words
spoken by humans and turn them into data. In smart house
applications, speech recognition can be applied as an interface
for users to control the motor or other electronic devices. In
this paper review, the writer implements speech recognition
system with MATLAB software. The characteristic extraction
process uses the melep method of frequency cepstral
coefficients (MFCC). As for the process of classification,
using the method of hidden markov model (HMM). The
output signal from the speech recognition process will be sent
to the microcontroller system and then used to control the
flame of the lamp and the room temperature (Air Conditioner)
which is part of the smart house implementation. Problems
encountered in this research is the noise (noise) from the room
when recording, as well as pronunciation commands that must
be clear and normal within one second. In this research, the
best accuracy is 88,889% in non-realtime test and 70%
accurate on realtime test. The system is made can be said to be
good because the results obtained relatively good accuracy.
Keywords: speech reecognition, hidden markov model,
speech-to-text, digital signal processor , smart house.
Introduction
Speech recognition is a technology used to translate words
spoken by humans into writing. Systems that implement
speech recognition are able to understand the instructions
spoken by humans and act according to the instructions given.
In a smart house application, the application of speech
recognition is used as an interface for the user to control
household electronic appliance [1]. This paper review project
is a further development of the last task made by one of
college student where in the paper using euclidean distance
classification method, the best accuracy level obtained on the
system is 63.33%, the cause of accuracy is quite low at the
final task is because the euclidean distance method can not
classify the sound other than the sound that exist in the system
database. In this review paper project using the method of
classification of hidden markov model which is expected to
produce a higher level of accuracy. In the method of hidden
markov model, the HMM parameter is built for each
command to be detected. This parameter is built by training
each command model with 1 or more speakers. After these
HMM parameters are trained, the test sound data obtained will
be observed by the system and calculated the probability
estimates for each command model of the voice data. In this
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method of hidden markov model because each observed
sound is calculated its probability to every existing command
model, whereas the euclidean distance method only compares
the closest distance between the received voice data and the
existing command model database. Speech Recognition is an
advance technique to be followed in the fields of Automation,
Artificial Intelligence and soon. The improvement in the
recognition accuracy, robustness may cause effects on the
performance of ASR. Linear Predictive Coding (LPC), Mel
Frequency Cepstral Coefficients (MFCC) and Perceptual
Linear Prediction (PLP) are the various methods of feature
extraction techniques available for ASR. MFCC and PLP are
most widely used feature extraction techniques which are
required to reconstruct the original signal [2]. The recent
focus of researchers involved in the implementation of ASR
into an embedded platform [3]. The speech-to-text conversion
is a useful technique which is helpful for handicapped peoples
[4]. The goal of speech recognition area is to developed
technique and system to developed for speech input to
macine.based on major advanced in statically modeling of
speech ,automatic speech recognition today find widespread
application in task that require human machine interface such
as automatic call processing [19].
Methods
1. Proposed Methodology
In this paper will explain how the implementation of HMM
method in smart house, but before we entering into the main
topic on the implementation of HMM for smart house, first we
take look at described the methodology used in HMM. The
speech signal has to be in wave format, then only the signal
has to be processed and feature extraction can be done. The
speech signal is sampled at 16000 Hz and the number of bits
per sample as 32 [5]. The reasonable up into number of
frames according to the frame length. The Proposed
methodology consists of the feature extraction module and the
codebook generation. The proposed architecture has been
shown in Figure 1.
Acoustic Feature Codebook Speech to
Wave Extraction Generatiom text
nn
 International Journal of Faculty Electrical Engineering ISSN 0001-0002 Volume 1 Number 1 (2017) pp. 0000-0000
© Telkom University Publications. http://www.telkomuniversity.ac.id

DHMM
Modelling
gg

Figure 1: Block Diagram of Proposed Methodoly[6]

The original signal has to be split up into number of frames

according to the frame length. The low frequency signals are
selected by blocking the low frequency in every frame. After
frame blocking of each and every frames, Hamming window
is applied to reduce the discontinuity of the signal. Determine
the DCT coefficients for the evaluation of campestral
coefficients by using filter bank spacing to each and every
windowed frame. Apply logarithmic values to get the DCT
values as a single value [7].

2. Feature Extraction Techniques

The speech feature extraction in a categorization problem is
about reducing the dimensionality of the input vector while
maintaining the discriminating power of the signal. As we
know from fundamental formation of speaker identification
and verification system, that the number of training and test
vector needed for the classification problem grows with the
dimension of the given input so we need feature extraction of
speech signal. Following are some feature extraction has
explained below:

Tabel 1: List of technique with their properties For Feature

extraction

Feature Extraction: Figure 2 describes the block diagram of

feature extraction techniques. The steps that are followed
for the desired system has explained below.

Pre- Frame
Emphasize Blocking

FFT Hamming
Windowing

Warping Log
Filter Bank

Cepstral DCT
Coefficients

Figure 2: Feature Extraction Techniques [8]

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© Telkom University Publications. http://www.telkomuniversity.ac.id
 Wrapping Filter Bank
Generally done by using a filterbank. Filterbank is one form
of filter that is done with the aim to find out the size of a
particular frequency band in the sound signal, on MFCC used
Mel Filterbank using mel scale. The mel scale is linear at
frequencies below 1000 Hz, and logarithms at frequencies
above 1000 Hz. To find the mel scale, the equations are used :
(2)
The Mel Filter bank spacing has to be applied to the FFT
values to get the conversion for the frequency domain into the
Mel scale. Triangular band pass filters are applied as a filter
bank spacing which his non -uniformly spaced on the linear
frequency axis and it is uniformly spaced on the linear
frequency axis, with the larger number of filters in the low
frequency region and lesser number of filters in the high
frequency region and is shown in Figure 3:
Figure 3: Mel-Scale Filter Bank
The bark-scale is originally defined in Zwicker (1961). A
distance of 1 on the bark scale is known as a critical band. The
implementation provided in this function is described in
Traunmuller (1990). An approximate expression for the Bark
scale frequency warping, due to Schroedinger is used in these
proposed method [12].
Figure 4: Bark-Scale Frequency Filter Bank
 Logarithm of Energies
To compute the log-energy, i.e.,the logarithm of the sum of
filtered components for each filter.Equation (3) expresses the
computing logarithm of weighted sum of spectral values in the
filter-bank channel. At this stage, the number of rows equal to
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the number of columns equal to the number of filters in the
filter bank.
(3)
 Discrete Cosine Transform
The next step is to convert the log spectrum into a time
domain. This result is called mel frequency cepstrum
coefficient (MFCC). The cepstral repre- sentation of the
speech spectrum provides a good reprentation of the local
spectral properties of the signal for known frame analysis.
Because the coefficient of mel spectrum is a real number [13].
By converting it into time domain using discrete cosine
transform (DCT). The cepstral analysis includes the
conversion of spatial domain to frequencydomain by applying
DCT to the Mel Scale values. DCTexpresses a finite set of
data points in terms of a sum ofcosine functions. The
conversion of DCT is similar to the DFT in the conversion
process, DCT is more preferablesince the value obtained will
provide us the accuracy for increasing the robustness of ASR
[14]. Equation (3.5) expresses the DCT:
(4)
 The Cepstral Coefficients
Obtained by applying DCT to the Mel scale values. The
coefficients which areobtained after the evaluation of DCT
called MelFrequency Cepstral Coefficients (MFCC). When
thenumber of coefficients increases the accuracy and alsothe
increase in the complex of the designer complexity,there is a
lag in design of ASR system [15].
3. Speech Recognition
3.1. Literature Survey of Speech Recognition:
3.1.1. 1920-1960s:
The earliest attempts to devise systems for automatic speech
recognition by machine were made in 1950s, when various
researchers tried to exploit the fundamental ideas of acoustic
phonetics. During 1950s[20], most of the speech recognition
systems investigated spectral resonances during the vowel
region of each utterance which were extracted from output
signals of an analogue filter bank and logic circuits. In 1952,
at Bell laboratories, Davis, Biddulph, and Balashek built a
system for isolated digit recognition for a single speaker [21].
The system relied heavily on measuring spectral resonances
during the vowel region of each digit. In an independent effort
at RCA Laboratories in 1956, Olson and Belar tried to
recognize 10 distinct syllables of a single talker, as embodied
in 10 monosyllabic words [22]. The system again relied on
spectral measurements (as provided by an analog filter bank)
primarily during vowel regions. In 1959, at University
College in England, Fry and Denes tried to build a phoneme
recognizer to recognize four vowels and nine consonants [23].
3.1.2 1960-1970:
 International Journal of Faculty Electrical Engineering ISSN 0001-0002 Volume 1 Number 1 (2017) pp. 0000-0000
© Telkom University Publications. http://www.telkomuniversity.ac.id
In the 1960s several fundamental ideas in speech recognition
surfaced and were published. In the 1960s since computers
were still not fast enough, several special purpose hardware
were built [24]. However, the decade started with several
Japanese laboratories entering the recognition arena and
building special purpose hardware as part of their systems. On
early Japanese system, described by Suzuki and Nakata of the
Radio Research Lab in Tokyo, was a hardware vowel
recognizer [25]. An elaborate filter bank spectrum analyzer
was used along with logic that connected the outputs of each
channel of the spectrum analyzer (in a weighted manner) to a
vowel decision circuit, and majority decisions logic scheme
was used to choose the spoken vowel. Another hardware
effort in Japan was the work of Sakai and Doshita of kyoto
University in 1962, who built a hardware phoneme recognizer
[25]. A hardware speech segmented was used along with a
zero crossing analysis of different regions of the spoken input
to provide the recognition output. A third Japanese effort was
the digit recognizer hardware of Nagata and coworkers at
NEC Laboratories in 1963[26].
3.1.3. 1970-1980:
In the 1970s speech recognition research achieved a number
of significant milestones. First the area of isolated word or
discrete utterance recognition became a viable and usable
technology based on fundamental studies by Velichko and
Zagoruyko in Russia[27], Cakoe and Chiba in Japan[28], and
Itakura in the united States. The Russian studies helped the
advance use of pattern recognition ideas in speech
recognition; the Japanese research showed how dynamic
programming methods could be successfully applied; and
Itakura s research showed how the ideas of linear predictive
coding (LPC), which had already been successfully used in
low bit rate speech coding, could be extended to speech
recognition systems through the use of an appropriate distance
measure based on LPC spectral parameters[29]. .Another
milestone of the 1970s was the beginning of a longstanding,
highly successful group effort in large vocabulary speech
recognition at IBM in which researchers studied three distinct
tasks over a period of almost two decades, namely the New
Raleigh language [30] for simple database queries, the laser
patent text language [31] for transcribing laser patents, and the
office correspondent tasks called Tangora[32].
3.1.4. 1980-1990:
Just as isolated word recognition was a key focus of research
in the 1970s, the problems of connected word recognition was
a focus of research in the 1980s. Here the goal was to create a
robust system capable of recognizing a fluently spoken string
of words(eg., digits) based on matching a concatenated pattern
of individual words. Moshey J. Lasry has developed a
featurebased speech recognition system in the beginning of
1980. Wherein his studies speech spectrograms of letters and
digits[33].A wide variety of the algorithm based on matching
a concatenated pattern of individual words were formulated
and implemented, including the two level dynamic
programming approach of Sakoe at Nippon Electric
Corporation (NEC)[34],the one pass method of Bridle and
Brown at Joint Speech Research Unit(JSRU) in UK[35], the
level building approach of Myers and Rabiner at Bell Labs
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[36], and the frame synchronous level building approach of
Lee and Rabiner at Bell Labs[37]. most practical speech
recognition systems are based on the statistical frame work
developed in the 1980s and their results, with significant
additional improvements have been made in the 1990s like
HMM, Neural Net, and Darpa Programs:
a) Hidden Markov Model (HMM):
HMM is one of the key technologies developed in the 1980s,
is the hidden Markov model(HMM) approach [38]. It is a
doubly stochastic process which as an underlying stochastic
process that is not observable (hence the term hidden), but
canbe observed through another stochastic process that
produces a sequence of observations. Although the HMM was
well known and understood in a few laboratories (primarily
IBM, Institute for Defense Analysis (IDA) and Dragon
Systems), it was not until widespread publication of the
methods and theory of HMMs in the mid-1980s that the
technique became widely applied in virtually every speech
recognition research laboratory in the world. In the early
1970s, Lenny Baum of Princeton University invented a
mathematical approach to recognize speech called Hidden
Markov Modeling (HMM). The HMM pattern-matching
strategy was eventually adopted by each of the major
companies pursuing the commercialization of speech
recognition technology (SRT).The U.S. Department of
Defense sponsored many practical research projects during the
70s that involved several contractors, including IBM, Dragon,
AT&T, Philips and others. Progress was slow in those early
years.
b) Neural Net:
Another new technology that was reintroduced in the late
1980s was the idea of applying neural networks to problems
in speech recognition. Neural networks were first introduced
in the 1950s, but they did not prove useful initially because
they had many practical problems. In the 1980s however, a
deeper understanding of the strengths and limitations of the
technology was achieved, as well as, understanding of the
technology to classical signal classification methods. Several
new ways of implementing systems were also proposed [39].
c) DARPA Program:
Finally, the 1980s was a decade in which a major impetus was
given to large vocabulary, continuous speech recognition
systems by the Defense Advanced Research Projects Agency
(DARPA) community, which sponsored a large research
program aimed at achieving high word accuracy for a 1000
word continuous speech recognition, database management
task.
3.1.5. 2000-2009:
A) General:
 International Journal of Faculty Electrical Engineering ISSN 0001-0002 Volume 1 Number 1 (2017) pp. 0000-0000
© Telkom University Publications. http://www.telkomuniversity.ac.id

Around 2000, Bayesian estimates (VB) were varied and 3.2. The Process of Classification of Voice Recognition
pooling techniques developed [40]. Unlike the maximum Speech recognition is a process by which the computer can
Perhaps, this VB approach is based on the back the identify the spoken words. The identification process is
required for the system to recognize a voice input so that it
distribution parameters Giuseppe Richardi [41] have develop
can be utilized. Voice recognition results can be used to
techniques to solve adaptive problems learning, in the perform various tasks such as controlling the machine,
introduction of automatic speech and even proposed Active accessing the database, and producing text [16].
learning algorithm for ASR. In 2005, several Enhancements
have been made in the large Vocabulary Continuous Speech
Recognition. Performance System repair. In 2007, the
difference in acoustic functionality between spontaneous
speech and large-scale reading the greeting database, CSJ was
analyzed. Sadaoki Furui studied SR methods that can adapt to
the word variations using a large number of trained models
based on pooling techniques In 2008, the authors explored
application of the conditional random field (CRF) to combine
local post estimates given by multilayer perceptions based on Figure 8: HMM Model left-right
the frame rate forecast of the phone and the classes are
In this calculations, HMM uses several notations, including:
phonologically linked. De-wachter et.al. , trying to cope with
time dependence, deep problems voice recognition using a
1. N is represents the number of hidden states (state to 1,
direct model corresponding method Xinwei Li et.al., he
2, ..., n) in HMM. All the states in the model are connected to
proposed a new one optimization method is semi-final
each other.
programming (SDP) a solving the problem of large margin
estimate (LME) HMM continuous density (CDHMM) in
2. M is the probability of appearing observed variables.
voice recognition. The training dissociates the acoustic model
3. T is the length of the observation circuit
to speak the recognition is offered under Massimo reciprocal
information (MMI). Around 2007, Rajesh M. Hegde et al.,
4. π represents the matrix π or π i with π i = P (q 1 = 1) ie the
proposed an alternative method for processing Fourier
probability distribution matrix at the initial stage resides in
transform for speech extraction function, which group delay
state i. In this case, Σ N i = 1 π i = 1.
process (GDF) that can be directed calculated by voice signal.

5. A is a transition opportunity between states. A = {a ij}

B) The DARPA Program:
where a ij = P (q t + 1 = j│q t = i) is the probability of being in
Affordable Accessible Speech (EARS) This program is made
state j at time t + 1 if at time t is in state i. in this case, a ij is
to develop speech-to-text (automatic transcription technology)
assumed to be free of time.
with the goal of achieving much richer and far more accurate
than output before. Activities include detection of phrase
6. B is the probability of occurrence of variables observed in a
limits, fillers, and fluids. The program focuses on nature,
state. B = {b j (k)} with b j (k) = P (v k when t if the state that
human speech that is not limited by broadcasts and foreigners
occurs is j.
voice conversations in various languages. The goal is to
allowing the machine to do a much better job detect, extract,
7. O represents the notation for observed values at time t, so
summarize and translate what counts information, thus
the observable symbol is O = O1, O2, O3, ..., O T with T
allowing humans to understand what they are She says being the length of the observation.
reading the transcript instead of listening to the audio signal.

C) Spontaneous voice recognition: 6. Design and Implementation

Despite reading speeches and similar types of speech, eg.
News Shipped to read the text can be accurately recognized
over 95% using state-of-the-art voice recognition technology
and recognition accuracy decreased drastically spontaneous
discourse Expanding voice application Recognition is very
important to increase recognition performance for
spontaneous speech.
In general, the purpose of designing the system made is to
send voice commands that have been processed in matlab to
the microcontroller system.
6.1. System Diagram Block
The system created in this final project can be described in
the following block diagram on figure 9:

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© Telkom University Publications. http://www.telkomuniversity.ac.id

Figure 10: Hardware Design

Notes:
1. Microphone Voice input device from the speaker / user is
then forwarded by interconnect from microphone to line in
laptop.
2. Interconnect line out laptop to DTMF Decode Is a media
conductor DTMF signal from line out laptop to block DTMF
Decoder.
3. DTMF Block Decoder This block translates the received
DTMF signal from the laptop into a 4 bit binary number.
4. ATMega 16 Minimum System This block processes 4 bit
data from the DTMF decoder and defines it as one of the
commands in the database and displays it on the LCD Display.
5. LCD Display Device to display the results of commands
defined by ATMega 16.
6. AC Adapter to DC 5V Device that produces 5 V output
from 220 V AC input as system power supply
Figure 9: Block Diagram System
7. Conclusion:
Speech is the primary, and the most convenient means of
6.2. Software Design
communication between people. Whether due to technological
Implementation of feature extraction methods and
curiosity to build machines that mimic humans or desire to
classification is done by using MATLAB. The system is
automate work with machines, research in speech and speaker
divided into 2 processes namely the process of training and
recognition, as a first step toward natural human-machine
test process. The training process aims to generate a database
communication, has attracted much enthusiasm over the past
of HMM parameters from the MFCC characteristic
five decades. we have also encountered a number of practical
coefficients to be used in the test process. The test process
limitations which hinder a widespread deployment of
works by using the HMM parameters that have been
application and services. In most speech recognition tasks,
generated by the training process to determine the word model
human subjects produce one to two orders of magnitude less
that best matches the MFCC characteristic coefficients of the
errors than machines. Conclusion Based on the review
test data.
implementation and testing of speech recognition system
using HMM methods, there are several things that can be
6.3. Test
concluded, among others. The speech signal can be processed
Test In this option there are two test methods are: realtime and
and that can be trained and compared with the feature vectors
non-realtime. The non-realtime test tests the previously
that are obtained by processing the speech.
recorded test data, and the realtime test tests the data recorded
at the time
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