International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 2 Issue: 8

ISSN: 2321-8169
2363 2367

_______________________________________________________________________________________________

Study of Speaker Verification Methods

Sneha M.Powar

Dr. V.V.Patil

Electronics & Telecommunication Department

Dr. J. J. Magdum College of Engineering, Jaysingpur
Kolhapur, India
Email: [email protected]

H.O.D. Electronics Department

Dr. J. J. Magdum College of Engineering
Jysingpur, India
Email: [email protected]

Abstract Speaker verification is a process to accept or reject the identity claim of a speaker by comparing a set of measurements of the
speakers utterances with a reference set of measurements of the utterance of the person whose identity is claimed.. In speaker verification, a
person makes an identity claim. There are two main stages in this technique, feature extraction and feature matching. Feature extraction is the
process in which we extract some useful data which can later to be used to represent the speaker. Feature matching involves identification of the
unknown speaker by comparing the feature extracted from the voice with the enrolled voices of known speakers.
Keywords- speaker verification, text dependent, text independent

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I.

INTRODUCTION

In our everyday lives there are many forms of communication,

for instance: body Language, textual language, pictorial
language and speech, etc. However amongst those forms
speech is always regarded as the most powerful form of
communication. From the signal processing point of view,
speech can be characterized in terms of the signal carrying
message information. The waveform could be one of the
representations of speech, and this kind of signal has been
most useful in practical applications. Extracting from speech
signal, we could get three main kinds of information: Speech
Text, Language, and Speaker Identity. Speech recognition
refers to the ability of a machine or program to recognize or
identify spoken words and carry out voice. The spoken words
are digitized into sequence of numbers, and matched against
coded dictionaries so as to identify the words. Speaker
recognition maybe defined as the process of recognizing a
person automatically using the information extracted from
speech signal of the person. This technique uses the voice of
the speaker to verify their identity to access to several services
such as accessing the computer or server from remote place,
voice dialing, accessing security services, mobile banking etc.
where security is the primary concern.[6]
II.

SPEAKER INFORMATION IN SPEECH SIGNAL

Speech is human beings primary means of communication,

and it contents essentially the meaning of information from a
speaker to a hearer, individual information representing
speakers identity and gender, and also sometimes the
emotions. In a speech production the properties of both
articulators, which produce the sound, and auditory organs,
which perceive the sound, should be involved in Speech
production can be divided into three principal components:
excitation production, vocal tract articulation, and lips' and/or

nostrils' radiation. Excitation powers the speech production

process. It is produced by the airflow from lungs, and then
carried by trachea through the vocal folds, during inspiration,
air is filled into lungs, and during expiration the energy will be
spontaneously released. The trachea conveys the resulting
provider to serve inputs to the vocal tract, and the volume of
air determines the amplitude of the sound. The vocal tract
works as a filter to shape the excitation sources. The
uniqueness of speaker voice not only depends on the physical
features of the vocal tract, but the speakers mental ability to
control the muscles of the organs in the vocal tract. It is not
easy for speaker to change the physical features intentionally.
However, these physical features are possible to be changed
with ageing. Speaker characteristics in the speech signal are
often difficult to carry out. Segmenting, labeling, and
measuring specific segmental speech events that characterize
speakers, such as nasalized speech sounds, is difficult because
of variable speech behavior and variable and distorted
recording and transmission conditions. Overall qualities, such
as breathiness, are difficult to correlate with specific speech
signal measurements and are subject to variability in the same
way as segmental speech events. The most important analysis
tool is short-time spectral analysis. It is no coincidence that
short-time spectral analysis also forms the basis for most
speech recognition systems. Short-time spectral analysis not
only resolves the characteristics that differentiate one speech
sound from another, but also many of the characteristics
already mentioned that differentiate one speaker from another.
There are two principal modes of short-time spectral analysis:
filter bank analysis and linear predictive coding (LPC)
analysis. In filter bank analysis, the speech signal is passed
through a bank of band pass filters covering the available
range of frequencies associated with the signal. Typically, this
range is 200 to 3,000 Hz for telephone band speech and 50 to
8,000 Hz for wide band speech. A typical filter bank for wide
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International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 2 Issue: 8

ISSN: 2321-8169
2363 2367

_______________________________________________________________________________________________
band speech contains 16 band pass filters spaced uniformly
500 Hz apart. The output of each filter is usually implemented
as a windowed, short-time Fourier transform [using fast
Fourier transform (FFT) techniques] at the center frequency of
the filter. LPC-based spectral analysis is widely used for
speech and speaker recognition. The LPC model of the speech
air stream to the larynx. Larynx refers as an energy signal
specifies that a speech sample at time t ,s .t/, can be
represented as a linear sum of the p previous samples plus an
excitation term, as follows:
S(t)= a1s(t 1) +a2s(t 2)+.ap(t p)+ G u(t)
The LPC coefficients ai are computed by solving a set of linear
equations resulting from the minimization of the mean-squared
error between the signal at time t and the linearly predicted
estimate of the signal. Two generally used methods for solving
the equations, the autocorrelation method and the covariance
method. [4]
III.

SPEAKER RECOGNITION

A. Speaker Identification
Speaker identification (SI) is the process of finding the identity
of an unknown speaker by comparing his/her voice with
voices of registered speakers in the database. Its a one-tomany comparison. The basic structure of SI system (SIS) is
shown in Figure. We notice that the core components in SIS
are the same as in SVS. In SIS, M speaker models are scored
in parallel and the most-likely one is reported. The core
components in SIS are the same as in SVS. In SIS, M speaker
models are scored in parallel and the most-likely one is
reported, and consequently decision will be one of the
speakers ID in the database, or will be none of the above if
and only if the matching score is below some threshold and
its in the case of a open-set SIS.

Different terms which have the same definition as SV could be

found in literature, such as voice verification, voice
authentication,
speaker/talker
authentication,
talker
verification. It performs a one-to-one comparison (it is also
called binary decision) between the features of an input voice
and those of the claimed voice that is registered in the system.
Three main components shown in this structure are: Front-end
Processing, Speaker Modeling, and Pattern Matching. To get
the feature vectors of incoming voice, front-end processing
will be performed, and then depending on the models used in
Pattern Matching, match scores will be calculated. If the score
is larger than a certain threshold, then as a result, claimed
speaker would be acknowledged. There are three main
components: Front-end Processing, Speaker Modeling, and
Pattern Matching. Front-end processing is used to highlight the
relevant features and remove the irrelevant ones. After the first
component, we will get the feature vectors of the speech
signal. Pattern Matching between the claimed speaker model
registered in the database and the imposter model will be
performed then, if the match is above a certain threshold, the

Fig. 2 Basic Structure of Speaker Verification

Identity claim is verified. Using a high threshold, system gets
high safety and prevents impostors to be accepted, but in the
mean while it also takes the risk of rejecting the genuine
person, and vice versa. [6]
IV.

METHODS OF SPEAKER VERIFICATION

Speaker verification systems typically operate in one of two

input modes: text dependent or text independent. In the textdependent mode, speakers must provide utterances of the same
text for both training and recognition trials. In the textindependent mode, speakers are not constrained to provide
specific texts in recognition trials. Since the text-dependent
mode can directly exploit the voice individuality associated
with each phoneme or syllable, it generally achieves higher
recognition performance than the text-independent mode.
A. Text-Dependent (Fixed Passwords)
Fig 1. Basic structure of Speaker Identification
B. Speaker verification
Speaker verification (SV) is the process of determining
whether the speaker identity is who the person claims to be.

The structure of a system using fixed passwords is rather

simple; input speech is time aligned with reference templates
or models created by using training utterances for the
passwords. If the fixed passwords are different from speaker to
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_______________________________________________________________________________________

International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 2 Issue: 8

ISSN: 2321-8169
2363 2367

_______________________________________________________________________________________________
speaker, the difference can also be used as additional
individual information. This helps to increase performance.
The most common approach to automatic speaker recognition
in the text-dependent mode uses representations that preserve
temporal characteristics. Each speaker is represented by a
sequence of feature vectors (generally, short-term spectral
feature vectors), analyzed for each test word or phrase. This
approach is usually based on template matching techniques in
which the time axes of an input speech sample and each
reference template of registered speakers are aligned, and the
similarity between them accumulated from the beginning to
the end of the utterance is calculated. Trial-to-trial timing
variations of utterances of the same talker, both local and
overall, can be normalized by aligning the analyzed feature
vector sequence of a test utterance to the template feature
vector sequence using a dynamic programming (DP) time
warping algorithm or DTW. Since the sequence of phonetic
events is the same for training and testing, there is an overall
similarity among these sequences of feature vectors. Ideally
the intra-speaker differences are significantly smaller than the
inter-speaker differences.
B. Text Independent (No Specified Passwords)
There are several applications in which predetermined
passwords cannot be used. In addition, human beings can
recognize speakers irrespective of the content of the utterance.
Therefore, text-independent methods have recently been
actively investigated. Another advantage of text-independent
recognition is that it can be done sequentially, until a desired
significance level is reached, without the annoyance of having
to repeat passwords again and again. In a text-independent
system, the words or phrases used in recognition trials
generally cannot be predicted. Therefore, it is impossible to
model or match speech events at the level of words or phrases.
Classical text-independent speaker recognition techniques are
based on measurements for which the time dimension is
collapsed. Recently text-independent speaker verification
techniques based on short duration speech events have been
studied. The new approaches extract and measure salient
acoustic and phonetic events. The bases for these approaches
lie in statistical techniques for extracting and modeling
reduced sets of optimally representative feature vectors or
feature vector sequences or segments. These techniques fall
under the related categories of vector quantization (VQ),
matrix and segment quantization, probabilistic mixture
models, and HMM.
A set of short-term training feature vectors of a speaker can be
used directly to represent the essential characteristics of that
speaker. However, such a direct representation is impractical
when the number of training vectors is large, since the
memory and amount of computation required become
prohibitively large. Therefore, efficient ways of compressing

the training data have been tried using VQ techniques. In this

method, VQ codebooks consisting of a small number of
representative feature vectors are used as an efficient means of
characterizing speaker-specific feature. A speaker specific
codebook is generated by clustering the training feature
vectors of each speaker. In the recognition stage, an input
utterance is vector-quantized using the codebook of each
reference speaker, and the VQ distortion accumulated over the
entire input utterance is used in making the recognition
decision.
A five-state ergodic linear predictive HMM is used for broad
phonetic categorization. After identifying frames belonging to
particular phonetic categories, feature selection is performed.
In the training phase, reference templates are generated and
verification thresholds are computed for each phonetic
category. In the verification phase, after the phonetic
categorization, a comparison with the reference template for
each particular category provides a verification score for that
category. The final verification score is a weighted linear
combination of the scores for each category. The weights are
chosen to reflect the effectiveness of particular categories of
phonemes in discriminating between speakers and are adjusted
to maximize the verification performance. The performances
of speaker recognition based on a VQ-based method and that
using discrete/continuous ergodic HMM-based methods have
been compared, in particular from the viewpoint of robustness
against utterance variations. It was shown that a continuous
ergodic HMM method is far superior to a discrete ergodic
HMM method, and that a continuous ergodic HMM method is
as robust as a VQ-based method when enough training data is
available. However, when little data is available, the VQ-based
method is more robust than a continuous HMM method.[6]
V.

APPLICATION

A. Secure Access via telephone

The most straightforward way to employ SV is in the cases
when one has to gain access to some secure place via
telephone. Voice is completely compatible with the existing
transmission protocols via telephone channels; therefore no
special adaptations of the system (besides the installment of a
SV system) are necessary.
B. Home Banking
Home banking is another application where SV can be
applied. For the time being such a service is restricted to
operations within the accounts maintained by a single
individual. One can for e.g. check the status of their account,
transfer money between ones own saving accounts, etc. The
security is pretty low in these cases, the users are verified only
by saying their PIN and FR almost never occurs (after all who
wants to play \robber" with his own savings!). Still, however,
it is being researched how secure it could be to use SV for
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International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 2 Issue: 8

ISSN: 2321-8169
2363 2367

_______________________________________________________________________________________________
transactions including a second and third party (i.e. the so
called high-risk bank transactions). It is always noted that the
security measures should be proportional to the value that
could be obtained by this service.

[3]

[4]

C. Home shopping
Home shopping (see for e.g. http://www.hsn.com) is the
service that is most uninteresting to an imposter. SV is here
being employed, though backed up by a human operator. In
this service people ring to order products that are later on
shipped to their home addresses. In cases when all lines are
busy, a customer can always choose to use the automatic
service. They just have to speak their telephone number and if
their identity is successfully verified they can start ordering
products. If they are rejected, they are redirected to a human
operator. But even if their identity is mistaken for someone
else and some products are sending to another customer, there
is no harm because these products cannot go to an
unauthorized party (i.e. a criminal).

[5]

[6]

[7]

Higgins, A.L., Bahler, L.andPorter, J., Speaker

verification using randomized phrase prompting,
Digital Signal Processing, 1, 89106, 1991.
Rabiner, L.R. and Juang, B.-H., Fundamentals of
Speech Recognition,Prentice-Hall
Englewood
Cliffs, NJ, 1993.
Rosenberg, A.E., Lee, C.-H. and Gokcen, S.,
Connected word talker verification using whole word
hidden Markov models, Proc. IEEE Intl. Conf.
Acoust., Speech, Signal Processing, Toronto, 381
384, 1991.
Technical University of Denmark Informatics and
Mathematical Modelling Master thesis by Ling Feng
IMM-THESIS: ISSN 1601-233X
R. A. Cole and colleagues, Survey of the State of the
Art in Human Language Technology, National
Science Foundation European Commission.
BIOGRAPHY

D. Forensics and Survelliance

Detection of speakers in forensic cases boils down, in most
situations, to deciding whether a given recording is really from
a suspect or not. This is exactly the case of the Hypothesis
Test. Leaving aside legal issues SV can help police discover
how many different individuals are involved in a conversation
on a tape.
VI.

DISCUSSION

The challenges for implementing practical and uniformly

reliable systems for speaker verification, are rooted in
problems associated with variability and insufficient data. As
described earlier, variability is associated with trial-to-trial
variations in recording and transmission conditions and
speaking behavior. The most serious variations occur between
enrollment sessions and subsequent test sessions resulting in
models that are mismatched to test conditions. Most
applications require reliable system operation under a variety
of environmental and channel conditions and require that
variations in speaking behavior will be tolerated. Insufficient
data refers to the unavailability of sufficient amounts of data to
provide representative models and accurate decision
thresholds.
[1]

[2]

Miss S.M.Powar received the BE degree in 2011 in Electronics from

Bharti Vidyapeeths College Of Engg Kolhapur & persuing ME
Electronics & Communication in Dr. J.J.Magdum College of Engg.
Jaysingpur.
Dr.Mrs.V.V.Patil received the BE degree in1994 & ME degree in
2004 in Electronics Engg. Dept. of Walchand College of Engg. sangli
& PHD degree from Electrical Engg. Dept. of I.I.T Bombay in 2014.
She is currently professor & Head in Electronics Engg. Dept. of Dr.
J.J.Magdum College of Engg. Jaysingpur.
Her research interests are in the area of speech & signal processing
applications.

REFERENCES
Matsui, T. and Furui, S., Concatenated phoneme
models for text-variable speaker recognition, Proc.
IEEE Intl. Conf. Acoust., Speech, Signal Processing,
II, 391394, 1992.
Matsui, T. and Furui, S., Speaker adaptation of tiedmixture-based phoneme models for text prompted
speaker recognition, Proc. IEEE Intl. Conf. Acoust.,
Speech, Signal Processing, I, 125128, 1994
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