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  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Using Vector Quantization for Universal Background Model in Automatic Speaker Verification</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Badji Mokhtar University Annaba Algeria</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff2">2</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Computer Science Department</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff2">2</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>LRS Laboratory</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff2">2</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>LRI Laboratory</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff2">2</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>, Laskri Mohamed Tayeb</institution>
        </aff>
        <aff id="aff1">
          <label>1</label>
          <institution>Badji Mokhtar University</institution>
          ,
          <addr-line>P-O Box 12, 23000 Annaba</addr-line>
          ,
          <country country="DZ">Algeria</country>
        </aff>
        <aff id="aff2">
          <label>2</label>
          <institution>Djellali Hayet</institution>
        </aff>
      </contrib-group>
      <pub-date>
        <year>2012</year>
      </pub-date>
      <fpage>112</fpage>
      <lpage>120</lpage>
      <abstract>
        <p>We aim to describe different approaches for vector quantization in Automatic Speaker Verification. We designed our novel architecture based on multiples codebook representing the speakers and the impostor model called universal background model and compared it to another vector quantization approach used for reducing training data. We compared our scheme with the baseline system, Gaussian Mixtures Models and Maximum a Posteriori Adaptation. The present study demonstrates that the multiples codebook gives more verification accuracy called equal error rate but this improvement also depends on the codebook size.</p>
      </abstract>
      <kwd-group>
        <kwd>Vector Quantization</kwd>
        <kwd>Speaker Verification</kwd>
        <kwd>Codebook</kwd>
        <kwd>false Acceptance</kwd>
        <kwd>False reject</kwd>
        <kwd>Universal Background Models</kwd>
        <kwd>Linde Buzo Gray</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1 Introduction</title>
      <p>The speaker verification is a field of speaker recognition which the main objective is
to authenticate a person’s claimed identity. The speaker voice is used to recognize
him (her), we create two models, the first one is the speaker model and the second is
the impostor model called universal background model UBM. The recorded speech is
preprocessed, compared to speaker and UBM model in order to compute the score
and finally compared to threshold.</p>
      <p>
        It has been proved that the variation factors like speaker identity, utterance length,
gender, session, transmission channel, speaking, affect the system performance
[
        <xref ref-type="bibr" rid="ref1">1</xref>
        ][
        <xref ref-type="bibr" rid="ref2">2</xref>
        ][
        <xref ref-type="bibr" rid="ref3">3</xref>
        ]. Intra speaker variability influences the verification performance system.
Thus, it is important to record each speaker at different time but also means the huge
speech data.
      </p>
      <p>The state of the art of text independent speaker recognition is Gaussian mixture model
and Maximum a posteriori adaptation. Speaker dependent GMM are derived from the
speaker independent model called universal background model (UBM) and Maximum
a posteriori adaptation MAP using target speaker speech data.</p>
      <p>
        Vector Quantization (VQ) model was introduced in 1980’s used in data compression
[
        <xref ref-type="bibr" rid="ref4">4</xref>
        ]. VQ is one of the simplest text independent speakers model, and often used for
computational technique. It also provides good accuracy when combined with
background model adaptation [
        <xref ref-type="bibr" rid="ref4">4</xref>
        ][
        <xref ref-type="bibr" rid="ref5">5</xref>
        ].
In VQ based speaker recognition, each speaker is characterized with the set of code
vectors and is referred to as that speaker’s codebook. Normally, a speaker’s codebook
is trained to minimize the quantization error for the training data from that speaker.
The most commonly used training algorithm is the Linde-Buzo-Gray (LBG)
algorithm [
        <xref ref-type="bibr" rid="ref6">6</xref>
        ].
      </p>
      <p>
        When the speaker speech data becomes huge, it involves the time consuming
problem. Gurmeet replaced the EM algorithm with LBG algorithm. Experimentally,
they found that the complexity of calculation can be reduced by 50% compared to the
EM algorithm. The reason is the LBG algorithm utilize apart of feature vectors for
classification [
        <xref ref-type="bibr" rid="ref7">7</xref>
        ].
      </p>
      <p>We applied Vector Quantization in Automatic Speaker Verification; usually, each
target speaker had his own codebook, when usually the speaker independent models
had two gender dependent codebook originates from impostor speakers (male,
female).</p>
      <p>Our approach aim to select the best universal background model UBM, we try another
way to model VQ UBM with set of sub UBM. We divide the features vectors
extracted from processing step (Mel cepstral coefficients: MFCC) in a equal size and
applied for each of them the LBG algorithm to obtain its codebook (cd1,cd2,…cdK)..
The aim is to get the best sub model with LBG algorithm for impostors (UBM) and
then compute the distortion error from optimal Sub UBM. We aim to reduce EER in
the presence of small training data of each client and select the best sub UBM.
We organized paper as follows, modeling speakers based on vector quantization and
MAP adaptation is introduced in Section 2, and the ASV architecture proposed in
Section 3 followed experiments in Section 4 and conclusion in section 5.</p>
    </sec>
    <sec id="sec-2">
      <title>2 Vector Quantization and MAP Adaptation</title>
      <p>We introduce vector quantization and Maximum a posteriori adaptation in Automatic
Speaker Verification:</p>
      <sec id="sec-2-1">
        <title>2.1 Vector Quantization</title>
        <p>
          Vector Quantization (VQ) is a pattern classification technique applied to speech data
to form a representative set of speaker features. It was introduced to speaker
recognition by Soong [
          <xref ref-type="bibr" rid="ref8">8</xref>
          ]. In speaker verification, Vector quantization (VQ) model
were applied in Soong and Rosenberg, It is one of the simplest text-independent
speaker models and usually used for computational speed-up techniques, it also
provides competitive accuracy when combined with background model adaptation
[
          <xref ref-type="bibr" rid="ref5">5</xref>
          ][
          <xref ref-type="bibr" rid="ref8">8</xref>
          ][
          <xref ref-type="bibr" rid="ref9">9</xref>
          ][
          <xref ref-type="bibr" rid="ref10">10</xref>
          ].
        </p>
        <p>
          In the training phase, a speaker-specific VQ codebook is generated for each known
speaker by clustering his training acoustic vectors. The distance from a vector to the
closet codeword of a codebook is called a VQ distortion [
          <xref ref-type="bibr" rid="ref4">4</xref>
          ][
          <xref ref-type="bibr" rid="ref11">11</xref>
          ].
        </p>
        <p>In the Test phase, an input utterance of a known voice is vector-quantized using
trained codebook from proclaimed identity and the speaker independent model
codebook (Universal Background Model). The total VQ distortion is computed.</p>
        <p>In principle, when we get a large amount of training vectors representing speaker
in the training vectors. We should reduce it by vector quantization. Suppose there are
N vectors, to be quantized, the average quantization error is given by
(1)</p>
        <p>
          The task of designing a codebook is to find a set of code vectors so that E is
minimized. However, the commonly used method is the LBG algorithm [
          <xref ref-type="bibr" rid="ref6">6</xref>
          ].
        </p>
        <p>In speaker verification, the codebook is used for classification and minimizing the
quantization error. We selected LBG algorithm defined as the iterative improvement
algorithm or the generalized Lloyd algorithm. Given a set of N training feature
vectors, {t1, t2,.tn} characterizing the variability of a speaker, we search a
partitioning of the feature vector space, {S1, S2,..., SM}, for that particular speaker
where S, the whole feature space, is represented as S = S1 U S2 U...U SM.</p>
        <p>The performance of a quantizer is designed by an average distortion between the
input vectors and the final vectors, where E represents the expectation operator
(equation 1).</p>
      </sec>
      <sec id="sec-2-2">
        <title>2.2 Gaussian Mixture Models &amp; MAP Adaptation</title>
        <p>GMM-UBM-Maximum Likelihood Modeling: this approach is based on training
UBM male model with Gaussian mixture model and the other female UBM (from
female speech). The model parameters (mean, covariance and weight of the Gaussian)
are trained with the EM algorithm (Expectation-Maximization).</p>
        <p>
          Maximum a Posteriori approach MAP resolve the problem of maximum likelihood
ML(can’t generalize well to unseen speech data in low training data). MAP use prior
knowledge of the distribution of the model parameters and insert it in modeling
process[
          <xref ref-type="bibr" rid="ref12">12</xref>
          ][
          <xref ref-type="bibr" rid="ref13">13</xref>
          ]. The Maximum A Posteriori MAP approach is to use the world model
and client training data to estimate the client model on the basis of these data and
MAP Adaptation [
          <xref ref-type="bibr" rid="ref12">12</xref>
          ][
          <xref ref-type="bibr" rid="ref13">13</xref>
          ] [
          <xref ref-type="bibr" rid="ref14">14</xref>
          ][
          <xref ref-type="bibr" rid="ref15">15</xref>
          ][
          <xref ref-type="bibr" rid="ref16">16</xref>
          ].
        </p>
        <p>
          The client model is derived from the world model by adapting the GMM
parameters (mean, covariance, weights) estimated. However, experimentally, only the
averages of GMM are adapted [
          <xref ref-type="bibr" rid="ref13">13</xref>
          ].
        </p>
      </sec>
    </sec>
    <sec id="sec-3">
      <title>3 Speaker Verification Architecture Based on Vector Quantization</title>
      <p>We proposed two VQ-UBM models, the first one is the baseline system, the second is
VQ Sub-UBM. We describe our new modeling UBM:</p>
      <sec id="sec-3-1">
        <title>3.1 Training Phase</title>
        <p>VQ Sub UBM
The acoustics vectors obtained in features extraction were split in subset of data with
the same dimension and served to create codebook {CDU1, CDU2, …, CDUk} for
world model UBM. We divide UBM speech data in N subsets instead of one global
UBM, in figure 1, after feature extraction, the MFCC vectors were the input of L.B.G
algorithm which provide K codebook.</p>
        <p>Codebook
There are several different approaches to finding an optimal codebook. The idea is to
begin with a vector quantizer and a codebook and improve upon the initial codebook
by iterating until the best codebook is found. We aim to reduce redundancy in UBM
data by clustering, to do that, we implement this algorithm:
Algorithm 1: VQ Sub-UBM</p>
        <sec id="sec-3-1-1">
          <title>Training Phase</title>
          <p>Input : MFCC vectors; Output: Codebook CDU(1..M).
We divide MFCC vector in equal sub matrix and applied
LBG algorithm for each of them.</p>
          <p>Input[ C ]= MFCC vector(Feature Extraction).</p>
          <p>Split C in M equal sub matrix Ci;
Train UBM of each Ci for different size of
codebook(k=16,32,64,128,256); Result= CDU (i=1..M).</p>
        </sec>
        <sec id="sec-3-1-2">
          <title>Test Phase</title>
          <p>In recognition phase, we compute Euclidean distance and
evaluate quantization error from each codebook and test
vector,
We choose the best codebook with minimal quantization
error.</p>
          <p>The quantization square error ESQ
(2)
Where</p>
          <p>; Xi :vector data; yK : centroid</p>
          <p>Fig 1. Vector Quantization Architecture: VQ Sub-UBM is applied for UBM only</p>
        </sec>
      </sec>
      <sec id="sec-3-2">
        <title>3.2 Test Phase</title>
        <p>We compute the threshold (CDT) from 8 male and 8 females’ speakers others than
UBM speakers and trained by LBG algorithm.</p>
        <p>9 Test Algorithm
CDU: UBM codebook; CDS: Speaker codebook;
VQdist : VQ distorsion
Input : X=speaker speech ,claimed identity,</p>
        <p>MFCC= Feature Extraction(X)</p>
        <p>For i=1 to M VQcdu=VQdistorsion(X,CDUi) End
CDUoptimal=CDU best cobdebook UBM where
Argmin(VQdistorsion(X,CDUi))
VQdist(speaker) = VQdist(X,CDS) – Vqdist(X,CDUoptimal)
If</p>
        <p>VQdist(speaker)&gt; VQdist(CDT) then client acces
Else</p>
        <p>reject</p>
      </sec>
    </sec>
    <sec id="sec-4">
      <title>4 Protocol Experiment</title>
      <p>In this section, we describe a set of experiments designed to evaluate the performance
of the proposed system under a variety of condition and compare it to baseline system
GMM MAP and standard VQ UBM.</p>
      <sec id="sec-4-1">
        <title>4.1 Database and Baseline System</title>
        <p>The Arabic database is recorded in Goldwave frequency 16KHz for a period of 60s
for each speaker when training and 30s in the testing phase. The UBM population is
15 men's and 15 women. Four sessions are recorded for each speaker at an interval of
1 month. Ten clients are registered in the database (5 men and 5 women).</p>
      </sec>
      <sec id="sec-4-2">
        <title>4.2 VQ Sub-UBM Model</title>
        <p>We extract MFCC vector for all acoustics data allowed to UBM training and applied
LBG algorithm for it. We obtain one centroid (NxT) by gender, where we try
different value of N=k=16, 32, 64, 128, 256. In recognition phase, we compute
Euclidean distance and evaluate quantization error (equation 1) from centroid and test
vector, we computed codebook for each target speaker and finally evaluate the score.</p>
      </sec>
      <sec id="sec-4-3">
        <title>4.3 Baseline VQ UBM Model</title>
        <p>We compute one codebook for the Baseline VQ UBM and evaluate LBG algorithm
for k=16, 32, 64, 128. We built UBM models from 30 Arabic speakers; UBM male
with 15 male speakers and UBM female from 15 female speakers. The global
threshold is computed from other database: 8 male and 8 female speakers.
CodeBook Size
CD32
CD64</p>
      </sec>
      <sec id="sec-4-4">
        <title>4.4 Baseline GMM MAP system</title>
        <p>We train universal background model UBM gender dependent(male, female) under
expectation maximization algorithm EM and create each target speaker model with
GMM MAP approach, we try different sizes of GMM (8, 16, 32,64,128) and evaluate
the value of false acceptance and false rejection.</p>
        <p>Fig 2. Comparaion of VQ Baseline and VQ SUB UBM</p>
      </sec>
    </sec>
    <sec id="sec-5">
      <title>5 Discussions</title>
      <p>We compare different modeling speaker techniques: VQ Sub-UBM, Baseline
VQUBM and GMM MAP their performances were evaluated using the same data and
front end processing.</p>
      <p>Table I shows the value of false acceptance and false rejection for different
codebook size (32, .., 256) in VQ SUB UBM approach and observe that the best value
is designed for 128 codebook size (7.14% and 22.73%). The result in table 1 provide
more accuracy recognition than table II for codebook size=32(FA=22.86%;
FR=23.86%) and worst for codebook size 64. We observe that the size of codebook
influences the performance and the multiple UBM provide better result.</p>
      <p>Figure 2 demonstrate the performance of VQ SUB UBM is worst than VQ UBM
in false rejection, however we tested only VQ UBM with 32 and 64 codebook size.</p>
      <p>In Baseline GMM MAP system, Equal error rate is 19.16% for 8 mixtures and
between [35%-36.12%] for model order M=16...128. The performances decreases
because the reduced speech data and didn’t apply normalization technique like
Tnorm.</p>
    </sec>
    <sec id="sec-6">
      <title>6 Conclusions</title>
      <p>VQ SUB UBM achieved (FA=7.14% and FR=22.73%) for 128 codebook size and
improved the performance of vector quantization applied in speaker verification
compared to baseline vector quantization. The codebook size influences the
verification accuracy. The size of speech data should be increased in order to validate
our experiments in large database.</p>
    </sec>
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