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  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>SpeeD @ MediaEval 2015: Multilingual Phone Recognition Approach to Query By Example STD</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Alexandru Caranica</string-name>
          <email>alexandru.caranica@speed.pub.ro</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Andi Buzo</string-name>
          <email>andi.buzo@upb.ro</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Horia Cucu</string-name>
          <email>horia.cucu@upb.ro</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Corneliu Burileanu</string-name>
          <email>corneliu.burileanu@upb.ro</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>SpeeD Research Laboratory, University Politehnica of Bucharest Bucharest</institution>
          ,
          <country country="RO">Romania</country>
        </aff>
      </contrib-group>
      <pub-date>
        <year>2015</year>
      </pub-date>
      <fpage>14</fpage>
      <lpage>15</lpage>
      <abstract>
        <p>In this paper, we attempt to solve the Spoken Term Detection (STD) problem for under-resourced languages by a phone recognition approach within the Automatic Speech Recognition (ASR) paradigm, with multilingual acoustic models from six languages (Albanian, Czech, English, Hungarian, Romanian and Russian). The Power Normalized Cepstral Coefficients (PNCC) features are used for improved robustness to noise, along with Phone Posteriorgrams in order to obtain content-aware acoustic features as independent as possible from speaker and acoustic environment.</p>
      </abstract>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1.1 Acoustic models</title>
      <p>
        In our approach, one thing we want to compare is the effect
of using PNCC features vs MFCC, along with the improvements a
robust phone recognizer based on neural network classifiers can
bring to our STD task. For the first comparison, we have built four
acoustic models, using internal audio resources for training, as
described in Table 1. The AM training and the phoneme
recognition are made in a conventional way, using Hidden
Markov Models (HMMs), in CMU Sphinx [
        <xref ref-type="bibr" rid="ref3">3</xref>
        ].
      </p>
      <p>
        We have built an AM for each language, (AM1 - AM3).
AM1 is trained with 8.7 hours of read speech. We could have
trained the Romanian AM with more data, but as we stated in the
introduction, we wanted to have balanced training data among
different languages, for an under-resourced task. AM2 is trained
with 4.1 hours of Albanian read speech and broadcast news. AM3
is trained with 3.9 hours of native English read speech from the
standard TIMIT database [
        <xref ref-type="bibr" rid="ref4">4</xref>
        ]. AM4 is trained with all the data
from the three languages above, phonemes that are common in
different languages were trained together, thus reducing the
number of phonemes to 98. This was necessary to try and keep
uncertainty as low as possible during the recognition phase. The
identification of the common phonemes was made based on
International Phonetic Alphabet (IPA) classification [
        <xref ref-type="bibr" rid="ref5">5</xref>
        ]. Two
speech features types are used in this first approach: the common
Mel Frequency Cepstral Coefficients (MFCC) and the Power
Normalized Cepstral Coefficients (PNCC) [
        <xref ref-type="bibr" rid="ref6">6</xref>
        ].
      </p>
      <p>No.
phones
34
36
75
98</p>
      <p>Training
data [h]
8.7
4.1
3.9
16.7</p>
      <p>
        For our second approach, we used phone posteriorgrams that
are output by the robust phoneme recognizer from BUT. This
phone recognizer uses a split temporal context (STC) based
feature extraction, with neural network classifiers [
        <xref ref-type="bibr" rid="ref7">7</xref>
        ] to output
phone posteriorgrams, while Viterbi algorithm is used for
phoneme string decoding. We can use the output of this tool in
our DTW search algorithm as input features, to do the matching.
      </p>
      <p>
        In order to use additional languages to build features for the
phoneme recognizer, we used the pre-trained systems available at
[
        <xref ref-type="bibr" rid="ref8">8</xref>
        ] and described in Table 2.
      </p>
      <p>No.
phonemes
45
61
52</p>
      <p>
        The languages used for training the systems described in
Table 2 are from the SpeechDat-E Eastern European Speech
Database [
        <xref ref-type="bibr" rid="ref9">9</xref>
        ]. Another incentive to use these systems is the
existence of trained non-speech events mapped to the following
tokens, which should prove useful with these years challenging
acoustic environment:
 “int” for intermittent noise
 “spk” for speaker noise
 “pau” for silent pause
      </p>
      <p>
        The STC approach is based on the theoretical study that
significant information about phoneme is spread over few
hundreds milliseconds and that an STC system can process two
parts of the phoneme independently. The trajectory representing a
phoneme feature can then be decorrelated by splitting them into
two parts, to limit the size of the model, in particular the number
of weights in the neural-net (NN). The system uses two blocks of
features, for left and right contexts (the blocks have one frame
overlap). Before splitting, the speech signal is filtered by applying
the Hamming window on the whole block, so that the original
central frame is emphasized. Dimensions of vectors are then
reduced by DCT and results are sent to two neural networks. The
posteriors from both contexts are, in the final stage, merged, after
the front-end neural networks are able to generate a three-state per
phoneme posterior model [
        <xref ref-type="bibr" rid="ref10">10</xref>
        ]. The above described features were
used in this work as input to our search algorithm, which is
described in the following section.
      </p>
    </sec>
    <sec id="sec-2">
      <title>1.2 Searching algorithm</title>
      <p>In ideal conditions, if an ASR system should output
utterances with a 100% accuracy and precision, then the STD
would be reduced to a simple character string search of a query
within a textual content. As the experimental results show, we are
far from the ideal case, hence we have to find within a content a
string which is similar to the query.</p>
      <p>Our proposed DTW String Search (DTWSS) uses the
Dynamic Time Warping to align a string (a query) within a
content. The search is not performed on the entire content, but
only on a part of it by means of a sliding window proportional to
the length of the query. The term is considered detected if the
DTW scores above a threshold. This method is refined by
introducing a penalization for the short queries and the spread of
the DTW match. The formula for the score s is given by equation
(1):
  −  
)(1 + 
  −  
 = (1 −  ℎ
)(1 +</p>
      <p>
        −    
where Lq is the length of the query, LQM = 18 and LQm = 4 are
the maximum and the minimum query lengths found in the
development data set, LW is the length of the sliding window, LS is
the length of the matched term in the content, while α and β are
the tuning parameters. For this task, α and β are set to 0.6, from
previous evaluations [
        <xref ref-type="bibr" rid="ref12">12</xref>
        ]. The penalizations in formula (1) are
motivated by the assumption that for two queries of different
length that match their respective contents by the same phone
error rate (PhER), the match of the longer query is more probable
to be the right one. Similarly the more compact DTW matches are
assumed to be more probable than the longer ones. This algorithm
is suitable for queries of type 1 and 2, because the DTW handles
inherently the small variations from the query, but it is not
suitable for queries of type 3 where word order may be inverted.
)
(1)
      </p>
    </sec>
    <sec id="sec-3">
      <title>2. EXPERIMENTAL RESULTS</title>
    </sec>
    <sec id="sec-4">
      <title>2.1 STD results</title>
      <p>For our first comparison, the results obtained on the
development database with the first two type of features (MFCC
and PNCC) are shown in Table 3.</p>
      <p>
        All our runs used our proposed DTW algorithm, described in
section 1.2. The metric used is the normalized cross entropy cost
(Cnxe) [
        <xref ref-type="bibr" rid="ref11">11</xref>
        ]. The results show almost no difference between the
three types of features. Although PNCC should offer better
accuracy in noisy conditions and the BUT systems had tokens for
noise, this was not reflected in the results, with no big
improvements obtained.
      </p>
    </sec>
    <sec id="sec-5">
      <title>2.2 Official run results</title>
      <p>The results obtained by the official runs on the evaluation
database are shown in Table 5. We selected the first five best
models, sorted after the Actual Cnxe metric. Because no tuning is
made based on the development data set, the results on the
evaluation data set are quite similar and the same conclusions can
be drawn. Table 5 shows also the results per query type.
*PNCC model</p>
      <p>It can be noticed that better results are obtained by query
type 2, as these queries are longer, which may have affected the
results. Posterior models (AM6/AM7) seem to offer minimal
performance improvements, so we cannot draw a conclusion that
posteriors are better suited for the STD task in difficult acoustical
environments. Going further, we think improvements can be
obtained by preprocessing the audio first, before extracting
features, to remove noise or possible reverberation.</p>
    </sec>
    <sec id="sec-6">
      <title>3. CONCLUSION</title>
      <p>We have approached STD with a two-step process. A
multilingual ASR is used as a phone recognizer for indexing the
database, while a DTW based algorithm is used for searching a
given query in the content database. We tested three types of
features (MFCC, PNCC and Posteriorgrams) with two approaches
to the phoneme recognizer (statistical HMMs and a neural STC
approach). The results show no big improvement between each
approach and feature types, in part because this year’s database
features very challenging acoustic environments, and our
phonetizers return a lot of “noise” tokens or repeated phonemes,
which reflected further upon our DTW algorithm.</p>
    </sec>
    <sec id="sec-7">
      <title>4. ACKNOWLEDGEMENTS</title>
      <p>The work has been funded by the Sectoral Operational
Programme Human Resources Development 2007-2013 of the
Ministry of European Funds through the Financial Agreements
POSDRU/159/1.5/S/132395 and in part by the PN II Programme
"Partnerships in priority areas" of MEN - UEFISCDI, through
project no. 332/2014.</p>
    </sec>
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